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SIP User`s Manual

1. 2 In the Domain Name field enter the host name to be translated You can enter a string of up to 31 characters long 3 In the First IP Address field enter the first IP address in dotted decimal format notation to which the host name is translated 4 Optionally in the Second IP Address Third IP Address and Second IP Address fields enter the next IP addresses to which the host name is translated 5 Click the Submit button to save your changes 6 To save the changes to flash memory refer to Saving Configuration on page 230 SIP User s Manual 186 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 7 4 7 Internal SRV Table The Internal SRV Table page provides a table for resolving host names to DNS A Records Three different A Records can be assigned to each host name Each A Record contains the host name priority weight and port If the Internal SRV table is configured the device initially attempts to resolve a domain name using this table If the domain name isn t found the device performs an Service Record SRV resolution using an external DNS server You can also configure the Internal SRV table using the ini file table parameter SRV2IP refer to Networking Parameters on page 260 gt To configure the Internal SRV table take these 9 steps 1 Open the Internal SRV Table page Configuration tab gt Protocol Configurati
2. This is a Welcome message WelcomeMessage 3 WHKEKKEKKEKEKKEKRKERREEKREREREREREEEKEEEN WelcomeMessage Note Each index represents a line of text in the Welcome message box Up to 20 indices can be defined 44 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 3 7 Getting Help The Web interface provides you with context sensitive Online Help The Online Help provides you with brief descriptions of most of the parameters you ll need to successfully configure the device The Online Help provides descriptions of parameters pertaining to the currently opened page gt To view the Help topic for a currently opened page take these 4 steps 1 Using the Navigation tree open the required page for which you want Help f 2 On the toolbar click the Help lt button the Help topic pertaining to the opened page appears as shown below Figure 3 25 Help Topic for Current Page faAudioC J MP 4t8 FXS amp FXO NA Sub met Q Bun 6 Help os lon oft TTI Stabs Centiguaten Menagemert 3 Diagnostics Defines the NTP Server IP Scorarios Search address v NTP Settings Basic Full NTP Server IP Address 0000 gt z Mo Ua Network Settings NTP UTC Offset ph Enables or disables the IP Settings embedded Telnet server Application Settings TOLER i Hours 24 Routing Table ITP Updated interval QoS Settings tidMecds Settings v Telnet Settings n a Secunty Setong Embedded Telnet Server
3. 0 Progress The device sends a PROGRESS message default 1 Alert The device sends an ALERT message upon receipt of a 183 response instead of an ISDN PROGRESS message Determines the numerical value that is sent in the Session Expires header in the first INVITE request or response if the call is answered The valid range is 1 to 86 400 sec The default is 0 i e the Session Expires header is disabled Defines the time in seconds that is used in the Min SE header This header defines the minimum time that the user agent refreshes the session The valid range is 10 to 100 000 The default value is 90 Determines the SIP method used for session timer updates 0 Re INVITE Uses Re INVITE messages for session timer updates default 1 UPDATE Uses UPDATE messages Notes The device can receive session timer refreshes using both methods The UPDATE message used for session timer is excluded from the SDP body Determines whether P Asserted Identity or P Preferred Identity is used in the generated INVITE request for Caller ID or privacy 0 Disabled None default 1 Adding PAsserted Identity 2 Adding PPreferred Identity The Asserted ID mode defines the header P Asserted Identity or P Preferred Identity that is used in the generated INVITE request The header also depends on the calling Privacy allowed or restricted The P Asserted Identity or P Preferred Identity headers a
4. If EnableProxyKeepAlive is set to 1 or 2 the device monitors the connection with the Proxies by using keep alive messages OPTIONS or REGISTER To use Proxy Redundancy you must specify one or more redundant Proxies When a port number is specified e g domain com 5080 DNS NAPTR SRV queries aren t performed even if ProxyDNSQueryType is set to 1 or 2 The transport type per Proxy server 0 UDP 1 TCP 2 TLS 1 Undefined Note If no transport type is selected the value of the global parameter SIPTransportType is used refer to SIP General Parameters on page 121 Enables the Proxy Load Balancing mechanism per Proxy Set ID 0 Disable Load Balancing is disabled default 1 Round Robin Round Robin 2 Random Weights Random Weights When the Round Robin algorithm is used a list of all possible Proxy IP addresses is compiled This list includes all IP addresses per Proxy Set after necessary DNS resolutions including NAPTR and SRV if configured After this list is compiled the Proxy Keep Alive mechanism according to parameters EnableProxyKeepAlive and ProxyKeepAliveTime tags each entry as offline or online Load balancing is only performed on Proxy servers that are tagged as online All outgoing messages are equally distributed across the list of IP addresses REGISTER messages are also distributed unless a RegistrarlP is configured The IP addresses list is
5. 1 Use 180 Ringing response to indicate call waiting For a description of this parameter refer to Supplementary Services on page 159 Sets the URI format in the SIP Diversion header 0 tel default 1 sip For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Advanced Parameters on page 151 Defines the timeout in msec between receiving a 100 Trying response and a subsequent 18x response If a 18x response is not received before this timer expires the call is disconnected The valid range is 0 to 32 000 The default value is 0 i e no timeout Determines the format of the Transparent coder representation in the SDP 0 clearmode default 1 X CCD For a description of this parameter refer to DTMF amp Dialing Parameters on page 147 This ini file table parameter determines a single or several up to 5 preferred transmit DTMF negotiation methods The format of this parameter is as follows TxDTMFOption FORMAT TxDTMFOption_Index TxDTMFOption_Type TxDTMF Option For example TxDTMFOption TxDTMFOption 0 1 TxDTMF Option Notes DTMF negotiation methods are prioritized according to the 294 Document LTRT 68808 SIP User s Manual Parameter DisableAutoDTMFMute EnablelmmediateTrying FirstCallIRBTld EnableReasonHeader 3xxBehavior EnablePChargingVector
6. 40 E1 NI2 ISDN 41 E1 CAS R15 Note The device simultaneously supports different variants of CAS and PRI protocols on different E1 T1 spans no more than four simultaneous PRI variants The device simultaneously supports different BRI variants Determines the Tx clock source of the E1 T1 line 0 Recovered Generate the clock according to the Rx of the E1 T1 line default 1 Generated Generate the clock according to the internal TDM bus Notes The source of the internal TDM bus clock is determined by the parameter TDMBusClockSource For detailed information on configuring the device s clock settings refer to Clock Settings on page 393 Defines the trunk priority for auto clock fallback per trunk parameter 0 to 99 priority 0 is the highest default 100 the SW never performs a fallback to that trunk usually used to mark untrusted source of clock Note Fallback is enabled when the TDMBusPSTNAutoClockEnable parameter is set to 1 Use to select B8ZS or AMI for T1 spans and HDB3 or AMI for E1 spans 0 B8ZS use B8ZS line code for T1 trunks only default 1 AMI use AMI line code 2 HDB3 use HDB3 line code for E1 trunks only Defines the line build out loss for the selected T1 trunk 86 Document LTRT 68808 SIP User s Manual 3 Web Based Management ini File Field Name Web Parameter Name LineBuildOut Loss Trace Level TraceLevel Framin
7. 7 15 1 a Ba E E Version 5 6 377 November 2008 A EA AudioCodes Mediant 2000 7 15 2 Call Transfer There are two types of call transfers Consultation Transfer The common way to perform a consultation transfer is as follows In the transfer scenario there are three parties Party A transferring Party B transferred Party C transferred to e A Calls B e B answers e A presses the hook flash button and places B on hold party B hears a hold tone e AdialsC e After A completes dialing C A can perform the transfer by on hooking the A phone e After the transfer is complete B and C parties are engaged in a call The transfer can be initiated at any of the following stages of the call between A and C e Just after completing dialing C phone number transfer from setup e While hearing Ringback transfer from alert e While speaking to C transfer from active Blind Transfer Blind transfer is performed after we have a call between A and B and party A decides to immediately transfer the call to C without speaking with C The result of the transfer is a call between B and C just like consultation transfer only skipping the consultation stage Note The device doesn t initiate call transfer it only responds to call transfer requests SIP User s Manual 378 Document LTRT 68808 SIP User s Manual 8 Networking Capabilities 8 8 1 Networking Capabilities Ethernet Interface Configura
8. 1 to 5 Tx DTMF Option field INFO Nortel Web interface refer to DTMF amp Dialing Parameters on page 147 Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 DTMF Transport Type field DTMF Mute Web interface Using INFO message according to Cisco s mode DTMF digits are carried to the remote side in INFO messages To enable this mode define the following e RxDTMFOption 0 ini file Declare RFC 2833 in SDP field No Web interface refer to DTMF amp Dialing Parameters on page 147 e TxDTMFOption 3 ini file 1 to 5 Tx DTMF Option field INFO Cisco Web interface refer to DTMF amp Dialing Parameters on page 147 Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 DTMF Transport Type field DTMF Mute Web interface Using NOTIFY messages according to lt draft mahy sipping signaled digits 01 txt gt DTMF digits are carried to the remote side using NOTIFY messages To enable this mode define the following e RxDTMFOption 0 ini file Declare RFC 2833 in SDP field No Web interface refer to DTMF amp Dialing Parameters on page 147 e TxDTMFOption 2 ini file 1 to 5 Tx DTMF Option field NOTIFY Web interface refer to DTMF amp Dialing Parameters on page 147 Note that in this mode DTMF digits
9. 3 6 DHCP packets default default 4 10 BootP retries 30 sec 4 7 DHCP packets 5 20 BootP retries 60 sec 5 8 DHCP packets 6 40 BootP retries 120 sec 6 9 DHCP packets 7 100 BootP retries 300 sec 7 10 DHCP packets 15 BootP retries indefinitely 15 18 DHCP packets Enables the Selective BootP mechanism 1 Enabled 0 Disabled default The Selective BootP mechanism available from Boot version 1 92 enables the device s integral BootP client to filter unsolicited BootP DHCP replies accepts only BootP replies that contain the text AUDC in the vendor specific information field This option is useful in environments where enterprise BootP DHCP servers provide undesired responses to the device s BootP requests Note When working with DHCP DHCPEnable 1 the selective BootP feature must be disabled The interval between the device s startup and the first BootP DHCP request that is issued by the device 1 1 second default 2 3 second 3 6 second 4 30 second 5 60 second Note This parameter only takes effect from the next reset of the device 272 Document LTRT 68808 SIP User s Manual Parameter ExtBootPReqEnable Serial Parameters DisableRS232 SerialBaudRate SerialData SerialParity SerialStop SerialFlowControl Version 5 6 4 ini File Configuration Description 0
10. 400 401 402 Table 9 2 Mapping of SIP Response to ISDN Release Reason Description oer Bad request 31 Unauthorized 24 Payment required 21 SIP User s Manual 396 Description Normal unspecified Call rejected Call rejected Document LTRT 68808 SIP User s Manual R aR Description esponse 403 Forbidden 404 Not found 405 Method not allowed 406 Not acceptable 407 B ee 408 Request timeout 409 Conflict 410 Gone 411 Length required 413 Request entity too long 414 Request URI too long 415 Unsupported media type 420 Bad extension 480 Temporarily unavailable 481 E doesn t 482 Loop detected 483 Too many hops 484 Address incomplete 485 Ambiguous 486 Busy here 488 Not acceptable here 500 Server internal error 501 Not implemented 502 Bad gateway 503 Service unavailable 504 Server timeout 505 Version not supported 600 Busy everywhere 603 Decline 604 Does not exist anywhere 606 Not acceptable ISDN Release Reason 21 1 63 79 21 102 41 22 127 127 127 79 127 18 127 127 127 28 17 31 41 38 38 41 102 127 17 21 1 38 9 Advanced PSTN Configuration Description Call rejected Unallocated number Service option unavailable Service option not implemented Call rejected Recovery on timer expiry Temporary failure Number changed w o diagnostic Interworking Interworking Interworking Service option not implemented Interworking No user responding Interworking
11. After changing the subnet mask you must reset the device IP address of the default Gateway used by the device Enter the IP address in dotted decimal notation for example 10 8 0 1 Notes A warning message is displayed after clicking Submit if the entered value is incorrect After changing the default Gateway IP address you must reset the device For detailed information on multiple routers support refer to Multiple Routers Support on page 383 OAM Network Settings Available only in Multiple IP and Dual IP modes IP Address LocalOAMIPAddress Subnet Mask LocalOAMSubnetMask Default Gateway Address LocalOAMDefaultGW The device s source IP address in the operations administration maintenance and provisioning OAMP network The default value is 0 0 0 0 The device s subnet mask in the OAMP network The default subnet mask is 0 0 0 0 N A Use the IP Routing table instead refer to Configuring the IP Routing Table on page 62 Control Network Settings Available only in Multiple IP and Dual IP modes IP Address LocalControllPAddress Subnet Mask LocalControlSubnetMask Default Gateway Address LocalControlDefaultGW The device s source IP address in the Control network The default value is 0 0 0 0 The device s subnet mask in the Control network The default subnet mask is 0 0 0 0 N A Use the IP Routing table instead refer to Configuring the IP Routing Table on page
12. E AnHTTP based Embedded Web Server Web interface using any standard Web browser described in Web based Management on page 19 m A configuration file referred to as the ini file refer to ini File Configuration on page 255 m Simple Network Management Protocol SNMP browser software refer to the Product Reference Manual m AudioCodes Element Management System refer to AudioCodes EMS User s Manual or EMS Product Description To initialize the device by assigning it an IP address a firmware file cmp and a configuration file ini file you can use AudioCodes BootP TFTP utility which accesses the device using its MAC address refer to the Product Reference Manual Version 5 6 17 November 2008 A ge AudioCodes Mediant 2000 Reader s Notes SIP User s Manual 18 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 3 1 Web Based Management The device s Embedded Web Server Web interface provides FCAPS fault management configuration accounting performance and security functionality The Web interface allows you to remotely configure your device for quick and easy deployment including uploading of configuration software upgrade and auxiliary files and resetting the device The Web interface provides real time online monitoring of the device including display of alarms and their severity In addition it displays performance statistics of voice calls and related traff
13. SIP User s Manual Description Enables the use of a SIP Proxy server 0 No Proxy isn t used the internal routing table is used instead default 1 Yes Proxy is used Parameters relevant to Proxy configuration are displayed If you are using a Proxy server enter the IP address of the Proxy server in the Proxy Sets table refer to Proxy Sets Table on page 141 If you are not using a Proxy server you must configure the device s Tel to IP Routing table described in Tel to IP Routing Table on page 175 or Outbound IP Routing table if EnableSBC is set to 1 refer to Outbound IP Routing Table on page 178 Click the right pointing arrow e button to open the Proxy Sets Table page to configure groups of proxy addresses Alternatively you can open this page from the Proxy Sets Table page item refer to Proxy Sets Table on page 141 for a description of this page Note This button appears only if the Use Default Proxy parameter is enabled Defines the Home Proxy Domain Name If specified the Proxy Name is used as the Request URI in REGISTER INVITE and other SIP messages and as the host part of the To header in INVITE messages If not specified the Proxy IP address is used instead The value must be string of up to 49 characters Determines whether the device switches back to the primary Proxy after using a redundant Proxy 0 Parking device continues working with a redundant
14. This parameter can include up to 30 indices 0 29 For assigning various attributes such as Proxy Load Balancing to each Proxy Set ID refer to the ini file parameter ProxySet For configuring the Proxy Set ID table using the Web interface and for a description of the parameters of this ini file table refer to Proxy Sets Table on page 141 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 This ini file table parameter configures the Proxy Set table by assigning various attributes per Proxy Set ID The format of this parameter is as follows ProxySet FORMAT ProxySet_Index ProxySet_EnableProxyKeepAlive ProxySet_ProxyKeepAliveTime ProxySet_ProxyLoadBalancingMethod ProxySet_IsProxyHotSwap ProxySet 285 November 2008 ca AudioCodes Parameter UseSIPTgrp EnableGRUU UserAgentDisplayInfo SIPSDPSessionOwner RetryAfterTime EnablePAssociatedURIHeader EnableContactRestriction RemoveToTagInFailureRespo nse ReRegisterOnConnectionFailu re SourceNumberPreference EnableRTCPAttribute OPTIONSUserPart SIP User s Manual Mediant 2000 Description For example ProxySet FORMAT ProxySet_Index ProxySet_EnableProxyKeepAlive ProxySet_ProxyKeepAliveTime ProxySet_ProxyLoadBalancingMethod ProxySet_lsProxyHotSwap ProxySet 0 0 60 O 0 ProxySet 1 1 60 1 0 ProxySet Notes This table parameter can include u
15. released 1 Pl 1 8 Pl 8 Sends a 183 response to IP ISDN Transfer Capabilities Defines the IP to ISDN Transfer Capability of the Bearer Capability ISDNTransferCapability_ID IE in ISDN SETUP messages The D in the ini file parameter depicts the trunk number 1 Not Configured 0 Audio 3 1 Audio default 1 Speech Speech 2 Data Data Audio 7 Currently not supported Note If this parameter isn t configured or equals to 1 Audio 3 1 capability is used ISDN Flexible Behavior Parameters ISDN protocol is implemented in different Switches PBXs by different vendors Several implementations vary a little from the specification Therefore to provide a flexible interface that supports these ISDN variants the ISDN behavior parameters are used To configure the different behavior bits in the Web interface you can either enter the exact hexadecimal bits value in the field to the right of the relevant parameter or directly configure each bit field by completing the following steps 1 Click the arrow Lop button to the right of the relevant parameter the relevant behavior page appears 2 Modify each bit field according to your requirements 3 Click the Submit button to save your changes Q 931 Layer Response Bit field used to determine several behavior options that influence the Behavior behaviour of the Q 931 protocol To select the options click the arrow Version 5 6 89 November 20
16. 3 3 6 1 1 Replacing the Corporate Logo with an Image You can replace the logo that appears in the Web interface s Title bar using either the Web interface or the ini file gt To replace the default logo with a different image via the Web interface take these 7 steps Access the device s Web interface refer to Accessing the Web Interface on page 20 In the URL field append the case sensitive suffix AdminPage to the IP address e g http 10 1 229 17 AdminPage the Admin page appears On the left pane click Image Load to Device the Image Download page is displayed as shown in the figure below Figure 3 23 Image Download Screen Send Logo Image file from your computer to the device Browse MSSE Logo width 141 _Restore Default Images This button restores the default images Important Use the Save Configuration menu option to save loaded images to flash memory Click the Browse button and then navigate to the folder in which the logo image file that you want to use is located Click the Send File button the image file uploads to the device When loading is complete the page is automatically refreshed and the uploaded logo image is displayed in the Web interface s title bar If you want to modify the width of the image in the Logo Width field enter the new width in pixels and then click the Set Logo Width button To save the image to flash memory r
17. 4000 The first retransmission is sent after 500 msec The second retransmission is sent after 1000 2 500 msec The third retransmission is sent after 2000 2 1000 msec The fourth retransmission and subsequent retransmissions until SIPMaxRtx are sent after 4000 2 2000 msec The maximum interval in msec between retransmissions of SIP messages The default is 4000 Note The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx Maximum number of UDP transmissions first transmission plus retransmissions of SIP messages The range is 1 to 30 The default value is 7 3 4 7 1 2 Proxy amp Registration Parameters The Proxy amp Registration page allows you to configure parameters that are associated with Proxy and Registration Note To view whether the device or its endpoints have registered to a SIP Registrar Proxy server refer to Registration Status SIP User s Manual Document LTRT 68808 132 SIP User s Manual 3 Web Based Management gt To configure the Proxy amp Registration parameters take these 4 steps Open the Proxy amp Registration page Configuration tab gt Protocol Configuration menu gt Protocol Definition submenu gt Proxy amp Registration page item Figure 3 60 Proxy amp Registration Page Use Default Proxy Proxy Name Redundancy Mode Proxy IP List Re
18. EnableVMURI Version 5 6 4 ini File Configuration Description order of their appearance When out of band DTMF transfer is used 1 2 or 3 the parameter DTMFTransportType is automatically set to 0 DTMF digits are erased from the RTP stream When RFC 2833 4 is used the device 1 Negotiates RFC 2833 Payload Type PT using local and remote SDPs 2 Sends DTMF packets using RFC 2833 PT according to the PT in the received SDP 3 Expects to receive RFC 2833 packets with the same PT as configured by the parameter RFC2833PayloadType 4 Uses the same PT for send and receive if the remote party doesn t include the RFC 2833 DTMF PT in its SDP When TxDTMFOption is set to 0 the RFC 2833 PT is set according to the parameter RFC2833PayloadType for both transmit and receive For defining this parameter using the Web interface refer to DTMF amp Dialing Parameters on page 147 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 Enables disables the automatic muting of DTMF digits when out of band DTMF transmission is used 0 Automatic mute is used default 1 No automatic mute of in band DTMF When DisableAutoDTMFMute 1 the DTMF transport type is set according to the parameter DTMFTransportType and the DTMF digits aren t muted if out of band DTMF mode is selected TxDTMFOption 1 2 or 3 This enables the sending of
19. Figure 3 43 Figure 3 44 Figure 3 45 Figure 3 46 Figure 3 47 Figure 3 48 Figure 3 49 Figure 3 50 Figure 3 51 Figure 3 52 Figure 3 53 Figure 3 54 Figure 3 55 Figure 3 56 SIP User s Manual 8 Mediant 2000 Typical Application Enter Network Password Screen Main Areas of the Web Interface GUI Reset Displayed on Toolbar a Terminology for Navigation Tree Levels Ss ailing Wee in Panie ana Full View in pii Cobiotha Paceistar riba ia Editing Symbol after Modifying Parameter Value Mediant 2000 List of Figures Value Reverts to Previous Valid Value Adding an Index Entry to a Table Compacting a Web Interface Table Searched Result Screen es Scenario Creation Confirm Message B Box Creating a Scenario n needs Scenario Loading Message Box Scenario Example me Scenario File Page i EPE Scenario Loading Message Box Message Box for Confirming Scenario Deletion Confirmation Message Box for Exiting Scenario Mode Customizing Web Logo and Product Name Image Download Screen User Defined Web Welcome Message after Login Help Topic for Current Page Areas of the Home Page Qi E EEEE nie ES aaa ee Shortcut Menu for Assigning lt a Port Name Entering the Port Name cseeeee EEE AEPA AEE TEEN zs Click Module to which you want to Switch sseeeecessees snipes ERR a aia Confirmation Message Box for ARAS Module
20. For example CauseMapSIP2ISDN CauseMapSIP2ISDN 0 480 50 CauseMapSIP2ISDN 0 404 3 CauseMapSIP2ISDN When a SIP response is received from the IP side the device searches this mapping table for a match If the SIP response is found the Release Cause assigned to it is sent to the PSTN If no match is found the default static mapping is used Notes This parameter can appear up to 12 times For an explanation on ini file table parameters refer to Structure of ini File Table Parameters on page 257 This ini file table parameter maps Q 850 Release Causes to SIP Responses The format of this parameter is as follows CauseMapISDNZ2SIP FORMAT CauseMapISDN2SIP_Index CauseMapISDN2SIP_IsdnReleaseCause CauseMapISDN2SIP_SipResponse CauseMapISDN2SIP Where IsdnReleaseCause Q 850 Release Cause SipResponse SIP Response For example CauseMapISDN2SIP CauseMapISDN2SIP 0 50 480 CauseMapISDN2SIP 0 6 406 CauseMapISDN2SIP When a Release Cause is received from the PSTN side the device searches this mapping table for a match If the Q 850 Release Cause is found the SIP response assigned to it is sent to the IP side If no match is found the default static mapping is used Notes This parameter can appear up to 12 times For an explanation on ini file table parameters refer to Structure of ini File Table Parameters on page 257 308 Document LTRT 68808 SIP User s Manual
21. G 711 Fax Modem Transport Mode In this mode when the terminating device detects fax or modem signals CED or AnsAM it sends a Re INVITE message to the originating device requesting it to re open the channel in G 711 VBD with the following adaptations m Echo Canceller off m Silence Compression off m Echo Canceller Non Linear Processor Mode off m Dynamic Jitter Buffer Minimum Delay 40 m Dynamic Jitter Buffer Optimization Factor 13 After a few seconds upon detection of fax V 21 preamble or super G3 fax signals the device sends a second Re INVITE enabling the echo canceller the echo canceller is disabled only on modem transmission A gpmd attribute is added to the SDP according to the following format m For G 711A law a gpmd 0 vbd yes ecan on or off for modems E For G 711 p law a gpmd 8 vbd yes ecan on or off for modems The parameters FaxTransportMode and VxxModemTransportType are ignored and automatically set to the mode called transparent with events To configure fax modem transparent mode set IsFaxUsed to 2 Version 5 6 355 November 2008 A Ee AudioCodes Mediant 2000 7 5 2 7 Fax Fallback 7 5 3 7 5 3 1 In this mode when the terminating device detects a fax signal it sends a Re INVITE message to the originating device with T 38 If the remote device doesn t support T 38 replies with SIP response 415 Media Not Supported the device sends a new Re INVITE with G 711 VBD with
22. IPSecMode Remote Tunnel IP Address IPSecPolicyRemoteTunnellPAddress Remote Subnet Mask IPsecPolicyRemoteSubnetMask Remote IP Address IPSecPolicyRemotelPAddress Local IP Address Type IPSecPolicyLocallPAddressType Source Port IPSecPolicySrcPort Destination Port IPSecPolicyDstPort Protocol IPSecPolicyProtocol Related Key Exchange Method Index IPsecPolicyKeyExchangeMethodIndex SIP User s Manual Description Defines the IPSec mode of operation 0 Transport Default 1 Tunneling Defines the IP address of the remote IPSec tunneling device Note This parameter is available only if the parameter IPSecMode is set to Tunneling 1 Defines the subnet mask of the remote IPSec tunneling device The default value is 255 255 255 255 i e host to host IPSec tunnel Note This parameter is available only if the parameter IPSecMode is set to Tunneling 1 Destination IP address or FQDN to which the IPSec mechanism is applied Notes This parameter is mandatory When an FQDN is used a DNS server must be configured DNSPriServerIP Determines the local interface to which the encryption is applied applicable to multiple IPs and VLANs 0 OAM OAMP interface default 1 Control Control interface IPSec is applied to outgoing packets that match the values defined for these parameters Defines the source port to which the IPSec mechanism is appli
23. Notes f registration fails then the userpart in the INVITE Contact header contains the source party number The ContactUser parameter in the Account Table page overrides this parameter An example is shown below of a REGISTER message for registering endpoint 101 using registration Per Endpoint mode The SipGroupName in the request URI is taken from the IP Group table REGISTER sip SipGroupName SIP 2 0 Via SIP 2 0 UDP 10 33 37 78 branch z9hG4bKac862428454 From lt sip 101 GatewayName gt tag 1c862422082 To lt sip 101 GatewayName gt Ca METODES 079770625 20002 S2BA5 10 35 357 78 CSeq 3 REGISTER Contact lt sip 101 10 33 37 78 gt expires 3600 Expires 3600 User Agent Audiocodes Sip Gateway v 5 40A 008 002 Content Length 0 SIP User s Manual 200 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 7 6 3 Configuring the IP Groups The IP Group Table page allows you to create up to nine logical IP entities IP Groups that are later used in the call routing tables The IP Groups are typically implemented in Tel to IP call routing The IP Group can be used as a destination entity in the Tel to IP Routing table or Outbound IP Routing Table and Serving IP Group ID in the Trunk Group Settings refer to Configuring the Trunk Group Settings on page 197 and Account refer to Configuring the Account Table on page 204 tables These call routing tables are used for i
24. O 9 ae TT 8 9 o0000000 F Note The number of channels displayed in the Home page depends on the device s hardware configuration The table below describes the areas of the Home page Table 3 6 Description of the Areas of the Home Page Item Description Label 1 Displays the highest severity of an active alarm raised if any by the device Green No alarms Red Critical alarm Orange Major alarm Yellow Minor alarm You can also view a list of active alarms in the Active Alarms page refer to Viewing Active Alarms on page 245 by clicking the Alarms area 2 Blade Activity icon green Initialization sequence terminated successfully 3 Blade Fail icon gray Normal functioning red Blade failure SIP User s Manual 46 Document LTRT 68808 SIP User s Manual 3 Web Based Management Item Description Label 4 T1 E1 Trunk Status icons for trunks 1 through 8 gray Disable Trunk not configured not in use green Active OK Trunk synchronized yellow RAI Alarm Remote Alarm Indication RAI also known as the Yellow Alarm red LOS LOF Alarm Loss due to LOS Loss of Signal or LOF Loss of Frame blue AIS Alarm Alarm Indication Signal AIS also known as the Blue Alarm orange D Channel Alarm D channel alarm You can switch modules refer to Switching Between Modu
25. Pinholes open On startup the device sends a STUN Binding Request The information received in the STUN Binding Response IP address port is used for SIP signaling This information is updated every user defined period NATBindingDefaultTimeout At the beginning of each call and if STUN is required i e not an internal NAT call the media ports of the call are mapped The call is delayed until the STUN Binding Response that includes a global IP port for each media RTP RTCP and T 38 is received To enable STUN perform the following m Enable the STUN feature using either the Web interface refer to Configuring the Application Settings on page 57 or the ini file set EnableSTUN to 1 m Define the STUN server address using one of the following methods e Define the IP address of the primary and the secondary optional STUN servers using either the Web interface refer to Configuring the Application Settings on page 57 or the ini file STUNServerPrimaryIP and STUNServerSecondaryIP If the primary STUN server isn t available the device attempts to communicate with the secondary server e Define the domain name of the STUN server using the ini file parameter StunServerDomainName The STUN client retrieves all STUN servers with an SRV query to resolve this domain name to an IP address and port sort the server list and use the servers according to the sorted list m Use the ini file parameter NATBindingDefaultTimeou
26. This is especially useful when requiring more than one Scenario to represent different environment setups e g where one includes PBX interoperability and another not Once you create a Scenario and save it to your PC you can then keep on saving modifications to it under different Scenario file names When you require a specific network environment setup you can simply load the suitable Scenario file from your PC refer to Loading a Scenario to the Device on page 39 gt To save a Scenario to a PC take these 5 steps 1 On the Navigation bar click the Scenarios tab the Scenario appears in the Navigation tree 2 Click the Get Send Scenario File button located at the bottom of the Navigation tree the Scenario File page appears as shown below Figure 3 18 Scenario File Page Seemario File LLN Get the Scenario file from the device to your computer Get Scenario File Send Scenario file from your computer to the device Browse Send File SIP User s Manual 38 Document LTRT 68808 SIP User s Manual 3 Web Based Management Click the Get Scenario File button the File Download window appears Click Save and then in the Save As window navigate to the folder to where you want to save the Scenario file When the file is successfully downloaded to your PC the Download Complete window appears Click Close to close the Download Complete window 3 3 5 5 Loading a Scenario to the Device I
27. Transfer request is rejected When this parameter is set to 2 the XferPrefix parameter can be used to define different routing rules for redirected calls This parameter is disregarded if the parameter AlwaysSendToProxy is set to 1 Proxy Registrar Registration Parameters Note The proxy and registrar parameter fields appear only if Enable Registration is enabled Enable Registration IsRegisterNeeded Registrar Name RegistrarName Registrar IP Address RegistrarIP SIP User s Manual Enables the device to register to a Proxy Registrar server 0 Disable device doesn t register to Proxy Registrar default server 1 Enable device registers to Proxy Registrar server when the device is powered up and at every user defined interval configured by the parameter RegistrationTime Note The device sends a REGISTER request for each channel or for the entire device according to the AuthenticationMode parameter Registrar domain name If specified the name is used as the Request URI in REGISTER messages If it isn t specified default the Registrar IP address or Proxy name or IP address is used instead The valid range is up to 49 characters The IP address or FQDN and optionally port number of the SIP Registrar server The IP address is in dotted decimal notation e g 201 10 8 1 lt 5080 gt Notes If not specified the REGISTER request is sent to the primary Proxy ser
28. Trunk ID Trunk Configuration State Protocol Type Active E1 EURO ISDN w Trunk Configuration Clock Master Auto Clock Trunk Priority 0 Line Code Line Build Out Loss Trace Level Line Build Out Overwrite Framing Method HDB3 0 dB Full ISDN Trace OFF EXTENDED SUPER FRAME w ISDN Configuration ISDN Termination Side User side Q931 Layer Response Behavior 0x0 Outgoing Calls Behavior 0x400 Incoming Calls Behavior General Call Control Behavior NFAS Group Number IUA Interface ID NFAS Interface ID D channel Configuration PRIM ARY PSTN Alert Timeout 1 Enable ECT Disable Local ISDN Ringback Tone Source PEX Set PI in Rx Disconnect Message Not Configured ISDN Transfer Capabilities Not Configured Progress Indicator to ISDN Enable Receiving of Overlap Dialing RTP Only Mode Not Configured EIEEE Not Configured Disable B channel Negotiation Not Configured Out Of Service Behavior Default Play Ringback Tone to Trunk 4 lt 1 4 Don t Play Version 5 6 83 November 2008 A Ee AudioCodes Mediant 2000 On the top of the page a bar with Trunk number icons displays the status of each trunk according to the following color codes e Grey Disabled e Green Active e Yellow RAI alarm e Red LOS LOF alarm e Blue AIS alarm e O
29. Under the LOCK UNLOCK group from the Graceful Option drop down list select one of the following options e Yes The device is locked only after the user defined time in the Lock Timeout field refer to Step 3 expires or no more active traffic exists the earliest thereof In addition no new traffic is accepted 229 November 2008 A C al AudioCodes Mediant 2000 e _ No The device is locked regardless of traffic Any existing traffic is terminated immediately Note These options are only available if the current status of the device is in the Unlock state 3 In the Lock Timeout field relevant only if the parameter Graceful Option in the previous step is set to Yes enter the time in seconds after which the device locks Note that if no traffic exists and the time has not yet expired the device locks 4 Click the LOCK button a confirmation message box appears requesting you to confirm device Lock Figure 3 98 Device Lock Confirmation Message Box Microsoft Internet Explorer 2 Are you sure you want to Lock the Gateway so incoming calls wil be rejected and active calls will be closed when timeout expires Ce 5 Click OK to confirm device Lock if Graceful Option is set to Yes the lock is delayed and a screen displaying the number of remaining calls and time is displayed Otherwise the lock process begins immediately The Current Admin State field displays the c
30. When TDM Tunneling is enabled the originating device automatically initiates SIP calls from all enabled B channels pertaining to E1 T1 J1 spans that are configured with the Transparent protocol The called number of each call is the internal phone number of the B channel from where the call 208 Document LTRT 68808 SIP User s Manual Parameter Send Screening Indicator to IP ScreeningInd2IP Send Screening Indicator to ISDN ScreeningInd2ISDN Add IE in SETUP AddlEinSetup Trunk Groups to Send IE SendlEonTG Enable User to User IE for Tel to IP EnableUUITel2IP Version 5 6 3 Web Based Management Description originates The IP to Trunk Group routing table is used to define the destination IP address of the terminating device The terminating device automatically answers these calls if its E1 T1 protocol is set to Transparent ProtocolType 5 Overrides the calling party s number CPN screening indication in the received ISDN SETUP message for Tel to IP calls 1 Not Configured not configured interworking from ISDN to IP or set to 0 for CAS default 0 User Provided CPN set by user but not screened verified 1 User Passed CPN set by user verified and passed 2 User Failed CPN set by user and verification failed 8 Network Provided CPN set by network Note Applicable only if Remote Party ID RPID header is enabled Overrides the screening in
31. included in a TNS IE in the ISDN SETUP message For example INVITE sip 555666 cic 2345 100 2 3 4 sip 2 0 Note Currently this feature is supported only in the SIP to ISDN direction 0 Not used default 1 ISDN Advice of Charge AOC messages are interworked to SIP The device supports receipt of ISDN Euro ISDN AOC messages AOC messages can be received during a call FACILITY messages or at the end of a call DISCONNECT or RELEASE messages The device converts the AOC messages into SIP INFO during a call and BYE end of a call messages using a proprietary AOC SIP header The device supports both 310 Document LTRT 68808 SIP User s Manual Parameter PlayBusyTone2ISDN TrunkTransferMode_X CASTransportType CASAddressingDelimiters Version 5 6 4 ini File Configuration Description Currency and Pulse AOC messages For a description of this parameter refer to SIP General Parameters on page 121 Determines the supported trunk transfer method when a SIP REFER message is received 0 Not supported default 1 Supports CAS NFA DMS 100 transfer When a SIP REFER message is received the device performs a Blind Transfer by executing a CAS Wink waits for an acknowledged Wink from the remote side dials the Refer to number to the switch and then releases the call Note A specific NFA CAS table is required 2 Supports ISDN transfer RLT DMS 100 TBCT NI2 ECT EURO
32. sipping qsig tunnel 03 gt 0 Disable Disable default 1 Enable Enable QSIG tunneling from QSIG to SIP and vice versa When QSIG tunneling is enabled all QSIG messages are sent as raw data in corresponding SIP messages using a dedicated message body Notes QSIG tunneling must be enabled on both originating and terminating devices To enable this function set the parameter ISDNDuplicateQ931 BuffMode to 128 i e duplicate all messages 0 Disable Disable default 1 Using Header Enable ISDN Tunneling from SIP to ISDN PRI using a proprietary SIP header 2 Using Body Enable ISDN Tunneling from SIP to ISDN PRI using a dedicated message body When ISDN Tunneling is enabled the device extracts raw data received in a proprietary SIP header X ISDNTunnelingInfo or a dedicated message body application isdn in the SIP messages 210 Document LTRT 68808 SIP User s Manual Parameter ISDN Transfer On Connect SendISDNTransferOnConnect Remove CLI when Restricted RemoveCLIWhenResiricted Remove Calling Name RemoveCallingName Default Cause Mapping From ISDN to SIP DefaultCauseMapISDN2IP Add Prefix to Redirect Number Prefix2RedirectNumber Copy Destination Number to Redirect Number CopyDest2RedirectNumber Version 5 6 3 Web Based Management Description and sends the data as ISDN messages to the PSTN side This parameter is used for the ECT TBCT RLT Path
33. the NFS Settings page appears Figure 3 37 NFS Settings Page Add Apply Delete Host Or IP Root Path NFS Version Authentication Type User 10 G Vian Type NFS Version 3 v 1 Enable v 3 In the Add field enter the index number of the remote NFS file system and then click Add an empty entry row appears in the table 4 Configure the NFS parameters according to the table below 5 Click the Apply button the remote NFS file system is immediately applied which can be verified by the appearance of the NFS mount was successful message in the Syslog server 6 To save the changes to flash memory refer to Saving Configuration on page 230 SIP User s Manual 60 Document LTRT 68808 SIP User s Manual 3 Web Based Management To avoid terminating current calls a row must not be deleted or modified while the device is currently accessing files on that remote NFS file system The combination of HostOrlP and RootPath must be unique for each row in the table For example the table must include only one row with a Host IP of 192 168 1 1 and Root Path of audio For a description of the web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 30 You can also configure the NFS table using the ini file table parameter NFSServers refer to Networking Parameters on page 260 Table 3 10 Network Settings NFS Settings Parameters Paramet
34. without requiring a device reset However once you start uploading a cmp file the process must be completed with a device reset 4 Click the Browse button navigate to the cmp file and then click Send File the cmp file is loaded to the device and you re notified as to a successful loading as shown below Figure 3 104 CMP File Successfully Loaded Message CMP file INI file CPT tile File SIP_F5 304 012 005 cmp was VP file successfully loaded inte the device PRT file CAS file USRINF file FINISH 5 Click one of the following buttons Oy Y Reset the device resets with the newly loaded cmp and utilizing the current configuration and auxiliary files gt Next the Load an ini File wizard page opens Note that as you progress by clicking Next the relevant file name corresponding to the applicable Wizard page is highlighted in the file list on the left SIP User s Manual 238 Document LTRT 68808 SIP User s Manual 3 Web Based Management 6 In the Load an ini File page you can now choose to either e Click Browse navigate to the ini file and then click Send File the ini file is loaded to the device and you re notified as to a successful loading e __ Use the ini file currently used by the device by not selecting an ini file and by ensuring that the Use existing configuration check box is marked default e Return the device s configuration settings to factory defaults
35. 1 Enable product name change The text string that replaces the product name The default is Mediant 2000 The string can be up to 29 characters 3 3 6 3 Creating a Login Welcome Message You can create a Welcome message box alert message that appears after each successful login to the device s Web interface The ini file table parameter WelcomeMessage allows you to create the Welcome message Up to 20 lines of character strings can be defined for the message If this parameter is not configured no Welcome message box is displayed after login An example of a Welcome message is shown in the figure below Figure 3 24 User Defined Web Welcome Message after Login r Microsoft Internet Explorer SRR AR ARERR EER RE EER EER EE ER EER EE RE ER EE REE ER EE REE AE patat t i a t i i Welcome to the Embedded Web Server faata a a a a a a aa aa a a a aa aa a aa a ata aa ca ata aa a a aa aa a aa aa aa a E SEARCHER EE Table 3 5 ini File Parameter for Welcome Login Message Parameter WelcomeMessage SIP User s Manual Description Defines the Welcome message that appears after a successful login to the Web interface The format of this parameter is as follows WelcomeMessage FORMAT WelcomeMessage_Index WelcomeMessage_ Text WelcomeMessage For Example WelcomeMessage FORMAT WelcomeMessage_Index WelcomeMessage_ Text WelcomeMessage 1 WHEE KKKKEKEKKEKRERRREKRKERKEREREREREREREN WelcomeMessage 2
36. 26 44 H323 Return Code 103 Acct Session ID 7 IP Telephony Capabilities Value PLUTO Format Example Number of packets sent N f umeric during the call Physical port type of device on which the call is active 0 ting Asynchronous The reason for failing authentication 0 ok other number failed Numeric 0 Peguea accepted A unique accounting identifier match start amp String AAA Stop Acc Start Acc Stop Acc Stop Acc Stop Acc stop Below is an example of RADIUS Accounting where the non standard parameters are preceded with brackets Accounting Request 361 user name 111 acct session id 1 nas ip address 212 179 22 213 nas port type 0 acct status type 2 acct input octets acct output octets acct session time 1 acct input packets 122 acct output packets 220 called station id 201 calling station id 202 Accounting non standard parameters 4923 33 h323 gw id 4923 23 h323 remote address 212 179 22 214 4923 1 h323 ivr out h323 incoming conf id 02102944 600a1899 3 d61009 Oe2f3cc5 4923 30 h323 disconnect cause 4923 27 h323 call type VOIP 4923 26 h323 call origin Originate 4923 24 h323 conf id 02102944 600a1899 3f d61009 Oe2f3cc5 4841 8800 22 0x16 7 11 Version 5 6 Call Detail Record The Call Detail Record CDR contains vital statistic information on calls made by the d
37. Control Premium Priority VLANPremiumServiceClassControlPriority Gold Priority VLANGoldServiceClassPriority Bronze Priority VLANBronzeServiceClassPriority Description Defines the priority for Network Class of Service CoS content The valid range is 0 to 7 The default value is 7 Defines the priority for the Premium CoS content and media traffic The valid range is 0 to 7 The default value is 6 Defines the priority for the Premium CoS content and control traffic The valid range is 0 to 7 The default value is 6 Defines the priority for the Gold CoS content The valid range is 0 to 7 The default value is 4 Defines the priority for the Bronze CoS content The valid range is 0 to 7 The default value is 2 Differential Services For detailed information on IP QoS using Differentiated Services refer to IP QoS via Differentiated Services DiffServ on page 384 Network QoS NetworkServiceClassDiffServ Media Premium QoS SIP User s Manual Defines the DiffServ value for Network CoS content The valid range is 0 to 63 The default value is 48 Defines the DiffServ value for Premium Media CoS 64 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Description PremiumServiceClassMediaDiffServ content only if IPDiffServ is not set in the selected IP Profile The valid range is 0 to 63 The default value is 46 Note The value for the Premium Control DiffServ is determ
38. Display The following tables define the device s redirect number and calling name Display support for various PRI variants according to NT Network Termination TE Termination Equipment interface direction Table 9 3 Calling Name Display NT TE Interface DMS 100 NI 2 4 5ESS Euro ISDN QSIG NT to TE Yes Yes Yes Yes Yes TE to NT Yes Yes Yes No Yes Table 9 4 Redirect Number NT TE Interface DMS 100 NI 2 4 5ESS Euro ISDN QSIG NT to TE Yes Yes Yes Yes Yes TE to NT Yes Yes Yes Yes Yes When using ETSI DivertingLegInformation2 in a Facility IE not Redirecting Number IE Version 5 6 401 November 2008 A ge AudioCodes Mediant 2000 9 6 Automatic Gain Control AGC Automatic Gain Control AGC adjusts the energy of the output signal to a required level i e volume This feature compensates for near far gain differences AGC estimates the energy of the incoming signal from the IP or PSTN determined by the parameter AGCRedirection calculates the essential gain and then performs amplification Feedback ensures that the output signal is not clipped You can define the required Gain Slope in decibels per second using the parameter AGCGainSlop and the required signal energy threshold using the parameter AGCTargetEnergy When the AGC first detects an incoming signal it begins operating in Fast Mode which allows the AGC to adapt quickly when a conversation starts This means that the Gain Slope is 8 dB sec for the first
39. FXO for IP to Tel calls 0 Disable Disabled default 1 Enable Enable Digit Delivery feature for the device two stage dialing 153 November 2008 A C wl AudioCodes Mediant 2000 Parameter RTP Only Mode RTPOnlyMode PSTN Alert Timeout PSTNAlertTimeout Reanswer Time RegretTime Description If the called number in IP to Tel call includes the characters w or p the device places a call with the first part of the called number before w or p and plays DTMF digits after the call is answered If the character w is used the device waits for detection of dial tone before it starts playing DTMF digits For example if the called number is 1007766p100 the device places a call with 1007766 as the destination number then after the call is answered it waits 1 5 seconds p and plays the rest of the number 100 as DTMF digits Additional examples 1664wpp102 66644ppp503 and 7774w100pp200 Enables the device to start sending and or receiving RTP packets to and from remote endpoints without the need to establish a Control session The remote IP address is determined according to the Tel to IP Routing table refer to Tel to IP Routing Table on page 175 or Outbound IP Routing table refer to Outbound IP Routing Table on page 178 The port is the same port as the local RTP port set by BaseUDPPort and the channel on which the call is received 0 Disable Disable defa
40. For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 Determines the V 34 fax transport method 0 Transparent 1 Relay default 2 Bypass 8 Transparent with Events 327 November 2008 ca AudioCodes Parameter UserDefinedToneDetectorEn able BellModemTransportType InputGain VoiceVolume RTPRedundancyDepth RFC2198PayloadType EnableSilenceCompression IsCiscoSCEMode EnableEchoCanceller MaxEchoCancellerLength ECNLPMode EchoCancellerAggressiveNL P EnableNoiseReduction SIP User s Manual Mediant 2000 Description Enables or disables detection of User Defined Tones signaling 0 Disable default 1 Enable Determines the Bell modem transport method 0 Transparent default 2 Bypass 3 Transparent with events For a description of this parameter refer to Configuring the Voice Settings on page 66 For a description of this parameter refer to Configuring the Voice Settings on page 66 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the Voice Settings on page 66 Determines wh
41. IPAlertTimeout SIPPSessionExpires SessionExpiresMethod MINSE SIP User s Manual Mediant 2000 Description Determines the format of the AMR header 0 default Non standard multiple frames packing in a single RTP frame Each frame has a CMR amp TOC header 1 Reserved 2 AMR Header according to RFC 3267 Octet Aligned header format 3 AMR is passed using the AMR IF2 format 0 Only use coders from the coder list default 1 Use transparent coder for data calls according to RFC 4040 The Transparent coder can be used on data calls When the device receives a Setup message from the ISDN with TransferCapabilities data it can initiate a call using the coder Transparent even if the coder is not included in the coder list The initiated INVITE includes the following SDP attribute a rtpmap 97 CLEARMODE 8000 The default Payload Type is set according to the CoderName table If the Transparent coder is not set in the Coders table the default value is set to 56 The Payload Type is negotiated with the remote side i e the selected Payload Type is according to the remote side selection The receiving device must include the Transparent coder in its coder list For a description of this parameter refer to SIP General Parameters on page 121 Defines the port with relation to RTP port for sending and receiving T 38 packets 0 Use the RTP port 2 to sen
42. Interworking Interworking Invalid number format Unallocated number User busy Normal unspecified Temporary failure Network out of order Network out of order Temporary failure Recovery on timer expiry Interworking User busy Call rejected Unallocated number Network out of order Messages and responses were created because the ISUP to SIP Mapping draft doesn t specify their cause code mapping Version 5 6 397 November 2008 9 3 9 4 A ge AudioCodes Mediant 2000 ISDN Overlap Dialing Overlap dialing is a dialing scheme used by several ISDN variants to send and or receive called number digits one after the other or several at a time This is in contrast to en bloc dialing in which a complete number is sent The device can optionally support ISDN overlap dialing for incoming ISDN calls for the entire device by setting the ini file parameter ISDNRxOverlap to 1 or per E1 T1 span by setting ISDNRxOverlap_x to 1 where x represents the number of the trunk For configuring ISDN overlap dialing using the Web interface refer to Configuring the Trunk Settings on page 82 To play a Dial tone to the ISDN user side when an empty called number is received set ISDNINCallsBehavior 65536 bit 16 This results in the Progress Indicator to be included in the SetupAck ISDN message The device stops collecting digits for ISDN to IP calls when m The sending device transmits a sending complete IE in t
43. Replacement ISDN Transfer methods Usually the device requests the PBX to connect an incoming and outgoing call This parameter determines if the outgoing call from the device to the PBX must be connected before the transfer is initiated 0 Alert Enable ISDN Transfer if outgoing call is in Alert or Connect state default 1 Connect Enable ISDN Transfer only if outgoing call is in Connect state Determines for IP to Tel calls whether the Calling Number and Calling Name IEs are removed from the ISDN SETUP message if the presentation is set to Restricted 0 No IE aren t removed default 1 Yes IE are removed Enables the device to remove the Calling Name from SIP to ISDN calls 0 Disable Does not remove Calling Name default 1 Enable Remove Calling Name Defines a single default ISDN release cause that is used in ISDN to IP calls instead of all received release causes except when the following Q 931 cause values are received Normal Call Clearing 16 User Busy 17 No User Responding 18 or No Answer from User 19 The range is valid Q 931 release causes 0 to 127 The default value is 0 i e not configured static mapping is used Defines a string prefix that is added to the Redirect number received from the Tel side This prefix is added to the Redirect Number in the Diversion header The valid range is an 8 character string The default is an empty string Determines
44. after detecting the end of DTMF or MF digits at the Tel PSTN side when the DTMF Transport Type is either Relay or Mute Valid range is 0 to 2 000 msec The default is 1 000 msec For a description of this parameter refer to Configuring the Voice Settings on page 66 For a description of this parameter refer to DTMF amp Dialing Parameters on page 147 Determines the R1 MF protocol used for detection 0 ITU default 1 R1 5 Enables or disables detection of User Defined Tones signaling 0 Disable 1 Enable Defines the deviation in Hz allowed for the detection of each signal frequency The valid range is 1 to 50 The default value is 50 Defines the deviation in Hz allowed for the detection of each CPT signal frequency The valid range is 1 to 30 The default value is 10 329 November 2008 ca AudioCodes Parameter MGCPDTMFDetectionPoint KeyBlindTransfer KeyBlindTransferAddPrefix VoicePayloadFormat VQMonEnable VQMonBurstHR VQMonDelayTHR VQMonEOCRVaITHR VQMonGMin RTCPinterval SIP User s Manual Mediant 2000 Description 0 DTMF event is reported on the end of a detected DTMF digit 1 DTMF event is reported on the start of a detected DTMF digit default Keypad sequence that activates blind transfer for Tel to IP calls There are two possible scenarios Option 1 After this sequence is dialed the current call is put on hold using Re INVITE a
45. are listed e Items must be separated by a comma e The Format line must only include columns that can be modified i e parameters that are not specified as read only An exception is Index fields that are always mandatory e The Format line must end with a semicolon Data line s Contain the actual values of the parameters The values are interpreted according to the Format line e The first word of the Data line must be the table s string name followed by the Index field e Items must be separated by a comma e A Data line must end with a semicolon m End of Table Mark Indicates the end of the table The same string used for the table s title preceded by a backslash e g MY_TABLE_ NAME The following displays an example of the structure of an ini file table parameter Table Title This is the title of the table FORMAT Item Index Item Namel Item Name2 Item Name3 This is the Format line Item 0 valuel value2 value3 meem 1 valuen SS values These are the Data lines Table Title This is the end of the table mark Refer to the following notes Version 5 6 257 November 2008 A ge AudioCodes Mediant 2000 m Indices in both the Format and the Data lines must appear in the same order The Index field must never be omitted m The Format line can include a subset of the configurable fields in a table In this case all other fields are assigned
46. channel ID preferred only 572 USE A LAW When set the device sends G 711 A Law in outgoing voice calls When disabled the device sends the default G 711 Law in outgoing voice calls Applicable to E10 variant 1024 Numbering plan type for T1 IP to Tel calling numbers are defined according to the manipulation tables or according to the RPID header default Otherwise the plan type for T1 calls are set according to the length of the calling number 2048 When this bit is set the device accepts any IA5 character in the called_nb and calling_nb strings and sends any IA5 character in the called_nb and is not restricted to extended digits only i e 0 9 16384 DLC REVERSED OPTION Behavior bit used in the IUA interface groups to indicate that the reversed format of the DLCI field must be used Note When using the ini file to configure the device to support several ISDNOutCallsBehavior features add the individual feature values For example to support both 2 and 16 features set ISDNOutCallsBehavior 18 i e 2 16 91 November 2008 A ge AudioCodes Mediant 2000 ini File Field Name Web Parameter Name Incoming Calls Behavior ISDNInCallsBehavior General Call Control Behavior ISDNGeneralCCBehavior SIP User s Manual Valid Range and Description This is the bit field used to determine several behavior options that influence how the ISDN Stack INCOMING calls behave To select the
47. gai ish apsbotineel Telnet Server TCP Port dvencs Ap x Advance Applications Telnet Server idle Timeout SSH Server Enable Help Topic 3 To view a description of a parameter click the plus sign to expand the parameter To collapse the description click the minus amp sign 4 To close the Help topic click the close button located on the top right corner of the Help topic window Instead of clicking the Help button for each page you open you can open it once for a page and then simply leave it open Each time you open a different page the Help topic pertaining to that page is automatically displayed Version 5 6 45 November 2008 A gA AudioCodes Mediant 2000 3 3 8 Using the Home Page The Home page provides you with a graphical display of the device s front panel displaying color coded status icons for monitoring the functioning of the device By default the Home page is displayed when you access the device s Web interface When you are configuring the device in a configuration page you can always return to the Home page by simply clicking the Home icon on the toolbar The Home page also displays general device information in the General Information pane such as the device s IP address and firmware version gt To access the Home page take this step a m On the toolbar click the Home icon the Home page is displayed Figure 3 26 Areas of the Home Page o0o000000
48. memory flash However you can define your own default values instead of using the factory defaults This is performed using an ini file that includes the header ClientDefaults Below this header simply define new default values for the required ini file parameters The parameters are defined in the same format as in the standard ini file and loaded to the device using TFTP i e not via the Web interface An example of a ClientsDefault ini file for defining default values for Syslog server parameters is shown below ClientDefaults EnableSyslog 1 SyslogServerIP 10 13 2 20 gt To define default values for device parameters take these 2 steps 1 Configure the ClientDefaults ini file with new default parameter values as required 2 Load the ClientDefaults ini file to the device using TFTP refer to the Product Reference Manual gt To remove user defined defaults and restore factory default values take this step E Load an empty i e without any parameters ClientDefaults ini file to the device using TFTP 5 2 Restoring Factory Defaults You can restore all or most of the device s configuration settings to default settings m Restoring default settings except for the device s IP address and Web interface s login user name and password Load to the device an empty ini file without any parameters or with a semicolon preceding all lines When a parameter is absent from a loaded ini file the de
49. now active Proxy until the next failure after which it works with the next redundant Proxy default 1 Homing device always tries to work with the primary Proxy server i e switches back to the primary Proxy whenever it s available Note To use ProxyRedundancyMode enable Keep alive with Proxy option EnableProxyKeepAlive 1 or 2 Defines the time interval in seconds between each Proxy IP list refresh The range is 5 to 2 000 000 The default interval is 60 Determines whether the device falls back to the Tel to IP Routing table or Outbound IP Routing table if EnableSBC is set to 1 for call routing when Proxy servers are unavailable 0 Disable Fallback is not used default 1 Enable Tel to IP Routing table or Outbound IP Routing table is used when Proxy servers are unavailable When the device falls back to its Tel to IP Routing table or Outbound IP Routing table the device continues scanning for a Proxy When the device locates an active Proxy it switches from 134 Document LTRT 68808 SIP User s Manual Parameter Prefer Routing Table PreferRouteTable Use Routing Table for Host Names and Profiles AlwaysUseRouteTable Always Use Proxy AlwaysSendToProxy Redundant Routing Mode RedundantRoutingMode SIP ReRouting Mode SIPReroutingMode Version 5 6 3 Web Based Management Description internal routing back to Proxy routing Note To
50. of the call i e both directions For example one Proxy Set for the Internet Telephony Service provider ITSP interfacing with one leg of the device and another Proxy Set for the second SIP entity e g ITSP interfacing with the other leg of the device Note You can also configure the Proxy Sets table using the ini file table parameters ProxylP and ProxySet refer to SIP Configuration Parameters on page 284 gt To add Proxy servers and configure Proxy parameters take these 5 steps 1 Open the Proxy Sets Table page Configuration tab gt Protocol Configuration menu gt Protocol Definition submenu gt Proxy Sets Table page item Figure 3 61 Proxy Sets Table Page v Proxy Set ID 0 v Proxy Address Transport v Enable Proxy Keep Alive Disable Proxy Keep Alive Time 60 Proxy Load Balancing Method Disable Is Proxy Hot Swap No 2 From the Proxy Set ID drop down list select an ID for the desired group 3 Configure the Proxy parameters according to the following table Version 5 6 141 November 2008 A ge AudioCodes Mediant 2000 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 32 Proxy Sets Table Parameters Parameter Description Proxy Set ID The Proxy Set identification number The valid range is 0 to 5 i e up to 6 Pro
51. on page 300 as well as the detector detection sensitivity using the parameter AMDDetectionSensitivity refer to Configuring the IPmedia Settings on page 76 Upon every Answering Machine Detection activation the device can send a SIP INFO message to an Application server notifying it of one of the following m Human voice has been detected m Answering machine has been detected m Silence i e no voice detected has been detected Version 5 6 343 November 2008 A ge AudioCodes Mediant 2000 The table below shows the success rates of the AMD feature for correctly detecting live and fax calls Table 7 1 Approximate AMD Detection Sensitivity Based on North American English AMD Detection petiermance Sensitivity Success Rate for Live Calls Success Rate for Answering Machine 0 Best for Answering Machine 1 82 56 97 10 2 85 87 96 43 3 Default 88 57 94 76 4 88 94 94 31 5 90 42 91 64 6 90 66 91 30 7 Best for Live 94 72 76 14 Calls A pre requisite for enabling the AMD feature is to set the ini file parameter EnableDSPIPMDetectors to 1 In addition to enable voice detection required once the AMD detects the answering machine the ini file parameter EnableVoiceDetection must be setto 1 The device s AMD feature is based on voice detection for North American English If you want to implement AMD in a different language or region you must provide AudioCodes with a database of recorded voices in the lang
52. refer to Customizing the Web Interface on page 41 UseProductName UserProductName Version 5 6 Determines whether the UserProductName text string is displayed instead of the default product name 0 Disabled default 1 Enables the display of the user defined UserProductName text string in the Web interface interface and in the extracted ini file If enabled the UserProductName text string is displayed instead of the default product name Text string that replaces the default product name that appears in the Web interface upper right hand corner and the extracted ini file The default is Mediant 2000 The string can be up to 29 characters 275 November 2008 ca AudioCodes Parameter UseWebLogo WebLogoText LogoWidth LogoFileName 4 4 4 Mediant 2000 Description 0 Logo image is used default 1 Text string is used instead of a logo image If enabled AudioCodes default logo or any other logo defined by the LogoFileName parameter is replaced with a text string defined by the WebLogoText parameter Text string that replaces the logo image The string can be up to 15 characters Width in pixels of the logo image Note The optimal setting depends on the resolution settings The default value is 441 which is the width of AudioCodes displayed logo Name of the image file of type GIF JPEG or JPG containing the user s logo The logo file name can be used
53. set the parameter FaxCNGMode to 1 2 Events Only CNG is detected on the originating side and a fax session is started by the originating side using the Re INVITE message Usually T 38 fax session starts when the preamble signal is detected by the answering side Some SIP devices don t support the detection of this fax signal on the answering side and thus in these cases it is possible to configure the device to start the T 38 fax session when the CNG tone is detected by the originating side However this mode is not recommended Defines the maximum size of a T 38 datagram that the device can receive This value is included in the outgoing SDP when T 38 is in use The valid range is 122 to 1 024 The default value is 122 71 November 2008 A Ee AudioCodes Mediant 2000 3 4 2 3 Configuring the RTP RTCP Settings The RTP RTCP Settings page allows you to configure the Real Time Transport Protocol RTP and Real Time Transport RTP Control Protocol RTCP parameters gt To configure the RTP RTCP parameters take these 4 steps 1 Open the RTP RTCP Settings page Configuration tab gt Media Settings menu gt RTP RTCP Settings page item Figure 3 42 RTP RTCP Settings Page v General Settings Dynamic Jitter Buffer Minimum Delay Dynamic Jitter Buffer Optimization Factor RTP Redundancy Depth Packing Factor Basic RTP Packet Interval Default RTP Directional Contro
54. set to 1 default no alerts are issued Voice quality monitoring end of call low quality alert threshold if set to 1 default no alerts are issued Voice quality monitoring minimum gap size number of frames The default is 16 Determines whether RTCP XR reports are sent to the Event State Compositor ESC and if so defines the interval in which they are sent 0 Disable RTCP XR reports are not sent to the ESC default 1 End Call RTCP XR reports are sent to the ESC at the end of each call 2 End Call amp Periodic RTCP XR reports are sent to the ESC at the end of each call and periodically according to the parameter RTCPInterval Defines the time interval in msec between adjacent RTCP reports The interval range is 0 to 65 535 The default interval is 5 000 Controls whether RTCP report intervals are randomized or whether each report interval accords exactly to the parameter RTCPInterval 0 Disable Randomize default 1 Enable No Randomize IP address of the Event State Compositor ESC The device sends RTCP XR reports to this server using PUBLISH messages The address can be configured as a numerical IP address or as a domain name 75 November 2008 A Ee AudioCodes Mediant 2000 3 4 2 4 Configuring the IPmedia Settings The IPMedia Settings page allows you to configure the IP media parameters This includes Automatic Gain Control AGC parameters AGC equalizes the
55. tag with a port value rport 1001 the destination port of the response is the port indicated in the rport tag For a description of this parameter refer to Configuring the DSP Templates on page 79 Defines the format of the RTP header for VBR coders 0 Payload only no header no TOC no m factor similar to RFC 3558 Header Free format default 1 Supports RFC 2658 1 byte for interleaving header always 0 TOC no m factor 2 Payload including TOC only allow m factor 3 RFC 3558 Interleave Bundled format Determines the required number of silence frames at the beginning of each silence period when using the VBR Coder silence suppression The range is 0 to 255 The default value is 1 Defines the AMR WB AMR Redundancy depth according to RFC 3267 The valid range is 0 to 3 The default is 0 Determines the number of entries to be defined in the AMR Management Policy table Each entry defines the policy of a different rate The range is 0 9 The default is 0 Defines the one way delay value in msec that may cause the AMR Hand Out report 0 Hand Out report is disabled default 255 msec Defines the hysteresis of the Delay Threshold for AMR Hand out events in msec The valid values are 0 to 255 The default is 100 msec 291 November 2008 ca AudioCodes Parameter AMRCoderHeaderFormat TransparentCoderOnDataCall IsFaxUsed T38UseRTPPort DefaultReleaseCause
56. you can configure multiple IPs and VLANs using the individual ini file parameters as shown below VLAN Configuration VlanMode 1 VlanOamVlaniId 4 VlanNativeVlaniId 4 VlanControlVlaniId 5 VlanMediaVlanID 6 Multiple IPs Configuration EnableMultipleIPs 1 LocalMedialIPAddress 10 33 174 50 LocalMediaSubnetMask 255 255 0 0 LocalMediaDefaultGW 10 33 0 1 LocalControlIPAddress 10 32 174 50 LocalControlSubnetMask 255 255 0 0 LocalControlDefaultGW 0 0 0 0 LocalOAMPAddress 10 31 174 50 LocalOAMSubnetMask 255 255 0 0 LocalOAMDefaultGW 0 0 0 0 IP Routing table parameters RoutingTableDestinationsColumn 130 33 4 6 83 4 87 6 RoutingTableDestinationMasksColumn 255 255 255 255 255 255 255 0 RoutingTableGatewaysColumn 10 32 0 1 10 31 0 1 RoutingTableInterfacesColumn 1 0 RoutingTableHopsCountColumn 20 20 SIP User s Manual 392 Document LTRT 68808 SIP User s Manual 9 Advanced PSTN Configuration 9 Advanced PSTN Configuration This section discusses advanced PSTN configurations 9 1 Clock Settings In a traditional TDM service network such as PSTN both ends of the TDM connection must be synchronized If synchronization is not achieved voice frames are either dropped to prevent a buffer overflow condition or inserted to prevent an underflow condition In both cases connection quality and reliability is affected The device s clock settings can be configured to one of the following m Generate its
57. 0 Disable default 1 Enable disables TFTP and allows secure protocols such as HTTPS to fetch the device configuration Note For a detailed explanation on Secure Startup refer to the Product Reference Manual Determines the RSA public key for strong authentication to logging in to the Secure Shell SSH interface if enabled The value should be a base64 encoded string The value can be a maximum length of 511 characters For additional information refer to the Product Reference Manual Enables or disables RSA public keys for SSH 0 RSA public keys are optional if a value is configured for the ini file parameter SSHAdminKey default 1 RSA public keys are mandatory For a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring the General Security Settings on page 109 This ini file table parameter configures the IPSec SPD table The format of this parameter is as follows IPSEC_SPD_TABLE 277 November 2008 ca AudioCodes Parameter IKE Parameters IPSec_IKEDB_Table SIP User s Manual Mediant 2000 Description Format SPD_INDEX IPSecMode IPSecPolicyRemotelPAddress PSecPolicySrcPort IPSecPolicyDStPort IPSecPolicyProtocol IPSecPolicyLifelnSec PSecPolicyLifelnKB IPSecPolicyProposalEncryption_X IPSecPolicyProposalAuthentication_X IPSecPolicyKeyExchangeMethodIndex IPSecPol
58. 1 2 4 8 or 16 T1 spans supporting channel capacity as follows 24 Channels on 1 T1 span with gateway 1 only 48 Channels on 2 T1 spans with gateway 1 only 96 Channels on 4 T1 spans with gateway 1 only 192 Channels on 8 T1 spans with gateway 1 only 384 Channels on 16 T1 spans with gateway 1 and gateway 2 Note Channel capacity depends on configuration settings Voice amp Tone Characteristics Voice Compression G 711 PCM at 64 kbps p law A law EG 711 y law A law at 64 kbps G 723 1 MP MLQ at 5 3 or 6 3 kbps G 726 at 32 kbps ADPCM G 729 CS ACELP 8 kbps Annex A B EVRC AMR Transparent GSM Full Rate Microsoft GSM iLBC QCELP Silence Suppression G 723 1 Annex A G 729 Annex B PCM and ADPCM Standard Silence Descriptor SID with Proprietary Voice Activity Detection VAD and Comfort Noise Generation CNG Packet Loss G 711 appendix 1 G 723 1 G 729 a b Concealment Echo Cancellation G 165 and G 168 2000 configurable tail length per device from 32 to 128 msec DTMF Detection and Dynamic range 0 to 25 dBm compliant with TIA 464B and Bellcore TR Generation NWT 000506 Version 5 6 409 November 2008 ca AudioCodes Function DTMF Transport in band Answer Detector Answer Machine Detector Call Progress Tone Detection and Generation Output Gain Control Input Gain Control Mediant 2000 Specification Mute transfer in RTP payload or relay in compliance with RFC 2833 Answer detect
59. 195 Channel Select Mode The method in which IP to Tel calls are assigned to channels TrunkGroupSettings Chan pertaining to a Trunk Group nelSelectMode 0 By Dest Phone Number Selects the device s channel according to the called number defined in the Trunk Group Table refer to Configuring the Trunk Group Table on page 195 1 Cyclic Ascending default Selects the next available channel in an ascending cyclic order The next highest channel number in the Trunk Group is always selected When the highest channel number in the Trunk Group is reached the lowest channel number in the Trunk Group is selected and then it starts ascending again 2 Ascending Selects the lowest available channel The lowest channel number in the Trunk Group is always first selected and if that channel is unavailable the next highest channel is selected 3 Cyclic Descending Selects the next available channel in descending cyclic order The next lowest channel number in the Trunk Group is always first selected When the lowest channel number in the Trunk Group is reached it selects the highest channel number in the Trunk Group and then start descending again SIP User s Manual 198 Document LTRT 68808 SIP User s Manual Parameter Registration Mode TrunkGroupSettings Regis trationMode Version 5 6 3 Web Based Management Description 4 Descending Selects the highest available channel The highest chan
60. 1c1505895240 To sipp lt Sip Sipp 172 22 2 9 5060 gt tag 1 Call ID 1 29758 172 22 2 9 CSeq TETNEO Contact lt sip 56700 172 22 168 249 gt Supported em timer replaces path resource priority Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO SUB SCRIBE UPDATE User Agent Audiocodes Sip Gateway IPmedia 260 UN v 5 20A 040 004 Content Type application x detect Content Length 30 Type AMD SubType AUTOMATA 2 The device then detects the start of voice i e the greeting message of the answering machine and then sends the following to the Application server INFO sip Sipp 172 22 2 9 5060 SIP 2 0 Via SIP 2 0 UDP 172 22 168 249 branch z9hG4bKac482466515 Max Forwards 70 From sut lt sip 3000 172 22 168 249 5060 gt 5 tag 1c419779142 To sipp lt Sip Sipp 172 22 2 9 5060 gt tag 1 Call ID 1 29753 172 22 2 9 CSeq 1TNEO Contact lt sip 56700 172 22 168 249 gt Supported em timer replaces path resource priority Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO SUB SCRIBE UPDATE User Agent Audiocodes Sip Gateway IPmedia 260 UN v 5 20A 040 004 Content Type application x detect Content Length 34 Type PTT SubType SPEECH START Version 5 6 345 November 2008 A Ee AudioCodes Mediant 2000 3 Upon detection of the end of voice i e end of the greeting message of the answering machine the device sends the Application se
61. 2000 Description For a description of this parameter refer to Configuring the TDM Bus Settings on page 218 Defines the automatic fallback of the clock 0 Manual default 1 Auto Non Revertive 2 Auto Revertive Selects the fallback clock source on which the device synchronizes in the event of a clock failure 4 PSTN Network default 8 H 110A 9 H 110B 10 NetRef1 11 NetRef2 Determines the NetRef frequency for both generation and synchronization 0 8 kHz default 1 1 544 MHz 2 2 048 MHz For a description of this parameter refer to Configuring the Trunk Settings on page 82 Same as the description for parameter LineCode but for a specific trunk ID where 0 depicts the first trunk For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 CAS file name e g E_M_WinkTable dat that defines the C
62. 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 57 Management Settings Parameters Parameter Description Syslog Settings Syslog Server IP Address IP address in dotted decimal notation of the computer you are SyslogServerIP using to run the Syslog server The Syslog server is an application designed to collect the logs and error messages generated by the device Default IP address is 0 0 0 0 For information on Syslog refer to the Product Reference Manual Syslog Server Port Defines the UDP port of the Syslog server The valid range is 0 to 65 535 The default port is 514 SyslogServerPort For information on the Syslog refer to the Product Reference Manual Enable Syslog Sends the logs and error message generated by the device to the EnableSyslog Syslog server 0 Disable Logs and errors are not sent to the Syslog server default 1 Enable Enables the Syslog server Notes If you enable Syslog you must enter an IP address and a port number using SyslogServerlIP and SyslogServerPort parameters You can configure the device to send Syslog messages implementing Debug Recording refer to Debug Recording DR by using the SyslogOutputMethod ini file parameter Syslog messages may increase the network traffic To configure Syslog logging levels use the parameter GwDebugLevel as described in Advanc
63. 3 58 SNMP Trap Destinations Parameters Description eccceecceceeeeeeeeeeneeeeeeeeeeeeeeeees 223 Table 3 59 SNMP Community Strings Parameters Description 0 ccceeeeeeeeeeeeeteeeeeeeteeeeeeenaeeeens 225 Table 3 60 SNMP V3 Users PAraMGteTs s iscecesecissateissenocecdnsnddessaeeudes snssavendanmanessineedsiansnieteainennsanienien 226 Table 3 61 Auxiliary Files DESCTIIPtiONS ccccccccseccesececseeecsececeeecseeecsaeecesueceseeecsaueeesaeceeeeeeseeeses 231 Table 3 62 Ethernet Port Information Parameters cccccccceseceeceeeeeeeeeenneeeceeeeeseeeenaneeseeeeeeeeeeeaas 243 Table 3 63 Color Coding Icons for Trunk and Channel Status ccccccecsssecsseeecsseseessecesseeesteeeens 247 Table 3 64 Call Counters D SCripthoti c ccscicisseiessccanesioanierssnesioenimanbernesnsoeiix anaabirxbenededaneabernbeoencciuaciod 249 Table 3 65 Call Routing Status Parameters ccccccccccccsscecssesecsseceseeecseeeccseeceeeecseeseseseeseeeseeeees 251 Table 3 66 SAS Registered Users Parameters ccccccccccecssscecseesseeecsseeescseecseeeecseesesseseeeeeeseeeees 252 Table 3 67 IP Connectivity Prairies ca desetentidnns teiubertindnndeiasvhsddaabetthidnntdedwbe EENEN NEE A 253 Table 4 1 Networking ini File Parameters cccccccccecsceeeeeeeceeeeeceaeeeeeeeaeeeeaecaaeceaeseaeseeeeereseeeesseesaees 260 Table 4 2 System im File Paramabasa araka AAE AA AAE A aaa 268 Table 4 3 Web and Telnet ini Fi
64. 4 steps 1 2 3 4 On the Navigation bar click the Scenarios tab a message box appears requesting you to confirm Figure 3 19 Scenario Loading Message Box E a Microsoft Internet Explorer dd Loading Scenario PBX Interoperability Click OK the Scenario mode appears in the Navigation tree Click the Delete Scenario File button a message box appears requesting confirmation for deletion Figure 3 20 Message Box for Confirming Scenario Deletion Microsoft Internet Explorer 2 J This operation will delete the current scenario file are you sure Click OK the Scenario is deleted and the Scenario mode closes Note You can also delete a Scenario using the following alternative methods e Loading an empty dat file refer to Loading a Scenario to the Device on page 39 Loading an ini file with the ScenarioFileName parameter set to no value i e ScenarioFileName SIP User s Manual 40 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 3 5 7 3 3 6 3 3 6 1 Exiting Scenario Mode When you want to close the Scenario mode after using it for device configuration follow the procedure below gt To close the Scenario mode take these 2 steps 1 Simply click any tab besides the Scenarios tab on the Navigation bar or click the Cancel Scenarios button located at the bottom of the Navigation tree a message box appears requesting you to confirm exiting Scenario
65. Auxiliary Files Descriptions Description Provisions the device s parameters The Web interface enables practically full device provisioning but customers may occasionally require new feature configuration parameters in which case this file is loaded Note Loading this file only provisions those parameters that are included in the ini file Parameters that are not specified in the ini file are reset to factory default values Up to eight different CAS files containing specific CAS protocol definitions for digital modules These files are provided to support various types of CAS signaling The voice announcement file contains a set of Voice Prompts VP that are played by the device during operation Dial plan file This is a region specific telephone exchange dependent file that contains the Call Progress Tones CPT levels and frequencies that the device uses The default CPT file is U S A 231 November 2008 A ge AudioCodes Mediant 2000 File Type Description Prerecorded The dat PRT file enhances the device s capabilities of playing a wide range of Tones telephone exchange tones that cannot be defined in the Call Progress Tones file User Info The User Information file maps PBX extensions to IP numbers This file can be used to represent PBX extensions as IP phones in the global IP world gt To load an auxiliary file to the device using the Web interface take these 6 steps 1 Open the Load Auxiliary Files pa
66. CN Payload Type EnableStandardS IDPayloadType Comfort Noise Generation Negotiation ComfortNoiseNegotiation Version 5 6 Description Minimum delay in msec for the Dynamic Jitter Buffer The valid range is 0 to 150 The default delay is 10 Note For more information on Jitter Buffer refer to Dynamic Jitter Buffer Operation on page 360 Dynamic Jitter Buffer frame error delay optimization factor The valid range is 0 to 13 The default factor is 10 Notes Set to 13 for data fax and modem calls For more information on Jitter Buffer refer to Dynamic Jitter Buffer Operation on page 360 Determines whether the device generates redundant packets 0 0 Disable the generation of redundant packets default 1 1 Enable the generation of RFC 2198 redundancy packets N A Controlled internally by the device according to the selected coder N A Controlled internally by the device according to the selected coder N A Controlled internally by the device according to the selected coder N A Use the ini file parameter RFC2833PayloadType instead N A Use the ini file parameter RFC2833PayloadType instead RTP redundancy packet payload type according to RFC 2198 The range is 96 127 The default is 104 Note This parameter is applicable only if RTP Redundancy Depth 1 Determines the fax bypass RTP dynamic payload type The valid range is 96 to 120 The default value is 102 Determi
67. DTMF digits in band transparent of RFC 2833 in addition to out of band DTMF messages Note Usually this mode is not recommended Determines if and when the device sends a 100 Trying response to an incoming INVITE request 0 100 Trying response is sent upon receipt of Proceeding message from the PSTN 1 100 Trying response is sent immediately upon receipt of INVITE request default For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 295 November 2008 A ge AudioCodes Mediant 2000 Parameter Description MaxActiveCalls For a description of this parameter refer to Advanced Parameters on page 151 MaxCallDuration For a description of this parameter refer to Advanced Parameters on page 151 EnableBusyOut For a description of this parameter refer to Advanced Parameters on page 151 EnableDigitDelivery2IP For a description of this parameter refer to Advanced Parameters on page 151 EnableDigitDelivery For a description of this parameter refer to Advanced Parameters on page 151 SlTDetectorEnable Enables or disables Special Info
68. DTMF digits is reduced To enable this mode define the following e RxDTMFOption 0 ini file Declare RFC 2833 in SDP field No Web interface refer to DTMF amp Dialing Parameters on page 147 e TxDTMFOption 0 ini file 1 to 5 Tx DTMF Option field Disable Web interface refer to DTMF amp Dialing Parameters on page 147 e DTMFTransportType 2 DTMF Transport Type Transparent DTMF m Using INFO message according to Korea mode DTMF digits are carried to the remote side in INFO messages To enable this mode define the following e RxDTMFOption 0 ini file Declare RFC 2833 in SDP field No Web interface refer to DTMF amp Dialing Parameters on page 147 e TxDTMFOption 3 ini file 1 to 5 Tx DTMF Option field INFO Korea Web interface refer to DTMF amp Dialing Parameters on page 147 Note that in this mode DTMF digits are erased from the audio stream DTMFTransportType is automatically set to 0 DTMF Mute The device is always ready to receive DTMF packets over IP in all possible transport modes INFO messages NOTIFY and RFC 2833 in proper payload type or as part of the audio stream To exclude RFC 2833 Telephony event parameter from the device s SDP set RxDTMFOption to 0 in the ini file The following parameters affect the way the device handles the DTMF digits m TxDTMFOption RXDTMFOption and RFC2833PayloadType described in DTMF amp Diali
69. Disable default 1 Enable extended information to be sent in BootP request If enabled the device uses the vendor specific information field in the BootP request to provide device related initial startup information such as blade type current IP address software version etc For a full list of the vendor specific Information fields refer to the Product Reference Manual The BootP TFTP configuration utility displays this information in the Client Info column refer to the Product Reference Manual Note This option is not available on DHCP servers Enables or disables the device s RS 232 port 0 RS 232 serial port is enabled default 1 RS 232 serial port is disabled The RS 232 serial port can be used to change the networking parameters and view error notification messages For information on establishing a serial communications link with the device refer to the device s Installation Manual Determines the value of the RS 232 baud rate The valid range is any value It is recommended to use the following standard values 1200 2400 9600 default 14400 19200 38400 57600 115200 Determines the value of the RS 232 data bit 7 7 bit 8 8 bit default Determines the value of the RS 232 polarity 0 None default 1 Odd 2 Even Determines the value of the RS 232 stop bit 1 1 bit default 2 2 bit Determines the value of the RS 232 flow contr
70. EnableQS GTunneling Enable ISDN Tunneling IP to Tel EnablelSDNTunnelingIP2Tel SIP User s Manual Mediant 2000 Description Enables SIP to PRI ISDN interworking 0 Disable Disabled default 1 Enable Enable transfer of UUIE from SIP INVITE message to PRI SETUP message The device supports the following SIP to PRI ISDN interworking SIP INVITE to SETUP SIP 200 OK to CONNECT SIP INFO to USER INFORMATION SIP 18x to ALERT and SIP BYE to DISCONNECT Note The interworking of User to User IE to SIP INFO is supported only on 4ESS PRI variant Enables ISDN Tunneling 0 Disable Disable default 1 Using Header Enable ISDN Tunneling from ISDN PRI to SIP using a proprietary SIP header 2 Using Body Enable ISDN Tunneling from ISDN PRI to SIP using a dedicated message body When ISDN Tunneling is enabled the device sends all ISDN PRI messages using the correlated SIP messages The ISDN SETUP message is tunneled using SIP INVITE all mid call messages are tunneled using SIP INFO and ISDN DISCONNECT RELEASE is tunneled using SIP BYE messages The raw data from the ISDN is inserted into a proprietary SIP header X ISDNTunnelingInfo or a dedicated message body application isdn in the SIP messages Note It is necessary to set the parameter ISDNDuplicateQ931BuffMode to 128 i e duplicate all messages for this feature to function Enables QSIG tunneling over SIP according to lt draft elwell
71. For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 Determines whether the device sends RFC 2833 ANS ANSam events upon detection of fax and or modem answer tones i e CED tone 0 Disabled default 1 Enabled Note This parameter is applicable only when the fax or modem transport type is set to bypass or Transparent with Events For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 Modem Bypass dynamic payload type The range is 0 127 The default value is 103 Determines the fax gain control The ran
72. Group ID You can also configure the IP to Trunk Group Routing table using the ini file table parameter PSTNPrefix refer to Number Manipulation and Routing Parameters on page 313 gt To configure the IP to Trunk Group Routing table take these 5 steps 1 Open the IP to Trunk Group Routing page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt IP to Trunk Group Routing page item Figure 3 73 IP to Trunk Group Routing Table Page _ Matching Rules Tel Destination Rules 2 From the Routing Index drop down list select the range of entries that you want to add SIP User s Manual 182 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 Configure the table according to the table below 4 Click the Submit button to save your changes 5 To save the changes so they are available after a power failure refer to Saving Configuration on page 230 Table 3 46 IP to Trunk Group Routing Table Description Parameter IP to Tel Routing Mode RouteModelP2Tel Dest Host Prefix PstnPrefix_DestHostPrefix Source Host Prefix PstnPrefix_SrcHostPrefix Dest Phone Prefix PstnPrefix_DestPrefix Source Phone Prefix PstnPrefix_SourcePrefix Source IP Address PstnPrefix_SourceAddress Version 5 6 Description Determines whether to route IP calls to the Trunk Group before or after manipulation of destination number configured in Con
73. IP to Tel or Tel to IP calls the device receives a SIP request message using the X Detect header that the remote party wishes to detect events on the media stream For incoming IP to Tel calls the request must be indicated in the initial INVITE and responded to either in the 183 response for early dialogs or in the 200 OK response for confirmed dialogs For outgoing calls Tel to IP the request may be received in the 183 for early dialogs and responded to in the PRACK or received in the 200 OK for confirmed dialogs and responded to in the ACK 2 Once the device receives such a request it sends a SIP response message using the X Detect header to the remote party listing all supported events that can be detected The absence of the X Detect header indicates that no detections are available 3 Each time the device detects a supported event the event is notified to the remote party by sending an INFO message with the following message body e Content Type application K DETECT e Type AMD CPT FAX PTT e Subtype xxx according to the defined subtypes of each type Below is an example of SIP messages implementing the X Detect header INVITE sip 101 10 33 2 53 user phone SIP 2 0 Via SIP 2 0 UDP 10 33 2 53 branch z9hG4bKac5906 Max Forwards 70 From anonymous lt sip anonymous anonymous invalid gt tag 1c25298 To lt Sip 101 10 33 2 53 user phone gt Calil 1De 119230110 33 2 53 CSeq 1 INVITE Co
74. ISDN and Path Replacement QSIG When a SIP REFER message is received the device performs a transfer by sending FACILITY messages to the PBX with the necessary information on the call s legs that are to be connected The different ISDN variants use slightly different methods using FACILITY messages to perform the transfer 3 Supports CAS Normal transfer When a SIP REFER message is received the device performs a Blind Transfer by executing a CAS Wink dialing the Refer to number to the switch and then releasing the call 4 Supports QSIG Single Step transfer IP to Tel When a SIP REFER message is received the device performs a transfer by sending a FACILITY message to the PBX initiating Single Step transfer Once a success return result is received the transfer is completed Tel to IP When a FACILITY message initiating Single Step transfer is received from the PBX a REFER message is sent to the IP side Notes To use QSIG Path Replacement the parameter UserToUserHeaderFormat must be set to 1 To configure Trunk Transfer Mode using the Web interface refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Voice Settings on page 66 Determines if delimiters are added to the dialed address or dialed ANI parameters 0 Disable default 1 Enable When this parameter is enabled delimiters such as and ST are added to th
75. ISDNJapanNTTTimerT3JA EnablePatternDetector PDPattern PDThreshold Enable911LocationldIP2Tel EarlyAnswerTimeout Version 5 6 4 ini File Configuration Description T3_JA timer in seconds This parameter overrides the internal PSTN T301 timeout on the Users Side TE side If an outgoing call from the device to ISDN is not answered during this timeout the call is released The valid range is 10 to 240 The default value is 50 Applicable only to Japan NTT PRI variant ProtocolType 16 Note This timer is also affected by the parameter PSTNAlertTimeout For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 Defines the patterns that can be detected by the Pattern Detector The valid range is 0 to OxFF Defines the number of consecutive patterns to trigger the pattern detection event The valid range is 0 to 31 The default is 5 Enables interworking of Emergency Location Identification from SIP to PRI 0 Disabled default 1 Enabled When enabled the From header received in the SIP INVITE is translated into the following ISDN Information Elements IE Emergency Call Control IE Generic Information IE to carry the Location Identification Number information Generic Information IE to carry the Calling Geodetic Location information Note This capability is supported only for the NI 2 ISDN variant Defines the time in seconds
76. Inthe Navigation tree navigate to the desired page item the corresponding page opens in the Work pane c Inthe page select the required parameter s by marking the corresponding check box es d Click Next e Add or Remove Parameters a Inthe Navigation tree select the required Step the corresponding page opens in the Work pane b To add parameters select the check boxes corresponding to the desired parameters to remove parameters clear the check boxes corresponding to the parameters that you want removed c Click Next 37 November 2008 A ge AudioCodes Mediant 2000 e Edit the Step Name a Inthe Navigation tree select the required Step b Inthe Step Name field modify the Step name c Inthe page click Next e Edit the Scenario Name a Inthe Scenario Name field edit the Scenario name b In the displayed page click Next e Remove a Step a Inthe Navigation tree select the required Step the corresponding page opens in the Work pane b In the page clear all the check boxes corresponding to the parameters c Click Next 5 After clicking Next a message box appears notifying you of the change Click OK 6 Click Save amp Finish a message box appears informing you that the Scenario has been successfully modified The Scenario mode is exited and the menus of the Configuration tab appear in the Navigation tree 3 3 5 4 Saving a Scenario to a PC You can save a Scenario to a PC as a dat file
77. Mediant 2000 SIP gateway referred to throughout this manual as device The device is a SIP based Voice over IP VoIP media gateway the device enables voice fax and data traffic to be sent over the same IP network The device provides excellent voice quality and optimized packet voice streaming over IP networks The device uses the award winning field proven VolPerfect voice compression technology typically implemented in AudioCodes products The device incorporates 1 2 4 8 or 16 E1 T1 or J1 spans for direct connection to the Public Switched Telephone Network PSTN Private Branch Exchange PBX through digital telephony trunks The device also provides SIP trunking capabilities for Enterprises operating with multiple Internet Telephony Service Providers ITSP for VoIP services The device includes two 10 100Base TX Ethernet ports providing redundancy connection to the network The device supports up to 480 simultaneous VoIP or Fax over IP FoIP calls supporting various Integrated Services Digital Network ISDN Primary Rate Interface PRI protocols such as EurolSDN North American NI2 Lucent 4 5ESS Nortel DMS100 and others In addition it supports different variants of Channel Associated Signaling CAS protocols for E1 and T1 spans including MFC R2 E amp M immediate start E amp M delay dial start loop start and ground start The device best suited for large and medium sized VoIP applications is a compact devic
78. Modem Bypass Output Gain ModemBypassOutputGain SIP User s Manual Mediant 2000 Description Number of times that each fax relay payload is retransmitted to the network 0 No redundancy default 1 One packet redundancy 2 Two packet redundancy Note This parameter is applicable only to non V 21 packets Number of times that control packets are retransmitted when using the T 38 standard The valid range is 0 to 4 The default value is 0 Determines whether the Error Correction Mode ECM mode is used during fax relay 0 Disable ECM mode is not used during fax relay 1 Enable ECM mode is used during fax relay default Maximum rate in bps at which fax relay messages are transmitted outgoing calls 0 2400 2 4 kbps 1 4800 4 8 kbps 2 7200 7 2 kbps 3 9600 9 6 kbps 4 12000 12 0 kbps 5 14400 14 4 kbps default Note The rate is negotiated between the sides i e the device adapts to the capabilities of the remote side Coder used by the device when performing fax modem bypass Usually high bit rate coders such as G 711 should be used 0 G 711Alaw G 711 A law 64 default 1 G 711Mulaw G 711 p law Number of 20 msec coder payloads that are used to generate a fax modem bypass packet The valid range is 1 2 or 3 coder payloads The default value is 1 coder payload Defines the fax bypass output gain control The range
79. Parameter SITQ850Cause UserToUserHeaderFormat RemoveCLIWhenRestricted ScreeningInd2ISDN ProgressIndicator2ISDN_ID PiForDisconnectMsg_ID ConnectOnProgressind LocallISDNRBSource_ ID PSTNAlertTimeout TrunkPSTNAlertTimeout_ID ISDNTransferCapability_ID oo nelNegotiationForTrunk SendISDNTransferOnConnect ISDNSubAddressFormat Version 5 6 4 ini File Configuration Description Determines the Q 850 cause value specified in the Reason header that is included in a 4xx response when Special Information Tone SIT is detected on an IP to Tel call The valid range is 0 to 127 The default value is 34 Determines the format of the User to User header 0 X UserToUser default 1 User to User with Protocol Discriminator pd attribute User to User 3030373435313734313635353b313233343b3834 pd 4 This is in accordance with the definitions in draft jonnston sipping cc uui 04 2 User to User with pd embedded as the first byte User to User 043030373435313734313635353b313233343b3834 encoding hex For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 Enables
80. Parameters Page Enable RADIUS Access Control Accounting Server IP Address Accounting Port RADIUS Accounting Type 444 Indications Disable 0 0 0 0 1646 At Call Release None Configure the RADIUS accounting parameters according to the table below To save the changes to flash memory refer to Saving Configuration on page 230 2 3 Click the Submit button to save your changes 4 Table 3 55 RADIUS Parameters Description Parameter Enable RADIUS Access Control EnableRADIUS Accounting Server IP Address Description Enables or disables the RADIUS application 0 Disable disables RADIUS application default 1 Enable enables RADIUS application RADIUSAccServerIP Accounting Port RADIUSAccPort Version 5 6 IP address of the RADIUS accounting server Port of the RADIUS accounting server The default value is 1646 217 November 2008 A ge AudioCodes Mediant 2000 Parameter Description RADIUS Accounting Type Determines when the RADIUS accounting messages are sent to the RADIUSAccountingType RADIUS accounting server 0 At Call Release Sent at call release only default 1 At Connect amp Release Sent at call connect and release 2 At Setup amp Release Sent at call setup and release AAA Indications Determines the Authentication Authorization and Accounting AAA AAAIndications indications 0 None No indications default 3
81. Per Account in the Trunk Group Settings table refer to Configuring the Trunk Group Settings on page 197 for the specific Trunk Group The Host Name i e host name in SIP From To headers and Contact User user in From To and Contact headers are taken from this table upon a successful registration See the example below REGISTER sip audiocodes SIP 2 0 Via SIP 2 0 UDP 10 33 37 78 branch z9hG4bKac1397582418 From lt Sip ContactUser HostName gt tag 1c1397576231 To lt sip ContactUser HostName gt Call ID 1397568957261200022256 10 33 37 78 CSeq 1 REGISTER Contact lt sip ContactUser 10 33 37 78 gt expires 3600 Expires 3600 User Agent Audiocodes Sip Gateway v 5 40A 008 002 Content Length 0 Notes The Trunk Group account registration is not effected by the parameter IsRegisterNeeded You can configure this table so that a specific IP Group can register to multiple ITSP s This is performed by defining several rows in this table containing the same Served IP Group but with different Serving IP Groups user password Host Name and Contact User parameters f registration to an IP Group s fails for all the accounts defined in this table for a specific Trunk Group and if this Trunk Group includes all the channels in the Trunk refer to Configuring the Trunk Group Table on page 195 the 206 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Description Trunk Group is s
82. Phone Number The Trunk Group ID 1 99 assigned to the corresponding channels The same Trunk Group ID can be used for more than one group of channels Trunk Group ID is used to define a group of common channel behavior that are used for routing IP to Tel calls If an IP to Tel call is assigned to a Trunk Group the call is routed to the channel or channels that correspond to the Trunk Group ID You can configure the Trunk Group Settings table refer to Configuring the Trunk Group Settings on page 197 to determine the method in which new calls are assigned to channels within the Trunk Groups Note You must configure the IP to Trunk Group Routing Table page refer to IP to Trunk Group Routing on page 181 to assign incoming IP calls to the appropriate Trunk Group If you do not configure the IP to Trunk Group Routing Table calls do not complete The Tel profile ID refer to Tel Profile Settings on page 192 assigned to the channels defined in the Channels field 3 4 7 6 2 Configuring the Trunk Group Settings The Trunk Group Settings page is mainly used to select the method for which IP to Tel calls are assigned to channels within each Trunk Group If no method is selected for a specific Trunk Group the setting of the global parameter ChannelSelectMode in the SIP General Parameters page refer to SIP General Parameters on page 121 applies In addition this page also defines the method for registering Tr
83. Port Name a mm oc x 3 oe oooo0o0000 ee oo0o000000 Enter port name or description ApplyPortinfo 3 Type a brief description for the port and then click Apply Port Info 3 3 8 2 Viewing Trunk Settings The Home page allows you to view the settings of a selected port in the Trunk Settings screen Accessing this screen from the Home page provides an alternative to accessing it from the Advanced Configuration menu refer to Configuring the Trunk Settings on page 82 gt To view port settings take these 2 steps 1 On the Home page click a desired trunk port LED on the TP 1610 labeled as items 3 and 5 in the figure in Accessing the Home Page a shortcut menu appears 2 From the shortcut menu choose Port Settings the Trunk Settings screen opens 3 3 8 3 Switching Between Modules The device can house up to two modules as discussed in previous sections Since each module is a standalone gateway the Home page displays only one of the modules to which you are connected However you can easily switch to the second module by having the Web browser connect to the IP address of the other module gt To switch modules take these 3 steps 1 In the Home page click anywhere on the module to which you want to switch as shown below Figure 3 29 Click Module to which you want to Switch 83 oo0o000000 o0 o A confirmation message box appears requesting you to confirm sw
84. Registration server in the Expires header minus the value of the parameter RegistrationTimeThreshold The valid range is 0 to 2 000 000 The default value is 0 Enables immediate re registration if a failure response is received for an INVITE request sent by the device 0 Disable Disabled default 1 Enable Enabled Enables the device to perform SIP Re Registration upon TCP TLS connection failure 0 Disable default 1 Enable 137 November 2008 ca AudioCodes Parameter Miscellaneous parameters Gateway Name SIPGatewayName Gateway Registration Name GWRegistrationName DNS Query Type DNSQueryType Proxy DNS Query Type ProxyDNSQueryType SIP User s Manual Mediant 2000 Description Assigns a name to the device e g gateway1 com Ensure that the name you choose is the one with which the Proxy is configured to identify the device Note If specified the device name is used as the host part of the SIP URI in the From header If not specified the device s IP address is used instead default Defines the user name that is used in the From and To headers in REGISTER messages If no value is specified default for this parameter the UserName parameter is used instead Note This parameter is applicable only for single registration per device i e AuthenticationMode is set to 1 When the device registers each channel separately i e AuthenticationMode is set to 0 the us
85. Signaling Method Dynamic Jitter Buffer Minimum Delay msec Dynamic Jitter Buffer Optimization Factor RTP IP DiffServ Signaling DiffServ Voice Volume 32 to 31 dB DTMF Volume 31 to 0 dB Input Gain 32 to 31 dB Enable Digit Delivery Disable Echo Canceler Enable Flash Hook Period 700 Enable Early Media Disable Progress Indicator to IP Not Configured Disconnect Call on Detection of Busy Tone Enable Time For Reorder Tone sec 255 Coder Group Coder Group Default Coder G roup SIP User s Manual 192 Document LTRT 68808 SIP User s Manual 3 Web Based Management 2 From the Profile ID drop down list select the Tel Profile identification number you want to configure 3 In the Profile Name field enter an arbitrary name that enables you to easily identify the Tel Profile 4 From the Profile Preference drop down list select the priority of the Tel Profile where 1 is the lowest priority and 20 is the highest If both IP and Tel profiles apply to the same call the coders and other common parameters noted by an asterisk in the description of the parameter TelProfile of the preferred Profile are applied to that call If the Preference of the Tel and IP Profiles is identical the Tel Profile parameters are applied Note If the coder lists of both IP and Tel Profiles apply to the same cal
86. Size Characters PBX extension The relevant PBX extension number 10 Global phone The relevant global phone number 20 A string that represents the PBX extensions for the Display name Caller ID 30 Username A string that represents the user name for SIP 40 registration Password A string that represents the password for SIP 20 registration SIP User s Manual 340 Document LTRT 68808 SIP User s Manual 6 Auxiliary Configuration Files An example of a User Information file is shown in the figure below Figure 6 1 Example of a User Information File UserInformationFile1000 txt Notepad 5 x 401 6380001 DN401 UN401 401 402 6380002 DN402 UN402 401 408 6380008 DN408 UN408 401 Note The last line in the User Information file must end with a carriage return i e by pressing the lt Enter gt key The User Information file can be loaded to the device using the ini file UserlnfoFileName parameter described in Auxiliary Configuration Files Parameters on page 331 the Web interface refer to Loading Auxiliary Files on page 231 or by using the automatic update mechanism UserlnfoFileURL refer to the Product Reference Manual The maximum permissible size of the file is 108 000 bytes Each PBX extension registers separately a REGISTER message is sent for each entry only if AuthenticationMode is set to Per Endpoint using the IP number in the From To headers The REGISTER messages are sent graduall
87. Statistics on page 248 Version 5 6 241 November 2008 ca AudioCodes 3 6 1 Mediant 2000 Status amp Diagnostics The Status amp Diagnostics menu is used to view and monitor the device s channels Syslog messages hardware and software product information and to assess the device s statistics and IP connectivity information This menu includes the following page items 3 6 1 1 Message Log refer to Viewing the Device s Syslog Messages on page 242 Ethernet Port Information refer to Viewing Ethernet Port Information on page 243 Active IP Interfaces refer to Viewing Active IP Interfaces on page 244 Device Information refer to Viewing Device Information on page 244 Performance Statistics refer to Viewing Performance Statistics on page 245 Active Alarms refer to Viewing Active Alarms on page 245 Trunks amp Channels Status refer to Viewing Trunks amp Channels Status on page 246 Viewing the Device s Syslog Messages The Message Log page displays Syslog debug messages sent by the device You can select the Syslog messages in this page and then copy and paste them into a text editor such as Notepad This text file txt can then be sent to AudioCodes Technical Support for diagnosis and troubleshooting Note It s not recommended to keep a Message Log session open for a prolonged period This may cause the device to overload For prolonged and detailed debugging use an external Syslog serve
88. User s Manual 152 Document LTRT 68808 SIP User s Manual Parameter General IP Security SecureCallsFromIP Filter Calls to IP FilterCalls2IP Enable Digit Delivery to IP EnableDigitDelivery2IP Enable Digit Delivery to Tel EnableDigitDelivery Version 5 6 3 Web Based Management Table 3 35 Advanced Parameters Description Description Determines whether the device accepts SIP calls received from only IP addresses defined in the Tel to IP Routing table refer to Tel to IP Routing Table on page 175 or Outbound IP Routing table if EnableSBC is set to 1 refer to Outbound IP Routing Table on page 178 This is useful in preventing unwanted SIP calls or messages and or VoIP spam 0 Disable device accepts all SIP calls default 1 Enable device accepts SIP calls only from IP addresses defined in the Tel to IP Routing table or Outbound IP Routing table The device rejects all calls from unknown IP addresses Note Specifying the IP address of a Proxy server in the Tel to IP Routing table or Outbound IP Routing table enables the device to accept only calls originating from the Proxy server while rejecting all other calls that don t appear in this table Enables filtering of Tel to IP calls when a Proxy is used i e IsProxyUsed parameter is set to 1 refer to Proxy amp Registration Parameters on page 132 0 Don t Filter device doesn t
89. Web interface is restored You can schedule automatic loading of cmp ini and auxiliary files using HTTP HTTPS FTP or NFS Refer to the Product Reference Manual SIP User s Manual 236 Document LTRT 68808 SIP User s Manual 3 Web Based Management gt To use the Software Upgrade Wizard take these 11 steps 1 Stop all traffic on the device refer to the note above 2 Open the Software Upgrade Wizard Management tab gt Software Update menu gt Software Upgrade Wizard the Software Upgrade Wizard page appears Figure 3 102 Start Software Upgrade Wizard Screen Start Software Upgrade Click the button to start the software upgrade pri Warning Before clicking the button Start Software Upgrade verify that no traffic is running on the device Even if you choose to cancel the process in the middle the device will reset itself and the previous configuration burned to flash will be reloaded 3 Click the Start Software Upgrade button the Load a CMP file Wizard page appears Figure 3 103 Load CMP File Wizard Page S http 10 33 4 161 Software Update Wizard Microsoft Intern CMP file Load a CMP file from your computer to the device INI file Browse Send File CPT tile VP file PRT file CAS file USRINF file FINISH Version 5 6 237 November 2008 A ge AudioCodes Mediant 2000 Note At this stage you can quit the Software Update Wizard by clicking Cancel x
90. You can also access previously opened pages by clicking your Web browser s Back button until you have reached the required page This is useful if you want to view pages in which you have performed configurations in the current Web session You can also access certain pages from the Device Actions button located on the toolbar refer to Toolbar on page 21 To view all the menus in the Navigation tree ensure that the Navigation tree is in Full view refer to Displaying Navigation Tree in Basic and Full View on page 24 To get Online Help for the currently opened page refer to Getting Help on page 45 Certain pages may not be accessible if your Web user account s access level is low refer to Configuring the Web User Accounts on page 99 Viewing Parameters For convenience some pages allow you to view a reduced or expanded display of parameters A reduced display allows you to easily identify required parameters enabling you to quickly configure your device The Web interface provides you with two methods for handling the display of page parameters m Display of basic and advanced parameters refer to Displaying Basic and Advanced Parameters on page 27 m Display of parameter groups refer to Showing Hiding Parameter Groups on page 28 Note Certain pages may only be read only if your Web user account s access level is low refer to Configuring the Web User Accounts on page 99 If a page is read onl
91. a SIP request is sent to the destination The device also supports NAT traversal for the SIP clients that are behind NAT In this case the device must be defined with a global IP address Note This field is available only if EnableSBC is set to 1 refer to SBC Configuration on page 163 Brief string description of the IP Group The value range is a string of up to 29 characters The default is an empty field Selects the Proxy Set ID defined in Proxy Sets Table on page 141 to associate with the IP Group All INVITE messages configured to be sent to the specific IP Group are in fact sent to the IP address associated with this Proxy Set The range is 0 5 where 0 is the default Proxy Set Note The Proxy Set is only defined for SERVER type IP Groups The request URI host name used in INVITE and REGISTER messages that are sent to this IP Group or the host name in the From header of INVITE messages received from this IP Group If not specified the value of the global parameter ProxyName refer to Proxy amp Registration Parameters on page 132 is used instead The value range is a string of up to 49 characters The default is an empty field Note If the IP Group is of type USER this parameter is used internally as a hostname in the request URI for TDM to IP initiated calls For example if an incoming call from the device s T1 trunk is routed to a USER type IP Group the device first forms the request URI destin
92. above NFASGroupNumber 0 1 NFASGroupNumber 1 1 NFASGroupNumber 2 1 NFASGroupNumber 3 1 DchConfig 0 0 Primary T1 trunk DeliConmal cele aes B Channel NFAS trunk DchConfig 2 2 B Channel NFAS trunk Dencontig oie B channel NFAS trunk 9 4 3 Creating an NFAS Related Trunk Configuration The procedures for creating and deleting an NFAS group must be performed in the correct order as described below gt 1 To create an NFAS Group take these 3 steps If there s a backup secondary trunk for this group it must be configured first Configure the primary trunk before configuring any NFAS slave trunk Configure NFAS slave trunks SIP User s Manual 400 Document LTRT 68808 SIP User s Manual 9 Advanced PSTN Configuration gt To stop delete an NFAS Group take these 3 steps 1 Stop or delete by setting ProtocolType to 0 i e None all NFAS slave trunks 2 Stop or delete by setting ProtocolType to 0 i e None the backup trunk if a backup trunk exists 3 Stop or delete by setting ProtocolType to 0 i e None the primary trunk All trunks in the group must be configured with the same values for trunk parameters TerminationSide ProtocolType FramingMethod and LineCode After stopping or deleting the backup trunk delete the group and then reconfigure it NFAS groups cannot be configured on the fly 9 5 Redirect Number and Calling Name
93. and then in the DSP Template Number field enter the desired DSP template number 4 Click Apply to save your settings 5 To save the changes to flash memory refer to Saving Configuration on page 230 If you delete all the table entries the device uses the default DSP template For a description of the Web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 30 Table 3 18 DSP Templates Parameters Parameter Description DSP Template Number Determines the number of the DSP template load Each load has DSPVersionTemplateNumber a different coder list channel capacity and supported features For the list of supported DSP template numbers coders and channel capacity refer to the device s Release Notes The default is 0 DSP Resources Percentage Resource percentage used for the specified template SIP User s Manual 80 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 2 7 Configuring Media Security The Media Security page allows you to configure media security gt Toconfigure media security take these 4 steps 1 Open the Media Security page Configuration tab gt Media Settings menu gt Media Security page item Figure 3 46 Media Security Page General Media Security Settings l Media Security Disable Media Security Behavior Preferable Disable Authentication On Transmitted RTP Packets 0 Disable Encr
94. another index entry you must ensure that you have applied the previously added index entry by clicking Apply If you leave the Add field blank and then click Add the existing index entries are all incremented by one and the newly added index entry is assigned the index 0 gt To add a copy of an existing index table entry take these 3 steps 1 In the Index column select the index that you want to duplicate the Edit button appears 2 Click Edit the fields in the corresponding index row become available 3 Click Duplicate a new index entry is added with identical settings as the selected index in Step 1 In addition all existing index entries are incremented by one and the newly added index entry is assigned the index 0 gt To edit an existing index table entry take these 3 steps 1 In the Index column select the index corresponding to the table row that you want to edit 2 Click Edit the fields in the corresponding index row become available 3 Modify the values as required and then click Apply the new settings are applied Version 5 6 31 November 2008 A EA AudioCodes Mediant 2000 gt To organize the index entries in ascending consecutive order take the following step Click Compact the index entries are organized in ascending consecutive order starting from index 0 For example if you added three index entries 0 4 and 6 then the index entry 4 is re assigned index number 1 and the i
95. areas m Title bar Displays the corporate logo and product name For replacing the logo with another image or text refer to Replacing the Corporate Logo on page 41 For customizing the product name refer to Customizing the Product Name on page 44 m Toolbar Provides frequently required command buttons for configuration refer to Toolbar on page 21 m Navigation Pane Consists of the following areas e Navigation bar Provides tabs for accessing the configuration menus refer to Navigation Tree on page 23 creating a Scenario refer to Scenarios on page 34 and searching ini file parameters that have corresponding Web interface parameters refer to Searching for Configuration Parameters on page 32 e Navigation tree Displays the elements pertaining to the tab selected on the Navigation bar tree like structure of the configuration menus Scenario Steps or Search engine Work pane Displays configuration pages where all configuration is performed refer to Working with Configuration Pages on page 25 Version 5 6 21 November 2008 A ge AudioCodes Mediant 2000 3 3 1 Toolbar The toolbar provides command buttons for quick and easy access to frequently required commands as described in the table below Table 3 1 Description of Toolbar Buttons Icon Button Description Name Y Submit Applies parameter settings to the device refer to Saving Configuration on page 230 Note This icon is grayed out wh
96. as configured by the parameter ProxyKeepAliveTime If set to Using REGISTER the SIP REGISTER message is sent every user defined interval as configured by the parameter RegistrationTime Any response from the Proxy either success 200 OK or failure 4xx response is considered as if the Proxy is communicating correctly Notes For Survivability mode for USER type IP Groups this parameter must be enabled 1 or 2 This parameter must be set to Using OPTIONS when Proxy redundancy is used When this parameter is set to Using REGISTER the homing redundancy mode is disabled When the active proxy doesn t respond to INVITE messages sent by the device the proxy is tagged as offline The behavior is similar to a Keep Alive OPTIONS or REGISTER failure Defines the Proxy keep alive time interval in seconds between Keep Alive messages This parameter is configured per Proxy Set The valid range is 5 to 2 000 000 The default value is 60 Note This parameter is applicable only if the parameter EnableProxyKeepAlive is set to 1 OPTIONS When the parameter EnableProxyKeepAlive is set to 2 REGISTER the time interval between Keep Alive messages is determined by the parameter RegistrationTime Enables the Proxy Hot Swap redundancy mode per Proxy Set 0 No Disabled default 1 Yes Proxy Hot Swap mode is enabled If Proxy Hot Swap is enabled the SIP INVITE REGISTER message is initially sent to the f
97. authentic When an organizational PKI is used two way authentication may be desired both client and server should be authenticated using X 509 certificates This is achieved by installing a client certificate on the managing PC and loading the same certificate in base64 encoded X 509 format to the device s Trusted Root Certificate Store The Trusted Root Certificate file should contain both the certificate of the authorized user and the certificate of the CA Since X 509 certificates have an expiration date and time the device must be configured to use NTP refer to Simple Network Time Protocol Support on page 383 to obtain the current date and time Without the correct date and time client certificates cannot work gt To enable two way client certificates take these 5 steps 1 Set the parameter Secured Web Connection HTTPS to HTTPS Only 0 in Configuring the General Security Settings on page 109 to ensure you have a method of accessing the device in case the client certificate doesn t work Restore the previous setting after testing the configuration 2 Open the Certificates Signing Request page refer to Server Certificate Replacement on page 105 3 In the Certificates Files group click the Browse button corresponding to Send Trusted Root Certificate Store file navigate to the file and then click Send File 4 When the operation is complete set the ini file parameter HTTPSRequireClientCert
98. authentication process to succeed Notes The pre shared key forms the basis of IPSec security and should therefore be handled cautiously in the same way as sensitive passwords It is not recommended to use the same pre shared key for several connections Since the ini file is in plain text format loading it to the device over a secure network connection is recommended preferably over a direct crossed cable connection from a management PC For added confidentiality use the encoded ini file option described in Secured Encoded ini File on page 255 After it is configured the value of the pre shared key cannot be obtained via Web interface ini file or SNMP refer the Product Reference Manual IKE SA LifeTime sec Determines the time in seconds the SA negotiated in the first IKEPolicyLifelnSec IKE session main mode is valid After the time expires the SA is re negotiated The default value is 28800 i e 8 hours IKE SA LifeTime KB Determines the lifetime in kilobytes that the SA negotiated in the IKEPolicyLifelnKB first IKE session main mode is valid After this size is reached the SA is re negotiated The default value is 0 i e this parameter is ignored These lifetime parameters IKE SA LifeTime sec and IKE SA LifeTime KB determine the duration the SA created in the main mode phase is valid When the lifetime of the SA expires it s automatically renewed by performing the IKE first phase ne
99. behind a NAT that initiates a signaling path has problems in receiving incoming signaling responses they are blocked by the NAT server Furthermore the initiating device must notify the receiving device where to send the media To resolve these issues the following mechanisms are available m STUN refer to STUN on page 381 m First Incoming Packet Mechanism refer to First Incoming Packet Mechanism on page 382 m RTP No Op packets according to the avt rtp noop draft refer to No Op Packets on page 382 For information on SNMP NAT traversal refer to the Product Reference Manual STUN Simple Traversal of UDP through NATs STUN based on RFC 3489 is a client server protocol that solves most of the NAT traversal problems The STUN server operates in the public Internet and the STUN clients are embedded in end devices located behind NAT STUN is used both for the signaling and the media streams STUN works with many existing NAT types and does not require any special behavior STUN enables the device to discover the presence and types of NATs and firewalls located between it and the public Internet It provides the device with the capability to determine the public IP address and port allocated to it by the NAT This information is later embedded in outgoing SIP SDP messages and enables remote SIP user agents to reach the device It also discovers the binding lifetime of the NAT the refresh rate necessary to keep NAT
100. bits per second Line Access Protocol for the D channel Loss of Frame Alignment Loss of Frame Megabit per second Million bits per second Management Information Base Multilevel Precedence and Preemption Millisecond a thousandth part of a second Media Server Control Markup Language Network Termination ISDN Message Waiting Indicator Naming Authority Pointer Network Address Translation Non Facility Associated Signalling ISDN PRI Network File System Numbering Plan Indicator Network Time Protocol Operations Administration Maintenance and Provisioning Open Systems Interconnection Industry Standard Private Branch Exchange Personal Computer Interface Industry Standard Pulse Code Modulation Progress Indicator Public Key Infrastructures Plain Old Telephone System or Service Prerecorded Tones File Primary Rate Interface ISDN Public Switched Telephone Network Port VLAN ID VLAN ID assignment to Ethernet packet by switch Quality of Service 416 Document LTRT 68808 SIP User s Manual Term RAI RFC RTCP RTP SA SAS SDP SIP SMDI SME SNMP SRTP SRV SSH SSL STUN T1 TCP TCP IP TE TDM TFTP TLS TON UA UDP URI SIP URIs VBD VLAN VoIP VoP VP VPN u Law Version 5 6 13 Glossary Meaning Remote Alarm Indication Request for Comment issued by IETF Real Time Transport RTP Control Protocol Real Time Transport Protocol Security Associations contains encryption keys and profi
101. by not selecting an ini file and by clearing the Use existing configuration check box Figure 3 105 Load an ini File Wizard Page F http 10 33 4 161 Software Update Wizard Microsoft Intern e O CMP file Load an imi ble from your computer to the dence IMI file Browse an Send File i H E Use existing configuration VP file r i ERE The Device will revert to default configuraton if no DRT tile configuration is chosen CAS file USRINF filo FINISH 7 You can now choose to either e Click Reset the device resets utilizing the new cmp and ini file you loaded up to now as well as utilizing the other auxiliary files e Click Back the Load a cmp file page is opened again e Click Next the next page opens for loading the next consecutive auxiliary file listed in the Wizard 8 Follow the same procedure as for loading the ini file Step 6 for loading the auxiliary files Version 5 6 239 November 2008 A C al AudioCodes Mediant 2000 9 In the FINISH page complete the upgrade process by clicking Reset the device burns the newly loaded files to flash memory and then resets t he device After the device resets the End Process screen appears displaying the burned configuration files refer to the figure below Figure 3 106 End Process Wizard Page http 10 33 4 161 EndOfProcess Microsoft Internet Explorer A CMP Version ID 5 404 000 Call Progress Tone File Na
102. call 3 Connect amp End Call CDR report is sent to Syslog at connection and at the end of each call 4 Start amp Connect amp End Call CDR report is sent to Syslog at the start at connection and at the end of each call The CDR Syslog message complies with RFC 3161 and is identified by Facility 17 local1 and Severity 6 Informational Syslog debug logging level 0 0 Debug is disabled default 1 1 Flow debugging is enabled 2 2 Flow and device interface debugging are enabled 8 3 Flow device interface and stack interface debugging are enabled 4 4 Flow device interface stack interface and session manager debugging are enabled 5 5 Flow device interface stack interface session manager and device interface expanded debugging are enabled Note Usually set to 5 if debug traces are needed 1 Not Configured for ISDN spans the progress indicator PI that is received in ISDN Proceeding Progress and Alert messages is used as described in the options below default 0 No PI For IP to Tel calls the device sends 180 Ringing SIP response to IP after receiving ISDN Alert or for CAS after placing a call to PBX PSTN 1 PI 1 8 Pl 8 For IP to Tel calls if EnableEarlyMedia 1 the device sends 180 Ringing with SDP in response to an ISDN Alert or 156 Document LTRT 68808 SIP User s Manual Parameter Enable X Channel Header XChan
103. cannot be reset from the Web interface i e via AdminPage or by loading an ini file This ini file table parameter configures the Welcome message that appears after a Web interface login The format of this parameter is as follows WelcomeMessage FORMAT WelcomeMessage_Index WelcomeMessage_Text WelcomeMessage 1 WelcomeMessage 2 WelcomeMessage 3 WelcomeMessage For Example WelcomeMessage FORMAT WelcomeMessage_Index WelcomeMessage_Text WelcomeMessage 1 WHEEKKKKKEKKEKRKEREREKRERERERERKEEEKEEEE WelcomeMessage 2 This is a Welcome message WelcomeMessage 3 nkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkk i WelcomeMessage Notes Each index represents a line of text in the Welcome message box Up to 20 indices can be defined lf this parameter is not configured no Welcome message is 274 Document LTRT 68808 SIP User s Manual Parameter DisableWebConfig HTTPport ScenarioFileName Telnet Parameters TelnetServerEnable TelnetServerPort TelnetServerldleDisconnect SSHServerEnable SSHServerPort 4 ini File Configuration Description displayed Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 Determines whether the entire Web interface is in read only mode 0 Enables modifications of parameters default 1 Web interface in read only mode When in read o
104. click the arrow button and then for each required option select 1 to enable The default is 0 i e disable 2 data calls with interworking indication use 64 kbps B channels physical only 8 REVERSE CHAN ALLOC ALGO Channel ID allocation algorithm 16 The device clears down the call if it receives a NOTIFY message specifying User Suspended A NOTIFY User Suspended message is used by some networks e g in Italy or Denmark to indicate that the remote user has cleared the call especially in the case of a long distance voice call 32 CHAN ID 16 ALLOWED Applies only to ETSI E1 lines 30B D Enables handling the differences between the newer QSIG standard ETS 300 172 and other ETSI based standards ETS 300 102 and ETS 300 403 in the conversion of B channel ID values into timeslot values 1 In regular ETSI standards the timeslot is identical to the B channel ID value and the range for both is 1 to 15 and 17 to 31 The D channel is identified as channel id 16 and carried into the timeslot 16 2 In newer QSIG standards the channel id range is 1 to 30 but the timeslot range is still 1 to 15 and 17 to 31 The D channel is 92 Document LTRT 68808 SIP User s Manual ini File Field Name Web Parameter Name 3 Web Based Management Valid Range and Description not identified as channel id 16 but is still carried into the timeslot 16 When this bit is set the channel ID 16 is consider
105. default Defines the User Part value of the Request URI for outgoing SIP OPTIONS requests If no value is configured the configuration parameter Username value is used 286 Document LTRT 68808 SIP User s Manual Parameter TDMOverlPMinCallsForTrunkAc tivation UseGatewayNameForOptions IsProxyHotSwap HotSwapRtx ProxyRedundancyMode ProxyLoadBalancingMethod ProxylIPListRefreshTime IsFallbackUsed UserName Password Cnonce SIPChallengeCachingMode MutualAuthenticationMode IsRegisterNeeded RegistrarIP RegistrarTransportType Version 5 6 4 ini File Configuration Description A special value is empty indicating that no User Part in the Request URI Host Part only is used The valid range is a 30 character string The default value is an empty string Defines the minimal number of SIP dialogs that must be established when using TDM Tunneling to consider the specific trunk as active When using TDM Tunneling if calls from this number of B Channels pertaining to a specific Trunk fail i e SIP dialogs are not properly set up an AIS alarm is sent on this trunk toward the PSTN and all current calls are dropped The originator gateway continues the INVITE attempts When this number of calls succeed i e SIP dialogs are set up properly the AIS alarm is cleared The valid range is 0 to 31 The default value is 0 i e don t send AIS alarms For a description of this
106. detected the device performs a self test If the self test succeeds the problem is logical link down i e Ethernet cable disconnected on the switch side and the Busy Out mechanism is activated if enabled EnableBusyOut 1 If the self test fails the device restarts to overcome internal fatal communication error Note Enable LAN Watchdog is relevant only if the Ethernet connection is full duplex Enables or disables usage of the User Information loaded to the device in the User Information auxiliary file For a description on User Information refer to Loading Auxiliary Files on page 231 0 Disable Disabled default 1 Enable Enabled Determines the index of the first Ringback Tone in the CPT file This option enables an Application server to request the device to play a distinctive Ringback tone to the calling party according to the destination of the call The tone is played according to the Alert Info header received in the 180 Ringing SIP response the value of the Alert Info header is added to the value of this parameter The valid range is 1 to 1 000 The default value is 1 i e play standard Ringback tone Notes It is assumed that all Ringback Tones are defined in sequence in the CPT file Incase of an MLPP call the device uses the value of this parameter plus 1 as the index of the Ringback tone in the CPT file e g if this value is set to 1 then the index is 2 i e 1 1 158 D
107. detection time is 100 msec m Cadence A repeating sequence of on and off sounds Up to four different sets of on off periods can be specified Burst A single sound followed by silence Only the First Signal On time and First Signal Off time should be specified All other on and off periods must be set to zero The burst tone is detected after the off time is completed You can specify several tones of the same type These additional tones are used only for tone detection Generation of a specific tone conforms to the first definition of the specific tone For example you can define an additional dial tone by appending the second dial tone s definition lines to the first tone definition in the ini file The device reports dial tone detection if either of the two tones is detected Version 5 6 335 November 2008 A EA AudioCodes Mediant 2000 The Call Progress Tones section of the ini file comprises the following segments m NUMBER OF CALL PROGRESS TONES Contains the following key Number of Call Progress Tones defining the number of Call Progress Tones that are defined in the file m CALL PROGRESS TONE X containing the Xth tone definition starting from 1 and not exceeding the number of Call Progress Tones defined in the first section using the following keys SIP User s Manual Tone Type Call Progress Tone types 1 Dial Tone 2 Ringback Tone 8 Busy Tone 7 Reorder Tone 8 Confirm
108. device to which packets are AccessList_Start_Port sent AccessList_End_Port The valid range is 0 to 65535 Protocol Note When the protocol type isn t TCP or UDP the entire range must be provided The protocol type e g UDP TCP ICMP ESP or Any or the IANA AccessList_Protocol protocol number in the range of O Any to 255 Note This field also accepts the abbreviated strings SIP and HTTP Specifying these strings implies selection of the TCP or UDP protocols and the appropriate port numbers as defined on the device SIP User s Manual 104 Document LTRT 68808 SIP User s Manual Parameter Packet Size AccessList_Packet_Size Byte Rate AccessList_Byte_Rate Burst Bytes AccessList_Byte_Burst Action Upon Match AccessList_Allow_Type Match Count AccessList_MatchCount 3 Web Based Management Description Maximum allowed packet size The valid range is 0 to 65535 Note When filtering fragmented IP packets this field relates to the overall re assembled packet size and not to the size of each fragment Expected traffic rate bytes per second Tolerance of traffic rate limit number of bytes Action upon match i e Allow or Block A read only field providing the number of packets accepted rejected by the specific rule 3 4 6 4 Configuring the Certificates The Certificates page is used for the following m Replacing the server certificate refer to S
109. device using the Web interface refer to Backing Up and Restoring Configuration that was initially loaded as encoded to the device the file is retrieved as encoded and vice versa Note The procedure for loading an encoded ini file is identical to the procedure for loading an unencoded ini file Version 5 6 255 November 2008 A ge AudioCodes Mediant 2000 4 2 4 2 1 4 2 2 The ini File Structure The ini file can contain any number of parameters The ini file can contain the following types of parameters m Individual parameters which are conveniently grouped optional by their functionality refer to Structure of Individual ini File Parameters on page 256 m Table parameters which include multiple individual parameters refer to Structure of ini File Table Parameters on page 257 Structure Rules The ini file must adhere to the following format rules m The ini file name must not include hyphens or spaces if necessary use an underscore _ instead m Lines beginning with a semi colon are ignored These can be used for adding remarks in the ini file A carriage return i e Enter must be done at the end of each line The number of spaces before and after the equals sign is irrelevant Subsection names for grouping parameters are optional If there is a syntax error in the parameter name the value is ignored Syntax errors in the parameter s value can cause unexpected errors parameters ma
110. device using the Web interface or a TFTP session refer to Auxiliary Files on page 231 Before you load them to the device you need to specify these files in the ini file and whether they must be stored in the non volatile memory The table below lists the ini file parameters associated with these auxiliary files Table 4 14 Auxiliary Configuration ini File Parameters Parameter CallProgressTonesFilename CASFileName CASFileName_x CASTablesNum VoicePromptsFileName PrerecordedTonesFileName CasTrunkDialPlanName Version 5 6 Description The name of the file containing the Call Progress Tones definitions Refer to the Product Reference Manual for additional information on how to create and load this file This is the name of the file containing specific CAS protocol definition such as E_M_WinkTable dat These files are provided to support various types of CAS signaling CAS file name e g E_M_WinkTable dat that defines the CAS protocol It is possible to define up to eight different CAS files by repeating this parameter Each CAS file can be associated with one or more of the device trunks using the parameter CASTablelndex_x Number 1 to 8 Specifies how many CAS configuration files are loaded The name and path of the file containing the Voice Prompts definitions Refer to the Product Reference Manual for additional information on how to create and load this file The name and path of the fil
111. dial tone is played to the B channel and then phone number collection starts Option 2 A Hook Flash is pressed the current call is put on hold a dial tone is played to the B channel and then digit collection starts After this sequence is identified the device continues the collection of the destination phone number For both options after the phone number is collected it s sent to the transferee in a SIP REFER request without a Replaces header The call is then terminated and a confirmation tone is played to the B channel If the phone number collection fails due to a mismatch a reorder tone is played to the B channel Note It is possible to configure whether the KeyBlindTransfer code is added as a prefix to the dialed destination number by using the parameter KeyBlindTransferAddPrefix Determines whether the device adds the Blind Transfer code KeyBlindTransfer to the dialed destination number 0 Disable default 1 Enable Determines the bit ordering of the G 726 G 727 voice payload format 0 Little Endian default 1 Big Endian Note To ensure high voice quality when using G 726 G 727 both communicating ends should use the same endianness format Therefore when the device communicates with a third party entity that uses the G 726 G 727 voice coder and voice quality is poor change the settings of this parameter between Big Endian and Little Endian For a description of this parameter
112. different E1 T1 trunk to provide the device s clock or enable TDM Bus PSTN Auto Clock in the TDM Bus Settings page refer to Configuring the TDM Bus Settings on page 218 To delete a previously configured trunk set the parameter Protocol Type to None Table 3 20 Trunk E1 T1 J1 Configuration Parameters ini File Field Name Web Parameter Nana Valid Range and Description General Settings Protocol Type Defines the PSTN protocol for the trunk ProtocolType 0 NONE 1 E1 EURO ISDN 2 T1 CAS 8 T1 RAW CAS 4 T1 TRANSPARENT 5 E1 TRANSPARENT 31 6 E1 TRANSPARENT 30 7 E1 MFCR2 8 E1 CAS 9 E1 RAW CAS 10 T1 NI2 ISDN 11 T1 4ESS ISDN 12 T1 5ESS 9 ISDN Version 5 6 85 November 2008 ca AudioCodes ini File Field Name Web Parameter Name Trunk Configuration Clock Master ClockMaster Auto Clock Trunk Priority AutoClockTrunkPriority Line Code LineCode Line Build Out Loss SIP User s Manual Mediant 2000 Valid Range and Description 13 T1 5ESS 10 ISDN 14 T1 DMS100 ISDN 15 J1 TRANSPARENT 16 T1 NTT ISDN Japan Nippon Telegraph 17 E1 AUSTEL ISDN Australian Telecom 18 T1 HKT ISDN Hong Kong HKT 19 E1 KOR ISDN Korean operator 20 T1 HKT ISDN Hong Kong HKT over T1 21 E1 QSIG 23 T1 QSIG 30 E1 FRENCH VN6 ISDN 31 E1 FRENCH VN3 ISDN 85 T1 DMS100 Meridian ISDN
113. digits is stopped The value must be an integer The default value is 1 In some cases when the state machine handles the ANI collection not related to MFCR2 you can control the state machine to collect ANI or discard ANI 0 No Don t collect ANI 1 Yes Collect ANI 1 Default Default value Defines which Signaling System to use in both directions detection generation 0 DTMF Uses DTMF signaling 1 MF Uses MF signaling default 1 Default Default value SIP User s Manual 98 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 4 SS7 Configuration The SS7 Configuration menu allows you to configure the Signaling System 7 SS7 protocol parameters For a detailed description of SS7 configuration refer to the Product Reference Manual 3 4 5 Sigtran Configuration The Sigtran Configuration menu allows you to configure the SIGTRAN parameters For a detailed description of SIGTRAN configuration refer to the Product Reference Manual 3 4 6 Security Settings The Security Settings menu allows you to configure various security settings This menu contains the following page items m Web User Accounts refer to Configuring the Web User Accounts on page 99 m Web amp Telnet Access List refer to Configuring the Web and Telnet Access List on page 102 Firewall Settings refer to Configuring the Firewall Settings on page 103 m Certificates refer to Conf
114. filter calls when using a Proxy default 1 Filter Filtering is enabled When this parameter is enabled and a Proxy is used the device first checks the Tel to IP Routing table or Outbound IP Routing table before making a call through the Proxy If the number is not allowed i e number isn t listed in the table or a call restriction routing rule of IP address 0 0 0 0 is applied the call is released Note When no Proxy is used this parameter must be disabled and filtering is according to the Tel to IP Routing table or Outbound IP Routing table The Digit Delivery feature enables sending DTMF digits to the destination IP address after the Tel to IP call is answered 0 Disable Disabled default 1 Enable Enable digit delivery to IP To enable this feature modify the called number to include at least one p character The device uses the digits before the p character in the initial INVITE message After the call is answered the device waits for the required time number of p multiplied by 1 5 seconds and then sends the rest of the DTMF digits using the method chosen in band or out of band Note The called number can include several p characters 1 5 seconds pause for example 1001pp699 8888p9p300 Enables the Digit Delivery feature which sends DTMF digits of the called number to the device s B channel phone line after the call is answered line offhooked FXS or seized
115. from the voice stream default V 21 Modem Transport Type used by the device 0 Disable Disable Transparent default 1 Enable Relay N A 2 Enable Bypass 3 Events Only Transparent with Events V 22 Modem Transport Type used by the device 0 Disable Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events V 23 Modem Transport Type used by the device 0 Disable Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events V 32 Modem Transport Type used by the device 0 Disable Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events Note This option applies to V 32 and V 32bis modems V 90 V 34 Modem Transport Type used by the device 0 Disable Disable Transparent 1 Enable Relay N A 2 Enable Bypass default 3 Events Only Transparent with Events 69 November 2008 ca AudioCodes Parameter Fax Relay Redundancy Depth FaxRelayRedundancyDepth Fax Relay Enhanced Redundancy Depth FaxRelayEnhancedRedundancyDepth Fax Relay ECM Enable FaxRelayECMEnable Fax Relay Max Rate bps FaxRelayMaxRate Fax Modem Bypass Coder Type FaxModemBypassCoderType Fax Modem Bypass Packing Factor FaxModemBypassM Fax Bypass Output Gain FaxBypassOutputGain
116. gt To apply the loaded certificate for IPsec negotiations take these 2 steps 1 Open the IKE Table page refer to Configuring the IKE Table on page 117 the Loaded Certificates Files group lists the newly uploaded certificates as shown below Figure 3 55 IKE Table Listing Loaded Certificate Files Loaded Certificate Files Server Certificate File Loaded Trusted Root File Loaded mE J Policy Index O State Exasts bd Authentication Method Pie chared F ey Shared Key ums IKE SA LifeTime sec 28800 IKE SA LifeTime KB 0 First Proposal Encryption Type Trige DES CBC First Proposal Authentication Type HMAC SHA 1 35 First Proposal DH Group DH 1024 B1T Second Proposal Encryption Type Not Defined Second Proposal Authentcation Type Not Defined Second Proposal OH Group Not Defined Third Proposal Encryption Type Not Defined Third Proposal Authentication Type Not Defined Third Proposal DH Group Not Defined Fourth Proposal Encryption Type Not Defined Fourth Proposal Authentication Type Not Defined SIX SEISMIC Cis Fourth Proposal OH Group Not Defined 2 Click the Apply button to load the certificates future IKE negotiations are now performed using the new certificates Version 5 6 107 November 2008 A EA AudioCodes Mediant 2000 3 4 6 4 2 Client Certificates By default Web servers using SSL provide one way authentication The client is certain that the information provided by the Web server is
117. indices The parameter NumberMaplp2Tel_IsPresentationRestricted is not applicable Set its value to RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add LeaveFromRight NumberType and NumberPlan are applied if the called and calling numbers match the DestinationPrefix SourcePrefix and SourceAddress conditions The manipulation rules are executed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and Suffix2Add Parameters can be skipped using two dollar signs The Source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all addresses between 10 8 8 10 and 10 8 8 99 The Source IP address can include the asterisk wildcard to represent any number between 0 and 255 For example 10 8 8 represents all the addresses between 10 8 8 0 and 10 8 8 255 To configure manipulation of destination numbers for IP to Tel calls using the Web interface refer to Configuring the Number Manipulation Tables on page 164 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 319 November 2008 ca AudioCodes Parameter SourceNumberMapTel2IP SourceNumberMapIP2Tel SIP User s Manual Mediant 2000 Description This ini file table parameter manipulates the source phone number for Tel to IP calls The format of this parameter is as follows So
118. interface Version 5 6 19 November 2008 A Ee AudioCodes Mediant 2000 3 2 Accessing the Web Interface The Web interface can be opened using any standard Web browser refer to Computer Requirements on page 19 When initially accessing the Web interface use the default user name Admin and password Admin For changing the login user name and password refer to Configuring the Web User Accounts on page 99 gt To access the Web interface take these 4 steps 1 Open a standard Web browser application 2 In the Web browser s Uniform Resource Locator URL field specify the device s IP address e g http 10 1 10 10 the Web interface s Enter Network Password dialog box appears as shown in the figure below Figure 3 1 Enter Network Password Screen Enter Network Password This secure Web Site at 10 33 4 128 requires you to log on Please type the User Name and Password that you use for Realm1 UserName Em F Password ais IV Save this password in your password list Cancel 3 In the User Name and Password fields enter the case sensitive user name and password 4 Click the OK button the Web interface is accessed displaying the Home page for a detailed description of the Home page refer to Using the Home Page on page 46 Note If access to the device s Web interface is denied Unauthorized due to Microsoft Internet Explorer security settings perform the following
119. interface with the traditional PSTN network enabling PSTN fallback in case of IP network failure In addition the device supports multiple SIP trunks whereby if a connection to one ITSP goes down the call can immediately be transferred to another ITSP By allowing multiple SIP trunks where each trunk is designated for a specific ITSP the device can route calls to an ITSP based on call destination e g country code Therefore in addition to providing VolP communication within an Enterpise s LAN the device allows the Enterprise to communicate outside of the corporate LAN using SIP trunking The device interfaces between the Enterprise s IP PBX and ITSP allowing SIP trunking implementation by the Enterprise for example in the following scenarios m VoIP between headquarters and remote offices m VoIP between Enterprise and PSTN via their ITSP For a detailed explanation on configuring IP to IP call routing refer to the document P to P SIP Call Routing Application Note 7 2 Answer Machine Detector AMD Answering Machine Detection can be useful in automatic dialing applications In some of these applications it is important to detect if a human voice or answering machine is answering the call Answering Machine Detection can be activated and de activated only after a channel is already open The direction of the detection PSTN or IP can be configured using the parameter AMDDetectionDirection refer to Media Server Parameters
120. interworking of ALERT messages from PRI to SIP 0 Disabled default 1 Enabled When enabled if the device already sent a 183 response with an SDP included and an ALERT message is received from the Tel side with or without Progress Indicator the device does not send an additional 18x response and the voice channel remains open When disabled the device sends additional 18x responses as a result of receiving an ALERT message whether or not a 18x response was already sent For a description of this parameter refer to Supplementary Services on page 159 For a description of this parameter refer to Supplementary Services on page 159 Modifies the called number for numbers received with Microsoft s proprietary ext xxx parameter in the SIP INVITE URI user part Microsoft Office Communications Server sometimes uses this proprietary parameter to indicate the extension number of the called party For example if a calling party makes a call to telephone number 622125519100 Ext 104 the device receives the SIP INVITE from Microsoft s application with the URI user part as INVITE sip 622125519100 ext 104 10 1 1 10 or INVITE tel 622125519100 ext 104 If the parameter EnableMicrosofExt is enabled the device modifies the called number by adding an e as the prefix removing the ext parameter and adding the extension number as the suffix e g 622125519100104 Once modified the device can then manipulate the n
121. is to the remote side without being converted QSIGSSIPSQSIG The advantage of tunneling over QSIG to SIP interworking is that by using interworking QSIG functionality can only be partially achieved When tunneling is used all QSIG capabilities are supported whereas the tunneling medium the SIP network does not need to process these messages QSIG messages are transferred in SIP messages in a separate Multipurpose Internet Mail Extensions MIME body Therefore if a message contains more than one body e g SDP and QSIG multipart MIME must be used The Content Type of the QSIG tunneled message is application QSIG In addition the device adds a Content Disposition header in the following format Content Disposition signal handling required Call setup originating device The QSIG SETUP request is encapsulated in the SIP INVITE message without being altered After the SIP INVITE request is sent the device doesn t encapsulate the subsequent QSIG message until a SIP 200 OK response is received If the originating device receives a 4xx 5xx or 6xx response it disconnects the QSIG call with a no route to destination cause Call setup terminating device After the terminating device receives a SIP INVITE request with a Content Type application QSIG it sends the encapsulated QSIG SETUP message to the Tel side and sends a 200 OK response no 1xx response is sent to IP The 200 OK response includes an
122. is 31 to 31 dB in 1 dB steps The default is 0 i e no gain Defines the modem bypass output gain control The range is 31 dB to 31 dB in 1 dB steps The default is 0 i e no gain 70 Document LTRT 68808 SIP User s Manual Parameter Fax CNG Mode FaxCNGMode CNG Detector Mode CNGDetectorMode T 38 Max Datagram Size T38MaxDatagram Version 5 6 3 Web Based Management Description Determines the device s behavior upon detection of a CNG tone 0 Does not send a SIP Re INVITE upon detection of a fax CNG tone when CNGDetectorMode is set to 1 default 1 Sends a SIP Re INVITE upon detection of a fax CNG tone when CNGDetectorMode is set to 1 Determines whether the device detects the fax Calling tone CNG 0 Disable The originating device doesn t detect CNG the CNG signal passes transparently to the remote side default 1 Relay CNG is detected on the originating side CNG packets are sent to the remote side according to T 38 if IsFaxUsed 1 and the fax session is started A Re INVITE message isn t sent and the fax session starts by the terminating device This option is useful for example when the originating device is located behind a firewall that blocks incoming T 38 packets on ports that have not yet received T 38 packets from the internal network i e originating device To also send a SIP Re INVITE message upon detection of a fax CNG tone in this mode
123. is 141 which is the width of AudioCodes displayed logo Note The optimal setting depends on the screen resolution settings 3 3 6 1 2 Replacing the Corporate Logo with Text The corporate logo can be replaced with a text string instead of an image To replace AudioCodes default logo with a text string using the ini file configure the ini file parameters listed in the table below For a description on using the ini file refer to Modifying an ini File on page 259 Table 3 3 ini File Parameters for Replacing Logo with Text Parameter Description UseWebLogo 0 Logo image is used default 1 Text string used instead of a logo image WebLogoText Text string that replaces the logo image The string can be up to 15 characters Note When a text string is used instead of a logo image the Web browser s title bar displays the string assigned to the WebLogoText parameter Version 5 6 43 November 2008 A Ee AudioCodes Mediant 2000 3 3 6 2 Customizing the Product Name You can customize the product name text that appears in the Title bar using the ini file parameters listed in the table below For a description on using the ini file refer to Modifying an ini File on page 259 Table 3 4 ini File Parameters for Customizing Product Name Parameter UseProductName UserProductName Description Defines whether or not to change the product name 0 Don t change the product name default
124. maximum TCP resources While trying to send a SIP message connection reuse policy determines whether alive connections to the specific destination are re used Persistent TCP connection ensures less network traffic due to fewer setting up and tearing down of TCP connections and reduced latency on subsequent requests due to avoidance of initial TCP handshake For TLS persistent connection may reduce the number of costly TLS handshakes to establish security associations in addition to the initial TCP connection set up For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 284 Document LTRT 68808 SIP User
125. mode as shown below Figure 3 21 Confirmation Message Box for Exiting Scenario Mode Microsoft Internet Explorer 2 J This operation will cancel scenario mode are you sure 2 Click OK to exit Customizing the Web Interface You can customize the device s Web interface to suit your company preferences The following Web interface elements can be customized m Corporate logo displayed on the Title bar refer to Replacing the Corporate Logo on page 41 E Product s name displayed on the Title bar refer to Customizing the Product Name on page 44 m Login welcome message refer to Creating a Login Welcome Message on page 44 Replacing the Corporate Logo The corporate logo that appears in the Title bar can be replaced either with a different logo image refer to Replacing the Corporate Logo with an Image on page 42 or text refer to Replacing the Corporate Logo with Text on page 43 The figure below shows an example of a customized Title bar The top image displays the Title bar with AudioCodes logo and product name The bottom image displays a customized Title bar with a different image logo and product name Figure 3 22 Customizing Web Logo and Product Name Device i Device acions v I5 Home Help Log off es ws Widget 2000 Device Actions v A Home Help S Log off x 1 tcusionjzaci Customized altager Product Name Version 5 6 41 November 2008 A Ee AudioCodes Mediant 2000
126. name of any configured SNMPV3 user to associate with this trap destination This determines the trap format authentication level and encryption level By default the trap is associated with the SNMP trap community string For a description of this parameter refer to Configuring the SNMP Managers Table on page 222 For a description of this parameter refer to Configuring the SNMP Managers Table on page 222 For a description of this parameter refer to Configuring the Management Settings on page 220 SNMP Community String Parameters SNMPReadOnlyCommunityS tring_x SNMPReadWriteCommunityS tring_x SNMPTrapCommunityString SNMP v3 Users Parameters SNMPUsers Version 5 6 For a description of this parameter refer to Configuring the SNMP Community Strings on page 224 For a description of this parameter refer to Configuring the SNMP Community Strings on page 224 For a description of this parameter refer to Configuring the SNMP Community Strings on page 224 This ini file table parameter configures SNMP v3 users The format of this parameter is as follows SNMPUsers FORMAT SNMPUsers_Index SNMPUsers_Username SNMPUsers_AuthProtocol SNMPUsers_PrivProtocol SNMPUsers_AuthKey SNMPUsers_PrivKey SNMPUsers_ Group SNMPUsers For example SNMPUsers FORMAT SNMPUsers_Index SNMPUsers_ Username SNMPUsers_AuthProtocol SNMPUsers_PrivProtocol SNMPUsers_AuthKey SNMPUsers_PrivKey SNMPUsers_ Gro
127. of an account perform the following a In the field Current Password enter the current password b In the fields New Password and Confirm New Password enter the new password maximum of 19 case sensitive characters c Click Change Password if you are currently logged into the Web interface with this account the Enter Network Password dialog box appears requesting you to enter the new password For security it s recommended that you change the default user name and password A Web user with access level Security Administrator can change all attributes of all the Web user accounts Web users with an access level other than Security Administrator can only change their own password and user name To reset the two Web user accounts user names and passwords to default set the ini file parameter ResetWebPassword to 1 To access the Web interface with a different account click the Log off button located on the toolbar click any button or page item and then re access the Web interface with a different user name and password You can set the entire Web interface to read only regardless of Web user account s access level by using the ini file parameter DisableWebConfig refer to Web and Telnet Parameters on page 273 Access to the Web interface can be disabled by setting the ini file parameter DisableWebTask to 1 By default access is enabled You can define additional Web user accounts using
128. only a difference in the UDP port the sending addresses won t change If both the IP address and UDP port need to be compared then both parameters need to be set to 1 No Op Packets The device s No Op packet support can be used to verify Real Time Transport Protocol RTP and T 38 connectivity and to keep NAT bindings and Firewall pinholes open The No Op packets are available for sending in RTP and T 38 formats You can control the activation of No Op packets by using the ini file parameter NoOpEnable If No Op packet transmission is activated you can control the time interval in which No Op packets are sent in the case of silence i e no RTP or T 38 traffic This is performed using the ini file parameter NoOpInterval For a description of the RTP No Op ini file parameters refer to Networking Parameters on page 260 m RTP No Op The RTP No Op support complies with IETF s draft wing avt rtp noop 03 txt titled A No Op Payload Format for RTP This IETF document defines a No Op payload format for RTP The draft defines the RTP payload type as dynamic You can control the payload type with which the No Op packets are sent This is performed using the RTPNoOpPayloadType ini parameter refer to Networking Parameters on page 260 AudioCodes default payload type is 120 T 38 No Op T 38 No Op packets are sent only while a T 38 session is activated Sent packets are a duplication of the previously sent frame including dupl
129. options click the arrow button and then for each required option select 1 to enable The default is 0 i e disable 32 DATA CONN RS Sends a CONNECT answer message on NOT incoming Tel calls 64 VOICE CONN RS device sends a CONNECT answer message on incoming Tel calls 2048 CHAN ID IN FIRST RS Sends Channel ID in the first response to an incoming Q 931 Call Setup message Otherwise the Channel ID is sent only if the device requires changing the proposed Channel ID default 8192 CHAN ID IN CALL PROC Sends Channel ID in a Q 931 Call Proceeding message 65536 PROGR IND IN SETUP ACK Includes Progress Indicator PI 8 in Setup ACK message if an empty called number is received in an incoming SETUP message Applicable to overlap dialing mode The parameter also directs the device to play a dial tone for TimeForDialTone until the next called number digits are received 262144 NI 2 second redirect number You can select and use in INVITE messages the NI 2 second redirect number if two redirect numbers are received in Q 931 Setup for incoming Tel to IP calls Note When using the ini file to configure the device to support several ISDNInCallsBehavior features add the individual feature values For example to support both 2048 and 65536 features set ISDNInCallsBehavior 67584 i e 2048 65536 Bit field used to determine several general CC behavior options To select the options
130. or Tel to IP Calls Count page item the figure below shows the IP to Tel Calls Count page Figure 3 118 Calls Count Page Number of Attempted Calls Number of Established Calls Percentage of Successful Calls ASR 73 684211 Number of Calls Terminated due to a Busy Line Number of Calls Terminated due to No Answer Number of Calls Terminated due to Forward Number of Failed Calls due to No Route Number of Failed Calls due to No Matched Capabilities Number of Failed Calls due to No Resources Number of Failed Calls due to Other Failures Average Call Duration ACD sec Attempted Fax Calls Counter Successful Fax Calls Counter uo O OIN O O O oO oO O N Table 3 64 Call Counters Description Counter Description Number of Indicates the number of attempted calls It is composed of established and Attempted Calls failed calls The number of established calls is represented by the Number of Established Calls counter The number of failed calls is represented by the failed call counters Only one of the established failed call counters is incremented every time Number of Indicates the number of established calls It is incremented as a result of one of Established Calls the following release reasons if the duration of the call is greater than zero GWAPP_REASON_NOT_RELEVANT 0 GWAPP_NORMAL_CALL_CLEAR 16 GWAPP_NORMAL_UNSPECIFIED 31 And the intern
131. or light blue background To navigate between Scenario Steps you can perform one of the following m In the Navigation tree click the required Scenario Step SIP User s Manual 36 Document LTRT 68808 SIP User s Manual 3 Web Based Management In an opened Scenario Step i e page appears in the Work pane use the following navigation buttons gt e Next opens the next Step listed in the Scenario 4 e Previous opens the previous Step listed in the Scenario Note If you reset the device while in Scenario mode after the device resets you are returned once again to the Scenario mode 3 3 5 3 Editing a Scenario You can modify a Scenario anytime by adding or removing Steps i e pages or parameters and changing the Scenario name and the Steps names Version 5 6 Note Only users with access level of Security Administrator can edit a Scenario To edit a Scenario take these 6 steps On the Navigation bar click the Scenarios tab a message box appears requesting you to confirm Scenario loading Click OK the Scenario appears with its Steps in the Navigation tree Click the Edit Scenario button located at the bottom of the Navigation pane the Scenario Name and Step Name fields appear You can perform the following edit operations e Add Steps a On the Navigation bar select the desired tab i e Configuration or Management the tab s menu appears in the Navigation tree b
132. outgoing packets are tagged each according to its interface Control Media or OAMP If the device s native VLAN ID is identical to one of the other IDs usually to the OAMP s VLAN ID this ID e g OAMP is set to zero on outgoing packets VianSendNonTaggedOnNative set to 0 This method is called Priority Tagging p tag without Q tag If the parameter VlanSendNonTaggedOnNative is set to 1 the device sends regular packets with no VLAN tag E Incoming packets from the switch to the device The switch sends all packets intended for the device according to the switch s configuration to the device without altering them For packets whose VLAN ID is identical to the switch s PVID the switch removes the tag and sends a packet The device accepts only packets that have a VLAN ID identical to one of its interfaces Control Media or OAMP Packets with a VLAN ID that is 0 or untagged packets are accepted only if the device s native VLAN ID is identical to the VLAN ID of one of its interfaces In this case the packets are sent to the relevant interface All other packets are rejected Version 5 6 385 November 2008 A EA AudioCodes Mediant 2000 Media traffic type is assigned Premium media CoS Management traffic type is assigned Bronze CoS and Control traffic type is assigned Premium control CoS For example RTP RTCP traffic is assigned the Media VLAN ID and Premium media CoS whereas Web traffic is assigned
133. own timing signals Use an internal clock E Recover a clock from one of the PSTN E1 T1 trunks gt To use the device s internal clock source configure the following parameters TDMBusClockSource 1 m ClockMaster 1 for all trunks gt To use the recovered clock option configure the following parameters TDMBusClockSource 4 m ClockMaster_x 0 for all slave trunks connected to PBX 1 m ClockMaster_x 1 for all master trunks connected to PBX 2 The above assumes that the device recovers its internal clock from one of the slave trunks connected to PBX 1 and provides clock to PBX 2 on its master trunks In addition it s necessary to define from which of the slave trunks the device recovers its clock m TDMBusPSTNAutoClockEnable 1 device automatically selects one of the connected slave trunks Or m TDMBusLocalReference trunk number where 0 is the first trunk and the default To configure the TDM Bus Clock Source parameters using the Web interface refer to Configuring the TDM Bus Settings on page 218 When the device is used in a non span configuration the internal device clock must be used as explained above Version 5 6 393 November 2008 A ge AudioCodes Mediant 2000 9 2 9 2 1 9 2 2 ISDN Release Description Reason 1 2 3 6 Release Reason Mapping This section describes the available mapping mechanisms of SIP responses to
134. parameter refer to Configuring the Voice Mail VM Parameters on page 214 302 Document LTRT 68808 SIP User s Manual Parameter DigitPatternInternalCall DigitPatternExternalCall TelDisconnectCode DigitPatternDigitTolgnore 4 ini File Configuration Description For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 4 4 10 PSTN Parameters The PSTN related ini file configuration parameters are described in the table below Parameter PCMLawSelect ProtocolType ProtocolType_x TraceLevel FramingMethod FramingMethod_x TerminationSide TerminationSide_x ClockMaster ClockMaster_x TDMBusClockSource TDMBusPSTNAutoClockE nable TDMBusLocalReference AutoClockTrunkPriority Version 5 6 Table 4 10 PSTN ini File Parameters Description For a description of this parameter refer to Configuring the TDM Bus Settings on page 218 For a description of this parameter refer to Configuring the Trunk Settings on page 82 Same as the description for parameter ProtocolType but for a specific trunk ID x 0 7 For a description of this
135. parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 287 Novemb
136. prefix For example if you enter 3 and the phone number is 5551234 the new phone number is 1234 Number of digits to remove from the right of the telephone number prefix For example if you enter 3 and the phone number is 5551234 the new phone number is 5551 The number or string that you want added to the front of the telephone number For example if you enter 9 and the phone number is 1234 the new number is 91234 The number or string that you want added to the end of the telephone number For example if you enter 00 and the phone number is 1234 the new number is 123400 The number of digits that you want to retain from the right of the phone number The Numbering Plan Indicator NPI assigned to this entry 0 Unknown default 9 Private 1 E 164 Public 1 Not Configured value received from PSTN IP is used Notes This parameter is applicable only to Number Manipulation tables for IP to Tel calls For a detailed list of the available NPI TON values refer to Numbering Plans and Type of Number on page 169 The Type of Number TON assigned to this entry f you selected Unknown for the NPI you can select Unknown 0 f you selected Private for the NPI you can select Unknown 0 Level 2 Regional 1 Level 1 Regional 2 PISN Specific 3 or Level 0 Regional Local 4 f you selected E 164 Public for the NPI you can select Unknown 0 International 1 National 2 N
137. range is 1 to OXFFFFFF The default value is 300 5 minutes 1 Never expires 0 Each request requires RADIUS authentication Defines the vendor ID that the device accepts when parsing a RADIUS response packet The valid range is 0 to OXFFFFFFFF The default value is 5003 Defines the code that indicates the access level attribute in the Vendor Specific Attributes VSA section of the received RADIUS packet The valid range is 0 to 255 The default value is 35 N A Enables disables the Internet Protocol security IPSec on the device 0 Disable IPSec is disabled default 1 Enable IPSec is enabled Enables the Dead Peer Detection DPD keep alive mechanism according to RFC 3706 to detect loss of peer connectivity 112 Document LTRT 68808 SIP User s Manual Parameter TLS Settings TLS version TLSVersion TLS Client Re Handshake Interval TLSReHandshakelnterval TLS Mutual Authentication SIPSRequireClientCertificate Peer Host Name Verification Mode PeerHostNameVerificationMode Version 5 6 3 Web Based Management Description 0 Disabled default 1 Periodic message exchanges at regular intervals 2 On Demand message exchanges as needed i e before sending data to the peer If the liveliness of the peer is questionable the device sends a DPD message to query the status of the peer If the device has no traffic to send it never sends a D
138. refer to Saving Configuration on page 230 Table 3 7 Network Settings IP Settings Parameters Parameter Description IP Settings IP Networking Mode Determines the IP network scheme EnableMultiplelPs 0 Single IP Network Single IP network default 1 Multiple IP Networks Multiple IP networks OAMP Media and Control 1 Dual IP Media amp Control Multiple IP networks 1 Dual IP OAM amp Control Multiple IP networks 1 Dual IP OAM amp Medial Multiple IP networks Note This parameter is not relevant when using Multiple Interface tables activated by clicking the Multiple Interface Table button described below refer to Configuring the Multiple Interface Table on Version 5 6 51 November 2008 A ge AudioCodes Mediant 2000 Parameter Single IP Settings IP Address Subnet Mask Default Gateway Address Description page 53 For detailed information on Multiple IPs refer to Multiple IPs on page 384 IP address of the device Enter the IP address in dotted decimal notation for example 10 8 201 1 Notes A warning message is displayed after clicking Submit if the entered value is incorrect After changing the IP address you must reset the device Subnet mask of the device Enter the subnet mask in dotted decimal notation for example 255 255 0 0 Notes A warning message is displayed after clicking Submit if the entered value is incorrect
139. relay 2 G 711 Transport Initiates fax modem using the coder G 711 A law u law with adaptations refer to Note below 3 Fax Fallback Initiates T 38 fax relay If the T 38 negotiation fails the device re initiates a fax session using the coder G 711 A law p law with adaptations refer to the Note below Notes Fax adaptations for options 2 and 3 Echo Canceller On Silence Compression Off Echo Canceller Non Linear Processor Mode Off Dynamic Jitter Buffer Minimum Delay 40 Dynamic Jitter Buffer Optimization Factor 13 If the device initiates a fax session using G 711 option 2 and possibly 3 a gpmd attribute is added to the SDP in the following format For A law a gpmd 8 vbd yes ecan on For u law a gomd 0 vbd yes ecan on When IsFaxUsed is set to 1 2 or 3 the parameter FaxTransportMode is ignored When the value of IsFaxUsed is other than 1 T 38 might still be used without the control protocol s involvement To completely disable T 38 set FaxTransportMode to a value other than 1 For detailed information on fax transport methods refer to Fax Modem Transport Modes on page 351 Determines when the device initiates a T 38 session for fax transmission 0 Initiate T 38 on Preamble The device to which the called fax is connected initiates a T 38 session on receiving Preamble signal from the fax default 1 Initiate T 38 on CED The device to which the called fax is connected initi
140. requests to the proxy Normal Mode this port serves as the source port The valid range is 1 to 65 534 The default value is 5080 SAS Default Gateway IP The default gateway used in SAS Emergency Mode When an SASDefaultGatewayIP incoming SIP INVITE is received and the destination Address Of Record is not included in the SAS database the request is immediately sent to this default gateway The address can be configured as an IP address dotted decimal notation or as a domain name up to 49 characters The default is a null string which is interpreted as the local IP address of the gateway SIP User s Manual 162 Document LTRT 68808 SIP User s Manual Parameter SAS Registration Time SASRegisirationTime Short Number Length SASShortNumberLength SAS Local SIP TCP Port SASLocalSIPTCPPort SAS Local SIP TLS Port SASLocalSIPTLSPort SAS Proxy Set SASProxySet Redundant SAS Proxy Set RedundantSASProxySet 3 Web Based Management Description Determines the value of the SIP Expires header that is sent in a 200 OK response to an incoming REGISTER message when in SAS Emergency Mode The valid range is 10 to 2 000 000 The default value is 20 This parameter is obsolete instead use the parameter SASRegistrationManipulation Local TCP port used to send receive SIP messages for the SAS application The SIP entities in the local network need to send the registration requests to this port W
141. s Manual Parameter AlwaysSendToProxy PreferRouteTable SIPReroutingMode EnableProxyKeepAlive ProxyKeepAliveTime DNSQueryType ProxyDNSQueryType ProxylP ProxySet Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 This ini file table parameter configures the Proxy Set ID table for configuring up to six Proxy Sets each with up to five Proxy server IP addresses The format of this parameter is as follows ProxyIP FORMAT Proxylp_Index Proxylp_IpAddress Proxylp_TransportType Proxylp_ProxySetld ProxyIP For example ProxyIP FORMAT Proxylp_Index Proxylp_lpAddress Proxylp_TransportType Proxylp_ProxySetld Proxylp 0 10 33 37 77 1 0 Proxylp 1 10 8 8 10 0 2 Proxylp 2 10 5 6 7 1 1 ProxyIP Notes
142. snmpTargetAddrTable in the snmpTargetMIB For example mngr corp mycompany com The valid range is a 99 character string Activity Types to Report via Activity Log Messages The Activity Log mechanism enables the device to send log messages to a Syslog server for reporting on certain types of Web operations according to the below user defined filters Parameters Value Change ActivityListToLog PVC Auxiliary Files Loading ActivityListToLog AFL Device Reset ActivityListToLog DR Flash Memory Burning ActivityListToLog FB Device Software Update ActivityListToLog SWU Access to Restricted Domains ActivityListToLog ARD Non Authorized Access ActivityListToLog NAA Sensitive Parameters Value Change ActivityListToLog SPC SIP User s Manual Changes made on the fly to parameters Loading of auxiliary files e g via Certificate page Reset of device via the Maintenance Actions page Burning of files parameters to flash e g Maintenance Actions page cmp loading via the Software Upgrade Wizard Access to Restricted Domains which includes the following pages jini parameters AdminPage General Security Settings Configuration File IPSec IKE tables Software Upgrade Key Internal Firewall Web Access List Web User Accounts Attempt to access the Web interface with a false empty user name or password Changes made to sensitive par
143. that the device waits for a CONNECT message from the called party Tel side after sending a SETUP message If the timer expires the call is answered by sending a 200 OK message IP side The valid range is 0 to 600 The default value is 0 i e disabled 313 November 2008 A EA AudioCodes Mediant 2000 4 4 12 Number Manipulation and Routing Parameters The number manipulation and routing related ini file configuration parameters are described in the table below Table 4 12 Number Manipulation and Routing ini File Parameters Parameter Description TrunkGroup This ini file table parameter defines the device s Trunks and assigns them to Trunk Groups The format of this parameter is shown below TrunkGroup FORMAT TrunkGroup_Index TrunkGroup_TrunkGroupNum TrunkGroup_FirstTrunkld TrunkGroup_LastTrunkld TrunkGroup_FirstBChannel TrunkGroup_LastBChannel TrunkGroup_FirstPhoneNumber TrunkGroup_Profileld TrunkGroup_Module TrunkGroup For example TrunkGroup TrunkGroup 1 0 0 0 1 31 401 0 E1 span TrunkGroup 1 0 0 0 1 31 1 TrunkGroup 2 1 2 2 1 24 3000 T1 span TrunkGroup 1 2 0 7 1 20 1000 8 E1 spans 20 B channels TrunkGroup 1 0 0 3 1000 4 E1 spans all B channels TrunkGroup Notes The parameter TrunkGroup_Module is not applicable To represent all B channels use an asterisk For configuring this table in the Web interface refer to Configur
144. the calculated amounts of delay or packet loss exceed these thresholds the IP connection is disallowed m DNS resolution When host name is used instead of IP address for the destination route it is resolved to an IP address by a DNS server Connectivity and QoS are then applied to the resolved IP address 7 9 3 PSTN Fallback as a Special Case of Alternative Routing The PSTN Fallback feature enables the device to redirect PSTN originated calls back to the legacy PSTN network if a destination IP route is unsuitable disallowed for voice traffic at a specific time To enable PSTN fallback assign the device s IP address as an alternative route to the desired prefixes Note that calls now referred to as IP to Tel calls can be re routed to a specific trunk group using the Routing parameters refer to IP to Trunk Group Routing on page 181 7 9 4 Relevant Parameters The following parameters described in Routing General Parameters on page 171 are used to configure the Alternative Routing mechanism E AltRoutingTel2IPEnable m AltRoutingTel2IPMode m IPConnQoSMaxAllowedPL m IPConnQoSMaxAllowedDelay SIP User s Manual 362 Document LTRT 68808 SIP User s Manual 7 10 Supported RADIUS Attributes Use the following table for explanations on the RADIUS attributes contained in the communication packets transmitted between the device and a RADIUS Server Attribute Attribute Number Name Request Attributes 1 User Na
145. the Tel to IP Routing page appears instead refer to Tel to IP Routing Table on page 175 for a description of this page This table allows you to configure the device s routing rules for sending inbound IP calls matching some or all of the following criteria to a destination IP address or IP Group Source IP Group Source host prefix Destination host prefix Trunk Group Destination telephone prefix Source telephone prefix gt To configure Outbound IP Routing take these 5 steps 1 Open the Outbound IP Routing Table page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Tel to IP Routing page item Figure 3 72 Outbound IP Routing Page U ee J c Matching Rules Destination Rules 2 From the Routing Index drop down list select the range of entries that you want to add 3 Configure the Outbound IP Routing table according to the table below 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 45 Outbound IP Routing Table Description Parameter Description Tel to IP Routing Mode Determines whether to route the inbound IP calls to the IP RouteModeTel2IP destination before or after manipulation of destination number 0 Route calls before manipulation IP to IP calls are routed before the number manipulation rules are applied default 1 Route calls after manipula
146. the device To avoid this disable all new traffic before saving by performing a graceful lock refer to Locking and Unlocking the Device on page 229 Throughout the Web interface parameters preceded by the lightning symbol are not applied on the fly to the device and require that you reset the device refer to Resetting the Device on page 228 for them to take effect 3 5 2 Software Update The Software Update menu allows you to upgrade the device s software by loading a new cmp file compressed firmware along with the ini file and a suite of auxiliary files or to update existing auxiliary files The Software Update menu includes the following page items E Load Auxiliary Files refer to Loading Auxiliary Files on page 231 Software Update Key refer to Upgrading the Software Upgrade Key on page 233 m E Software Upgrade Wizard refer to Software Upgrade Wizard on page 236 E Configuration File refer to Backing Up and Restoring Configuration on page 240 3 5 2 1 Loading Auxiliary Files The Load Auxiliary Files page allows you to load to the device various auxiliary files described in the table below For detailed information on these files refer to Auxiliary Configuration Files on page 335 For information on deleting these files from the device refer to Device Information on page 244 File Type ini CAS Voice Prompts Dial Plan Call Progress Tones Version 5 6 Table 3 61
147. the PSTN The device with ISDN protocol type operates according to the parameter LocallSDNRBSource 1 If LocallSDNRBSource 1 the device plays an RBT and sends an ISDN Alert with PI 8 to the ISDN unless the parameter ProgressIndicator2ISDN_ID is configured differently 2 If LocallSDNRBSource 0 the device doesn t play an RBT No Pl is sent in the ISDN Alert message unless the parameter ProgressIndicator2ISDN_ID is configured differently In this case the PBX PSTN should play an RBT tone to the originating terminal by itself Note Receipt of a 183 response results in an ISDN Progress message unless SIP183Behaviour 1 If SIP183Behaviour 1 183 is handled the same way as a 180 SDP the device sends an Alert message with PI 8 without playing an RBT Determines the ISDN B Channel negotiation mode 1 Not Configured use per device configuration of BChannelNegotiation parameter default 0 Preferred Preferred 1 Exclusive Exclusive 2 Any Any Notes Applicable to ISDN protocols The option Any is only applicable if TerminationSide is set to 0 i e User side The Din the ini file parameter name represents the trunk number where 0 is the first trunk Enables the device to start sending and or receiving RTP packets to and from remote endpoints without the need to establish a Control session The remote IP address is determined according to the Tel to IP Routing tab
148. the following adaptations E Echo Canceller on Silence Compression off Echo Canceller Non Linear Processor Mode off Dynamic Jitter Buffer Minimum Delay 40 Dynamic Jitter Buffer Optimization Factor 13 When the device initiates a fax session using G 711 a gpmd attribute is added to the SDP according to the following format E For G 711A law a gpmd 0 vbd yes ecan on E For G 711 p law a gpmd 8 vbd yes ecan on In this mode the parameter FaxTransportMode is ignored and automatically set to transparent To configure fax fallback mode set IsFaxUsed to 3 Supporting V 34 Faxes Unlike T 30 fax machines V 34 fax machines have no relay standard to transmit data over IP to the remote side Therefore the device provides the following operation modes for transporting V 34 fax data over the IP m Using bypass mechanism for V 34 fax transmission refer to Using Bypass Mechanism for V 34 Fax Transmission on page 356 m Using relay mode i e fallback to T 38 refer to Using Relay mode for both T 30 and V 34 faxes on page 357 Using the ini file parameter V34FaxTransportType you can determine whether to pass V 34 Fax over T 38 fallback to T 30 or use Bypass over the High Bit Rate coder e g PCM A Law Note The CNG detector is disabled CNGDetectorMode 0 in all the subsequent examples Using Bypass Mechanism for V 34 Fax Transmission In this proprietary scenario the device uses bypass or N
149. the time after which the device resets Note that if no traffic exists and the time has not yet expired the device resets Click the Reset button a confirmation message box appears requesting you to confirm Figure 3 97 Reset Confirmation Message Box r Microsoft Internet Explorer 2 re you sure you want to RESET the Gateway Click OK to confirm device reset if the parameter Graceful Option is set to Yes in Step 3 the reset is delayed and a screen displaying the number of remaining calls and time is displayed When the device begins to reset a message appears notifying you of this Throughout the Web interface parameters preceded by the lightning symbol are not applied on the fly to the device and require that you reset the device for them to take effect If you modify parameters that only take effect after a device reset after you click the Submit button the toolbar displays the word Reset refer to Toolbar on page 21 to remind you to later reset the device 3 5 1 3 2 Locking and Unlocking the Device The Lock and Unlock options allow you to lock the device so that it doesn t accept any new incoming calls This is useful when for example you are uploading new software files to the device and you don t want any traffic to interfere with the process gt 1 Version 5 6 To lock the device take these 5 steps Open the Maintenance Actions page refer to Maintenance Actions on page 228
150. to Configuring the RTP RTCP Settings on page 71 or to specific IP destinations using the IP Profile feature refer to IP Profile Settings on page 193 To enable RTP Multiplexing set the parameter RemoteBaseUDPPort to a nonzero value Note that the value of RemoteBaseUDPPort on the local device must equal the value of BaseUDPPort of the remote device The device uses these parameters to identify and distribute the payloads from the received multiplexed IP packet to the relevant channels In RTP Multiplexing mode the device uses a single UDP port for all incoming multiplexed packets and a different port for outgoing packets These ports are configured using the parameters L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort When RTP Multiplexing is used call statistics aren t available since there is no RTCP flow RTP Multiplexing must be enabled on both devices When VLANs are imlemented the RTP Multiplexing mechanism is not supported 7 8 Dynamic Jitter Buffer Operation Voice frames are transmitted at a fixed rate If the frames arrive at the other end at the same rate voice quality is perceived as good In many cases however some frames can arrive slightly faster or slower than the other frames This is called jitter delay variation and degrades the perceived voice quality To minimize this problem the device uses a jitter buffer The jitter buffer collects voice packets stores them and sends them to the voice proce
151. updated If the device is configured to obtain the date and time from an SNTP server refer to Configuring the Application Settings on page 57 the fields on this page are read only and cannot be modified For an explanation on SNTP refer to Simple Network Time Protocol Support on page 383 After performing a hardware reset the date and time are returned to their defaults and therefore should be updated Version 5 6 227 November 2008 A C al AudioCodes Mediant 2000 3 5 1 3 Maintenance Actions The Maintenance Actions page allows you to perform the following operations Reset the device refer to Resetting the Device on page 228 Lock and unlock the device refer to Locking and Unlocking the Device on page 229 Save the configuration to the device s flash memory refer to Saving Configuration on page 230 To access the Maintenance Actions page take this step On the Navigation bar click the Management tab and then in the Navigation tree select the Management Configuration menu and then choose the Maintenance Actions page item Figure 3 96 Maintenance Actions Page v Reset Configuration Reset Board Burn To FLASH Yes v Graceful Option No w LOCK UNLOCK Lock LOCK Graceful Option No v Current Admin State UNLOCKED wv Save Configuration Burn To FLASH 3 5 1 3 1 Resetting the Device The Maintenance Actions page allows you to remotely reset th
152. used in Remote Party ID RPID header by using the EnableRPiHeader and AddTON2RPI parameters To configure manipulation of source numbers for Tel to IP calls using the Web interface refer to Configuring the Number Manipulation Tables on page 164 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 This ini file table parameter manipulates the source number for IP to Tel calls The format of this parameter is as follows SourceNumberMaplp2Tel 320 Document LTRT 68808 SIP User s Manual 4 ini File Configuration Parameter Description FORMAT SourceNumberMaplp2Tel_ Index SourceNumberMaplp2Tel_ DestinationPrefix SourceNumberMaplp2Tel_SourcePrefix SourceNumberMaplp2Tel_ SourceAddress SourceNumberMaplp2Tel_NumberType SourceNumberMaplp2Tel_ NumberPlan SourceNumberMaplp2Tel_ RemoveFromLeft SourceNumberMaplp2Tel_RemoveFromRight SourceNumberMaplp2Tel_LeaveFromRight SourceNumberMaplp2Tel_ Prefix2Add SourceNumberMaplp2Tel_ Suffix2Add SourceNumberMaplp2Tel_IsPresentationRestricted SourceNumberMaplp2Tel For example SourceNumberMaplp2Tel SourceNumberMaplp2Tel 0 22 03 2 667 SourceNumberMaplp2Tel 1 034 01 1 1 1 1 0 2 972 10 SourceNumberMaplp2Tel Notes RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add LeaveFromRight NumberType and NumberPlan are applied if the called and calling numbers matc
153. used to define various routing rules To use this feature you must configure the Trunk Group IDs refer to Configuring the Trunk Group Table on page 195 Determines whether the Trunk ID is added as a prefix to the called number for Tel to IP calls 0 No Don t add Trunk ID as prefix default 1 Yes Enable add Trunk ID as prefix If enabled the Trunk ID single digit in the range 1 to 8 is added as a prefix to the called destination phone number This option can be used to define various routing rules Determines whether the internal channel number is used as the destination number if the called number is missing 0 No default 1 Yes Note Applicable only for Tel to IP calls and if the called number is missing Determines whether Numbering Plan Indicator NPI and Type of Numbering TON are added to the Calling Number 172 Document LTRT 68808 SIP User s Manual Parameter AddNPlandTON2CallingNumber Add NPI and TON to Called Number AddNPlandTON2CalledNumber IP to Tel Remove Routing Table Prefix RemovePrefix Source IP Address Input SourcelPAddressInput Version 5 6 3 Web Based Management Description for Tel to IP calls 0 No Do not change the Calling Number default 1 Yes Add NPI and TON to the Calling Number ISDN Tel to IP call For example After receiving a Calling Number of 555 NPI of 1 and TON of 3 the modified number becomes 13555 Th
154. used to depict any source host prefix The called telephone number prefix The prefix can be 1 to 49 digits long Note For notations representing multiple numbers refer to Dialing Plan Notation on page 168 The calling telephone number prefix The prefix can be 1 to 49 digits long Note For notations representing multiple numbers refer to Dialing Plan Notation on page 168 The source IP address of an IP to IP call obtained from the Contact header in the SIP INVITE message Notes The source IP address can include the letter x wildcard to represent single digits For example 10 8 8 xx represents all the addresses between 10 8 8 10 to 10 8 8 99 The source IP address can include the asterisk wildcard to represent any number between 0 and 255 For example 10 8 8 represents all addresses between 10 8 8 0 and 10 8 8 255 Inbound SIP IP calls matching all or any combination of the above routing rules are subsequently assigned to the IP Group selected below Trunk Group ID PstnPrefix_TrunkGroupld IP Profile ID PstnPrefix_Profileld Version 5 6 Identifies these calls as IP to IP calls when set to 1 IP profile configured in IP Profile Settings on page 193 assigned to the inbound IP to IP call 185 November 2008 A ge AudioCodes Mediant 2000 Parameter Description Source IP Group ID The IP Group 1 9 to which you want to assign this inbound IP to IP PstnPrefix_SrclPGroupID call Th
155. v Digit Patterns Forward on Busy Digit Pattern Internal Forward on No Answer Digit Pattern Internal Forward on Do Not Disturb Digit Pattern Internal Forward on No Reason Digit Pattern Internal Forward on Busy Digit Pattern External Forward on No Answer Digit Pattern External Forward on Do Not Disturb Digit Pattern External Forward on No Reason Digit Pattern External Internal Call Digit Pattern External Call Digit Pattern Disconnect Call Digit Pattern Digit To Ignore Digit Pattern Message Waiting Indication MWI MWI Off Digit Pattern MWI On Digit Pattern MWI Suffix Pattern MWI Source Number v SMDI Enable SMDI SMDI Timeout msec 2000 SIP User s Manual 214 Document LTRT 68808 SIP User s Manual 3 Web Based Management 2 Configure the voice mail parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 54 Voice Mail Parameters Parameter General Voice Mail Interface VoiceMaillnterface Digit Patterns Description Enables the voice mail application on the device and determines the communication method used between the PBX and the device 0 None default 1 DTMF 2 SMDI 8 QS
156. whether the device copies the received ISDN called number to the outgoing SIP Diversion header for Tel to IP calls even if a Redirecting Number IE is not received in the ISDN Setup message Therefore the called number is used as a redirect number Call redirection information is typically used for Unified Messaging and voice mail services to identify the recipient of a message 0 Don t copy Disable default 1 Copy after phone number manipulation Copies the called number after manipulation The device first performs Tel to IP destination phone number manipulation i e on the SIP To header and only then copies the manipulated called number to the SIP Diversion header for the Tel to IP call Therefore with this option the called and redirected numbers are identical 2 Copy before phone number manipulation Copies the called number before manipulation The device first copies the original called number to the SIP Diversion header and then 211 November 2008 ca AudioCodes Parameter Enable Calling Party Category EnableCallingPartyCategory Digital Out Of Service Behavior DigitalOOSBehavior SIP User s Manual Mediant 2000 Description performs Tel to IP destination phone number manipulation Therefore this allows you to have different numbers for the called i e SIP To header and redirected i e SIP Diversion header numbers Notes f the incoming ISDN to IP call includes a Redirect Nu
157. which sends the call directly to the PSTN This is important for routing emergency numbers such as 911 in North America directly to the PSTN This is applicable to SAS operating in Normal and Emergency modes 297 November 2008 ca AudioCodes Parameter Profile Parameters CoderName SIP User s Manual Mediant 2000 Description Up to four emergency numbers can be defined where each number can be up to four digits This ini file table parameter defines the device s coder list This includes up to five groups of coders consisting of up to five coders per group that can be associated with IP or Tel profiles Coder Group Settings page in the Web interface refer to Coder Group Settings on page 190 The first group of coders indices 0 through 4 is the default coder list and default coder group The format of this parameter is as follows CoderName FORMAT CoderName_Index CoderName_Type CoderName_Packetinterval CoderName_ rate CoderName_PayloadType CoderName_Sce CoderName Where Type Coder name Packetlnterval Packetization time Rate Packetization rate PayloadType Payload type Sce Silence suppression mode For example CoderName CoderName 0 g711Alaw64k 20 0 CoderName 1 g726 3 38 0 CoderName 2 g729 40 255 255 1 CoderName Notes This parameter can include up to 25 indices i e five coders per five coder groups The coder name is ca
158. zero take the following step Click the Reset Statistics button Version 5 6 245 November 2008 A gA AudioCodes Mediant 2000 3 6 1 6 Viewing Active Alarms The Active Alarms page displays a list of currently active alarms For each alarm the following information is provided m Severity severity level of the alarm e Critical alarm displayed in red e Major alarm displayed in orange e Minor alarm displayed in yellow Source unit from which the alarm was raised Description brief explanation of the alarm Date date and time that the alarm was generated You can also access this page from the Home page refer to Using the Home Page on page 46 gt To view the list of alarms take this step E Open the Active Alarms page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Active Alarms page item Figure 3 114 Active Alarms Page Severity 3 6 1 7 Viewing Trunks amp Channels Status The Trunks amp Channels Status page displays the status of the device s Trunks and the channels pertaining to these trunks gt To view the status of the device s trunks and the trunks channels take the following step E Open the Trunks amp Channels Status page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Trunks amp Channels Status page item Figure 3 115 Trunks amp Channels Status Page Trunks Channels Status OF 1 23 4 5 6 7 8 9 1011121
159. 0 2 In the SNMP Community String field click the right pointing arrow 4 button the SNMP Community String page appears Figure 3 92 SNMP Community Strings Page Delete Community String Access Level Read Only Read Only Read Only Read Only Read Only Read Write Read Write Read Write Read Write Read Write Trap Community String trapuser 3 Configure the SNMP community strings parameters according to the table below 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 230 Note To delete a community string select the Delete check box corresponding to the community string that you want to delete and then click Submit SIP User s Manual 224 Document LTRT 68808 SIP User s Manual 3 Web Based Management Table 3 59 SNMP Community Strings Parameters Description Parameter Description Community String Read Only SNMPReadOnlyCommunityString_x Up to five read only community strings up to 19 characters each The default string is public Read Write SNMPReadWriteCommunityString_x Up to five read write community strings up to 19 characters each The default string is private Trap Community String Community string used in traps up to 19 characters SNMPTrapCommunityString The defaul
160. 0 0 Note Do not include read only parameters in the ini file table parameter as this can cause an error when trying to load the file to the device SIP User s Manual 258 Document LTRT 68808 SIP User s Manual 4 ini File Configuration 4 2 4 Example of an ini File Below is an example of an ini file for the VoIP device Channel Params DJBufMinDelay 75 RTPRedundancyDepth 1 IsProxyUsed 1 Broxy T PET ILA 6 1VA2 7s CoderName FORMAT CoderName Index CoderName Type CoderName PacketInterval CoderName rate CoderName PayloadType CoderName Sce CoderName 1 g7231 90 CoderName List of serial B channel numbers TrunkGroup FORMAT TrunkGroup Index TrunkGroup TrunkGroupNum TrunkGroup FirstTrunkId TrunkGroup LastTrunkId TrunkGroup FirstBChannel TrunkGroup LastBChannel TrunkGroup FirstPhoneNumber TrunkGroup Profileld TrunkGroup Module INE 1 O 0 0 i1 24 1900 IMA ulD 2 O i 1 1 24 20005 IewialErovjo 3 O0 2 2 i1 24 3000 Inevialdercoujo 4 O 2 5 1 24 4000 TrunkGroup CallProgressTonesFilename CPUSA dat SaveConfiguration 1 4 3 Modifying an ini File You can modify an ini file currently used by a device Modifying an ini file instead of loading an entirely new ini file preserves the device s current configuration including factory default values gt To modify an ini file take these 4 steps 1 Save the ini file from the device to your PC using the Web i
161. 0 PRIMARY Primary Trunk default contains a D channel that is used for signaling 1 BACKUP Backup Trunk contains a backup D channel that is used if the primary D channel fails 2 NFAS NFAS Trunk contains only 24 B channels without a signaling D channel Note This parameter is applicable only to T1 ISDN protocols Enables Rx ISDN overlap per trunk ID 0 Disable Disabled default 1 Enable Enabled Notes If enabled the device receives ISDN called number that is sent in the Overlap mode The SETUP message to IP is sent only after the number including the Sending Complete IE is fully received via SETUP and or subsequent INFO Q 931 messages The MaxDigits parameter can be used to limit the length of the collected number for device ISDN overlap dialing if sending complete is not received fa digit map pattern is defined DigitMapping the device collects digits until a match is found e g for closed numbering schemes or until a timer expires e g for open numbering schemes If a match is found or the timer expires the digit collection process is terminated even if Sending Complete wasn t received Determines whether Ringback tone is played to the ISDN by the PBX PSTN or by the device 0 PBX PBX PSTN default 1 Gateway This parameter is applicable to ISDN protocols It is used 88 Document LTRT 68808 SIP User s Manual 3 Web Based Management
162. 0 33 174 50 16 10 35 0 1 6 Mecha InterfaceTable 0 2 O 10 32 174 50 16 0 0 0 0 5 Comexolls InterfaceTable VLAN related parameters VlanMode 1 VlanNativeVlaniId 4 Routing Table Configuration IP Routing table parameters RoutingTableDestinationsColumn 130 33 4 6 83 4 87 6 RoutingTableDestinationMasksColumn 255 255 255 255 255 255 25050 RoutingTableGatewaysColumn 10 32 0 1 10 31 0 1 RoutingTableInterfacesColumn 2 0 RoutingTableHopsCountColumn 20 20 Class Of Service parameters VlanNetworkServiceClassPriority 7 VlanPremiumServiceClassMediaPriority VlanPremiumServiceClassControlPriority VlanGoldServiceClassPriority 4 VlanBronzeServiceClassPriority 2 NetworkServiceClassDiffServ 48 PremiumServiceClassMediaDiffServ 46 PremiumServiceClassControlDiffServ 40 GoldServiceClassDiffServ 26 BronzeServiceClassDiffServ 10 Application Type for applications EnableDNSasOAM 1 EnableSCTPasControl 1 EnableNTPasOAM 1 IOA Ov 2 Use the BootP TFTP utility refer to the Product Reference Manual to load and burn the firmware version and the ini file you prepared in the previous step to the device multiple IPs and VLANs support is available only when the firmware is burned to flash 3 Reset the device after disabling it on the BootP TFTP utility Version 5 6 391 November 2008 A Ee AudioCodes Mediant 2000 Instead of using the ini file table parameter InterfaceTable
163. 0 and 1 use the E amp M Winkstart CAS protocol while trunks 2 and 3 use the E amp M Immediate Start CAS protocol Note For additional CAS table ini file parameters CASFileName_0 CASFileName_1 CASFileName_7 and CASTablesNum refer to E1 T1 Configuration Parameters on page 303 Ar OO The Dial Plan name that is used on a specific trunk The range is up to 11 character strings Alert Timeout ISDN T301 timer in seconds for outgoing calls to PSTN This timer is used between the time that a SETUP message is sent to the Tel side IP to Tel call establishment and a CONNECT message is received If ALERT is received the timer is restarted In the ini file parameter ID depicts the trunk number where 0 is the first trunk The range is 1 to 600 The default is 180 Determines the method for setting digital trunks to Out Of Service state per trunk 1 Not Configured Use the settings of the DigitalOOSBehavio parameter for per device default 93 November 2008 ca AudioCodes ini File Field Name Web Parameter Name Play Ringback Tone to Trunk PlayRBTone2Trunk_ID SIP User s Manual Mediant 2000 Valid Range and Description 0 Default Uses default behavior for each trunk see note below 1 Service Sends ISDN In or Out of Service only for ISDN protocols that support Service message 2 D Channel Takes D Channel down or up ISDN only 8 Alarm Sends or cleans PSTN AIS Alarm ISDN and
164. 0 dB sec 15 6 00 dB sec 16 7 00 dB sec 17 8 00 dB sec 18 9 00 dB sec 19 10 00 dB sec 20 11 00 dB sec 21 12 00 dB sec 22 13 00 dB sec 23 14 00 dB sec 24 15 00 dB sec 25 20 00 dB sec 26 25 00 dB sec 27 30 00 dB sec 28 35 00 dB sec 29 40 00 dB sec 30 50 00 dB sec 31 70 00 dB sec Determines the AGC direction 0 0 AGC works on signals from the TDM side default 1 1 AGC works on signals from the IP side Determines the signal energy value dBm that the AGC attempts to attain The valid range is 0 to 63 dBm The default value is 19 dBm N A N A N A Enables or disables the activation of the Pattern Detector PD Valid options include 0 Disable Disable default 1 Enable Enable 78 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 2 5 Configuring the General Media Settings The General Media Settings page allows you to configure various media parameters gt Toconfigure general media parameters take these 4 steps 1 Open the General Media Settings page Configuration tab gt Media Settings menu gt General Media Settings page item Figure 3 44 General Media Settings Page General Settings Max Echo Canceller Length Default lg Enable Continuity Tones Disable 2 Configure the general media parameters according to the table below 3 Click the Submit bu
165. 00rel em Content Length 0 m F7 10 8 201 10 gt gt 10 8 201 108 200 OK SIP 2 0 200 OK Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacRKCVBud From lt sip 8000 10 8 201 108 gt tag 1c5354 To lt Sip 1000 10 8 201 10 gt tag 1c7345 Call ID 534366556655skKw 8000 1000 10 8 201 108 Server Audiocodes Sip Gateway Mediant 2000 v 5 40 010 006 CSeq 18154 BYE Supported 100rel em Content Length 0 7 14 2 SIP Authentication Example The device supports basic and digest MD5 authentication types according to SIP RFC 3261 standard A proxy server might require authentication before forwarding an INVITE message A Registrar Proxy server may also require authentication for client registration A proxy replies to an unauthenticated INVITE with a 407 Proxy Authorization Required response containing a Proxy Authenticate header with the form of the challenge After sending an ACK for the 407 the user agent can then resend the INVITE with a Proxy Authorization header containing the credentials User agent redirect or registrar servers typically use 401 Unauthorized response to challenge authentication containing a WWW Authenticate header and expect the re INVITE to contain an Authorization header The following example describes the Digest Authentication procedure including computation of user agent credentials 1 The REGISTER request is sent to Registrar Proxy server for registration as follows Version 5 6 37
166. 08 ca AudioCodes ini File Field Name Web Parameter Name ISDNIBehavior SIP User s Manual Mediant 2000 Valid Range and Description button and then for each required option select 1 to enable The default is 0 i e disable 1 NO STATUS ON UNKNOWN IE Q 931 Status message isn t sent if Q 931 received message contains an unknown unrecognized IE s By default the Status message is sent Note Applicable only to PRI variants in which sending of Status message is optional 2 NO STATUS ON INV OP IE Q 931 Status message isn t sent if an optional IE with invalid content is received By default the Status message is sent Note Applicable only to PRI variants in which sending of Status message is optional 4 ACCEPT UNKNOWN FAC IE Accepts unknown unrecognized Facility IE Otherwise the Q 931 message that contains the unknown Facility IE is rejected default Note Applicable only to PRI variants where a complete ASN1 decoding is performed on Facility IE 128 SEND USER CONNECT ACK Connect ACK message is sent in response to received Q 931 Connect Otherwise the Connect ACK is not sent default Note Applicable only to Euro ISDN User side outgoing calls 512 EXPLICIT INTERFACE ID Enables to configure T1 NFAS Interface ID refer to the parameter ISDNNFASInterfacelD_x Note Applicable to 4 5ESS DMS NI 2 and HKT variants 2048 ALWAYS EXPLICIT Always set the Channel Identification IE to explic
167. 1 102 111 127 Description Identified channel does not exist Suspended call exists but this call identity does not Call identity in use No call suspended Call having the requested call identity has been cleared User not member of CUG Incompatible destination Invalid transit network selection Invalid message Mandatory information element is missing Message type non existent or not implemented Message not compatible with call state or message type non existent or not implemented Information element non existent or not implemented Invalid information elements contents Message not compatible with call state Recovery of timer expiry Protocol error Interworking unspecified Mediant 2000 SIP E Response Description 502 Bad gateway 503 Service unavailable 503 Service unavailable 503 Service unavailable 408 Request timeout 503 Service unavailable 503 Service unavailable 502 Bad gateway 503 Service unavailable 409 Conflict 480 Temporarily not available 409 Conflict 480 Not found 501 Not implemented 503 Service unavailable 408 Request timeout 500 Server internal error 500 Server internal error Messages and responses were created because the ISUP to SIP Mapping draft doesn t specify their cause code mapping 9 2 3 Fixed Mapping of SIP Response to ISDN Release Reason The following table describes the mapping of SIP response to ISDN release reason SIP Response
168. 1 in the IP Group table refer to Configuring the IP Groups on page 201 the request SIP URI host name in the INVITE message is set to the value of the parameter Dest IP Address if defined otherwise it is set to the value of the parameter SIP Group Name defined in the IP Group table Note This parameter is also used as the Serving IP Group in the Account table for acquiring authentication user password for this call IP Profile ID defined in IP Profile Settings on page 193 assigned to the outbound IP call This allows you to assign many different configuration attributes e g voice coders to this IP Group outbound routing rule A read only field representing the Quality of Service of the destination IP address n a Alternative Routing feature is disabled OK IP route is available Ping Error No ping to IP destination route is not available QoS Low Bad QoS of IP destination route is not available DNS Error No DNS resolution only when domain name is used instead of an IP address 3 4 7 4 4 IP to Trunk Group Routing Table The IP to Trunk Group Routing Table page provides a table for routing incoming IP calls to groups of channels E1 T1 B channels called Trunk Groups Trunk Group ID s are assigned to the device s channels in the Trunk Group Table page refer to Configuring the Trunk Group Table on page 195 You can add up to 24 IP to Trunk Group routing ru
169. 1 5 seconds After this period the Gain Slope is changed to the user defined value You can disable or enable the AGC s Fast Mode feature using the ini file parameter AGCDisableFastAdaptation After Fast Mode is used the signal should be off for two minutes in order to have the feature turned on again To configure AGC refer to Configuring the Pmedia Settings on page 76 SIP User s Manual 402 Document LTRT 68808 SIP User s Manual 10 Tunneling Applications 10 10 1 Tunneling Applications This section discusses TDM and QISG tunneling supported by the device TDM Tunneling The device s TDM Tunneling feature allows you to tunnel groups of digital trunk spans or timeslots B channels over the IP network TDM Tunneling utilizes the device s internal routing without Proxy control capabilities to receive voice and data streams from TDM E1 T1 J1 spans or individual timeslots convert them into packets and then transmit them over the IP network using point to point or point to multipoint device distributions A device opposite it or several devices when point to multipoint distribution is used converts the IP packets back into TDM traffic Each timeslot can be targeted to any other timeslot within a trunk in the opposite device When TDM Tunneling is enabled Enable TDM Tunneling parameter is set to Enable on the originating device refer to Configuring the Digital Gateway Parameters on page 207 the originati
170. 1 6 Viewing Active Alarms 0 ccceeee ee mani re eens manaii 3 6 1 7 Viewing Trunks amp Channels Status mies ee ee ame 246 3 6 2 Gateway Statistics T E E A E E S 3 6 2 1 Call Counters sabes N SORAR T E en 3 6 2 2 Call Routing Status EIE EEEN A E IAE A AEEA E OA D 3 6 2 3 SAS SBC Registered Users AEE A AEAT AAE EA we 3 6 2 4 IP Connectivity OE AA E ana neice al 4 iM File Contigurati sssaaa AEREA AEREA SARANANE 255 4 1 Secured Encoded ini File E nee ae er nee AEN ee a2 The mi File SiGe acsearle eed OO AAT OPUS RUES aeaa TUTTE TES ert Cet RN eet mn eee d 256 4 2 2 Structure of Individual ini File Parameters 2 ccccceeeceeeceeeeeeeeeeeeeeeeeeeeeeenetees 256 423 Structure of ini File Table Parameters icc ccecc cscs causes aennctieamenennnecee 257 SIP User s Manual 4 Document LTRT 68808 SIP User s Manual Contents 52 Restonng Factory TA sti ca ce scccetiarc act terieieeenae ome msibnnebameaeccenedaeree 6 Auxiliary Configu rat io n Fi les SERRE RRR RRR RRR RRR 335 6 1 6 2 6 4 PEE EEA E EEE EE A AEE PE EEE E 6 5 Dial Plan File P EE AE ca eed eid E Nese aa nc cesar E a EA maaan 6 6 User Information Fil 7 IP Telephony Capabi li ities SEER EERE BERR RRR RR RRR RRR RRR 343 7 1 IP to IP Rigel baal i pia EAEE LA PE EA E E S E A E E E Version 5 6 5 November 2008 3 wt Aud ioCodes Mediant 2000 7 6 Event Notification using X Detect Header TI
171. 1 November 2008 A ge AudioCodes Mediant 2000 REGISTER sip 10 2 2 222 SIP 2 0 Via SIP 2 0 UDP 10 1 1 200 From lt Sip 122 10 1 1 200 gt tag 1c17940 Weg eios L22200 1 1 200s Call ID 634293194 10 1 1 200 User Agent Audiocodes Sip Gateway Mediant 2000 v 5 40 010 006 CSeq 1 REGISTER ComicaAcies SijaslA2 iO 1 i200 Expires 3600 2 Upon receipt of this request the Registrar Proxy returns 401 Unauthorized response SIP 2 0 401 Unauthorized Via SIP 2 0 UDP 10 2 1 200 rom lt sljosl2Z LO 2 2 222 Sricacgeilcily7 S40 Mos lt gijosil22 10 2 2 222 gt Call ID 634293194 10 1 1 200 Cseq 1 REGISTER Date Mon 30 Jul 2001 15 33 54 GMT Server Columbia SIP Server 1 17 Content Length 0 WWW Authenticate Digest realm audiocodes com nonce 11432d6bce58ddf02e3b5e1c77c010d2 stale FALSE algorithm MD5 3 According to the sub header present in the WWW Authenticate header the correct REGISTER request is formed 4 Since the algorithm is MD5 then e The username is equal to the endpoint phone number 122 e The realm return by the proxy is audiocodes com e The password from the ini file is AudioCodes e The equation to be evaluated is according to RFC this part is called A1 122 audiocodes com AudioCodes e The MD5 algorithm is run on this equation and stored for future usage e The result is a8f17d4b41ab8dab6c95d3c14e34a9e1 5 Next the par called A2 needs to be evaluated e The
172. 10 1 1 11 NTP UTC Offset Hours 0 Minutes NTP Updated Interval Hours 24 Minutes w Telnet Settings Embedded Telnet Server Enable Unsecured Telnet Server TCP Port 23 Telnet Server Idle Timeout 0 SSH Server Enable Disable 6 SSH Server Port 22 v DNS Settings I DNS Primary Server IP DNS Secondary Server IP v STUN Settings I Enable STUN Disable STUN Server Primary IP 0 0 0 0 STUN Server Secondary IP 0 0 0 0 v NFS Settings NFS Table gt v DHCP Settings Enable DHCP Disable 2 Configure the Applications parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Version 5 6 57 November 2008 ca AudioCodes Mediant 2000 Table 3 9 Application Settings Parameters Parameter Description NTP Settings For detailed information on Network Time Protocol NTP refer to Simple Network Time Protocol Support on page 383 NTP Server IP Address NTPServerIP NTP UTC Offset NTPServerUTCOffset NTP Update Interval NTPUpdatelnterval Telnet Settings Embedded Telnet Server TelnetServerEnable Telnet Server TCP Port TelnetServerPort Telnet Server Idle Timeout TelnetServerldleDisconnect SSH Serve
173. 2 From the Routing Index drop down list select the range of entries that you want to add 3 Configure the table according to the table below 4 Click the Submit button to save your changes 5 To save the changes so they are available after a power fail refer to Saving Configuration on page 230 Table 3 47 Inbound IP Routing Table Description Parameter IP to Tel Routing Mode RouteModelP2Tel Dest Host Prefix PstnPrefix_DestHostPrefix_ Source Host Prefix PstnPrefix_SrcHostPrefix Dest Phone Prefix PstnPrefix_DestPrefix Source Phone Prefix PstnPrefix_SourcePrefix Source IP Address PstnPrefix_SourceAddress Description Determines whether to route the IP calls before or after manipulation of the destination number configured in Configuring the Number Manipulation Tables on page 164 0 Route calls before manipulation IP to IP calls are routed before the number manipulation rules are applied default 1 Route calls after manipulation IP to IP outbound calls are routed after the number manipulation rules are applied The Request URI host name prefix of the incoming SIP INVITE message If this routing rule is not required leave the field empty The asterisk symbol can be used to depict any destination host prefix The From header URI host name prefix of the incoming INVITE message If this routing rule is not required leave the field empty The asterisk symbol can be
174. 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 Digit Patterns The following digit pattern parameters apply only to VM applications that use the DTMF communication method For the available pattern syntaxes refer to the CPE Configuration Guide for Voice Mail DigitPatternForwardOnBusy DigitPatternForwardOnNoAnswer DigitPatternForwardOnDND DigitPatternForwardNoReason DigitPatternForwardOnBusyExt DigitPatternForwardOnNoAnswerExt DigitPatternForwardOnDNDExt DigitPatternForwardNoReasonExt SIP User s Manual For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this
175. 255 If the source IP address SourceAddress includes an FQDN DNS resolution is performed according to DNSQueryType For available notations that represent multiple numbers refer to Dialing Plan Notation on page 168 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to IP to Trunk Group Routing on page 181 For a description of this parameter refer to Tel to IP Routing Table on page 175 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 Determines the SIP headers containing the source number after manipulation 0 Both SIP From and P Asserted Id headers contain the source number after manipulation default 1 Only SIP From header contains the source number after manipulation while the P Asserted Id header contains the source number before manipulation 317 November 2008 ca AudioCodes Parameter SwapTel2IPCalled amp CallingNum bers AddTON2RPI NumberMapTel2IP SIP User s Manual Mediant 2000 Description If enabled the device swaps the calling and called numbers received from the Tel side The INVITE message contains the
176. 2IP_Port3 SRV2IP For example SRV2IP SRVZ2IP 0 SrvDomain 0 Dnsnamet1 1 1 500 Dnsname2 2 2 501 0 0 0 SRV2IP Notes This parameter can include up to 10 indices Ifthe Internal SRV table is used the device first attempts to resolve a domain name using this table If the domain name isn t located the device performs an SRV resolution using an external DNS server To configure the Internal SRV table using the Web interface and for a description of the parameters in this ini file table parameter refer to Internal SRV Table on page 187 261 November 2008 A Ee AudioCodes Mediant 2000 Parameter EnableSTUN STUNServerPrimaryIP STUNServerSecondarylIP STUNServerDomainName NATBindingDefaultTimeo ut DisableNAT EnablelPAddrTranslation SIP User s Manual Description Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 For a description of this parameter refer to Configuring the Application Settings on page 57 For a description of this parameter refer to Configuring the Application Settings on page 57 For a description of this parameter refer to Configuring the Application Settings on page 57 Defines the domain name for the Simple Traversal of User Datagram Protocol STUN server s address used for retrieving all STUN servers with an SRV query The STUN client can perform the required SRV query to resol
177. 3 111 Active IP Interfaces Page Index Application Type Address Type interface Mode IP Address ene na jal IP IPy4 Manual 10 13 4 13 10 13 01 0 All Gateway WLAN 10 Interiave ame dis VLAN Mode 3 6 1 4 Viewing Device Information The Device Information page displays the device s specific hardware and software product information This information can help you to expedite troubleshooting Capture the page and e mail it to AudioCodes Technical Support personnel to ensure quick diagnosis and effective corrective action This page also displays any loaded files used by the device stored in the RAM and allows you to remove them gt To access the Device Information page take this step m Open the Device Information page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Device Information page item Figure 3 112 Device Information Page v General Settings MAC Address 00908f049345 Serial Number 299845 Board Type 24 Device Up Time Od 4h 16m 27s 67th Device Administrative State Unlocked Device Operational State Enabled Flash Size bytes 8388608 RAM Size bytes 134217728 CPU Speed MHz 200 wv Versions Version ID 5 304 015 DSP Type 2 DSP Software Version 54012 DSP Software Name 6244E3 Flash Version 192 Module Firmware 0x31 v Loaded Files Call Progress Tones File
178. 3 14 15 16 17 18 19 20 21 22 23 24 25 26 27 26 29 30 31 Tunk ggg ggg gag agg ag gg ag gg agg gg gg ag ap agg ig gg ag ag Trunk 2g ggg gg ag agg agg ggg ig gg ag gl ggg ig gag ag gl ig gl gl ag gl Trunk as gig gig gg gag agg ag ag gt gl gl lg ig gg ggg ag ag ag ag ag ag ig igh ra ag gag gag gg gg gl gl agg agg agg agg gl gl gl ig gl gl gl gl gl ag gl agg ra gpg agg ggg gg agg agg agg ag agg agg gg gg gl gg ag gl agg Trunk eg gl ggg gg gg agg agg agg agg agg agg gg gl ig gl gg ag gl ag gl rk 7g gl gg ggg gg agg agg agg gl agg agg gl ag gl eg gl ag gl ag gl agg Trak gg gl ag gh ag gl gl gl gl gl gg gg gl gg gh gl gh gl gh gl gh gl gl gl gl alg SIP User s Manual 246 Document LTRT 68808 SIP User s Manual 3 Web Based Management Note The number of trunks and channels displayed on the page depends on the system configuration The page initially displays the first eight trunks and their channels The page displays eight consecutive trunks at a time You can view the next eight trunks by performing the procedure below gt To view the next eight trunks take this step Click the Go To Page J icon Figure 3 116 Example of a Selected Page Icon for Displaying Trunks 17 24 Page Number GoTo cj EEE os eh Ch ty i ry Page Selected Page Icon Showing Trunks 17 24 The Trunks and Channels Status page uses the following color coding icons to indicate the status of the trunks and chann
179. 3 40 Dialing Plan Notations Description Represents a range of numbers Note Range of letters is not supported Represents multiple numbers Up to three digits can be used to denote each number Represents any single digit Represents the end of a number Represents any number SIP User s Manual Example 5551200 5551300 represents all numbers from 5551200 to 5551300 123 100 200 represents all numbers from 123100 to 123200 2 3 4 5 6 represents a one digit number that starts with 2 3 4 5 or 6 11 22 33 xxx represents a four digit number that starts 11 22 or 33 111 222 xxx represents a four digit number that starts 111 or 222 54324 represents any number that starts with 54324 54324xx represents a 7 digit number that starts with 54324 represents any number i e all numbers 168 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 7 3 2 The device matches the rules starting at the top of the table i e top rules take precedence over lower rules For this reason enter more specific rules above more generic rules For example if you enter 551 in entry 1 and 55 in entry 2 the device applies rule 1 to numbers that start with 551 and applies rule 2 to numbers that start with 550 552 553 554 555 556 557 558 and 559 However if you enter 55 in entry 1 and 551 in entry 2 the device applies rule 1 to all numbers that start with 55 includi
180. 348 7 3 1 Configuring SAS For configuring the device to operate with SAS perform the following configurations m lIsProxyUsed 1 m ProxyIP 0 lt SAS agent s IP address i e the device gt m ProxylIP 1 lt external Proxy server IP address gt m IsRegisterNeeded 1 for the device m RegistrarlP m SIPDestinationPort 5080 m IsUserPhone 0 don t use user phone in SIP URL m IsUserPhonelnFrom 0 don t use user phone in From Header m IsFallbackUsed 0 m EnableProxyKeepAlive 1 enables keep alive with Proxy using OPTIONS E EnableSAS 1 E SASLocalSIPUDPPort default 5080 E SASRegistrationTime lt expiration time that SAS returns in the 200 OK to REGISTER in Emergency mode gt default 20 m SASDefaultGatewayIP lt SAS gateway IP address gt E SASProxySet 1 Version 5 6 347 November 2008 A ge AudioCodes Mediant 2000 7 3 2 Configuring Emergency Calls The device s SAS agent can be configured to detect a user defined emergency number e g 911 in North America which it then redirects the call directly to the PSTN through its E1 T1 trunk The emergency number is configured using the ini file parameter SASEmergencyNumbers for a detailed description refer to SIP Configuration Parameters on page 284 Figure 7 1 Device s SAS Agent Redirecting Emergency Calls to PSTN IP Centrex AudioCodes Mediant 2000 PSTN Network Emergency Calls e g 911 Route
181. 5 The default value is 1 i e no redundant Proxy Set 3 4 7 2 4 SBC Configuration The SBC Settings page allows you to enable the device s IP to IP call routing feature To enable IP to IP capabilities the following prerequisites must be fulfilled m The device must be loaded with the Feature Key that includes the SBC feature refer to Upgrading the Software Upgrade Key on page 233 m The device must be running SIP version 5 4 or later Version 5 6 163 November 2008 A Ee AudioCodes Mediant 2000 gt To configure the SBC parameters take these 4 steps 1 Open the SBC Settings page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt SBC Configuration page item Figure 3 67 SBC Settings Page v Enable SBC SBC Registration Time 2 Configure the SBC parameters according to the following table 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration Table 3 38 SBC Parameters Parameter Description Enable SBC Enables or disables the SBC feature EnableSBC 0 Disable default 1 Enable SBC Registration Time Configures the value in sec sent in the expires when the device replies SBCRegistrationTime with a SIP 200 OK in response to Registration requests The default is 20 Note This parameter is applicable only to clients belonging to IP groups of type USER 3 4 7 3 C
182. 505 Disable Enable 0 5060 Yes No Disabl 180 Disab Yes Disab No No Disable Don t Play Play According to Early Media Disable Disable He ASSES SS AudiocodesGW Don t Play g None Disable Forward Disable Disable 0 Disable Parallel handling Enable Batin Retransmission Parameters SIP T1 Retransmission Timer msec SIP T2 Retransmission Timer msec SIP Maximum RTX 500 4000 7 121 November 2008 A ge AudioCodes Mediant 2000 2 Configure the parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 30 SIP General Parameters Protocol Definition Parameter PRACK Mode PRACKMode Channel Select Mode ChannelSelectMode SIP User s Manual Description PRACK Provisional Acknowledgment mechanism mode for 1xx SIP reliable responses 0 Disable 1 Supported default 2 Required Notes The Supported and Required headers contain the 100rel tag The device sends PRACK messages if the 180 183 response is received with 100rel in the Supported or Required headers P
183. 62 Media Network Settings Available only in Multiple IP and Dual IP modes IP Address LocalMedialPAddress Subnet Mask LocalMediaSubnetMask SIP User s Manual The device s source IP address in the Media network The default value is 0 0 0 0 The device s subnet mask in the Media network The default subnet mask is 0 0 0 0 52 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Description Default Gateway Address The device s default Gateway IP address in the Media network LocalMediaDefaultGW The default value is 0 0 0 0 Multiple Interface Settings Multiple Interface Table Click the right pointing arrow u button to open the Multiple Interface Table page For a description of configuring multiple IP interfaces refer to Configuring the Multiple Interface Table on page 53 VLAN For detailed information on the device s VLAN implementation refer to VLANS and Multiple IPs on page 384 VLAN Mode Enables the VLAN functionality VIANMode 0 Disable default 1 Enable Note This parameter cannot be changed on the fly and requires a device reset VALN ID Settings Native VLAN ID Defines the native VLAN identifier Port VLAN ID PVID VLANNativeVlanID The valid range is 1 to 4094 The default value is 1 OAM VLAN ID Defines the OAMP VLAN identifier VLANOamVlanID The valid range is 1 to 4094 The default value is 1 Control VLAN ID Defines the Control VLAN i
184. 7 266 Document LTRT 68808 SIP User s Manual Parameter Differential Services 4 ini File Configuration Description For detailed information on IP QoS via Differentiated Services refer to IP QoS via Differentiated Services DiffServ on page 384 NetworkServiceClassDiffS erv PremiumServiceClassMed iaDiffServ PremiumServiceClassCon trolDiffServ GoldServiceClassDiffServ BronzeServiceClassDiffSe rv NFS Table Parameter NFSServers Version 5 6 For a description of this parameter refer to Configuring the QoS Settings on page 63 For a description of this parameter refer to Configuring the QoS Settings on page 63 For a description of this parameter refer to Configuring the QoS Settings on page 63 For a description of this parameter refer to Configuring the QoS Settings on page 63 For a description of this parameter refer to Configuring the QoS Settings on page 63 This ini file table parameter defines Network File Systems NFS so that the device can access a remote server s shared files and directories for loading cmp ini and auxiliary files using the Automatic Update mechanism The format of this ini file table parameter is as follows NFSServers FORMAT NFSServers_Index NFSServers_HostOrlP NFSServers_RootPath NFSServers_NfsVersion NFSServers_AuthType NFSServers_UID NFSServers_ GID NFSServers_VlanType NFSServers For example NFSServers FORMAT NFSSer
185. 7 8 9 SIP User s Manual 252 Document LTRT 68808 SIP User s Manual Column Name IP Address Host Name Connectivity Method Connectivity Status Quality Status Quality Info DNS Status Version 5 6 3 Web Based Management Table 3 67 IP Connectivity Parameters Description The IP address can be one of the following P address defined as the destination IP address in the Tel to IP Routing table or Outbound IP Routing Table page IP address resolved from the host name defined as the destination IP address in the Tel to IP Routing table or Outbound IP Routing Table page Host name or IP address as defined in the Tel to IP Routing table or Outbound IP Routing Table page The method according to which the destination IP address is queried periodically ICMP ping or SIP OPTIONS request The status of the IP address connectivity according to the method in the Connectivity Method field OK Remote side responds to periodic connectivity queries Lost Remote side didn t respond for a short period Fail Remote side doesn t respond Init Connectivity queries not started e g IP address not resolved Disable The connectivity option is disabled i e parameter Alt Routing Tel to IP Mode AltRoutingTel2IPMode ini is set to None or QoS refer to Routing General Parameters on page 171 Determines the QoS accord
186. ADGgEOADCCAQkCggEAPqd4MziR4spW1dGRx8bQrhZkon WnNm Yhb7 4067ecf1janH7GcN SXsf x7jJprewuL 7v7Cvpr4R7qlJcmdHIntm 7 JPM5n6cDBv17uSW63er7NkVnMFHWK1QaGFLMybFkzaeGrvFm4k3 1RefixDmu0e FhJd gHYezYHf44LvPRPwhSrzi9 Aq308pWDguJuZDIUP1F1jMa LPwvREXf FCcUW w END CERTIFICATE 6 Set the parameter Secured Web Connection HTTPS to HTTPS Only 0 refer to Configuring the General Security Settings on page 109 to ensure you have a method of accessing the device in case the new certificate doesn t work Restore the previous setting after testing the configuration SIP User s Manual 106 Document LTRT 68808 SIP User s Manual 3 Web Based Management 7 In the Certificates Files group click the Browse button corresponding to Send Server Certificate navigate to the cert txt file and then click Send File 8 When the loading of the certificate is complete save the configuration refer to Saving Configuration on page 230 and restart the device the Web interface uses the provided certificate The certificate replacement process can be repeated when necessary e g the new certificate expires It is possible to use the IP address of the device e g 10 3 3 1 instead of a qualified DNS name in the Subject Name This is not recommended since the IP address is subject to changes and may not uniquely identify the device The server certificate can also be loaded via ini file using the parameter HTTPSCertFileName
187. AS protocol It is possible to define up to eight different CAS files by repeating this parameter Each CAS file can be associated with one or more of the device trunks using the parameter CASTablelndex_x 1 to 8 Indicates how many CAS protocol configurations files are loaded For a description of this parameter refer to Configuring the TDM Bus Settings on page 218 For a description of this parameter refer to Configuring the TDM Bus Settings on page 218 304 Document LTRT 68808 SIP User s Manual Parameter LineBuildOut Loss ISDNRxOverlap_x ISDNRxOverlap R2Category CallPriorityMode MLPPDefaultNamespace SIPDefaultCallPriority MLPPDiffserv PreemptionToneDuration MLPPNormalizedServiceD omain MLPPDefaultServiceDoma in Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 0 Disabled default 1 Enabled Any number bigger than one Number of digits to receive Notes f enabled the device receives ISDN called number that is sent in the Overlap mode The INVITE to IP is sent only after the number including Sending Complete Info Element was fully received in SETUP and or subsequent INFO Q 931 messages For detailed information on ISDN overlap dialing refer to ISDN Overlap Dia
188. ASa5h64 IR laOkeEbDSAddF 8938s KeTIAddFSc iss O2x1aOkeTJIAdGF 8c ts TJQINGSaShbtyx1eO0keXZIAddFSahss a inthe Send Upgrade Key file field click the Browse button and navigate to the folder in which the Software Upgrade Key text file is located on your PC b Click the Send File button the new key is loaded to the device and validated If the key is valid it is burned to memory and displayed in the Current Key field 5 Verify that the Software Upgrade Key file was successfully loaded to the device by using one of the following methods e In the Key features group ensure that the features and capabilities activated by the installed string match those that were ordered e Access the Syslog server refer to the Product Reference Manual and ensure that the following message appears in the Syslog server S N____ Key Was Updated The Board Needs to be Reloaded with ini file n 6 Reset the device the new capabilities and resources are active Note If the Syslog server indicates that the Software Upgrade Key file was unsuccessfully loaded i e the SN_ line is blank perform the following preliminary troubleshooting procedures 1 Open the Software Upgrade Key file and check that the S N line appears If it does not appear contact AudioCodes 2 Verify that you ve loaded the correct file Open the file and ensure that the first line displays LicenseKeys 3 Verify that the contents of the file has not b
189. Accounting Only Only accounting indications are used 3 4 9 Configuring the TDM Bus Settings The device s Time Division Multiplexing TDM bus settings can be performed in the TDM Bus Settings page as described in the procedure below gt To configure the TDM Bus settings take these 5 steps 1 Open the TDM Bus Settings page Configuration tab gt TDM Configuration menu gt TDM Bus Settings page item Figure 3 89 TDM Bus Settings Page v PCM Law Select MuLaw I TDM Bus Clock Source Internal TDM Bus PSTN Auto Clock Disable amp TDM Bus PSTN Auto Clock Reverting Disable l Idle PCM Pattern 255 Idle ABCD Pattern OxOF ra TDM Bus Local Reference 1 2 Configure the TDM bus parameters according to the table below 3 Click the Submit button to save your changes 4 Save the changes to flash memory refer to Saving Configuration on page 230 5 Reset the device refer to Resetting the Device on page 228 Table 3 56 TDM Bus Settings Parameters Description Parameter Description PCM Law Select Determines the type of PCM companding law in input output TDM bus PCMLawSelect 1 Alaw Alaw default 3 MuLaw MuLaw Note Typically A Law is used for E1 spans and u Law for T1 J1 spans SIP User s Manual 218 Document LTRT 68808 SIP User s Manual Parameter Idle PCM Pattern IdleP gt CMPattern Idle ABCD Pattern IdleABCDPattern TDM Bus Local Re
190. AccountingType DefaultAccessLevel RadiusLocalCacheMode RadiusLocalCacheTimeout Version 5 6 Description For a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring RADIUS Accounting Parameters on page 217 For a description of this parameter refer to Configuring the General Security Settings on page 109 Number of concurrent calls that can communicate with the RADIUS server optional The valid range is 0 to 240 The default value is 240 For a description of this parameter refer to Configuring the General Security Settings on page 109 Number of retransmission retries The valid range is 1 to 10 The default value is 3 Determines the time interval measured in seconds the device waits for a response before a RADIUS retransmission is issued The valid range is 1 to 30 The default value is 10 For a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring RADIUS Accounting Parameters on page 217 For a description of this parameter refer to Configuring RADIUS Accounting Parameters on page 217 For a description of this parameter refer to Configuring RADIUS Accounting Parameters on page 217 For
191. AudioCodes Mediant 2000 8 8 8 9 8 9 1 The NTP client follows a simple process in managing system time the NTP client requests an NTP update receives an NTP response and then updates the local system clock based on a configured NTP server within the network The client requests a time update from a specified NTP server at a specified update interval In most situations this update interval is every 24 hours based on when the system was restarted The NTP server identity as an IP address and the update interval are user defined using either the Web interface refer to Configuring the Application Settings on page 57 the ini file NTPServerlP and NTPUpdatelnterval respectively or an SNMP MIB object refer to the Product Reference Manual When the client receives a response to its request from the identified NTP server it must be interpreted based on time zone or location offset that the system is to a standard point of reference called the Universal Time Coordinate UTC The time offset that the NTP client uses is configurable using the Web interface refer to Configuring the Application Settings on page 57 the ini file NTPServerUTCOffset or via an SNMP MIB object refer to the Product Reference Manual If required the clock update is performed by the client as the final step of the update process The update is performed in such a way as to be transparent to the end users For instance the response of the server may in
192. BC Registered Users The SAS Registered Users page displays a list of up to 250 Stand Alone Survivability SAS and or IP to IP registered users The SAS feature is configured in the SAS Configuration page refer to Stand Alone Survivability on page 161 The IP to IP feature is configured enabled in the SBC Configuration page refer to SBC Configuration on page 163 gt To view the SAS registered users take this step m Open the SAS Registered Users page Status amp Diagnostics tab gt Gateway Statistics menu gt SAS SBC Registered Users page item Figure 3 120 SAS Registered Users Page Address Of Record Contact lt sip 2400 Proxies ac gt lt sip 2400 10 8 210 5 gt expires 160 lt sip 2401 Proxies ac gt lt sip 2401 10 6 210 5 gt expires 160 lt sip 2500 Proxies ac gt lt sip 2500 10 8 210 5 gt expires 180 lt sip 2402 Proxies ac gt lt sip 2402 10 8 210 5 gt expires 160 lt sip 2403 Proxies ac gt lt sip 2403 10 8 210 5 gt expires 160 lt sip 2404 Proxies ac gt lt sip 2404 10 6 210 5 gt expires 160 lt sip 2405 Proxies ac gt lt sip 2405 10 8 210 5 gt expires 160 Version 5 6 251 November 2008 A Ee AudioCodes Mediant 2000 Table 3 66 SAS Registered Users Parameters Column Name Description Address of An address of record AOR is a SIP or SIPS URI that points to a domain with a Record location service that can map the URI to another URI Contact where the user might be a
193. BD transport mode the V 152 implementation can use alternative relay fax transport methods e g fax relay over IP using T 38 The preferred V 152 transport method is indicated by the SDP pmft attribute Omission of this attribute in the SDP content means that VBD mode is the preferred transport mechanism for voice band data To configure T 38 mode use the CoderName parameter 7 6 Event Notification using X Detect Header The device supports the sending of notifications to a remote party notifying the occurrence or detection of certain events on the media stream Event detection and notifications is performed using the X Detect SIP message header and only when establishing a SIP dialog For supporting some events certain device configurations need to be performed The table below lists the support event types and subtypes and the corresponding device configurations if required Table 7 2 Supported X Detect Event Types Events Type Subtype Required Configuration AMD voice EnableDSPIPMDetectors 1 automatic _ silence AMDTimeout 2000 msec unknown CPT SIT SITDetectorEnable 1 UserDefinedToneDetectorEnable 1 FAX CED IsFaxUsed 0 or IsFaxUsed 0 and FaxTransportMode 0 modem VxxModemtTransportT ype 3 PTT voice start EnableDSP IPMDetectors 1 voice end SIP User s Manual 358 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities The X Detect event notification process is as follows 1 For
194. CAS 4 Block Blocks trunk CAS only Notes The default behavior value 0 is as follows ISDN Use Service messages on supporting variants and use Alarm on non supporting variants CAS Use Alarm When updating this parameter value at run time you must stop the trunk and then restart it for the update to take effect To determine the method for setting Out Of Service state for all trunks i e per device use the DigitalOOSBehavior parameter refer to Configuring the Digital Gateway Parameters on page 207 The Din the ini file parameter name represents the trunk number where 0 is the first trunk Determines the method for playing a ringback tone RBT to the Trunk side In the ini file parameter ID depicts the Trunk number where 0 is the first trunk 1 Not configured use the value of the parameter PlayRBTone2Tel 0 Don t Play The device configured with ISDN CAS protocol type doesn t play an RBT No PI is sent to the ISDN unless the parameter ProgressIndicator2ISDN_ID is configured differently 1 Play on Local The device configured with CAS protocol type plays a local RBT to PSTN upon receipt of a 180 Ringing response with or without SDP Note Receipt of a 183 response doesn t cause the device configured with CAS to play an RBT unless SIP183Behaviour 1 The device configured with ISDN protocol type operates according to the parameter LocallSDNRBSource 1 If the devic
195. Codes Mediant 2000 The figure below illustrates a typical device applications VoIP network Figure 1 1 Mediant 2000 Typical Application Mediant 2000 Digital Gateway SIP Service Node MediaPack Analog Gateway Mediant 2000 Digital Gateway Phones T7 Phones Nee 1 1 SIP Overview Session Initiation Protocol SIP is an application layer control signaling protocol used on the gateway for creating modifying and terminating sessions with one or more participants These sessions can include Internet telephone calls media announcements and conferences SIP invitations are used to create sessions and carry session descriptions that enable participants to agree on a set of compatible media types SIP uses elements called Proxy servers to help route requests to the user s current location authenticate and authorize users for services implement provider call routing policies and provide features to users SIP also provides a registration function that enables users to upload their current locations for use by Proxy servers SIP implemented in the gateway complies with the Internet Engineering Task Force IETF RFC 3261 refer to http www ietf org SIP User s Manual 16 Document LTRT 68808 SIP User s Manual 2 Configuration Concepts 2 Configuration Concepts You can configure the device s parameters including upgrading the software and uploading configuration and auxiliary files using the following tools
196. D InterfaceName 0 10 33 174 50 16 10 33 0 1 Media I b Click the Submit button to save your changes Note Configure the OAM parameters only if the OAM networking parameters are different from the networking parameters used in the Single IP Network mode 5 Configure the IP Routing table to define static routing rules for the OAMP and Control networks since a default gateway isn t supported on these networks Version 5 6 389 November 2008 A ge AudioCodes Mediant 2000 a Open the IP Routing Table page refer to Configuring the IP Routing Table on page 62 Figure 8 6 Static Routes for OAM Control in IP Routing Table annaas Delete Destination IP Gateway IP gt Hop Row Address Destination Mask Address mL Count Interface b Use the Add New Entry to add the routing rules listed in the following table Destination IP Address Destination Mask Gateway IP Address Hop Count Interface 87 66 15 8 255 255 255 255 10 13 0 1 20 Control 85 44 115 50 255 255 255 0 10 31 0 1 20 OAMP 6 Save your changes to flash memory refer to Saving Configuration on page 230 and reset the device refer to Resetting the Device on page 228 8 9 3 2 Integrating Using the ini File The procedure below describes how to integrate the device into a multiple IPs network with VLANs using the ini file The procedure below is based on the example setup described in Gettin
197. DN termination side choose User side If the D channel alarm is indicated choose Network Side Indicates the NFAS group number NFAS member for the selected trunk x identifies the Trunk ID 0 Non NFAS trunk default 1 to 9 NFAS group number Trunks that belong to the same NFAS group have the same number With ISDN Non Facility Associated Signaling you can use single D 87 November 2008 A Ee AudioCodes Mediant 2000 ini File Field Name Web Parameter Name NFAS Interface ID ISDNNFASIinterfacelD_x D channel Configuration DChConfig_x Enable Receiving of Overlap Dialing ISDNRxOverlap_x Local ISDN Ringback Tone Source LocalISDNRBSource_ D SIP User s Manual Valid Range and Description channel to control multiple PRI interfaces Notes This parameter is applicable only to T1 ISDN protocols Fora detailed description on NFAS refer to ISDN Non Facility Associated Signaling NFAS on page 398 Defines a different Interface ID for each T1 trunk The valid range is 0 to 100 The default interface ID equals to the trunk s ID x identifies the trunk ID Notes To set the NFAS interface ID configure ISDNIBehavior_x to include 512 feature per T1 trunk Fora detailed description on NFAS refer to ISDN Non Facility Associated Signaling NFAS on page 398 Defines primary backup optional and B channels only The ini file parameter x represents the Trunk ID
198. DN is used DNS resolution is performed according to DNSQueryType Ifthe string ENUM is specified for the destination IP address an ENUM query containing the destination phone number is sent to the DNS server The ENUM reply includes a SIP URI used as the Request URI in the outgoing INVITE and for routing if Proxy is not used The IP address can include wildcards The x wildcard is used to represent single digits e g 10 8 8 xx represents all addresses between 10 8 8 10 to 10 8 8 99 The wildcard represents any number between 0 and 255 e g 10 8 8 represents all addresses between 10 8 8 0 and 10 8 8 255 For available notations refer to Dialing Plan Notation on page 168 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 This ini file table parameter configures the routing of IP to Tel calls to Trunk Groups or Inbound IP Routing for IP to IP calls The format of this parameter is as follows PSTNPrefix FORMAT PsinPrefix_Index PstnPrefix_DestPrefix PstnPrefix_TrunkGroupld PstnPrefix_SourcePrefix PstnPrefix_SourceAddress PstnPrefix_Profileld PstnPrefix_SrclIPGroupID PstnPrefix_DestHostPrefix PstnPrefix_SrcHostPrefix PSTNPrefix For example PSTNPrefix FORMAT PstnPrefix_Index PstnPrefix_DestPrefix PstnPrefix_TrunkGroupld PstnPrefix_SourcePrefix PstnPrefix_SourceAddress PstnPrefix_Profileld PstnPrefix_SrclPGroupID
199. DP Connection line to the device s own IP address and adds a a sendonly line to the SDP For a description of this parameter refer to SIP General Parameters on page 121 This ini file table parameter configures the Account table for registering and or authenticating digest a Trunk Group e g IP PBX to a Serving IP Group e g Internet Telephony Service Provider ITSP The format of this parameter is as follows Account FORMAT Account_Index Account_ServedTrunkGroup Account_Served IPGroup Account_ServinglPGroup Account_Username Account_Password Account_HostName Account_Register Account_ContactUser Account For example Account FORMAT Account_Index Account_ServedTrunkGroup Account_ServedIPGroup Account_Serving PGroup Account_Username Account_Password Account_HostName Account_Register Account_ContactUser Account 0 1 1 1 user 1234 acl 1 ITSP1 Account Notes This table can include up to 10 indices You can define multiple table indices having the same 288 Document LTRT 68808 SIP User s Manual Parameter IPGroup NumberOfActiveDialogs PrackMode AssertedidMode PAssertedUserName Version 5 6 4 ini File Configuration Description ServedTrunkGroup with different ServinglPGroups username password HostName and ContactUser This provides the capability for registering the same Trunk Group to several ITSP s i e Serving IP Groups For configur
200. Dela rites chee le Backup TI trunk DchConfig 2 2 24 B channel NFAS trunk DeEnNContigi ORSR 24 B channel NFAS trunk 9 4 1 The NFAS parameters are described in PSTN Parameters on page 303 NFAS Interface ID Several ISDN switches require an additional configuration parameter per T1 trunk that is called Interface Identifier In NFAS T1 trunks the Interface Identifier is sent explicitly in Q 931 Setup Channel Identification IE for all NFAS trunks except for the B channels of the Primary trunk refer to note below The Interface ID can be defined per member T1 trunk of the NFAS group and must be coordinated with the configuration of the Switch The default value of the Interface ID is identical to the number of the physical T1 trunk O for the first trunk 1 for the second T1 trunk and so on up to the maximum number of trunks To define an explicit Interface ID for a T1 trunk that is different from the default use the following parameters m ISDNIBehavior_x 512 x 0 to the maximum number of trunks identifying the device s physical trunk Em ISDNNFASInterfacelD_x ID x 0 to 255 Usually the Interface Identifier is included in the Q 931 Setup Channel Identification IE only on T1 trunks that doesn t contain the D channel Calls initiated on B channels of the Primary T1 trunk by default don t contain the Interface Identifier Setting the parameter ISDNIBehavior_x to 2048 forces the inclusion of
201. EEEE RELE T eee 358 7 7 RTP Multiplexing ThroughPacket isictindininicsniniieiinimeiiaaaansna OGO 7 8 Dynamic Jitter Buffer Operation ivicicdiisiaiiaicmusmcimiceavainen E o i 7 9 Configuring Alternative Routing Based on Tonne dii Qos EREE EARE TE 361 79 1 Alternative Routing Mechanism ssssenswicrssorasninena ikna GO 7 9 2 Determining the Availability of Destination IP Addresses E AAEE 7 9 3 PSTN Fallback as a Special Case of Alternative Routing Aleem 794 Relevant PSU os cranes da mraivoual stan aA S EAEE EREA 362 7 10 Supported RADIUS Altria saccscisieceicirsvasa oderesencioioce taearcvrstcconortense OO 7 11 Call Detail Record EAE A EEEE EEE AET E E E L EAEE E PAE E ols 7 12 Trunk to Trunk Routing Eai EEE E ceeteneas ETE ce eerie renee 367 7 13 Proxy or Registrar Registration Example casicisicisrcisscicieisiaininesisnaacieee 7 14 Configuration Examples aii cecinccrccstcvrcrcmeceinccaevcctieenis EE A PEE E TET m PAT PET ON aaa a a a a a jaan 7 14 2 SIP Authentication Example ere 7 14 3 SIP Trunking between Enterprise and ITSPs BRRR Sunni 7 15 Working with Supplementary Services E AEE AES EA ees 377 Se Caloi and RENE caren es ect wnteaaternenniaiebicraandubivinlancdiamudeosimdsiembontiaamaneneies 377 Pl I TN aa EAO E aa 8 Networking Capabilities sss ssisissssnisisssinisnnunaninnnunaninnnunssusnnunaninnnununannnananinnns O O 8 1 Ethernet Interface Configuration 2 04 AET TE EAE eee 379 8 2
202. Ethernet Interface Redundancy E E E EEE E EE E E A 8 3 NAT Network Address Translation SUP OM cere ees BBO fone Mes Ll ieee ce teenaee cee reme eee a cOepenT ret ri 381 8 3 2 First Incoming Packet Mechanism cccccccccceeeeeeeeeeeeeceeeeeeeeeeeeeeeeeeneeneeeee 382 Pei NEUD Pak acco rsh hts acca eles aaee iaae aaka ENE dol lacie 382 8 4 IP Multicasting Ar TA A AT EEE TET T EE T A TATE EE 383 8 5 Rob stRecepton of RIP SUCAMS iiiiiisciicahinicinasaisisisiibiainianiiininninili O 86 Multiple Routers Suppo iis chacs eros aati ease ee 8 7 Simple Network Time Protocol Supari A E E E S ees ere ee E A 383 8 8 IP QoS via Differentiated Services DiffServ ccccccseseeeeeeeeseeeeeeeeeeeee OOF 69 VLANS and Multiple IPS a ncuiiscixidecatnee aaan Go FTP IP Sn a a aaa ai 6 9 2 JEEE 802 1p Q VLANs and Priority iriserai 8 9 3 Getting Started with VLANS and Multiple IPS cccccccccssecesseecsteeeesseeeseees A 8 9 3 1 Integrating Using the Web Interface ba 8 9 3 2 Integrating Using the ini File se ssiccssesccssasscocascnccesicseensenmtereussererten eee 9 Advanced PSTN Configuration ssssssscccssssssssssesssscsesssssssssensssseeessess 9 1 Clock Settings ene eee ee eee PAETE RI E EE eT PEPA A 393 92 Release Reason Mapping oii disiiasdsisianiaisisnasianannininuetinuneimamaDeS 921 Roason Heddr escassas aa aaa ee 9 2 2 Fixed Mapping of ISDN Release Reason to SIP RESPONSE sesssesecesseece
203. Fa Wa wi AudioCodes Mediant Media Gateways SIP Mediant 2000 User s Manual Version 5 6 Document LTRT 68808 November 2008 SP Uses Manual 00ers Table of Contents 11 SIP CSW ceceni a a ai O 2 Configuration GOMGR OIE eiiscsisiiosisisbicassasiesiensdbimmndccmmuaavanmneabimeniioimnninscinuusnabes TO 3 1 3 4 Configura n De E E EE vl 3 wi AudioCodes Mediant 2000 34 2 6 Configuring the DSP Templates ccc cscccesissnincndenniieneencenanienes 80 3 4 2 7 Configuring Media Security T 3 4 3 PSTN Settings 3 4 3 1 Configuring the Trunk Settings 3 4 3 2 Configuring the CAS State Machines 3 4 4 SS7 Configuration ete ee sees E ees 3 4 5 Sigtran Configuration E E E Cr te rrr eer rrr er errr Saa basne SAS Secunity SOUINOS ssiri ani a t Sanaa E R 3 4 6 1 Configuring the Web User ACCOUNTS oosa s saans pans 3 4 6 2 Configuring the Web and Telnet ACCESS LiSt scesssssssseeveeeeeeee 346 3 Configuring the Firewall Settings i csissccsnsienssiacpiadenimnnnae nie SAGA Contigquring the Certificats renerrien RRA 105 3 4 6 5 Configuring the General eea Settings E EAT 3 4 6 6 Configuring the IPSec Table NEA EAA E 3 4 6 7 Configuring the IKE Table 3 4 7 Protocol Configuration 3 4 7 1 Configuring the Protocol Definition Parameters 3 4 7 2 Configuring the SIP Advanced Parameters adn raien enanta anea 3 4 7 3 Configuring the Number Manipul
204. Group Table Page Trunk IP Profile Source Dest Host Prefix Source Most Prefix Dest Phone Prefix E cove Group tO PGroup ID 8 In the Tel to IP Routing page refer to Tel to IP Routing Table on page 175 configure Tel to IP routing rules for calls to ITSPs see 1 below and to local PSTN see 2 below Figure 7 10 Configuring Tel to IP Routing to ITSPs in Tel to IP Routing Table Page Sre Trunk Dest Phone Prefix Source Phone Prefix x Dest IP Address k IP Profile 10 Group ID gt Gro Gx 1 1 0 3 4 5 1 O 6 7 8 O 3l 02 10 13 413 SIP User s Manual 376 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 7 15 Working with Supplementary Services The device supports the following supplementary services Call Hold and Retrieve refer to Call Hold and Retrieve on page 377 Call Transfer refer to Call Transfer on page 377 Call Forward when a callRerouting IE is received in a FACILITY message in response to an outgoing SETUP message the device sends a 3xx response to the IP side including the callRerouting destination number only applicable to QSIG protocol Call Waiting The device SIP users are only required to enable the Hold and Transfer features By default the Call Forward supporting 30x redirecting responses and Call Waiting receipt of 182 response features are enabled All call participants must support the specific supplementary service that is used Whe
205. IG 4 SETUP Only ISDN 5 MATRA AASTRA QSIG The following digit pattern parameters apply only to voice mail applications that use the DTMF communication method For the available patterns syntaxes refer to the CPE Configuration Guide for Voice Mail Forward on Busy Digit Pattern Internal DigitPatternForwardOnBusy Forward on No Answer Digit Pattern Internal DigitPatternForwardOnNoAnswer Forward on Do Not Disturb Digit Pattern Internal DigitPatternForwardOnDND Forward on No Reason Digit Pattern Internal DigitPatternForwardNoReason Forward on Busy Digit Pattern External DigitPatternForwardOnBusyExt Forward on No Answer Digit Pattern External DigitPatternForwardOnNoAnswerExt Version 5 6 Determines the digit pattern used by the PBX to indicate call forward on busy when the original call is received from an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on no answer when the original call is received from an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on do not disturb when the original call is received from an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward with no reason when the original call is rec
206. IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 This ini file table parameter configures the Multiple Interface table for configuring logical IP addresses The format of this parameter is as follows InterfaceTable FORMAT InterfaceTable_Index InterfaceTable_ApplicationTypes InterfaceTable_IPv6InterfaceMode InterfaceTable_IPAddress InterfaceTable_PrefixLength InterfaceTable_Gateway InterfaceTable_VlanlID InterfaceTable_InterfaceName InterfaceTable 0 6 0 192 168 85 14 16 192 168 0 1 1 myAll InterfaceTable For example InterfaceTable FORMAT InterfaceTable_Index InterfaceTable_ApplicationTypes InterfaceTable_IPv6InterfaceMode InterfaceTable_IPAddress InterfaceTable_PrefixLength InterfaceTable_Gateway InterfaceTable_VlanID InterfaceTable_InterfaceName InterfaceTable 0 0 0 192 168 85 14 16 0 0 0 0 1 ManagementIF InterfaceTable 1 2 0 200 200 85 14 24 0 0 0 0 200 myControllF InterfaceTable 2 1 0 211 211 85 14 24 211 211 85 1 211 myMedialF InterfaceTable The above example configures three network interfaces OAMP Control and Media applications Notes To configure the Multiple Interface table using the Web interface refer to Configuring the Multiple Interface Table on page 53 Fora description of configuring ini file table parameters refer to Structure of ini File Table Parameters on page 25
207. IP address or FQDN from which it is loaded For example http server_name file https server_nameffile Note The maximum length of the URL address is 99 characters Specifies the name of the Voice Prompts file and the location of the server IP address or FQDN from which it is loaded For example http server_nameffile https server_nameffile Note The maximum length of the URL address is 99 characters Specifies the name of the CAS file and the location of the server IP address or FQDN from which it is loaded For example http server_name file https server_nameffile Note The maximum length of the URL address is 99 characters Specifies the name of the TLS trusted root certificate file and the location URL from where it s downloaded Specifies the name of the TLS certificate file and the location URL from where it s downloaded Specifies the name of the User Information file and the location of the server IP address or FQDN from which it is loaded For example http server_name file https server_name file Note The maximum length of the URL address is 99 characters Enables disables the Automatic Update mechanism for the cmp file 0 The Automatic Update mechanism doesn t apply to the cmp file default 1 The Automatic Update mechanism includes the cmp file Determines the number of minutes the device waits between automatic updates The default value is 0 the update at fixed interva
208. IP packet the device searches this table for an entry that matches the requested destination host network If such an entry is found the device sends the packet to the indicated router If no explicit entry is found the packet is sent to the default gateway configured in the IP Settings page refer to Configuring the IP Settings on page 50 gt To configure static IP routing take these 3 steps 1 Open the IP Routing Table page Configuration tab gt Network Settings menu gt IP Routing Table page item Figure 3 38 IP Routing Table Page Delete Row Destination IP Address Destination Mask Gateway IP Address Metric Interface Delete Selected Entries Destination IP Address Destination Mask Gateway IP Address Interface o Add New Entry 2 In the Add a new table entry group add a new static routing rule according to the parameters described in the table below 3 Click Add New Entry the new routing rule is added to the IP routing table To delete a routing rule from the table select the Delete Row check box that corresponds to the routing rule entry and then click Delete Selected Entries Table 3 11 IP Routing Table Description Parameter Description Destination IP Address Specifies the IP address of the destination host RoutingTableDestinationsColumn network Destination Mask
209. ISDN E 164 Network Specific 1 4 Public ISDN E 164 Subscriber 1 6 Public ISDN E 164 Abbreviated For NI 2 and DMS 100 ISDN variants the valid combinations of TON and NPI for calling and called numbers are Plan Type 0 0 Unknown Unknown 1 1 International number in ISDN Telephony numbering plan 1 2 National number in ISDN Telephony numbering plan 1 4 Subscriber local number in ISDN Telephony numbering plan 9 4 Subscriber local number in Private numbering plan SecureCallsFromIP For a description of this parameter refer to Advanced Parameters on page 151 AltRouteCauseTel2IP This ini file table parameter configures SIP call failure reason values received from the IP side If a call is released as a result of one of these reasons the device attempts to locate an alternative route to the call in the Tel to IP Routing table if Proxy is not used or used as a redundant Proxy when Proxy is used The format of this parameter is as follows AltRouteCauseTel2IP FORMAT AltRouteCauseTel2IP_Index AltRouteCauseTel2IP_ReleaseCause AltRouteCauseTel2IP For example AltRouteCauseTel2IP AltRouteCauseTel2IP 0 486 Busy Here AltRouteCauseTel2IP 1 480 Temporarily Unavailable AltRouteCauseTel2IP 2 408 No Response AltRouteCauseTel2IP Notes The 408 reason can be used to specify no response from the remote party to the INVITE request This parameter can include up t
210. K response closing the REGISTER transaction SIP 2 0 200 OK Via SIP 2 0 UDP 10 1 1 200 Woms lt io LA2 LO i 1 200 gt itag le2s 940 Mos gifs a PACOS Call ID 654982194 10 1 1 200 Cseq 1 REGISTER Date Thu 26 Jul 2001 09 34 42 GMT Server Columbia SIP Server 1 17 Content Length 0 Contact lt sip 122 10 1 1 200 gt expires Thu 26 Jul 2001 10 34 42 GMT action proxy q 1 00 Contact lt 122 10 1 1 200 gt expires Tue 19 Jan 2038 03 14 07 GMT action proxy q 0 00 Expires Thu 26 Jul 2001 10 34 42 GMT 7 14 3 SIP Trunking between Enterprise and ITSPs By implementing the device s enhanced and flexible routing configuration capabilities using Proxy Sets IP Groups and Accounts you can design complex routing schemes This section provides an example of an elaborate routing scheme for SIP trunking between an Enterprise s PBX and two Internet Telephony Service Providers ITSP using AudioCodes device Version 5 6 373 November 2008 A EA AudioCodes Mediant 2000 Scenario In this example the Enterprise wishes to connect its TDM PBX to two different ITSPs by implementing a device in its network environment It s main objective is for the device to route Tel to IP calls to these ITSPs according to a dial plan The device is to register on behalf of the PBX to each ITSP which implements two servers for redundancy and load balancing The Register messages are to use different URI s in the From
211. Loeding Device Reset z Device Software Update a Parameter Grou P Access to Restricted Domans Non Aorced Access Sens ve Parameters Value Change yY suma mansgemer somnos SA Bove Parameter List a Syslog Semngs Syslog Server IP Address Syslog Server Port Enable Syslog Analog Ports Miter v SN Setengs SNMP Trap Destinatons SNMP Correnenty String SNMP V3 Table SNMP Trosed Managers Cosabdle SNMP Trap Meneger Host Nome a Achyty Types to Report vis Achvity Log Messages 3 3 3 3 Modifying and Saving Parameters When you change parameter values on a page the Edit symbol appears to the right of these parameters This is especially useful for indicating the parameters that you have currently modified before applying the changes After you save your parameter modifications refer to the procedure described below the Edit symbols disappear SIP User s Manual 28 Document LTRT 68808 SIP User s Manual 3 Web Based Management Figure 3 9 Editing Symbol after Modifying Parameter Value wv Priority Settings Network Priority Media Premium Priority Control Premium Priority Gold Priority Bronze Priority a Differential Services gt To save configuration changes on a page to the device s volatile memory RAM take this step m Click the Submit y button which is located near the bottom of the page in which you are working modifications to parameters with on the fly capa
212. NAPTR query is not performed Note When enabled NAPTR SRV queries are used to discover Proxy servers even if the parameter DNSQueryType is disabled Number of retransmitted INVITE REGISTER messages before the call is routed hot swap to another Proxy Registrar The valid range is 1 to 30 The default value is 3 Note This parameter is also used for alternative routing using the Tel to IP Routing table or Outbound IP Routing table if EnableSBC is set to 1 If a domain name in the table is resolved into two IP addresses and if there is no response for HotSwapRtx retransmissions to the INVITE message that is sent to the first IP address the device immediately initiates a call to the second IP address Determines whether the device uses its IP address or gateway name in keep alive SIP OPTIONS messages 0 No Use the device s IP address in keep alive OPTIONS messages default 1 Yes Use Gateway Name SIPGatewayName in keep alive OPTIONS messages The OPTIONS Request URI host part contains either the device s IP address or a string defined by the parameter SIPGatewayName The device uses the OPTIONS request as a keep alive message to its primary and redundant Proxies i e the parameter EnableProxyKeepAlive is set to 1 User name used for Registration and Basic Digest authentication with a Proxy Registrar server The parameter doesn t have a default value empty string Note Applicable only if sin
213. Name M2K_usa_tones dat Delete Coder Table File Name codertable test dat Delete SIP User s Manual 244 Document LTRT 68808 SIP User s Manual 3 Web Based Management The Board Type field number depicts the following devices m Mediant 2000 31 m TP 1610 24 gt To delete any of the loaded files take this step Click the Delete button corresponding to the files that you want to delete Deleting a file takes effect only after the device is reset refer to Resetting the Device on page 228 3 6 1 5 Viewing Performance Statistics The Performance Statistics page provides read only device performance statistics This page is refreshed with new statistics every 60 seconds The duration that the current statistics has been collected is displayed above the statistics table gt To view performance statistics take the following step m Open the Performance Statistics page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Performance Statistics page item Figure 3 113 Performance Statistics Page Performance statisties LSNA Statisti Statistics for 2811 seconds Active TDM channels Active DSP resources Active analog channels Active G 711 channels Average voice delay ms Average voice jitter ms Total RTP packets TX _ Total RTP packets RX Total call attempts Reset Statistics gt To reset the performance statistics to
214. Notation on page 168 The source IP address of an IP to Tel call obtained from the Contact header in the INVITE message that can be used for routing decisions Notes You can configure from where the source IP address is obtained using the parameter SourcelPAddressInput refer to Routing General Parameters on page 171 The source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all the addresses between 10 8 8 10 to 10 8 8 99 The source IP address can include the asterisk wildcard to 183 November 2008 A ge AudioCodes Mediant 2000 Parameter Description represent any number between 0 and 255 For example 10 8 8 represents all addresses between 10 8 8 0 and 10 8 8 255 Trunk Group ID The Trunk Group to which incoming SIP calls are assigned that PstnPrefix_TrunkGroupld match all or any combination including only a single parameter of the parameters described above Profile ID The IP Profile configured in IP Profile Settings on page 193 that PstnPrefix_Profileld is assigned to the routing rule Source IP Group ID The source IP Group 1 9 associated with the incoming IP to Tel PstnPrefix_SrclPGroupID call This is the IP Group from where the INVITE message originated This IP Group can later be used as the Serving IP Group in the Account table refer to Configuring the Account Table on page 204 for obtaining authentication user name pass
215. P calls define the IP address as 0 0 0 0 The IP address 127 0 0 1 can be used when the IP address of the device itself is unknown for example when DHCP is used The destination port 180 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Transport Type PREFIX_TransportType Dest IP Group ID PREFIX_DestIPGroupID IP Profile ID PREFIX_Profileld Status Description The transport layer type for sending the outbound SIP IP calls 1 Not Configured 0 UDP 1 TCP 2 TLS Note When Not Configured is selected the transport type defined by the parameter SIPTransportType refer to SIP General Parameters on page 121 is used The IP Group 1 to 9 to where you want to route the outbound IP to IP call The INVITE messages are sent to the IP address es defined for the Proxy Set that is associated with this IP Group If you select an IP Group it is unnecessary to configure a destination IP address in the Dest IP Address field above However if both parameters are configured the INVITE message is sent only to the IP Group If the destination IP Group is of type USER the device searches for a match between the request URI of the received INVITE to an AOR registration record in the device s internal database The INVITE is then sent to the IP address of the registered contact If the parameter AlwaysUseRouteTable AlwaysUseRouteTable is set to Enable
216. P responses are sent to the port specified in the Via header Determines whether to add user phone string in SIP URI 0 No user phone string isn t used in SIP URI 1 Yes user phone string is part of the SIP URI default 125 November 2008 ca AudioCodes Parameter Use user phone in From Header IsUserPhonelnFrom Use Tel URI for Asserted Identity UseTelURIForAssert edID Tel to IP No Answer Timeout IPAlertTimeout Enable Remote Party ID EnableRPlheader Add Number Plan and Type to RPI Header AddTON2RPI SIP User s Manual Mediant 2000 Description Determines whether to add user phone string in the From header 0 No Doesn t use user phone string in From header default 1 Yes user phone string is part of the From header Determines the format of the URI in the P Asserted Identity and P Preferred Identity headers 0 Disable sip default 1 Enable tel Defines the time in seconds that the device waits for a 200 OK response from the called party IP side after sending an INVITE message If the timer expires the call is released The valid range is 0 to 3600 The default value is 180 Enables Remote Party ID RPI headers for calling and called numbers for Tel to IP calls 0 Disable default 1 Enable RPI headers are generated in SIP INVITE messages for both called and calling numbers Determines whether the TON PLAN parameter
217. PD message For detailed information on DPD refer to the Product Reference Manual Defines the supported versions of SSL TLS Secure Socket Layer Transport Layer Security 0 SSL 2 0 3 0 and TLS 1 0 SSL 2 0 SSL 3 0 and TLS 1 0 are supported default 1 TLS 1 0 Only only TLS 1 0 is used When set to 0 SSL TLS handshakes always start with SSL 2 0 and switch to TLS 1 0 if both peers support it When set to 1 TLS 1 0 is the only version supported clients attempting to contact the device using SSL 2 0 are rejected Defines the time interval in minutes between TLS Re Handshakes initiated by the device The interval range is 0 to 1 500 minutes The default is 0 i e no TLS Re Handshake Determines the device s behavior when acting as a server for TLS connections 0 Disable The device does not request the client certificate default 1 Enable The device requires receipt and verification of the client certificate to establish the TLS connection Notes The SIPS certificate files can be changed using the parameters HTTPSCertFileName and HTTPSRootFileName This parameter cannot be changed on the fly and requires a device reset Determines whether the device verifies the Subject Name of a remote certificate when establishing TLS connections 0 Disable Disable default 1 Server Only Verify Subject Name only when acting as a server for the TLS connection 2 Server am
218. PREFIX DestinationPrefix PREFIX DestAddress PREFIX SourcePrefix PREFIX Profileld PREFIX MeteringCode PREFIX DestPort Prefix 1 10 8 24 12 PREFIX al 5 5 5 5 IP address of the device in the opposite location Channel selection by Phone number ChannelSelectMode 0 Profiles can be used do define different coders per B channels such as Transparent coder for B channels timeslot 16 that carries PRI signaling TrunkGroup FORMAT TrunkGroup Index TrunkGroup TrunkGroupNum TrunkGroup FirstTrunkId TrunkGroup LastTrunkId TrunkGroup_FirstBChannel TrunkGroup_LastBChannel TrunkGroup FirstPhoneNumber TrunkGroup Profileld TrunkGroup Module TrunkGroup TrunkGroup TrunkGroup TrunkGroup TrunkGroup TrunkGroup TrunkGroup TrunkGroup 1 TrunkGroup CoderName FORMAT CoderName Index CoderName Type CoderName PacketInterval CoderName rate CoderName PayloadType CoderName Sce CoderName o moi 2srnie CoderName 1 Transparent CoderName 5 USAS IL p CoderName 6 Transparent CoderName TelProfile FORMAT TelProfile Index TelProfile ProfileName TelProfile TelPreference TelProfile CodersGroupID TelProfile IsFaxUsed TelProfile JitterBufMinDelay TellProf ile JitterBufOptEacCtor Lellprort leur PDair kSermw TelProfile SigIPDiffServ TelProfile DtmfVolume TelProfile InputGain TelProfile VoiceVolume TelProfile EnableReversePolarity TelProfile EnableCurrentDisconnect TelPr
219. Parameter Forking Handling Mode ForkingHandlingMod e Enable Reason Header EnableReasonHeade r Mediant 2000 Description The valid range is a string of up to 10 characters The default is an empty string Determines how the device reacts to forking of outgoing INVITE messages by the Proxy 0 Sequential handling The device opens a voice stream toward the first 18x SIP response that includes an SDP and disregards any 18x response with an SDP received thereafter default 1 Parallel handling The device opens a voice stream toward the first 18x SIP response that includes an SDP and re opens the stream toward any subsequent 18x responses with an SDP Note Regardless of the ForkingHandlingMode value once a SIP 200 OK response is received the device uses the RTP information and re opens the voice stream if necessary Enables disables the usage of the SIP Reason header 0 Disable 1 Enable default Retransmission Parameters SIP T1 Retransmission Timer msec SipT1Rtx SIP T2 Retransmission Timer msec SipT2Rtx SIP Maximum RTX SIPMaxRtx The time interval in msec between the first transmission of a SIP message and the first retransmission of the same message The default is 500 Note The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx For example assuming that SipT1Rtx 500 and SipT2Rtx
220. Play Ringback tone isn t played 1 Play Local Ringback tone is played to the Tel side of the call when 180 183 response is received 128 Document LTRT 68808 SIP User s Manual Parameter Use Tgrp Information UseSIPTgrp Enable GRUU EnableGRUU Version 5 6 3 Web Based Management Description 2 Play According to Early Media Ringback tone is played to the Tel side of the call if no SDP is received in 180 183 responses If 180 183 with SDP message is received the device cuts through the voice channel and doesn t play ringback tone default Determines whether the SIP tgrp parameter which specifies the Trunk Group to which the call belongs is used according to RFC 4904 For example INVITE sip 163055501 00 tgrp 1 trunk context example com 10 1 0 3 user phone SIP 2 0 0 Disable The tgrp parameter isn t used default 1 Send Only The Trunk Group number is added to the tgrp parameter value in the Contact header of outgoing SIP messages If a Trunk Group number is not associated with the call the tgrp parameter isn t included If a tgrp value is specified in incoming messages it is ignored 2 Send and Receive The functionality of outgoing SIP messages is identical to the functionality described in option 1 In addition for incoming SIP messages if the Request URI includes a tgrp parameter the device routes the call according to that value if possible If the Con
221. Prefix PREFIX_Profileld PREFIX_MeteringCode PREFIX_DesitPort PREFIX_SrclPGroupID PREFIX_DestHostPrefix PREFIX_DestIPGroupID PREFIX_SrcHostPrefix PREFIX_TransportType PREFIX_SrcTrunkGroupID PREFIX For example PREFIX FORMAT PREFIX_Index PREFIX_DestinationPrefix PREFIX_DestAddress PREFIX_SourcePrefix PREFIX_Profileld PREFIX_MeteringCode PREFIX_DestPort PREFIX_SrclPGroupID PREFIX_DestHostPrefix PREFIX_DestIPGroupID PREFIX_SrcHostPrefix 315 November 2008 ca AudioCodes Parameter PSTNPrefix SIP User s Manual Mediant 2000 Description PREFIX_TransportType PREFIX_SrcTrunkGroupID PREFIX 0 quest 0 255 1 1 1 1 PREFIX 1 20 10 33 37 77 0 255 1 2 0 1 PREFIX 2 30 10 33 37 79 1 255 1 1 2 1 PREFIX Notes This parameter can include up to 50 indices Fora description of these parameters refer to the corresponding Web parameters in Tel to IP Routing Table on page 175 or Outbound IP Routing Table on page 178 The parameter PREFIX_MeteringCode is not applicable The destination and source phone prefixes PREFIX_DestinationPrefix and PREFIX_SourcePrefix respectively can be a single number or a range of numbers Parameters can be skipped using two dollar symbols for example Prefix 10 2 10 2 202 1 The destination IP address PREFIX_DestAddress can be either in dotted decimal notation or FQDN If an FQ
222. Proxy Set ID refer to Proxy Sets Table on page 141 associated with this Serving IP Group The Request URI hostname in the INVITE and REGISTER messages except for Per Account registration modes is set to the value of the field SIP Group Name defined in the IP Group table refer to Configuring the IP Groups on page 201 If no Serving IP Group ID is selected the INVITE messages are sent to the default Proxy or according to the Tel to IP Routing Table refer to Tel to IP Routing Table on page 175 or Outbound IP Routing Table if EnableSBC is set to 1 refer to Outbound IP Routing Table on page 178 Note If the parameter PreferRouteTable is set to 1 refer to Proxy amp Registration Parameters on page 132 the routing rules in the Tel to IP Routing Table or Outbound IP Routing Table prevail over the selected Serving IP Group ID Gateway Name The host name used in the From header in INVITE messages and TrunkGroupSettings Gate as a host name in From To headers in REGISTER requests If not wayName configured the global parameter SIPGatewayName is used instead Contact User This is used as the user part in the Contact URI in INVITE TrunkGroupSettings Cont messages and as a user part in From To and Contact headers in actUser REGISTER requests This is applicable only if the field Registration Mode is set to Per Account and the Registration through the Account table is successful
223. PstnPrefix_DestHostPrefix PstnPrefix_SrcHostPrefix 316 Document LTRT 68808 SIP User s Manual Parameter RemovePrefix RouteModelP2Tel RouteModeTel2IP SwapRedirectNumber Prefix2RedirectNumber SourceManipulationMode Version 5 6 4 ini File Configuration Description PstnPrefix 0 100 1 200 0 2 PstnPrefix 1 2 1 3 acl joe PSTNPrefix Notes This parameter can include up to 24 indices Fora description of these parameters refer to the corresponding Web parameters in IP to Trunk Group Routing Table on page 181 or Inbound IP Routing Table on page 184 for IP to IP calls To support the In Call Alternative Routing feature you can use two entries that support the same call but assigned with a different Trunk Group The second entry functions as an alternative selection if the first rule fails as a result of one of the release reasons listed in the AltRouteCauselP2Tel table Selection of Trunk Groups for IP to Tel calls is according to destination number source number and source IP address The source IP address SourceAddress can include the x wildcard to represent single digits For example 10 8 8 xx represents all IP addresses between 10 8 8 10 and 10 8 8 99 The source IP address SourceAddress can include the asterisk wildcard to represent any number between 0 and 255 For example 10 8 8 represents all addresses between 10 8 8 0 and 10 8 8
224. Q 850 Release Causes and vice versa The existing mapping of ISDN Release Causes to SIP Responses is described in Fixed Mapping of ISDN Release Reason to SIP Response on page 394 and Fixed Mapping of SIP Response to ISDN Release Reason on page 396 To override this hard coded mapping and flexibly map SIP responses to ISDN Release Causes use the ini file CauseMapISDN2SIP and CauseMapSIP2ISDN as described in ISDN and CAS Interworking Related Parameters on page 307 or the Web interface refer to Release Cause Mapping on page 189 It is also possible to map the less commonly used SIP responses to a single default ISDN Release Cause Use the parameter DefaultCauseMapISDN2IP described in ISDN and CAS Interworking Related Parameters on page 307 to define a default ISDN Cause that is always used except when the following Release Causes are received Normal Call Clearing 16 User Busy 17 No User Responding 18 or No Answer from User 19 This mechanism is only available for Tel to IP calls Reason Header The device supports the Reason header according to RFC 3326 The Reason header conveys information describing the disconnection cause of a call m Sending Reason header If a call is disconnected from the Tel side ISDN the Reason header is set to the received Q 850 cause in the appropriate message BYE CANCEL final failure response and sent to the SIP side If the call is disconnected because of a SIP reason the Reason heade
225. Registration per Account in Trunk Group Settings Page376 Figure 7 8 Configuring Accounts for PBX Registration to ITSPs in Account Table Page 4 376 Figure 7 9 Configuring ITSP to Trunk Group 1 Routing in IP to Trunk Group Table Page 376 Figure 7 10 Configuring Tel to IP Routing to ITSPs in Tel to IP Routing Table Page c ee 376 Figure G91 NAT Arie si xs a rated cea OITU GIAO OAA OIETAN 380 Figure 8 2 Multiple Network Interfaces and VLANS ccsssiedissccessssccensncseeienseonsndeoumscarsmsieune 386 Figure 8 3 VLAN Configuration in the IP Settings Page ssssssssesssisssrssssrsrersssrnaarieeresensaarsnasasssnnaanna 388 Figure 8 4 OAM Control Media IP Configuration in the IP Settings Page eceeeeeeeeeteeeeeee 389 Figure 8 5 Multiple uses lea Tae Fa eerrerrere treme ete rere tr rere ee tere Petree T tertyy eter ype tr ery 389 Figure 8 6 Static Routes forOAM Control in IP Routing Table s cisccccsseeesesvessacetsesseiteeaaien 390 SIP User s Manual 10 Document LTRT 68808 List of Tables bis 3 Table Table Table Table lt Table Table Table Table Table Table Table Table Table lt Table 2 Table lt Table lt Table lt Table lt Table lt Table lt Table lt Table lt Table lt 1e 3 wi AudioCodes Mediant 2000 Table 3 57 Management Settings Parameters ccccceccecececeeeeeeeeeeeeeaeceaeceaeeeeeeecaeseaeenereseseesneetaees 221 Table
226. Release Cause Mapping Page Release Cause Mapping from ISDN to SIP 9 650 Cause SIP Response Release Cause Mapping from SIP to ISDN 850 Cause SIP Response oe SEES bone In the Release Cause Mapping from ISDN to SIP group map up to 12 different Q 850 Release Causes to SIP Responses In the Release Cause Mapping from SIP to ISDN group map up to 12 different SIP Responses to Q 850 Release Causes Click the Submit button to save your changes To save the changes so they are available after a power fail refer to Saving Configuration on page 230 189 November 2008 A C al AudioCodes Mediant 2000 3 4 7 5 Configuring the Profile Definitions The Profile Definitions submenu includes the following page items m Coder Group Settings refer to Coder Group Settings on page 190 m Tel Profile Settings refer to Tel Profile Settings on page 192 m P Profile Settings refer to IP Profile Settings on page 193 Implementing the device s Profile features provides the device with high level adaptation when connected to a variety of equipment at both Tel and IP sides and protocols each of which requires different system behavior You can assign different Profiles behavior per call using the call routing tables m Tel to IP Routing page refer to Tel to IP Routing Table on
227. Routing table 2 Serving IP Group The device sends the SIP INVITE to the selected Serving IP Group If no Serving IP Group is selected the default IP Group is used If the Proxy server s associated with the 203 November 2008 A Ee AudioCodes Mediant 2000 Parameter Description destination IP Group is not alive the device uses the Outbound IP Routing table if the parameter IsFallbackUsed is set 1 i e fallback enabled refer to Proxy amp Registration Parameters on page 132 3 Request URI The device sends the SIP INVITE to the IP address according to the received SIP Request URI host name Note This field is available only if EnableSBC is set to 1 refer to SBC Configuration on page 163 SIP Re Routing Mode Determines the routing mode after a call redirection i e a 3xx SIP response is received or transfer i e a SIP REFER request is received 0 Standard INVITE messages that are generated as a result of Transfer or Redirect are sent directly to the URI according to the Refer To header in the REFER message or Contact header in the 3xx response default 1 Proxy Sends a new INVITE to the Proxy Note Applicable only if a Proxy server is used and the parameter AlwaysSendtoProxy is set to 0 2 Routing Table Uses the Routing table to locate the destination and then sends a new INVITE to this destination Notes When this parameter is set to 1 and the INVITE sent to t
228. SCEMode Version 5 6 147 November 2008 A Ee AudioCodes Mediant 2000 3 4 7 1 5 DTMF amp Dialing Parameters The DTMF amp Dialing page is used to configure parameters associated with dual tone multi frequency DTMF and dialing gt To configure the DTMF and dialing parameters take these 4 steps 1 Open the DTMF amp Dialing page Configuration tab gt Protocol Configuration menu gt Protocol Definition submenu gt DTMF amp Dialing page item ist Tx DTMF Option 2nd Tx DTMF Option 3rd Tx DTMF Option 4th Tx DTMF Option Sth Tx DTMF Option Digit Mapping Rules Max Digits In Phone Num Inter Digit Timeout sec Declare RFC 2833 in SDP RFC 2833 Payload Type Figure 3 63 DTMF amp Dialing Page 30 4 No RFC 2833 MAES 96 Default Destination Number 1000 Special Digit Representation Special 2 Configure the DTMF and dialing parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Parameter Max Digits in Phone Num MaxDigits Inter Digit Timeout for Overlap Dialing sec TimeBetweenDigits Declare RFC 2833 in SDP RxDTMFOption SIP User s Manual Table 3 34 DTMF and Dialing Parameters Description Defines the maximum number of collected destination number digits that can be received from the
229. SDNIBehavior features add the individual feature values For example to support both 512 and 2048 features set ISDNIBehavior 2560 i e 512 2048 This parameter determines several behaviour options that influence the behaviour of the ISDN Stack outgoing calls To select options click the arrow button and then for each required option select 1 to enable The default is 0 i e disable 2 USER SENDING COMPLETE When this bit is set the device doesn t automatically generate the information element Sending Complete IE in the SETUP message If this bit is not set the device generates it automatically in the SETUP message only 16 USE MU LAW When set the device sends G 711 m Law in outgoing voice calls When disabled the device sends G 711 A Law in outgoing voice calls Applicable only to the Korean variant 128 DIAL WITH KEYPAD When enabled the device uses the Keypad IE to store the called number digits instead of the CALLED_NB IE Only applicable to the KOR variant Korean network Useful for Korean switches that don t accept the CALLED_NB IE 256 STORE CHAN ID IN SETUP When this bit is set the device forces the sending of a Channel ld IE in an outgoing SETUP message even if it s not required by the standard i e optional and no Channel ld has been specified in the establishment request This is useful for improving required compatibility with switches On PRI lines it indicates an unused
230. SE mode to transmit V 34 faxes enabling the full utilization of its speed Configure the following parameters to use bypass mode for both T 30 and V 34 faxes E FaxTransportMode 2 Bypass m V34ModemTransportType 2 Modem bypass m V32ModemTransportType 2 SIP User s Manual 356 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 7 5 3 2 7 5 4 m V23ModemTransportType 2 m V22ModemTransportType 2 Configure the following parameters to use bypass mode for V 34 faxes and T 38 for T 30 faxes E FaxTransportMode 1 Relay V34ModemtTransportType 2 Modem bypass V32ModemTransportType 2 V23ModemtTransportType 2 V22ModemtTransportType 2 Using Relay mode for both T 30 and V 34 faxes In this scenario V 34 fax machines are forced to use their backward compatibility with T 30 faxes and operate in the slower T 30 mode Use the following parameters to use T 38 mode for both V 34 faxes and T 30 faxes E FaxTransportMode 1 Relay V34ModemtTransportType 0 Transparent V32ModemtTransportType 0 V23ModemtTransportType 0 V22ModemtTransportType 0 Supporting V 152 Implementation The device supports the ITU T recommendation V 152 Procedures for Supporting Voice Band Data over IP Networks Voice band data VBD is the transport of modem facsimile and text telephony signals over a voice channel of a packet network with a codec appropriate for such signals For V 152 capability t
231. SETUP message doesn t contain a valid Precedence Level value the default value is used in the Resource Priority header of the outgoing SIP INVITE request In this scenario the character string is sent without translation to a numerical value Defines the duration in seconds in which the device plays a preemption tone to both the Tel and IP sides if a call is preempted The valid range is 0 to 60 The default is 3 Note If set to 0 no preemption tone is played 213 November 2008 A ge AudioCodes Mediant 2000 3 4 8 3 4 8 1 Advanced Applications The Advanced Applications menu allows you to configure advanced SIP based applications This menu includes the following page items HB Voice Mail Settings refer to Configuring the Voice Mail Parameters on page 214 m RADIUS Parameters refer to Configuring RADIUS Accounting Parameters on page 217 Configuring the Voice Mail VM Parameters The Voice Mail Settings page allows you to configure the voice mail parameters The voice mail application applies only to CAS interfaces For detailed information on voice mail refer to the CPE Configuration Guide for Voice Mail User s Manual gt To configure the Voice Mail parameters take these 4 steps 1 Open the Voice Mail Settings page Configuration tab gt Advanced Applications menu gt Voice Mail Settings page item Figure 3 87 Voice Mail Settings Page v General voice Mail Interface
232. SIP 200 OK is received from the IP side The digit pattern is a pre defined DTMF sequence that is used to indicate an answer signal e g for billing The valid range is 1 to 8 characters Note This parameter is applicable to FXO and CAS Determines whether the device releases the call if RTP packets are not received within a user defined timeout 0 No 154 Document LTRT 68808 SIP User s Manual Parameter nnection Broken Connection Timeout BrokenConnectionEven tTimeout Disconnect Call on Silence Detection EnableSilenceDisconne ct Silence Detection Period sec FarEndDisconnectSilen cePeriod Silence Detection Method FarEndDisconnectSilen ceMethod Enable Fax Re Routing EnableFaxReRouting Version 5 6 3 Web Based Management Description 1 Yes default Notes The timeout is set by the parameter BrokenConnectionEventTimeout This feature is applicable only if the RTP session is used without Silence Compression If Silence Compression is enabled the device doesn t detect a broken RTP connection During a call if the source IP address from where the RTP packets are sent is changed without notifying the device the device filters these RTP packets To overcome this set DisconnectOnBrokenConnection to 0 the device doesn t detect RTP packets arriving from the original source IP address and switches after 300 msec to the RTP packets arriving from the new source IP addr
233. Sample ini file for Mediant 2000 T1 device Default loadable Call Progress Tones dat file Call Progress Tones ini file used to create dat file Sample loadable Voice Prompts dat file TrunkPack Downloadable Conversion Utility to create Call Progress Tones Voice Prompts and CAS files Syslog server BootP TFTP configuration utility Used for various signaling types such as E_M_WinkTable dat MIB library for SNMP browser Utility that is used to convert CAS traces to textual form Utility that is used to convert ISDN traces to textual form 407 November 2008 A ge AudioCodes Mediant 2000 Reader s Notes SIP User s Manual 408 Document LTRT 68808 SIP User s Manual 12 Selected Technical Specifications 12 Selected Technical Specifications The technical specifications of the Mediant 2000 is listed in the table below Note All specifications in this document are subject to change without prior notice Table 12 1 Mediant 2000 Functional Specifications Function Specification Trunk amp Channel Capacity Capacity with E1 1 2 4 8 or 16 E1 spans supporting channel capacity as follows 30 Channels on 1 E1 span with gateway 1 only 60 Channels on 2 E1 spans with gateway 1 only 120 Channels on 4 E1 spans with gateway 1 only 240 Channels on 8 E1 spans with gateway 1 only 480 Channels on 16 E1 spans with gateway 1 and gateway 2 Note Channel capacity depends on configuration settings Capacity with T1
234. Shell SSH server 0 Disable default 1 Enable Defines the port number for the embedded SSH server Range is any valid port number The default port is 22 IP address of the primary DNS server Enter the IP address in dotted decimal notation for example 10 8 2 255 Note To use Fully Qualified Domain Names FQDN in the Tel to IP Routing table or Outbound IP Routing table if EnableSBC is set to 1 you must define this parameter IP address of the second DNS server Enter the IP address in dotted decimal notation for example 10 8 2 255 58 Document LTRT 68808 SIP User s Manual Parameter STUN Settings Enable STUN EnableSTUN STUN Server Primary IP STUNServerPrimaryIP STUN Server Secondary IP STUNServerSecondaryIP NFS Settings NFS Table DHCP Settings Enable DHCP DHCPEnable Version 5 6 3 Web Based Management Description Determines whether Simple Traversal of UDP through NATs STUN is enabled 0 Disable default 1 Enable When enabled the device functions as a STUN client and communicates with a STUN server located in the public Internet STUN is used to discover whether the device is located behind a NAT and the type of NAT In addition it is used to determine the IP addresses and port numbers that the NAT assigns to outgoing signaling messages using SIP and media streams using RTP RTCP and T 38 STUN works with many existing NAT types and does n
235. Specifies the subnet mask of the destination host RoutingTableDestinationMasksColumn network SIP User s Manual 62 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Description The address of the host network you want to reach is determined by an AND operation that is applied to the fields Destination IP Address and Destination Mask For example to reach the network 10 8 x x enter 10 8 0 0 in the field Destination IP Address and 255 255 0 0 in the field Destination Mask As a result of the AND operation the value of the last two octets in the field Destination IP Address is ignored To reach a specific host enter its IP address in the field Destination IP Address and 255 255 255 255 in the field Destination Mask Gateway IP Address The IP address of the router next hop to which the RoutingTableGatewaysColumn packets are sent if their destination matches the rules in the adjacent columns Note The Gateway address must be in the same subnet on which the address is configured on the Multiple Interface Table page refer to Configuring the Multiple Interface Table on page 53 Metric The maximum number of allowed routers hops RoutingTableHopsCountColumn between the device and destination Note This parameter must be set to 1 for the routing rule to be valid Routing entries with Hop Count equals 0 are local routes set automatically by the device Interface Spec
236. TP or HTTPS Telnet 412 Document LTRT 68808 SIP User s Manual Function Type Approvals Telecommunication Standards Safety and EMC Standards Environmental Version 5 6 12 Selected Technical Specifications Specification IC CS03 FCC part 68 Chassis and Host telecom card comply with IC CS03 FCC part 68 CTR 4 CTR 12 amp CTR 13 JATE TS 016 TSO Anatel Mexico Telecom Russia CCC ASIF S016 ASIF S038 UL 60 950 1 FCC part 15 Class B Class A with SUN 2080 CPU card CE Mark EN 55022 Class B Class A with SUN 2080 CPU card EN 60950 1 EN 55024 EN 300 386 TS001 NEBS Level 3 GR 63 Core GR 1089 Core Type 1 amp 3 Approved for DC powered version Complies with ETS 301019 ETS 300019 1 2 3 T 1 1 T 2 3 T3 2 Approved for AudioCodes or DC powered versions 413 November 2008 A ge AudioCodes Mediant 2000 Reader s Notes SIP User s Manual 414 Document LTRT 68808 SIP User s Manual 13 Glossary 13 Glossary Term ADPCM AIS A law AMD AOR AWG bps BootP CAS CoS CMP cPCI CPT dB DHCP DID DiffServ DNS DR DS1 DSP DTMF E1 ETSI FQDN GRUU ICMP IE IETF IKE IP IPSec Version 5 6 Table 13 1 Glossary of Terms Meaning Adaptive Differential PCM voice compression Alarm Indication Signal Standard companding algorithm used in European digital communications systems to optimize the dynamic range of an analog signal for digitizing Answe
237. TR and Service Record SRV queries to discover Proxy servers 0 A Record A Record default 1 SRV SRV 2 NAPTR NAPTR If set to A Record 0 no NAPTR or SRV queries are performed If set to SRV 1 and the Proxy IP address parameter contains a domain name without port definition e g ProxylIP domain com 138 Document LTRT 68808 SIP User s Manual Parameter Number of RTX Before Hot Swap HotSwapRtx Use Gateway Name for OPTIONS UseGatewayNameForOption s User Name UserName Password Password Version 5 6 3 Web Based Management Description an SRV query is performed The SRV query returns up to four Proxy host names and their weights The device then performs DNS A record queries for each Proxy host name according to the received weights to locate up to four Proxy IP addresses Therefore if the first SRV query returns two domain names and the A record queries return two IP addresses each no additional searches are performed If set to NAPTR 2 an NAPTR query is performed If it is successful an SRV query is sent according to the information received in the NAPTR response If the NAPTR query fails an SRV query is performed according to the configured transport type If the Proxy IP address parameter contains a domain name with port definition e g ProxyIP domain com 5080 the device performs a regular DNS A record query If a specific Transport Type is defined a
238. Tel side when Tel to IP ISDN overlap dialing is performed When the number of collected digits reaches the maximum the device uses these digits for the called destination number The valid range is 1 to 49 The default value is 30 Note Digit Mapping Rules can be used instead Defines the time in seconds that the device waits between digits that are received from the Tel side when Tel to IP overlap dialing is performed ISDN uses overlap dialing When this inter digit timeout expires the device uses the collected digits to dial the called destination number The valid range is 1 to 10 The default value is 4 Defines the supported Receive DTMF negotiation method 0 No Don t declare RFC 2833 telephony event parameter in SDP 148 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Description 3 Yes Declare RFC 2833 telephony event parameter in SDP default The device is designed to always be receptive to RFC 2833 DTMF relay packets Therefore it is always correct to include the telephony event parameter as default in the SDP However some devices use the absence of the telephony event in the SDP to decide to send DTMF digits in band using G 711 coder If this is the case you can set RxDTMFOption to 0 1 to 5 Tx DTMF Option Determines a single or several preferred transmit DTMF negotiation TxDTMFOption methods 0 Not Supported No negotiation DTMF digits are sent accordi
239. This parameter can include up to 240 indices For configuring Trunk Group Settings using the Web interface refer to Configuring Trunk Group Settings on page 197 For a description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to Routing General Parameters on page 171 0 Leave Source Number empty default 1 If the Source Number of a Tel to IP call is empty the Destination Number is copied to the Source Number For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 This ini file table parameter configures the Tel to IP Routing table for routing Tel to IP calls and the Outbound IP Routing table for IP to IP calls The format of this parameter is as follows PREFIX FORMAT PREFIX_Index PREFIX_DestinationPrefix PREFIX_DestAddress PREFIX_Source
240. To and Contact headers per ITSP In addition all calls dialed from the Enterprise PBX with prefix 02 is sent to the local PSTN The figure below illustrates the example setup Figure 7 3 Example Setup for Routing Between ITSP and Enterprise PBX PSTN Network Local PSTN 4 Network Proxy Set 1 P 10 33 37 77 IP 10 33 37 79 ITSP 1 IP Group 1 Trunk Group ID 2 Trunk Group ID 1 AudioCodes Mediant 2000 Registratio to IP Group 2 ITSP 2 IP Group 2 Proxy Set 2 IP 10 8 8 40 IP 10 8 8 10 Y Network gt To configure call routing between Enterprise and two ITSPs using the device take these 8 steps 1 Enable the device to register to a Proxy Registrar server using the parameter IsRegisterNeeded in the Proxy amp Registration page refer to Proxy amp Registration Parameters on page 132 SIP User s Manual 374 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 2 In the Proxy Sets Table page refer to Proxy Sets Table on page 141 configure two Proxy Sets and for each enable Proxy Keep Alive using SIP OPTIONS and round robin load balancing method e Proxy Set 1 includes two IP addresses of the first ITSP ITSP 1 10 33 37 77 and 10 33 37 79 and using UDP e Proxy Set 2 includes two IP addresses of the second ITSP ITSP 2 10 8 8 40 and 10 8 8 10 and using TCP The figure below displays the configuration of Proxy Set ID 1 Perform simila
241. To create your own auxiliary file it s recommended to modify the supplied usa_tone ini file in any standard text editor to suit your specific requirements and to convert the modified ini file into binary format using the TrunkPack Downloadable Conversion Utility For the description of the procedure on how to convert CPT ini file into a binary dat file refer to the Product Reference Manual To load the Call Progress Tones dat file to the device use the Web interface or ini file refer to Loading Auxiliary Files on page 231 Note Only the dat file can be loaded to the device You can create up to 32 different Call Progress Tones each with frequency and format attributes The frequency attribute can be single or dual frequency in the range of 300 to 1980 Hz or an Amplitude Modulated AM In total up to 64 different frequencies are supported Only eight AM tones in the range of 1 to 128 kHz can be configured the detection range is limited to 1 to 50 kHz Note that when a tone is composed of a single frequency the second frequency field must be set to zero The format attribute can be one of the following m Continuous e g dial tone a steady non interrupted sound Only the First Signal On time should be specified All other on and off periods must be set to zero In this case the parameter specifies the detection period For example if it equals 300 the tone is detected after 3 seconds 300 x 10 msec The minimum
242. Trunk Trunk Group ID L mi mo Jh o 2 vist 2200 e e Channels Phone Number IP Profile ID 5 In the Trunk Group Settings page refer to Configuring the Trunk Group Settings on page 197 configure Per Account registration for Trunk Group ID 1 without serving IP Group Figure 7 7 Configuring Trunk Group 1 for Registration per Account in Trunk Group Settings Page Serving Trunk Channel Select Mode Registration m Contact User Group IO Mode Gateway Name ontact U 1 1 Cyclic Ascending w Per Account v usemame 6 In the Account Table page refer to Configuring the Account Table on page 204 configure the two Accounts for PBX trunk registration to ITSPs using the same Trunk Group i e ID 1 but different serving IP Groups 1 and 2 For each account define user name password and hostname and ContactUser The Register messages use different URI s Hostname and ContactUser in the From To and Contact headers per ITSP Enable registration for both accounts Figure 7 8 Configuring Accounts for PBX Registration to ITSPs in Account Table Page HostName Register ContactUser 7 Inthe IP to Trunk Group Routing page refer to IP to Trunk Group Routing on page 181 configure IP to Tel routing for calls from ITSPs to Trunk Group ID 1 see 1 below and from the device to the local PSTN see 2 below Figure 7 9 Configuring ITSP to Trunk Group 1 Routing in IP to Trunk
243. Voice gain control in decibels This parameter sets the level for VoiceVolume the transmitted IP to PSTN signal The valid range is 32 to 31 dB The default value is 0 dB Input Gain Pulse code modulation PCM input gain control in decibels This InputGain parameter sets the level for the received PSTN to IP signal The valid range is 32 to 31 dB The default value is 0 dB Silence Suppression Silence Suppression is a method for conserving bandwidth on VoIP EnableSilenceCompression calls by not sending packets when silence is detected 0 Disable Silence Suppression is disabled default 1 Enable Silence Suppression is enabled 2 Enable without Adaptation A single silence packet is sent during a silence period applicable only to G 729 Note If the selected coder is G 729 the following rules determine the value of the annexb parameter of the fmtp attribute in the SDP f EnableSilenceCompression is 0 annexb no f EnableSilenceCompression is 1 annexb yes f EnableSilenceCompression is 2 and IsCiscoSCEMode is 0 annexb yes SIP User s Manual 66 Document LTRT 68808 SIP User s Manual Parameter Echo Canceler EnableEchoCanceller DTMF Transport Type DTMFTransportType MF Transport Type MFTransportType DTMF Volume 31 to 0 dB DTMFVolume CAS Transport Type CASTransportType DTMF Generation Twist DTMFGenerationTwist Version 5 6 3 Web Based Mana
244. a AudioCodes Parameter SyslogOutputMethod BaseUDPport RemoteBaseUDPPort L1L1ComplexTxUDPPort L1L1ComplexRxUDPPort NTPServerlP NTPServerUTCOffset NTPUpdaitelnterval Description Determines the method used for Syslog messages 0 Send all Syslog messages to the defined Syslog server default 1 Send all Syslog messages using the Debug Recording mechanism 2 Send only Error and Warning level Syslog messages using the Debug Recording mechanism For a detailed description on Debug Recording refer to Debug Recording DR For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTOP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTOP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTOP Settings on page 71 For a description of this parameter refer to Configuring the Application Settings on page 57 For a description of this parameter refer to Configuring the Application Settings on page 57 For a description of this parameter refer to Configuring the Application Settings on page 57 Mediant 2000 IP Routing Table parameters The IP routing ini file parameters are array parameters Each parameter configures a specific column in the IP routing table The first entry in e
245. a RADIUS server refer to the Product Reference Manual For secured HTTP connection HTTPS refer to the Product Reference Manual Version 5 6 101 November 2008 A C al AudioCodes Mediant 2000 3 4 6 2 Configuring the Web and Telnet Access List The Web amp Telnet Access List page is used to define up to ten IP addresses that are permitted to access the device s Web and Telnet interfaces Access from an undefined IP address is denied If no IP addresses are defined this security feature is inactive and the device can be accessed from any IP address The Web and Telnet Access List can also be defined using the ini file parameter WebAccessList_x refer to Web and Telnet Parameters on page 273 gt To add authorized IP addresses for Web and Telnet interfaces access take these 4 steps 1 Open the Web amp Telnet Access List page Configuration tab gt Security Settings menu gt Web amp Telnet Access List page item Figure 3 51 Web amp Telnet Access List Page Add New Entry Add New Entry 2 To add an authorized IP address in the Add a New Authorized IP Address field enter the required IP address and then click Add New Address the IP address you entered is added as a new entry to the Web amp Telnet Access List table Figure 3 52 Web amp Telnet Access List Table Delete Authorized IP Row Address Delete Selected Addresses Note Delete all rows to allow access from any IP addre
246. a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring the General Security Settings on page 109 281 November 2008 ca AudioCodes Parameter RadiusVSAVendorID RadiusVSAAccessAttribute 4 4 6 Mediant 2000 Description For a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring the General Security Settings on page 109 SNMP Parameters The SNMP related ini file configuration parameters are described in the table below Parameter DisableSNMP SNMPPort SNMPTrustedMGR_x KeepAliveTrapPort SendKeepAliveTrap SNMPSysOid SNMPTrapEnterpriseOid acUserInputAlarmDescriptio n acUserlInputAlarmSeverity AlarmHistoryTableMaxSize SIP User s Manual Table 4 6 SNMP ini File Parameters Description For a description of this parameter refer to Configuring the Management Settings on page 220 The device s local UDP port used for SNMP Get Set commands The range is 100 to 3999 The default port is 161 Up to five IP addresses of remote trusted SNMP managers from which the SNMP agent accepts and processes get and set requests Notes f no values are assigned to these parameters any manager ca
247. abled Uses RADIUS queries for Web and Telnet interface authentication 0 Disable default 1 Enable When enabled logging in to the device s Web and Telnet embedded servers is performed via a RADIUS server The device contacts a predefined server and verifies the given user name and password pair against a remote database ina secure manner Notes The parameter EnableRADIUS must be set to 1 RADIUS authentication requires HTTP basic authentication meaning the user name and password are transmitted in clear text over the network Therefore it s recommended to set the parameter HttpsOnly to 1 to force the use of HTTPS since the transport is encrypted fusing RADIUS authentication when logging in to the CLI only the primary Web User Account which has Security Administration access level can access the device s CLI refer to Configuring the Web User Accounts on page 99 IP address of the RADIUS authentication server Port number of the RADIUS authentication server The default value is 1645 111 November 2008 ca AudioCodes Parameter RADIUS Shared Secret SharedSecret General RADIUS Authentication Default Access Level DefaultAccessLevel Device Behavior Upon RADIUS Timeout BehaviorUponRadiusTimeout Local RADIUS Password Cache Mode RadiusLocalCacheMode Local RADIUS Password Cache Timeout RadiusLocalCacheTimeout RADIUS VSA Vendor ID RadiusVSAVendorID RADIUS VSA A
248. ach parameter refers to the first row in the IP routing table the second entry to the second row and so forth In the following example two rows are configured when the device is in network 10 31 x x RoutingTableDestinationsColumn 130 33 4 6 83 4 87 6 RoutingTableDestinationMasksColumn 255 255 255 255 255 255 255 0 RoutingTableGatewaysColumn 10 31 0 1 10 31 0 112 RoutingTablelnterfacesColumn 0 1 RoutingTableHopsCountColumn 20 20 RoutingTableDestinations Column For a description of this parameter refer to Configuring the IP Routing Table on page 62 RoutingTableDestination MasksColumn For a description of this parameter refer to Configuring the IP Routing Table on page 62 RoutingTableGatewaysCo lumn For a description of this parameter refer to Configuring the IP Routing Table on page 62 Routing TableHopsCountC olumn For a description of this parameter refer to Configuring the IP Routing Table on page 62 RoutingTablelInterfacesCo For a description of this parameter refer to Configuring the IP lumn Routing Table on page 62 VLAN Parameters VLANMode For a description of this parameter refer to Configuring the IP Settings on page 50 SIP User s Manual 264 Document LTRT 68808 SIP User s Manual Parameter VLANNativeVLANID VLANOamVLANID VLANControlVLANID VLANMediaVLANID VLANNetworkServiceClas sPriority VLANPremiumServiceCla ssMediaPriority VLANPremiumServic
249. addition you can enable or disable Error Correction Mode ECM fax mode using the FaxRelayECMEnable parameter When using T 38 mode you can define a redundancy feature to improve fax transmission over congested IP networks This feature is activated using the FaxRelayRedundancyDepth and FaxRelayEnhancedRedundancyDepth parameters Although this is a proprietary redundancy scheme it should not create problems when working with other T 38 decoders 7 5 2 1 1 Switching to T 38 Mode using SIP Re INVITE In the Switching to T 38 Mode using SIP Re INVITE mode upon detection of a fax signal the terminating device negotiates T 38 capabilities using a Re INVITE message If the far end device doesn t support T 38 the fax fails In this mode the parameter FaxTransportMode is ignored Version 5 6 351 November 2008 A EA AudioCodes Mediant 2000 7 5 2 1 2 7 5 2 2 To configure T 38 mode using SIP Re INVITE messages set IsFaxUsed to 1 Additional configuration parameters include the following E FaxRelayEnhancedRedundancyDepth m FaxRelayRedundancyDepth E FaxRelayECMEnable m FaxRelayMaxRate Automatically Switching to T 38 Mode without SIP Re INVITE In the Automatically Switching to T 38 Mode without SIP Re INVITE mode when a fax signal is detected the channel automatically switches from the current voice coder to answer tone mode and then to T 38 compliant fax relay mode To configure automatic T 38 mode perform the following confi
250. age 171 For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to Routing General Parameters on page 171 323 November 2008 ca AudioCodes Parameter Phone Context Parameters AddPhoneContextAsPrefix PhoneContext SIP User s Manual Mediant 2000 Description For a description of this parameter refer to Mapping NPI TON to Phone Context on page 170 This ini file table parameter defines the Phone Context table The format for this parameter is as follows PhoneContext FORMAT PhoneContext_Index PhoneContext_Npi PhoneContext_Ton PhoneContext_Context PhoneContext Where Npi Number Plan Ton Type of Number Context Phone Context value When a call is received from the ISDN the NPI and TON are compared to the table and the Phone Context value is used in the outgoing SIP INVITE message The same mapping occurs when an INVITE with a Phone Context attribute is received The Phone Context parameter appears in the standard SIP headers where a phone number is used Request URI To From Diversion For example PhoneContext PhoneContext 0 0 0 unknown com PhoneContext 1 1 1 host com PhoneContext 2 9 1 na e164 host com PhoneContext Notes This parameter can include up to 20 indices Several entries with the same NPI TON or Phone Context are allowed In this scenario a Tel to IP call uses the f
251. al reasons RELEASE _BECAUSE_UNKNOWN_REASON RELEASE BECAUSE _REMOTE_CANCEL_CALL RELEASE BECAUSE _MANUAL_DISC RELEASE _BECAUSE_SILENCE_DISC RELEASE_BECAUSE_DISCONNECT_CODE Note When the duration of the call is zero the release reason GWAPP_NORMAL_CALL_CLEAR increments the Number of Failed Calls due to No Answer counter The rest of the release reasons increment the Number of Failed Calls due to Other Failures counter Percentage of The percentage of established calls from attempted calls Successful Calls ASR Version 5 6 249 November 2008 ca AudioCodes Counter Number of Calls Terminated due to a Busy Line Number of Calls Terminated due to No Answer Number of Calls Terminated due to Forward Number of Failed Calls due to No Route Number of Failed Calls due to No Matched Capabilities Number of Failed Calls due to No Resources Number of Failed Calls due to Other Failures Average Call Duration ACD sec Attempted Fax Calls Counter Successful Fax Calls Counter Mediant 2000 Description Indicates the number of calls that failed as a result of a busy line It is incremented as a result of the following release reason GWAPP_USER_BUSY 17 Indicates the number of calls that weren t answered It s incremented as a result of one of the following release reasons GWAPP_NO_USER_RESPONDING 18 GWAPP_NO_ANSWER_FROM_USER_ALERTED 19 GWAPP_NORMAL_CALL_CLEAR 16 when
252. all to the PSTN because it has no available channels in a specific trunk group g all trunk group s channels are occupied or the trunk group s spans are disconnected or out of sync it uses the Internal Release Cause 3 No Route to Destination This cause can be used in the AltRouteCauselP2Tel table to define routing to an alternative trunk group For defining the Reasons for Alternative Routing table using the Web interface refer to Reasons for Alternative Routing on page 188 Foran explanation on usng ini file table parameters refer to Structure of ini File Table Parameters on page 257 Defines the method in which the Redirect Number is passed toward the Tel side 0 Q 931 Redirecting Number Information Element IE default 1 ETSI DivertingLegInformation2 in a Facility IE For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to Routing General Parameters on page 171 For a description of this parameter refer to Routing General Parameters on p
253. alues the device maps the Redirect phone number to the SIP target parameter and the Redirect number reason to the SIP cause parameter in the Request URI Redirecting Reason gt gt SIP Response Code Unknown gt gt 404 User busy gt gt 486 No reply gt gt 408 Deflection gt gt 487 480 Unconditional gt gt 302 Others gt gt 302 If the device receives a Request URI that includes a target and cause parameter the target is mapped to the Redirect phone number and the cause is mapped to Redirect number reason Determines the time in seconds used in the Retry After header when a 503 Service Unavailable response is generated by the device The time range is 0 to 3 600 The default value is 0 Determines the device usage of the P Associated URI header This header can be received in 200 OK responses to REGISTER requests When enabled the first URI in the P Associated URI header is used in subsequent requests as the From P Asserted Id headers value 0 Disable default 1 Enable Note P Associated URIs in registration responses is handled only if the device is registered per endpoint using the User Information file Determines the SIP header used to determine the Source Number in incoming INVITE messages empty string Use device s internal logic for header preference default FROM Use the Source Number received in the From header 131 November 2008 ca AudioCodes
254. ame field enter an arbitrary name that allows you to easily identify the IP Profile 4 From the Profile Preference drop down list select the priority of the IP Profile where 1 is the lowest priority and 20 is the highest If both IP and Tel profiles apply to the same call the coders and other common parameters noted by an asterisk in the description of the parameter IPProfile of the preferred Profile are applied to that call If the Preference of the Tel and IP Profiles is identical the Tel Profile parameters are applied Note If the coder lists of both IP and Tel Profiles apply to the same call only the coders common to both are used The order of the coders is determined by the preference SIP User s Manual 194 Document LTRT 68808 SIP User s Manual 3 Web Based Management 5 Configure the IP Profile s parameters according to your requirements For detailed information on each parameter refer to the description on the page in which it is configured as an individual parameter Parameters that are unique to IP Profile are described in the table below 6 From the Coder Group drop down list select the coder group you want to assign to the Profile You can select the device s default coders refer to Coders on page 144 or one of the coder groups you defined in the Coder Group Settings page refer to Coder Group Settings on page 190 7 Repeat steps 2 through 6 for the next IP Profiles optional 8 Click the Su
255. ameters 1 IP Address 2 Subnet Mask 3 Default Gateway IP Address 4 ActivityListToLog 222 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 5 1 1 1 Configuring the SNMP Trap Destinations Table The SNMP Trap Destinations page allows you to configure up to five SNMP trap managers gt To configure the SNMP Trap Destinations table take these 5 steps 1 Access the Management Settings page as described in Configuring the Management Settings on page 220 2 In the SNMP Trap Destinations field click the right pointing arrow e button the SNMP Trap Destinations page appears Figure 3 91 SNMP Trap Destinations Page IP Address Trap Port Trap Enable SNMP Manager 1 0 8 2 26 162 Enable v m E l SNMP Manager 0 0 0 0 162 Enable SNMP Manager 0 0 0 0 162 Enable SNMP Manager 0 0 0 0 Enable oO E Ea o 3 4 5 SNMP Manager 0 0 0 0 Enable Configure the SNMP trap managers parameters according to the table below Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 230 Note Only table row entries whose corresponding check boxes are selected are applied when clicking Submit otherwise settings revert to their defaults Table 3 58 SNMP Trap Destinations Parameters Description Parameter Description SNMP Manager Determines the validity of
256. ameters are available and can be modified 2 Configure the desired trunk parameters as described in the table below SIP User s Manual 84 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 Click the Apply Trunk Settings button to apply the changes to the selected trunk or click Apply to All Trunks to apply the changes to all trunks the Stop Trunk button replaces Apply Trunk Settings and the Trunk Configuration State displays Active 4 To save the changes to flash memory refer to Saving Configuration on page 230 5 To reset the device refer to Resetting the Device on page 228 If the Protocol Type field displays NONE i e no protocol type selected and no other trunks have been configured after selecting a PRI protocol type you must reset the device The displayed parameters on the page depend on the protocol selected in the Protocol Type field All trunks must be of the same line type i e either E1 or T1 However different variants of the same line type can be configured on different trunks for example E1 Euro ISDN and E1 CAS subject to the constraints in the device s Release Notes If the trunk protocol type is CAS you can assign or modify a dial plan in the Dial Plan field and perform this without stopping the trunk If the trunk can t be stopped because it provides the device s clock assuming the device is synchronized with the E1 T1 clock assign a
257. an ENUM query is performed and the response is used to correctly route the INVITE If no response is received from the ENUM server the INVITE is routed to the default gateway 0 Disable default 1 Enable This ini file table parameter is used by the SAS application to manipulate the User Part of an incoming REGISTER request AoR the To header before saving it to the registered users database The format of this table parameter is as follows SASRegistrationManipulation FORMAT SASRegistrationManipulation_Index SASRegistrationManipulation RemoveFromRight SASRegistrationManipulation_LeaveFromRight SASRegistrationManipulation RemoveFromRight number of digits removed from the right side of the User Part before saving to the registered user database LeaveFromRight number of digits to keep from the right side If both RemoveFromRight and LeaveFromRight are defined the RemoveFromRight is applied first The registered database contains the AoR before and after the manipulation The range of both RemoveFromRight and LeaveFromRight is 0 to 30 Note This table can include only one index entry Defines emergency numbers for the device s SAS application When the device s SAS agent receives a SIP INVITE from an IP phone that includes one of the emergency numbers in the SIP user part it forwards the INVITE to the default gateway configured by the parameter SASDefaultGatewayIP i e the device itself
258. ange User Name Access Level Administrator A Change Access Level w Fill in the following 3 fields to change the password Current Password New Password Confirm New Password Change Password SIP User s Manual 100 Document LTRT 68808 SIP User s Manual 3 Web Based Management Note If you are logged into the Web interface as the Security Administrator both Web user accounts are displayed on the Web User Accounts page as shown above If you are logged in with the secondary user account only the details of the secondary account are displayed on the page 2 To change the access level of the secondary account a From the Access Level drop down list select the new access level b Click Change Access Level the new access level is applied immediately The access level of the primary Web user account is Security Administrator which cannot be modified The access level of the secondary account can only be modified by the primary account user or a secondary account user with Security Administrator access level 3 To change the user name of an account perform the following a Inthe field User Name enter the new user name maximum of 19 case sensitive characters b Click Change User Name if you are currently logged into the Web interface with this account the Enter Network Password dialog box appears requesting you to enter the new user name 4 To change the password
259. arameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 Determines whether the called number is set in the user part of the To header 0 Sets the destination number to the user part of the Request URI for IP to Tel calls and sets the Contact header to the source number for Tel to IP calls default 1 Sets the destination number to the user part of the To header for IP to Tel calls and sets the Contact header to the username parameter for Tel to IP calls For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 Determines whether the ptime header is included in the SDP 0 Remove the ptime header from SDP 1 Include the ptime header in SDP default For a description of this parameter refer to Advanced Parameters on page 151 Determines whether the device uses the value of the incoming SIP Reason header for Release Reason mapping 0 Disregard Reason header in incom
260. are allowed In such a scenario a Tel to IP call uses the first match Phone Context is a unique case as it doesn t appear in the Request URI as a Phone Context parameter Instead it s added as a prefix to the phone number The isn t removed from the phone number in the IP to Tel direction You can also configure the Phone Context table using the ini file table parameter PhoneContext refer to Number Manipulation and Routing Parameters on page 313 Table 3 42 Phone Context Parameters Description Parameter Description Add Phone Context As Prefix Determines whether the received Phone Context parameter is AddPhoneContextAsPrefix added as a prefix to the outgoing ISDN SETUP message with Called and Calling numbers 0 Disable Disable default 1 Enable Enable SIP User s Manual 170 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Description NPI TON Select the Number Plan assigned to this entry 0 Unknown Unknown default 1 E 164 Public E 164 Public 9 Private Private For a detailed list of the available NPI TON values refer to Numbering Plans and Type of Number on page 169 Select the Number Type assigned to this entry f you selected Unknown as the NPI you can select Unknown 0 If you selected Private as the NPI you can select Unknown 0 Level 2 Regional 1 Level 1 Regional 2 PSTN Specific 3 or Level 0 Regional Loca
261. are erased from the audio stream DTMFTransportType is automatically set to 0 DTMF Transport Type field DTMF Mute Web interface Using RFC 2833 relay with Payload type negotiation DTMF digits are carried to the remote side as part of the RTP stream in accordance with RFC 2833 standard To enable this mode define the following e RxDTMFOption 3 ini file Declare RFC 2833 in SDP field Yes Web interface refer to DTMF amp Dialing Parameters on page 147 e TxDTMFOption 4 ini file 1 to 5 Tx DTMF Option field RFC 2833 Web interface refer to DTMF amp Dialing Parameters on page 147 Note that to set the RFC 2833 payload type with a different value other than its default 96 configure the RFC2833PayloadType RFC 2833 Payload Type parameter The device negotiates the RFC 2833 payload type using local and remote SDP and sends packets using the payload type from the received SDP The device expects to receive RFC 2833 packets with the same payload type as configured by the RFC2833PayloadType parameter If the remote side doesn t include telephony event in its SDP the device sends DTMF digits in transparent mode as part of the voice stream 349 November 2008 A Ee AudioCodes Mediant 2000 E Sending DTMF digits in RTP packets as part of the audio stream DTMF Relay is disabled This method is typically used with G 711 coders with other low bit rate LBR coders the quality of the
262. ased on Connectivity and QoS on page 361 Determines the event s reason for triggering Alternative Routing 0 None Alternative routing is not used 1 Connectivity Alternative routing is performed if ping to initial destination fails 2 QoS Alternative routing is performed if poor QoS is detected 3 Both Alternative routing is performed if either ping to initial destination fails poor Quality of Service is detected or DNS host name is not resolved default Notes QoS is quantified according to delay and packet loss calculated according to previous calls QoS statistics are reset if no new data is received within two minutes For information on the Alternative Routing feature refer to Configuring Alternative Routing Based on Connectivity and QoS on page 361 To receive quality information displayed in the Quality Status and Quality Info fields in IP Connectivity on page 252 per destination this parameter must be set to 2 or 3 Determines the method used by the device for periodically querying the connectivity status of a destination IP address 0 ICMP Ping default Internet Control Message Protocol ICMP ping messages 1 SIP OPTIONS The remote destination is considered offline if the latest OPTIONS transaction timed out Any response to an OPTIONS request even if indicating an error brings the connectivity status to online Defines the time interval i
263. ates a T 38 session on receiving a CED answer tone from the fax This option can only be used to relay fax signals as the device sends T 38 Re INVITE on detection of any fax modem Answer tone 2100 Hz amplitude modulated 2100 Hz or 2100 Hz with phase reversals The modem signal fails when using T 38 for fax relay Notes For this parameter to take effect you must reset the device This parameters is applicable only if the ini file parameter IsFaxUsed is set to 1 or 3 124 Document LTRT 68808 SIP User s Manual Parameter SIP Transport Type SIPTransportType SIP UDP Local Port LocalSIPPort SIP TCP Local Port TCPLocalS PPort SIP TLS Local Port TLSLocalSIPPort Enable SIPS EnableSIPS Enable TCP Connection Reuse EnableTCPConnectio nReuse TCP Timeout SIPTCPTimeout SIP Destination Port SIPDestinationPort Use user phone in SIP URL IsUserPhone Version 5 6 3 Web Based Management Description Determines the default transport layer for outgoing SIP calls initiated by the device 0 UDP default 1 TCP 2 TLS SIPS Notes It s recommended to use TLS for communication with a SIP Proxy and not for direct device to device communication For received calls i e incoming the device accepts all these protocols The value of this parameter is also used by the SAS application as the default transport layer for outgoing SIP calls Local UDP port for SIP messa
264. ation Tables IEEE A EATE T U4 3 4 7 4 Configuring the Routing Tables cccccssescesssessscesscssncseecssesecseeeeees 3 4 7 5 Configuring the Profile Definitions 3 4 7 6 Configuring the Trunk and IP Groups 3 4 7 7 Configuring the Digital Gateway Parameters 3 4 8 Advanced Applications PEIEE E T 3 4 8 1 Configuring the Voice Mail VM Parameters scccccsssscsseeeeeessesn ETETE 21 3 4 8 2 Configuring RADIUS Accounting Parameters EPEA 3 4 9 Configuring the TDM Bus Settings cccccccccescscssssssssvesesssecessssnessessseseeeesee EEO Z 3 5 Management Tab 00 en eer fer ena PEE LEARE EPEN TE 35 1 Management Configuratio M sineasta eein aR E 3 5 1 1 Configuring the Management ia eusebubqumntuamsiese 3 5 1 2 Configuring the R TAN EEE EA EAA TES NT 3 5 1 3 Maintenance Actions Garduri 3 5 2 Software Update 3 5 2 1 Loading Auxiliary Files 3 5 2 2 Upgrading the Software Upgrade Key 3 5 2 3 Software Upgrade Wizard 3 5 2 4 Backing Up and Restoring Configuration E E E T 30 Status amp Diagnostics TAD vcecicetes cstcecorescenceineccrnerans EEL AA EEE EATE EIR 3 6 1 Status amp Diagnostics cceeseeeeeerees S A 3 6 1 1 Viewing the Device s Syslog Messages A PTE 3 6 1 2 Viewing the Ethernet Port Information 0 cccsceeee E EET koa 3 6 1 3 Viewing Active IP Interfaces c eee ee ee ieee 244 3 6 1 4 Viewing Device Information 3 6 1 5 Viewing Performance Statistics 3 6
265. ation Tone Applicable only to Analog devices 9 Call Waiting Tone Applicable only to Analog devices 15 Stutter Dial Tone Applicable only to Analog devices 16 Off Hook Warning Tone Applicable only to Analog devices 17 Call Waiting Ringback Tone 23 Hold Tone Tone Modulation Type Either Amplitude Modulated 1 or regular 0 Tone Form The tone s format can be one of the following Continuous 1 Cadence 2 Burst 3 Low Freq Hz frequency in Hz of the lower tone component in case of dual frequency tone or the frequency of the tone in case of single tone This is not relevant to Amplitude Modulated AM tones High Freq Hz frequency in Hz of the higher tone component in case of dual frequency tone or zero 0 in case of single tone not relevant to AM tones Low Freq Level dBm generation level 0 dBm to 31 dBm in dBm not relevant to AM tones High Freq Level generation level 0 to 31 dBm The value should be set to 32 in the case of a single tone not relevant to AM tones First Signal On Time 10 msec Signal On period in 10 msec units for the first cadence on off cycle For be continuous tones this parameter defines the detection period For burst tones it defines the tone s duration First Signal Off Time 10 msec Signal Off period in 10 msec units for the first cadence on off cycle for cadence tones For burst tones this parameter defines the off time r
266. ation_number SIPGroupName and then it searches the User s internal database for a match Defines the user part for the From To and Contact headers of SIP REGISTER messages and the user part for the Contact header of 202 Document LTRT 68808 SIP User s Manual Parameter Serving IP Group ID Enable Survivability Routing Mode Version 5 6 3 Web Based Management Description INVITE messages that are received from this IP Group and forwarded by the device to another IP Group Notes This parameter is applicable only for USER type IP Groups This parameter is overridden by the Contact User parameter if configured in the Account table refer to Configuring the Account Table on page 204 If configured INVITE messages initiated from the IP Group are sent to this Serving IP Group range 1 to 9 In other words the INVITEs are sent to the address defined for the Proxy Set associated with this Serving IP Group The Request URI host name in these INVITE messages are set to the value of the parameter SIP Group Name defined for the Serving IP Group Notes This field is available only if EnableSBC is set to 1 refer to SBC Configuration on page 163 If the parameter PreferRouteTable is set to 1 the routing rules in the Outbound IP Routing table takes precedence over this Serving IP Group ID parameter If this parameter is not configured the INVITE messages are se
267. atus The Call Routing Status page provides you with information on the current routing method used by the device This information includes the IP address and FQDN if used of the Proxy server with which the device currently operates SIP User s Manual 250 Document LTRT 68808 SIP User s Manual 3 Web Based Management gt To view the call routing status take this step E Open the Call Routing Status page Status amp Diagnostics tab gt Gateway Statistics menu gt Calls Routing Status page item Figure 3 119 Call Routing Status Page 7 Current Call Routing Method Routing Table Current Proxy Not Used Current Proxy State Table 3 65 Call Routing Status Parameters Parameter Description Current Call Routing Proxy Proxy server is used to route calls Method Routing Table preferred to Proxy The Tel to IP Routing table or Outbound IP Routing Table if EnableSBC is set to 1 takes precedence over a Proxy for routing calls Prefer Routing Table parameter is set to Yes as described in Proxy amp Registration Parameters on page 132 Current Proxy Not Used Proxy server isn t defined IP address and FQDN if exists of the Proxy server with which the device currently operates Current Proxy State N A Proxy server isn t defined OK Communication with the Proxy server is in order Fail No response from any of the defined Proxies 3 6 2 3 SAS S
268. bilities are immediately applied to the device and take effect other parameters displayed on the page with the lightning symbol are not changeable on the fly and require a device reset refer to Resetting the Device on page 228 before taking effect Parameters saved to the volatile memory by clicking Submit revert to their previous settings after a hardware or software reset or if the device is powered down Therefore to ensure parameter changes whether on the fly or not are retained you need to save burn them to the device s non volatile memory i e flash refer to Saving Configuration on page 230 If you modify a parameter value and then attempt to navigate away from the page without clicking Submit a message box appears notifying you of this Click Yes to save your modifications or No to ignore them Version 5 6 29 November 2008 A ge AudioCodes Mediant 2000 3 3 3 4 3 3 3 5 If you enter an invalid parameter value e g not in the range of permitted values and then click Submit a message box appears notifying you of the invalid value In addition the parameter value reverts to its previous value and is highlighted in red as shown in the figure below Figure 3 10 Value Reverts to Previous Valid Value lt etg l w Priority Settings Value Reverted Network Priority to Previous i Media Premium Priority _ Valid Value Control Premium Priority Gold Priority Bronze Priorit
269. bjectName defined in the remote side certificate when establishing TLS connections If the SubjectAltName of the received certificate is not equal to any of the defined Proxies Host names IP addresses and is not marked as critical the Common Name CN of the Subject field is compared with this value If not equal the TLS connection is not established If the CN uses a domain name the certificate can also use wildcards to replace parts of the domain name The valid range is a string of up to 49 characters Note This parameter is applicable only if the parameter PeerHostNamevVerificationMode is set to 1 or 2 3 4 6 6 Configuring the IPSec Table The IPSec Table page allows you to configure the Security Policy Database SPD parameters for IP security IPSec Note You can also configure the IPSec table using the ini file table parameter IPSEC_SPD_TABLE refer to Security Parameters on page 276 SIP User s Manual 114 Document LTRT 68808 SIP User s Manual gt To configure the IPSec SPD table take these 5 steps 3 Web Based Management 1 Open the IPSec Table page Configuration tab gt Security Settings menu gt IPSec Table page item Figure 3 57 IPSec Table Page v Policy Index 0 State Does not exist IPSec table row does not exist v IPSec Mode Remote IP Address Local IP Address Type Source Port Destination Port Protocol Releat
270. ble Encryption On On a secured RTP session this parameter determines whether Transmitted RTP Packets to enable Encryption on transmitted RTP packets RTPEncryptionDisableTx 0 Enable default 1 Disable Disable Encryption On On a secured RTP session this parameter determines whether Transmitted RTCP Packets to enable Encryption on transmitted RTCP packets RTCPEncryptionDisableTx 0 Enable default 1 Disable SRTP Settings Master Key Identifier MKI Size Determines the size in bytes of the Master Key Identifier MKI SRTPTxPacketMkISize in SRTP Tx packets The range is 0 to 4 The default value is 0 3 4 3 PSTN Settings The PSTN Settings menu allows you to configure various PSTN settings and includes the following page items m Trunk Settings refer to Configuring the Trunk Settings on page 82 m CAS State Machines refer to Configuring the CAS State Machines on page 97 3 4 3 1 Configuring the Trunk Settings The Trunk Settings page allows you to configure the device s trunks For configuring the trunks using the ini file parameters refer to PSTN Parameters on page 303 gt To configure the Trunks take these 7 steps 1 Open the Trunk Settings page Configuration tab gt PSTN Settings menu gt Trunk Settings page item SIP User s Manual 82 Document LTRT 68808 SIP User s Manual Figure 3 47 Trunk Settings Page 3 Web Based Management J i il OD General Settings
271. ble Parameters on page 257 Secure Hypertext Transport Protocol HTTPS Parameters HTTPSOnly HTTPSPort HTTPSCipherString WebAuthMode HTTPSRequireClientCertificate Version 5 6 For a description of this parameter refer to Configuring the General Security Settings on page 109 Determines the local Secured HTTPS port of the device The valid range is 1 to 65535 other restrictions may apply within this range The default port is 443 Defines the Cipher string for HTTPS in OpenSSL cipher list format For the valid range values refer to URL http www openssl org docs apps ciphers html The default is EXP RC4 For a description of this parameter refer to Configuring the General Security Settings on page 109 Requires client certificates for HTTPS connection The client certificate must be preloaded to the device and its matching private key must be installed on the managing PC Time and date must be correctly set on the device for the client certificate to be verified 0 Client certificates are not required default 1 Client certificates are required 279 November 2008 ca AudioCodes Parameter HTTPSRootFileName HTTPSPkeyFileName HTTPSCertFileName Internal Firewall Parameters AccessList SIP User s Manual Mediant 2000 Description Defines the name of the HTTPS trusted root certificate file to be loaded via TFTP The file must be in base64 encoded PEM Privacy Enhanc
272. ble parameter IPProfile refer to SIP Configuration Parameters on page 284 Version 5 6 193 November 2008 ca AudioCodes gt To configure the IP Profile settings take these 9 steps Mediant 2000 1 Open the IP Profile Settings page Configuration tab gt Protocol Configuration menu gt Profile Definitions submenu gt IP Profile Settings page item Figure 3 81 IP Profile Settings Page v Profile ID Profile Name wv Profile Parameters Profile Preference Fax Signaling Method Dynamic Jitter Buffer Minimum Delay msec 10 Dynamic Jitter Buffer Optimization Factor 10 RTP IP DiffServ 46 Signaling DiffServ 40 RTP Redundancy Depth 0 Remote RTP Base UDP Port CNG Detector Mode Modems Transport Type NSE Mode Play Ringback Tone to IP Enable Early Media Progress Indicator to IP Echo Canceler Media Security Behavior Number of Calls Limit Copy Destination Number to Redirect Number Disconnect on Broken Connection 0 Disable Enable Bypass Disable Don t Play Disable Not Configured Enable Preferable SNK NS SNS NSN Disable Yes Coder Group Coder Group Default Coder Group 2 From the Profile ID drop down list select an identification number for the IP Profile 3 In the Profile N
273. bmit button to save your changes 9 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 48 Description of Parameter Unique to IP Profile Parameter Description Number of Calls Limit Maximum number of concurrent calls If the profile is set to some limit the device maintains the number of concurrent calls incoming and outgoing pertaining to the specific profile A limit value of 1 indicates that there is no limitation on calls for that specific profile default A limit value of 0 indicates that all calls are rejected When the number of concurrent calls is equal to the limit the device rejects any new incoming and outgoing calls belonging to that profile 3 4 7 6 Configuring the Trunk and IP Groups The Trunk IP Group menu allows you to configure groups of channels This submenu includes the following page items Trunk Group refer to Configuring the Trunk Group Table on page 195 Trunk Group Settings refer to Configuring the Trunk Group Settings on page 197 IP Group Table refer to Configuring the IP Groups on page 201 Account Table refer to Configuring the Account Table on page 204 3 4 7 6 1 Configuring the Trunk Group Table The Trunk Group Table page provides you with a table for enabling device channels by assigning them telephone numbers Trunk Groups and Profiles Trunk Groups are used for routing IP to Tel calls with common rules Channels that are not defined ar
274. ccess Level Attribute RadiusVSAAccessAtiribute EtherDiscover Setting EtherDiscover Operation Mode IPSec Setting Enable IP Security EnablelPSec Dead Peer Detection Mode IPSecDPDMode SIP User s Manual Mediant 2000 Description Secret used to authenticate the device to the RADIUS server Should be a cryptographically strong password Defines the default access level for the device when the RADIUS authentication response doesn t include an access level attribute The valid range is 0 to 255 The default value is 200 Security Administrator Defines device behavior upon a RADIUS timeout 0 Deny Access Denies access 1 Verify Access Locally Checks password locally default Defines the device s mode of operation regarding the timer configured by the parameter RadiusLocalCacheTimeout that determines the validity of the user name and password verified by the RADIUS server 0 Absolute Expiry Timer when you access a Web page the timeout doesn t reset but instead continues decreasing 1 Reset Timer Upon Access upon each access to a Web page the timeout always resets reverts to the initial value configured by RadiusLocalCacheTimeout Defines the time in seconds the locally stored user name and password verified by the RADIUS server are valid When this time expires the user name and password become invalid and a must be re verified with the RADIUS server The valid
275. ce attempts to recover the clock from the trunk defined by the parameter TDMBusLocalReference Note This parameter is relevant only if the parameter TDMBusClockSource is set to 4 Enables or disables the PSTN trunk auto fallback reverting feature If enabled and a trunk returning to service has an AutoClockTrunkPriority parameter value refer to Configuring the Trunk Settings on page 82 that is higher than the priority of the local reference trunk set in the TDMBusLocalReference parameter the local reference reverts to the trunk with the higher priority that has returned to service for the device s clock source 0 Disable default 1 Enable Note This parameter is applicable only when the TDMBusPSTNAutoClockEnable parameter is set to 1 Selects the clock source to which the device synchronizes 1 Internal Generate clock from local source default 4 Network Recover clock from PSTN line For detailed information on configuring the device s clock settings refer to Clock Settings on page 393 219 November 2008 7 wa ei AudioCodes Mediant 2000 3 5 Management Tab The Management tab on the Navigation bar displays all menus related to device management These menus appear in the Navigation tree and include the following m Management Configuration refer to Management Configuration on page 220 m Software Update refer to Software Update on page 231 3 5 1 Management Configuration The Ma
276. ch As long as delays in the network do not change jitter by more than 10 msec from one packet to the next there is always a sample in the buffer for the coder to use If there is more than 10 msec of delay at any time during the call the packet arrives too late The coder tries to access a frame and is not able to find one The coder must produce a voice sample even if a frame is not available It therefore compensates for the missing packet by adding a Bad Frame Interpolation BFI packet This loss is then flagged as the buffer being too small The dynamic algorithm then causes the size of the buffer to increase for the next voice session The size of the buffer may decrease again if the device notices that the buffer is not filling up as much as expected At no time does the buffer decrease to less than the minimum size configured by the Minimum delay parameter For certain scenarios the Optimization Factor is set to 13 One of the purposes of the Jitter Buffer mechanism is to compensate for clock drift If the two sides of the VoIP call are not synchronized to the same clock source one RTP source generates packets at a lower rate causing under runs at the remote Jitter Buffer In normal operation optimization factor 0 to 12 the Jitter Buffer mechanism detects and compensates for the clock drift by occasionally dropping a voice packet or by adding a BFI packet Fax and modem devices are sensitive to small packet losses or to added BFI packe
277. ch page in the Scenario is referred to as a Step For each Step you can select up to 25 parameters in the page that you want available in the Scenario Therefore the Scenario feature is useful in that it allows you quick and easy access to commonly used configuration parameters specific to your network environment When you login to the Web interface your Scenario is displayed in the Navigation tree thereby facilitating your configuration Instead of creating a Scenario you can also load an existing Scenario from a PC to the device refer to Loading a Scenario to the Device on page 39 3 3 5 1 Creating a Scenario The Web interface allows you to create one Scenario with up to 20 configuration pages as described in the procedure below gt To create a Scenario take these 10 steps 1 On the Navigation bar click the Scenarios tab a message box appears requesting you to confirm creation of a Scenario Figure 3 14 Scenario Creation Confirm Message Box Microsoft Internet Explorer A Create a new scenario Note If a Scenario already exists the Scenario Loading message box appears 2 Click OK the Scenario mode appears in the Navigation tree as well as the menus of the Configuration tab Note If a Scenario already exists and you wish to create a new one click the Create Scenario button and then click OK in the subsequent message box 3 In the Scenario Name field enter an arbitrary name for the Scenario 4 On
278. coming INVITE and REGISTER requests are forwarded to the defined Proxy list in SASProxySet in Normal mode and handled by the SAS application in Emergency mode default 1 Always Emergency The SAS application does not use Keep Alive messages towards the SASProxySet and instead always operates in Emergency mode as if no Proxy in the SIP User s Manual 296 Document LTRT 68808 SIP User s Manual Parameter SASBindingMode SASEnableENUM SASRegistrationManipulation SASEmergencyNumbers Version 5 6 4 ini File Configuration Description SASProxySet is available 2 Ignore REGISTER Use regular SAS Normal Emergency logic Same as option 0 but when in Normal mode incoming REGISTER requests are ignored Determines the SAS application database binding mode 0 URI If the incoming AoR in the INVITE requests is using a tel URI or user phone is defined the binding is performed according to the user part of the URI only Otherwise the binding is according to the entire URI i e User Host default 1 User Part only The binding is always performed according to the User Part only Determines whether the SAS application uses ENUM queries to route incoming INVITE requests when in Emergency mode Once an INVITE is received in Emergency mode the SAS database of registered users is searched for a matching AoR If not found the Redundant SAS servers are searched If there is still no match
279. ctive Version 5 6 Contents November 2008 Gg wi AudioCodes mca Figure 3 115 rinks amp Channels Stats Page scien seantsncadadeticiesisadinabaeniedaienendandieiantiinieksdannmannadin 246 Figure 3 116 Example of a Selected Page Icon for Displaying Trunks 17 24 247 Figure 3 117 Basic Channel nformmation Pages ssis aninion Ai 248 FOUE ot aa OE a 2s ae ania a a a aa 249 Figure 3 119 Call Routing Status Page iisisissrsisinsirininieinideni inini ennienni anin Eae Ei ia 251 Figure 3 120 SAS Registered Users Page eseeseenerrriecrrerrrcerreeriursrerrreerrrnedeenreseunnnnttanaeiinnaaeennaaaaa 251 Figure 3 12 EIP Connecti PAGE iiirss cccitecsntincionedhsserthattine dean tds itaccaanals aaema rn dendeasauadienies 252 Figure 6 1 Example of a User Information File sacs ss cscs chncesesshnnweds rman tues cindenatineadnlssipeeducannednnsnbeeabacennadies 341 Figure 7 1 Device s SAS Agent Redirecting Emergency Calls to PSTN ssessesrsseeerreseerrsseerene 348 Foue a SPERO a ae espa Shc cera vienna ned naa ls a a antenatal 369 Figure 7 3 Example Setup for Routing Between ITSP and Enterprise PBX ccseeseeeeeeeeees 374 Figure 7 4 Configuring Proxy Set ID 1 in the Proxy Sets Table Page 375 Figure 7 5 Configuring IP Groups 1 and 2 in the IP Group Table Page e eee 375 Figure 7 6 Assign the Trunk to Trunk Group ID 1 in the Trunk Group Table Page 376 Figure 7 7 Configuring Trunk Group 1 for
280. d receive T 38 packets default 1 Use the same port as the RTP port to send receive T 38 packets Notes For this parameter to take effect you must reset the device When the device is configured to use V 152 to negotiate audio and T 38 coders the UDP port published in SDP for RTP and for T38 must be different Therefore set the the parameter T38UseRTPPort to 0 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 292 Document LTRT 68808 SIP User s Manual Parameter SIPMaxRtx SipT1Rtx SipT2Rtx EnableEarlyMedia IgnoreAlertAfterEarlyMedia EnableTransfer XferPrefix EnableMicrosofExt XferPrefixIP2Tel Version 5 6 4 ini File Configuration Description For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 Determines the device s
281. d TelProfile JitterBufMinDelay TelProfile JitterBu OptFactor TelProfile IPDiff Serv TelProfile SigIPDiffServ TelProfile DtmfVolume TelProfile InputGain TelProfile VoiceVolume TelProfile EnableReversePolarity TelProfile EnableCurrentDisconnect TelProfile EnableDigitDelivery TelProfile EnableEC TelProfile MWIAnalog TelProfile MWIDisplay TelProfile FlashHookPeriod TelProfile EnableEarlyMedia TelProfile ProgressIndicator2IP TelProfile 1 voice 1 88 89 88 98 88 TelProfile 2 data 2 8 88 98 88 98 Tel Profile rN A On O iL Siah p LOO 15 Lp Si AOOO if Ly SOOO Lz 4000 1 16 000 25 16 OWL 4 x 16 7 002 25 16 VOOS 25 WNRORNEFO ei ed SS east omen eee ly vo tou wd wt wt wowed WNHROWNEHFO OoOOO0O0C00 10 2 QSIG Tunneling The device supports QSIG tunneling over SIP according to IETF draft Tunnelling of QSIG over SIP draft elwell sipping qsig tunnel 03 and the ECMA 355 ISO IEC 22535 standard This method enables all QSIG messages to be sent as raw data in corresponding SIP messages using a dedicated message body This mechanism is useful for two QSIG subscribers connected to the same or different QSIG PBX to communicate with each other over an IP network Tunneling is supported in both directions Tel to IP and IP to Tel Version 5 6 405 November 2008 A Ee AudioCodes Mediant 2000 The term tunneling means that messages are transferred as
282. d dynamic tables of other applications are empty but static tables are not A table is defined as a secret table i e concealed if it contains at least one secret data field or if it depends on another secret table For example in the IPSec application IPSec tables are defined as secret tables as the IKE table contains a pre shared key that must be concealed Therefore the SPD table that depends on the IKE table is defined as a secret table as well Secret tables are always concealed when loading an ini file to the device However there is a commented title that states that the secret table exists in the device but is not to be revealed Secret tables are always stored in the device s non volatile memory and can be overwritten by new tables that are provided in a new ini file If a secret table appears in an ini file it replaces the current table regardless of its content To delete a secret table from the device include an empty table of the same type with no data lines as part of a new ini file The ini file table parameter is composed of the following elements E Title of the table The name of the table in square brackets e g MY_TABLE_NAME E Format line Specifies the columns parameters of the table by their string names that are to be configured e The first word of the Format line must be FORMAT followed by the Index field name and then an equal sign After the equal sign the names of the parameters items
283. d IP profiles are identical the Tel Profile parameters are applied Two adjacent dollar signs indicate that the parameter s default value is used PProfile can be used in the Tel to IP Routing or Outbound IP Routing Table if EnableSBC is set to 1 and IP to Trunk Group Routing tables Prefix and PSTNPrefix parameters The Profile Name assigned to a Profile index must enable users to identify it intuitively and easily To configure the IP Profile table using the Web interface refer to IP Profile Settings on page 193 299 November 2008 A ge AudioCodes Mediant 2000 Parameter Description Fora description of using ini file table parameters refer to Structure of ini File Table Parameters on page 257 TelProfile This ini file table parameter configures the Tel Profile Settings table The format of this parameter is as follows TelProfile FORMAT TelProfile_Index TelProfile_ProfileName TelProfile_TelPreference TelProfile_CodersGroupID TelProfile_IsFaxUsed TelProfile_JitterBufMinDelay TelProfile_JitterBufOptFactor TelProfile_IPDiffServ TelProfile_SigIPDiffServ TelProfile_DtmfVolume TelProfile_InputGain TelProfile_VoiceVolume N A N A TelProfile_EnableDigitDelivery TelProfile EnableEC N A N A TelProfile_FlashHookPeriod TelProfile_EnableEarlyMedia TelProfile_ProgressIndicator2IP TelProfile_TimeForReorderTone N A N A N A TelProfile Indicates com
284. d on the interface 3 OAM amp Media Only OAMP and Media RTP applications are allowed on the interface 4 OAM amp Control Only OAMP and Call Control applications are allowed on the interface 5 Media amp Control Only Media RTP and Call Control applications are allowed on the interface Application Type 6 All All the applications are allowed on the interface Notes Only one IPv4 interface of OAM can be configured Only one IPv4 interface of Control can be configured Atleast one interface with Media must be configured The IPv4 IP address in dotted decimal notation IP Address Note Each interface must be assigned a unique IP address Version 5 6 55 November 2008 A ge AudioCodes Mediant 2000 Parameter Prefix Length Gateway VLAN ID Interface Name General Parameters VLAN Mode VIANMode Native VLAN ID VLANNativeVlanID SIP User s Manual Description This column lists the number of 1 bits in the subnet mask i e replaces the standard dotted decimal representation of the subnet mask for IPv4 interfaces For example A subnet mask of 255 0 0 0 is represented by a prefix length of 8 i e 11111111 00000000 00000000 00000000 anda subnet mask of 255 255 255 252 is represented by a prefix length of 30 i e 11111111 11111111 11111111 11111100 The prefix length is a Classless Inter Domain Routing CIDR style presentation of a dotted decimal subnet nota
285. d to PSTN Phones To configure support for emergency calls configure the parameters below The device and the SAS feature are configured independently If the device and the SAS agent use different proxies then the device s proxy server is defined using the Use Default Proxy parameter while the SAS proxy agent is defined using the Proxy Set table and SASProxySet parameter E EnableSAS 1 E SASLocalSIPUDPPort default 5080 E IsProxyUsed 1 E ProxylP 0 lt external proxy IP address device gt E ProxylP 1 lt external proxy IP address SAS gt E IsRegisterNeeded 1 for the device E IsFallbackUsed 0 E SASRegistrationTime lt expiration time that SAS returns in the 200 OK to REGISTER in Emergency mode gt default 20 E SASDefaultGatewayIP lt SAS gateway IP address gt E SASProxySet 1 SIP User s Manual 348 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 7 4 Configuring the DTMF Transport Types You can control the way DTMF digits are transported over the IP network to the remote endpoint by using one of the following modes Version 5 6 Using INFO message according to Nortel IETF draft DTMF digits are carried to the remote side in INFO messages To enable this mode define the following e RxDTMFOption 0 ini file Declare RFC 2833 in SDP field No Web interface refer to DTMF amp Dialing Parameters on page 147 e TxDTMFOption 1 ini file
286. d to configure the port for best performance and highest bandwidth i e Full Duplex with 100Base TX but at the same time adhering to the guidelines listed above Note that when remote configuration is performed the device should be in the correct Ethernet setting prior to the time this parameter takes effect When for example the device is configured using BootP TFTP the device performs many Ethernet based transactions prior to reading the ini file containing this device configuration parameter To resolve this problem the device always uses the last Ethernet setup mode configured In this way if you want to configure the device to operate in a new network environment in which the current Ethernet setting of the device is invalid you should first modify this parameter in the current network so that the new setting holds next time the device is restarted After reconfiguration has completed connect the device to the new network and restart it As a result the remote configuration process that occurs in the new network uses a valid Ethernet configuration Version 5 6 379 November 2008 8 2 8 3 A EA AudioCodes Mediant 2000 Ethernet Interface Redundancy The device supports an Ethernet redundancy scheme At the beginning of the start up procedure the device tests whether the primary Ethernet interface is connected by checking the existence of the Ethernet link carrier If it s connected the start up procedure commences as u
287. de activated only after a channel is already open The direction of the detection PSTN or IP can also be configured The range is 0 to 7 where 0 is the best detection of an answering machine and 7 is the best detection of a live call i e voice detected The default is 3 For a detailed description on AMD refer to Answer Machine Detector AMD on page 343 Note To enable the AMD feature set the ini file parameter EnableDSPIPMDetectors to 1 Activates the Automatic Gain Control AGC mechanism The AGC mechanism adjusts the level of the received signal to maintain a steady configurable volume level 0 Disable default 1 Enable Note For a description on AGC refer to Automatic Gain Control AGC on page 401 Determines the AGC convergence rate 0 0 25 dB sec 1 0 50 dB sec 2 0 75dB sec 3 1 00 dB sec default 4 1 25 dB sec 5 1 50 dB sec 6 1 75 dB sec 7 2 00 dB sec 8 2 50 dB sec 9 3 00 dB sec 10 3 50 dB sec 77 November 2008 ca AudioCodes Parameter AGC Redirection AGCRedirection AGC Target Energy AGCTargetEnergy Enable Energy Detector EnableEnergyDetector Energy Detector Quality Factor EnergyDetectorQualityFactor Energy Detector Threshold EnergyDetectorThreshold Enable Pattern Detector EnablePatternDetector SIP User s Manual Mediant 2000 Description 11 4 00 dB sec 12 4 50 dB sec 13 5 00 dB sec 14 5 5
288. dem signal in the current voice coder refer to Fax Modem Transparent Mode on page 354 E Transparent with events passing the fax modem signal in the current voice coder with adaptations refer to Fax Modem Transparent with Events Mode on page 355 G 711 Transport switching to G 711 when fax modem is detected refer to G 711 Fax Modem Transport Mode on page 355 m Fax fallback to G 711 if T 38 is not supported refer to Fax Fallback on page 356 Adaptations refer to automatic reconfiguration of certain DSP features for handling fax modem streams differently than voice 7 5 2 1 1 38 Fax Relay Mode In Fax Relay mode fax signals are transferred using the T 38 protocol T 38 is an ITU standard for sending fax across IP networks in real time mode The device currently supports only the T 38 UDP syntax T 38 can be configured in the following ways m Switching to T 38 mode using SIP Re INVITE messages refer to Switching to T 38 Mode using SIP Re INVITE on page 351 m Automatically switching to T 38 mode without using SIP Re INVITE messages refer to Automatically Switching to T 38 Mode without SIP Re INVITE on page 352 When fax transmission ends the reverse switching from fax relay to voice is automatically performed at both the local and remote endpoints You can change the fax rate declared in the SDP using the parameter FaxRelayMaxRate this parameter doesn t affect the actual transmission rate In
289. dentifier VLANControlVlanID The valid range is 1 to 4094 The default value is 2 Media VLAN ID Defines the Media VLAN identifier VLANMediaVlanID The valid range is 1 to 4094 The default value is 3 NAT Settings NAT IP Address Global public IP address of the device to enable static Network StaticNatIP Address Translation NAT between the device and the Internet 3 4 1 2 Configuring the Multiple Interface Table The Multiple Interface Table page allows you to configure up to three logical network interfaces each with its own IP address unique VLAN ID if enabled interface name and application types i e Control Media and or Operations Administration Maintenance and Provisioning OAMP permitted on the interface In addition this page provides VLAN related parameters for enabling VLANs and for defining the Native VLAN ID VLAN ID to which incoming untagged packets are assigned For assigning VLAN priorities and Differentiated Services DiffServ for the supported Class of Service CoS refer to Configuring the QoS Settings on page 63 Version 5 6 53 November 2008 A EA AudioCodes Mediant 2000 Once you access the Multiple Interface Table page the IP Settings page is no longer available You can view all added IP interfaces that are currently active in the IP Active Interfaces page refer to Viewing Active IP Interfaces on page 244 You can also configure this table using the ini file
290. dentifying the IP Group from where the INVITE is sent for obtaining a digest user password from the Account table if there is a need to authenticate subsequent SIP requests in the call The IP Group can also be implemented in IP to Tel call routing or inbound IP routing as a source IP Group The IP Groups are assigned various entities such as a Proxy Set ID which represents an IP address created in Proxy Sets Table on page 141 You can also assign the IP Group with a host name and other parameters that reflect parameters sent in SIP Request From To By default if you disable the use of a proxy i e IsProxyUsed is set to 0 then only one IP Group is defined and working with multiple IP Groups is not valid You can also configure the IP Groups table using the ini file table parameter IPGroup refer to SIP Configuration Parameters on page 284 gt To configure IP Groups take these 4 steps 1 Open the IP Group Table page Configuration tab gt Protocol Configuration menu gt Trunk IP Group submenu gt IP Group Table page item Figure 3 84 IP Group Table Page SIP Re Ra Routing Mode Mod terxdied Standard 2 Configure the IP group parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Version 5 6 201 November 2008 A Ee AudioCodes Mediant 2000 Parameter Type Descrip
291. dicate that the clock is running too fast on the client The client slowly robs bits from the clock counter to update the clock to the correct time If the clock is running too slow then in an effort to catch the clock up bits are added to the counter causing the clock to update quicker and catch up to the correct time The advantage of this method is that it does not introduce any disparity in the system time that is noticeable to an end user or that could corrupt call timeouts and timestamps IP QoS via Differentiated Services DiffServ DiffServ is an architecture providing different types or levels of service for IP traffic DiffServ according to RFC 2474 offers the capability to prioritize certain traffic types depending on their priority thereby accomplishing a higher level QoS at the expense of other traffic types By prioritizing packets DiffServ routers can minimize transmission delays for time sensitive packets such as VoIP packets The device can be configured to set a different DiffServ value to IP packets according to their class of service Network Premium Media Premium Control Gold and Bronze The DiffServ parameters are described in Networking Parameters on page 260 For the mapping of an application to its class of service refer to IEEE 802 1p Q VLANs and Priority on page 385 VLANS and Multiple IPs Multiple IPs Media Control and Management OAMP traffic in the device can be assigned one of the followin
292. dicator of the calling party s number for IP to Tel ISDN calls 1 Not Configured Not configured interworking from IP to ISDN default 0 User Provided user provided not screened 1 User Passed user provided verified and passed 2 User Failed user provided verified and failed 3 Network Provided network provided Adds an optional Information Element IE data in hex format to ISDN SETUP messages For example to add IE 0x20 0x02 0x00 0xe1 enter the following value for this parameter 200200e1 Note This IE is sent from the Trunk Group IDs defined by the parameter SendIEonTG Defines Trunk Group IDs up to 50 characters from where the optional ISDN IE defined by the parameter AddlEinSetup is sent For example 1 2 4 10 12 6 Enables ISDN PRI to SIP interworking 0 Disable Disabled default 1 Enable Enable transfer of User to User Information Element UUIE from PRI to SIP The device supports the following ISDN PRI to SIP interworking SETUP to SIP INVITE CONNECT to SIP 200 OK USER INFORMATION to SIP INFO ALERT to SIP 18x response and DISCONNECT to SIP BYE response messages Note The interworking of User to User IE to SIP INFO is supported only on the 4ESS PRI variant 209 November 2008 ca AudioCodes Parameter Enable User to User IE for IP to Tel EnableUUIIP2Tel Enable ISDN Tunneling Tel to IP EnablelSDNTunnelingTel2IP Enable QSIG Tunneling
293. disconnect cause Stop 26 Disconnect 30 code Numeric Ace Cause Start 26 H323 Gw ID 33 Name of the gateway String SIPIDString aa Acc Start 26 SIP Call ID 34 SIP Call ID String abcde ac com d Acc The call s terminator Call PSTN terminated call Stop 2 Terminator Yes IP terminated call Suing ge Ne Acc No String 8004567145 Start Called cC i Station ID Destination phone Stop String 2427456425 number Acc Start Calling Calling Party Number Acc 31 Station ID ANI String 5135672127 Stop Acc Account Request Type Start Acct Status start or stop Acc 40 Note start isn t Numeric 1 start 2 stop Type Stop supported on the Calling Acc Card application eed Start No of seconds tried in 41 pert Delay sending a particular Numeric 5 AGE Time Stop record A cc Number of octets 42 pate received for that call Numeric ri duration Acct Output Number of octets sent for Stop Octets that call duration Nuniene Acc Start Acct A unique accounting Acc 44 identifier match start amp String 34832 Session ID Stop stop A cc Acct For how many seconds Sto 46 Session the user received the Numeric A p f cc Time service Acct Input Number of packets Stop a Packets received during the call Mameng Acc SIP User s Manual 364 Document LTRT 68808 SIP User s Manual Attribute Number 48 61 VSA No Attribute Name Acct Output Packets NAS Port Type Response Attributes
294. e comprising a 19 inch 1U chassis with optional dual AC or single DC power supplies The deployment architecture can include several devices in branch or departmental offices connected to local PBXs Call routing is performed by the devices using internal routing or SIP Proxy s The device enables users to make cost effective long distance or international telephone fax calls between distributed company offices using their existing telephones fax These calls can be routed over the existing network using state of the art compression techniques ensuring that voice traffic uses minimum bandwidth The device can also route calls over the network using SIP signaling protocol enabling the deployment of Voice over Packet solutions in environments where access is enabled to PSTN subscribers by using a trunking device This provides the ability to transmit voice and telephony signals between a packet network and a TDM network The device is offered as a 1 module up to 240 channels or 8 trunk spans or 2 module for 480 channels or 16 trunk spans only platform The latter configuration supports two TrunkPack modules each having its own IP address Configuration instructions in this document relate to the device as a 1 module platform and must be repeated for the second module as well For channel capacity refer to the device s specifications in Selected Technical Specifications on page 409 Version 5 6 15 November 2008 A C al Audio
295. e default 1 Detailed self test mode full test of DSPs PCM Switch LAN PHY and Flash 2 A quicker version of the Detailed self test mode full test of DSPs PCM Switch LAN PHY but partial test of Flash For detailed information refer to the Product Reference Manual For a description of this parameter refer to Advanced Parameters on page 151 The Activity Log mechanism enables the device to send log messages to a Syslog server that report certain types of Web actions according to a pre defined filter The following filters are available PVC Parameters Value Change Changes made on the fly to parameters AFL Auxiliary Files Loading Loading of auxiliary files e g via Certificate screen DR Device Reset Reset of device via the Maintenance Actions screen FB Flash Memory Burning Burning of files parameters to flash in Maintenance Actions screen SWU Device Software Update cmp loading via the Software Upgrade Wizard ARD Access to Restricted Domains Access to Restricted Domains The following screens are restricted 1 ini parameters AdminPage 2 General Security Settings 3 Configuration File 4 IPSec IKE tables 5 Software Upgrade Key 6 Internal Firewall 7 Web Access List 8 Web User Accounts NAA Non Authorized Access Attempt to access the Web interface with a false empty user
296. e 109 For a description of this parameter refer to Configuring the General Security Settings on page 109 276 Document LTRT 68808 SIP User s Manual Parameter PeerHostNameVerificationMode VerifyServerCertificate TLSRemoteSubjectName OCSPEnable OCSPServerlP OCSPServerPort OCSPDefaultResponse EnableSecureStartup SSHAdminKey SSHRequirePublicKey IPSec Parameters EnablelPSec IPSecDPDMode IPSEC_SPD_TABLE Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring the General Security Settings on page 109 Enables or disables certificate checking using Online Certificate Status Protocol OCSP 0 Disable default 1 Enable Defines the IP address of the OCSP server The default IP address is 0 0 0 0 Defines the OCSP server s TCP port number The default port number is 2560 Determines the default OCSP behavior when the server cannot be contacted 0 Rejects peer certificate default 1 Allows peer certificate Enables the Secure Startup mode In this mode downloading the ini file to the device is restricted to a URL provided in initial configuration see parameter IniFileURL or using DHCP
297. e 4 8 Media Server ini File Parameters Parameter AMRCoderHeaderFormat EnableAGC AGCGainSlope AGCRedirection AGCTargetEnergy AGCMinGain AGCMaxGain AGCDisableFastAdaptation AMDDetectionSensitivity AMDTimeout AMDDetectionDirection Version 5 6 Description Determines the format of the AMR header 0 Non standard multiple frames packing in a single RTP frame Each frame has a CMR and TOC header 1 Reserved 2 AMR Header according to RFC 3267 Octet Aligned header format 3 AMR is passed using the AMR IF2 format For a description of this parameter refer to Configuring the IPmedia Settings on page 76 For a description of this parameter refer to Configuring the IPmedia Settings on page 76 For a description of this parameter refer to Configuring the IPmedia Settings on page 76 For a description of this parameter refer to Configuring the IPmedia Settings on page 76 Defines the minimum gain in dB by the AGC when activated The range is 0 to 31 The default is 20 Defines the maximum gain in dB by the AGC when activated The range is 0 to 18 The default is 15 Disables the AGC Fast Adaptation mode 0 Disable default 1 Enable For a description of this parameter refer to Configuring the IPmedia Settings on page 76 Timeout in msec between receiving CONNECT messages from the ISDN and sending Answering Machine Det
298. e containing the Prerecorded Tones The Dial Plan name up to 11 character strings that is used on the specific trunk 331 November 2008 A Ee AudioCodes Mediant 2000 Parameter Description DialPlanFileName The name and path of the file containing dial plan configuration for CAS and SIP protocols This file should be constructed using the TrunkPack Conversion Utility refer to the Product Reference Manual supplied as part of the software package on the CD accompanying the device UserinfoFileName The name and path of the file containing the User Information data SetDefaultOnIniFileProcess Determines if all the device s parameters are set to their defaults before processing the updated ini file 0 Disable parameters not included in the downloaded ini file are not returned to default settings i e retain their current settings 1 Enable default SaveConfiguration Determines if the device s configuration parameters and files is saved to flash non volatile memory 0 Configuration isn t saved to flash memory 1 Configuration is saved to flash memory default SIP User s Manual 332 Document LTRT 68808 SIP User s Manual 5 Default Settings 5 Default Settings You can restore the device s factory default settings or define your own default settings for the device 5 1 Defining Default Settings The device is shipped with factory default configuration values stored on its non volatile
299. e device In addition before resetting the device you can choose the following options Save the device s current configuration to the device s flash memory non volatile Perform a graceful shutdown i e device reset starts only after a user defined time expires i e timeout or after no more active traffic exists the earliest thereof To reset the device take these 6 steps Open the Maintenance Actions page refer to Maintenance Actions on page 228 Under the Reset Configuration group from the Burn To FLASH drop down list select one of the following options e Yes The device s current configuration is saved burned to the flash memory prior to reset default e __ No Resets the device without saving the current configuration to flash discards all unsaved modifications SIP User s Manual 228 Document LTRT 68808 SIP User s Manual 3 Web Based Management Under the Reset Configuration group from the Graceful Option drop down list select one of the following options e Yes Reset starts only after the user defined time in the Shutdown Timeout field refer to Step 4 expires or after no more active traffic exists the earliest thereof In addition no new traffic is accepted e _ No Reset starts regardless of traffic and any existing traffic is terminated at once In the Shutdown Timeout field relevant only if the Graceful Option in the previous step is set to Yes enter
300. e devices both devices start using G 711 packets with standard payload type 8 for G 711 A Law and 0 for G 711 Mu Law In this mode no Re INVITE messages are sent The voice channel is optimized for fax modem transmission Same as for usual bypass mode The parameters defining payload type for the proprietary AudioCodes Bypass mode FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass When configured for NSE mode the device includes in its SDP the following line a rtpmap 100 X NSE 8000 where 100 is the NSE payload type The Cisco gateway must include the following definition modem passthrough nse payload type 100 codec g711alaw Version 5 6 353 November 2008 A Ee AudioCodes Mediant 2000 To configure NSE mode perform the following configurations IsFaxUsed 0 FaxTransportMode 2 NSEMode 1 NSEPayloadType 100 V21ModemtTransportType 2 V22ModemtTransportType 2 V23ModemtTransportType 2 V32ModemtTransportType 2 V34ModemTransportType 2 BellModemTransportType 2 7 5 2 4 Fax Modem Transparent Mode In this mode fax and modem signals are transferred using the current voice coder without notifications to the user and without automatic adaptations It s possible to use the Profiles mechanism refer to Configuring the Profile Definitions on page 190 to apply certain adaptations to the channel used for fax modem e g to use the coder G 711 to set the jitter buffer optimizatio
301. e dialed address or dialed ANI parameters When it is disabled the address and ANI strings remain without delimiters 311 November 2008 ca AudioCodes Parameter CASDelimitersPaddingUsage CasStateMachineGenerateDigi tOnTime CasStateMachineGeneratelnte rDigitTime CasStateMachineDTMFMaxOn DetectionTime CasStateMachineDTMFMinOn DetectionTime CasStateMachineMaxNumOflin comingAddressDigits CasStateMachineMaxNumOflin comingANIDigits CasStateMachineCollectANl CasStateMachineDigitSignalin gSystem EnableDSPIPMDetectors XChannelHeader AddlEinSetup SendlEonTG ISDNDMSTimerT310 SIP User s Manual Mediant 2000 Description Defines the digits string delimiter padding usage per trunk 0 default default address string padding XXX where XXX is the digit string that begins with and ends with when using padding 1 special use of asterisks delimiters XXX YYY where XXX is the address YYY is the source phone number and is the only delimiter padding For a description of this parameter refer to Configuring the CAS State Machines on page 97 For a description of this parameter refer to Configuring the CAS State Machines on page 97 For a description of this parameter refer to Configuring the CAS State Machines on page 97 For a description of this parameter refer to Configuring the CAS State Machines on page 97 For a description of this parameter refer
302. e disabled You can add up to 24entries in this table Version 5 6 Note You can also configure the Trunk Groups using the ini file table parameter TrunkGroup_x to refer to Number Manipulation and Routing Parameters on page 313 195 November 2008 A ge AudioCodes Mediant 2000 gt To configure the Trunk Group table take these 4 steps 1 Open the Trunk Group Table page Configuration tab gt Protocol Configuration menu gt Trunk IP Group submenu gt Trunk Group page item Figure 3 82 Trunk Group Table Page x Add Phone Context s Prefix Disable Trunk Group Index 1 12 Group To Trunk lindex ink Trunk T aikai Phone sini Group 1D 1 Profile ID i2 M 6000 7000 6000 2 Configure the Trunk Group according to the table below 3 Click the Submit button to save your changes 4 To save the changes to the flash memory refer to Saving Configuration on page 230 Table 3 49 Trunk Group Table Description Parameter Description From Trunk Starting physical Trunk number The number of listed Trunks TrunkGroup_FirstTrunkld depends on the device s hardware configuration To Trunk Ending physical Trunk number The number of listed Trunks TrunkGroup_LastTrunkld depends on the device s hardware configuration Channels The device s Trunk B channels To enable the channels enter TrunkG
303. e doesn t contain a called party number and no phone number Special Digit Representation UseDigitForSpecialDTMF is configured in the Trunk Group table refer to Configuring the Trunk Group Table on page 195 The parameter is used as a starting number for the list of channels comprising all trunk groups in the device The default value is 1000 Defines the representation for special digits and that are used for out of band DTMF signaling using SIP INFO NOTIFY 0 Special Uses the strings and default 1 Numeric Uses the numerical values 10 and 11 3 4 7 2 Configuring the SIP Advanced Parameters Version 5 6 The SIP Advanced Parameters submenu allows you to configure advanced SIP control protocol parameters This submenu contains the following page items Advanced Parameters refer to General Parameters on page 151 Supplementary Services refer to Supplementary Services on page 159 Stand Alone Survivability refer to Stand Alone Survivability on page 161 SBC Configuration refer to SBC Configuration on page 163 151 November 2008 ca AudioCodes 3 4 7 2 1 Advanced Parameters Mediant 2000 The Advanced Parameters page allows you to configure general control protocol parameters gt To configure the advanced general protocol parameters take these 4 steps 1 Open the Advanced Parameters page Configuration tab gt Protocol Configurat
304. e number of additional digits by a comma E Empty lines are ignored m Lines beginning with a semicolon are ignored Multiple dial plans may be specified in one file A name in square brackets on a separate line indicates the beginning of a new dial plan Up to eight dial plans can be defined m Asterisks and number signs can be specified as part of the prefix m Numeric ranges are allowed in the prefix E A numeric range is allowed in the number of additional digits Note The prefixes must not overlap Attempting to process an overlapping configuration in the TrunkPack Conversion Utility results in an error message specifying the problematic line 6 6 User Information File The User Information file is a text file that maps PBX extensions connected to the device to global IP numbers In this context a global IP phone number alphanumerical serves as a routing identifier for calls in the IP World The PBX extension uses this mapping to emulate the behavior of an IP phone Note The mapping mechanism is disabled by default and must be activated using the parameter EnableUserlnfoUsage refer to Advanced Parameters on page 151 Each line in the file represents a mapping rule of a single PBX extension Up to 1 000 rules can be configured Each line includes five items separated with commas The items are described in the table below Table 6 1 User Information Items Item Description Maximum
305. e performed due to one of the following scenarios Physically disconnected from the network i e Ethernet cable is disconnected The Ethernet cable is connected but the device can t communicate with any host Note that LAN Watch Dog must be activated EnableLANWatchDog 1 The device can t communicate with the proxy according to the Proxy keep alive mechanism and no other alternative exists to send the call The IP Connectivity mechanism is enabled using AltRoutingTel2IPEnable and there is no connectivity to any destination IP address Note The Busy Out behavior varies between different protocol types Defines the default Release Cause sent to IP for IP to Tel calls when the device initiates a call release and an explicit matching cause for this release isn t found The default release cause is NO ROUTE_TO_DESTINATION 3 Other common values include NO_CIRCUIT_AVAILABLE 34 DESTINATION _OUT_OF_ORDER 27 etc Notes The default release cause is described in the Q 931 notation and is translated to corresponding SIP 40x or 50x values e g 3 to SIP 404 and 34 to SIP 503 When the Trunk is disconnected or is not synchronized the internal cause is 27 This cause is mapped by default to SIP 502 For mapping SIP to Q 931 and Q 931 to SIP release causes refer to 157 November 2008 ca AudioCodes Parameter Delay After Reset sec GWAppDelayTime Max Number of Active Calls MaxActiveCal
306. e receives a 180 Ringing response with or without SDP and LocallSDNRBSource 1 it plays an RBT and sends an Alert with PI 8 unless the parameter ProgressIndicator2ISDN_ID is configured differently 2 If LocallSDNRBSource 0 the device doesn t play an RBT and an Alert message without PI is sent to the ISDN In this case the PBX PSTN should play the RBT to the originating terminal by itself Note Receipt of a 183 response doesn t cause the device with ISDN protocol type to play an RBT the device issues a Progress message unless SIP183Behaviour 1 If SIP183Behaviour 1 the 183 response is treated the same way as a 180 Ringing response 2 Prefer IP Play according to Early Media default If a 180 response is received and the voice channel is already open due to a previous 183 early media response or due to an SDP in the 94 Document LTRT 68808 SIP User s Manual ini File Field Name Web Parameter Name B Channel Negotiation BChannelNegotiationForTr unk_ID RTP Only Mode RTPOnlyModeForTrunk_ID Version 5 6 3 Web Based Management Valid Range and Description current 180 response the device with ISDN CAS protocol type doesn t play the RBT PI 8 is sent in an ISDN Alert message unless the parameter ProgressIndicator2ISDN_ID is configured differently If a 180 response is received but the early media voice channel is not opened the device with CAS protocol type plays an RBT to
307. eCla ssControIPriority VianGoldServiceClassPrio rity VLANBronzeServiceClass Priority EnableDNSasOAM EnableNTPasOAM VLANSendNonTaggedOn Native Multiple IPs Parameters EnableMultiplelPs LocalMedialPAddress LocalMediaSubnetMask LocalMediaDefaultGW Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the QoS Settings on page 63 For a description of this parameter refer to Configuring the QoS Settings on page 63 For a description of this parameter refer to Configuring the QoS Settings on page 63 For a description of this parameter refer to Configuring the QoS Settings on page 63 For a description of this parameter refer to Configuring the QoS Settings on page 63 This parameter applies to both Multiple IPs and VLAN mechanisms Multiple IPs Determines the network type for DNS services VLAN Determines the traffic type for DNS services 1 OAMP default 0 Control This parameter applies to both Multiple IPs and VLAN mechanisms Multiple IPs Determines the network type f
308. eb Interface on page 20 Configuration Tab The Configuration tab on the Navigation bar displays all menus related to device configuration These menus appear in the Navigation tree and include the following Network Settings refer to Network Settings on page 50 Media Settings refer to Media Settings on page 65 PSTN Settings refer to PSTN Settings on page 82 SS7 Configuration refer to SS7 Configuration on page 99 Sigtran Configuration refer to Sigtran Configuration on page 99 Security Settings refer to Security Settings on page 99 Protocol Configuration refer to Protocol Configuration on page 120 Advanced Applications refer to Advanced Applications on page 213 TDM Configuration refer to Configuring the TDM Bus Settings on page 218 gt To access the menus of the Configuration tab take this step m On the Navigation bar click the Configuration tab the Navigation tree displays the configuration menus pertaining to the Configuration tab Network Settings The Network Settings menu allows you to configure various networking parameters This menu contains the following page items m P Settings refer to Configuring the IP Settings on page 50 m Application Settings refer to Configuring the Application Settings on page 57 m P Routing Table refer to Configuring the IP Routing Table on page 62 mM QOS Settings refer to Configuring the QoS Settings on page 63 Conf
309. econds after which the call is released Determines whether call transfer is enabled 0 Disable Disable the call transfer service 1 Enable The device responds to a REFER message with the Referred To header to initiate a call transfer default Notes To use call transfer the devices at both ends must support this option To use Call transfer set the parameter EnableHold to 1 Defines the string that is added as a prefix to the transferred forwarded called number when the REFER 3xx message is received Notes The number manipulation rules apply to the user part of the REFER TO Contact URI before it is sent in the INVITE message This parameter can be used to apply different manipulation rules to differentiate transferred number from the originally dialed number Determines whether Call Forward is enabled 0 Disable Disable the Call Forward service 1 Enable Enable Call Forward service default The device doesn t initiate call forward it can only respond to call forward requests Note To use this service the devices at both ends must support this option Determines whether Call Waiting is enabled 0 Disable Disable the Call Waiting service 1 Enable Enable the Call Waiting service default If enabled when the device initiates a Tel to IP call to a destination that is busy it plays a Call Waiting Ringback tone to the caller Notes The device s Call Progress T
310. ection AMD results The valid range is 1 to 30 000 The default is 2 000 i e 2 seconds Determines the AMD Answer Machine Detector detection direction 0 Detection from the PSTN side 1 Detection from the IP side 301 November 2008 ca AudioCodes 4 4 9 Voice Mail Parameters Mediant 2000 The voice mail related ini file configuration parameters are described in the table below For detailed information on the Voice Mail application refer to the CPE Configuration Guide for Voice Mail Table 4 9 Voice Mail ini File Parameters Parameter VoiceMaillnterface SMDI SMDITimeOut LineTransferMode WaitForDialTime MW1IOnCode MWI1OffCode MWi1SuffixCode MW1SourceNumber Description For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 214 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page
311. ed This parameter cannot be changed on the fly and requires a device reset Determines the remote UDP port to where the multiplexed RTP packets are sent and the local UDP port used for incoming multiplexed RTP packets applies to RTP multiplexing The valid range is the range of possible UDP ports 6 000 to 64 000 The default value is 0 i e RTP multiplexing is disabled 74 Document LTRT 68808 SIP User s Manual Parameter RTCP XR Settings 3 Web Based Management Description This parameter cannot be changed on the fly and requires a device reset Note All devices that participate in the same RTP multiplexing session must use this same port Note For a detailed description of RTCP XR reports refer to the Product Reference Manual Enable RTCP XR VQMonEnable Burst Threshold VQMonBurstHR Delay Threshold VQMonDelayTHR R Value Delay Threshold VQMonEOCRVaITHR Minimum Gap Size VQMonGMin RTCP XR Report Mode RTCPXRReportMode RTCP XR Packet Interval RTCPinterval Disable RTCP XR Interval Randomization DisableRTCPRandomize RTCP XR Collection Server RTCPXREscIP Version 5 6 Enables voice quality monitoring and RTCP Extended Reports RTCP XR 0 Disable Disable default 1 Enable Enables Voice quality monitoring excessive burst alert threshold if set to 1 default no alerts are issued Voice quality monitoring excessive delay alert threshold if
312. ed The default value is 0 i e any port Defines the destination port to which the IPSec mechanism is applied The default value is 0 i e any port Defines the protocol type to which the IPSec mechanism is applied 0 Any protocol default 17 UDP 6 TCP Any other protocol type defined by IANA Internet Assigned Numbers Authority Determines the index for the corresponding IKE entry Note that several policies can be associated with a single IKE entry The valid range is 0 to 19 The default value is 0 116 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Name Description IKE Second Phase Parameters Quick Mode SA Lifetime sec Determines the time in seconds that the SA negotiated PsecPolicyLifelnSec in the second IKE session quick mode is valid After the time expires the SA is re negotiated The default value is 28 800 i e 8 hours SA Lifetime KB Determines the lifetime in kilobytes that the SA IPSecPolicyLifelnKB negotiated in the second IKE session quick mode is valid After this size is reached the SA is re negotiated The default value is 0 i e this parameter is ignored These lifetime parameters SA Lifetime sec and SA Lifetime KB determine the duration for which an SA is valid When the lifetime of the SA expires it is automatically renewed by performing the IKE second phase negotiations To refrain from a situation where the SA exp
313. ed Key Exchange Method Index SA Lifetime sec SA Lifetime KB First Proposal Encryption Type First Proposal Authentication Type Second Proposal Encryption Type Second Proposal 4uthentication Type Third Proposal Encryption Type Third Proposal Authentication Type Fourth Proposal Encryption Type Fourth Proposal Authentication Type Transport Control 0 0 0 0 28800 0 Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined SIS SSS S S 2 From the Policy Index drop down list select the rule you want to edit up to 20 policy rules can be configured 3 Configure the IPSec SPD parameters according to the table below 4 Click the button Create the IPSec rule is applied on the fly to the device 5 To save the changes to flash memory refer to Saving Configuration on page 230 If no IPSec methods are defined Encryption Authentication the default settings shown in the following table are applied Table 3 26 Default IKE Second Phase Proposals Proposal Encryption Proposal 0 3DES Proposal 1 3DES Proposal 2 DES Proposal 3 DES Version 5 6 115 Authentication SHA1 MD5 SHA1 MD5 November 2008 ca AudioCodes Mediant 2000 Table 3 27 IPSec SPD Table Configuration Parameters Parameter Name IPSec Mode
314. ed Mail format The valid range is a 47 character string Note This parameter is only relevant when the device is loaded via BootP TFTP For information on loading this file via the Web interface refer to the Product Reference Manual Defines the name of a private key file in unencrypted PEM format to be loaded from the TFTP server Defines the name of the HTTPS server certificate file to be loaded via TFTP The file must be in base64 encoded PEM format The valid range is a 47 character string Note This parameter is only relevant when the device is loaded using BootP TFTP For information on loading this file via the Web interface refer to the Product Reference Manual This ini file table parameter configures the device s access list firewall which defines network traffic filtering rules The format of this parameter is as follows ACCESSLIST FORMAT AccessList_Index AccessList_Source_IP AccessList_Net_Mask AccessList_Start_Port AccessList_End_ Port AccessList_Protocol AccessList_Packet_Size AccessList_Byte_Rate AccessList_Byte_Burst AccessList_Allow_Type ACCESSLIST For example ACCESSLIST FORMAT AccessList_Index AccessList_Source_IP AccessList_Net_Mask AccessList_Start_Port AccessList_End_Port AccessList_Protocol AccessList_Packet_Size AccessList_Byte_Rate AccessList_Byte_Burst AccessList_Allow_Type AccessList 10 mgmt customer com 255 255 255 255 0 80 tcp 0 0 0 allow Acce
315. ed Parameters on page 151 For information on the Syslog refer to the Product Reference Manual SNMP Settings For detailed information on the SNMP parameters that can be configured via the ini file refer to SNMP Parameters on page 282 For detailed information on developing an SNMP based program to manage your device refer to the Product Reference Manual SNMP Trap Destinations Click the arrow L button to configure the SNMP trap destinations refer to Configuring the SNMP Trap Destinations Table on page 222 SNMP Community String Click the arrow button to configure the SNMP community strings refer to Configuring the SNMP Community Strings on page 224 SNMP V3 Table Click the arrow gt button to configure the SNMP V3 users refer to Configuring SNMP V3 Table on page 225 Version 5 6 221 November 2008 ca AudioCodes Parameter SNMP Trusted Managers Enable SNMP DisableSNMP Trap Manager Host Name SNMPTrapManagerHostName Mediant 2000 Description Click the arrow u button to configure the SNMP Trusted Managers refer to Configuring SNMP Trusted Managers on page 226 0 Enable SNMP is enabled default 1 Disable SNMP is disabled and no traps are sent Defines an FQDN of a remote host that is used as an SNMP manager The resolved IP address replaces the last entry in the Trap Manager table defined by the parameter SNMPManagerTablelP_x and the last trap manager entry of
316. ed as a valid B channel ID but timeslot values are converted to reflect the range 1 to 15 and 17 to 31 This is the new QSIG mode of operation When this bit is not set default the channel_id 16 is not allowed as for all ETSI like standards 64 USE T1 PRI PRI interface type is forced to T1 128 USE E1 PRI PRI interface type is forced to E1 256 START WITH B CHAN OOS B channels start in the Out Of Service state OOS 512 CHAN ALLOC LOWEST CC allocates B channels starting from the lowest available B channel id 1024 CHAN ALLOC HIGHEST CC allocates B channels starting from the highest available B channel id Note When using the ini file to configure the device to support several ISDNGeneralCCBehavior features add the individual feature values For example to support both 16 and 32 features set ISDNGeneralCCBehavior 48 i e 16 32 CAS Configuration These parameters only appear if the protocol Type is CAS Table CASTablelndex_x Dial Plan CasTrunkDialPlanName Miscellaneous PSTN Alert Timeout TrunkPSTNAlertTimeout_ D Digital Out Of Service Behavior DigitalOOSBehaviorFor Trunk_ID Version 5 6 Defines CAS protocol for each trunk ID from a list of CAS protocols defined by the parameter CASFileName_Y For example CASFileName_0 E_M_WinkTable dat CASFileName_1 E_M_ImmediateTable dat CASTablelndex_0 CASTablelndex_1 CASTablelndex_2 CASTablelndex_3 Trunks
317. ed w o diagnostic 410 Gone 26 Non selected user clearing 404 Not found 27 Destination out of order 502 Bad gateway 28 Address incomplete 484 Address incomplete 29 Facility rejected 501 Not implemented 30 Response to status enquiry 501 Not implemented 31 Normal unspecified 480 Temporarily unavailable 34 No circuit available 503 Service unavailable 38 Network out of order 503 Service unavailable 41 Temporary failure 503 Service unavailable 42 Switching equipment congestion 503 Service unavailable 43 Access information discarded 502 Bad gateway 44 Requested channel not available 503 Service unavailable 47 Resource unavailable 503 Service unavailable 49 QoS unavailable 503 Service unavailable 50 Facility not subscribed 503 Service unavailable 55 Incoming calls barred within CUG 403 Forbidden 57 Bearer capability not authorized 403 Forbidden 58 Bearer capability not presently available 503 Service unavailable 63 Service option not available 503 Service unavailable 65 Bearer capability not implemented 501 Not implemented 66 Channel type not implemented 480 Temporarily unavailable 69 Requested facility not implemented 503 Service unavailable 70 ee ae intonation Beater 503 Service unavailable 79 Service or option not implemented 501 Not implemented 81 Invalid call reference value 502 Bad gateway Version 5 6 395 November 2008 ca AudioCodes ISDN Release Reason 82 83 84 85 86 87 88 91 95 96 97 98 99 100 10
318. een altered in any way 3 5 2 2 1 Loading via BootP TFTP The procedure below describes how to load a Software Upgrade Key to the device using AudioCodes BootP TFTP Server utility for a detailed description on the BootP utility refer to the Product Reference Manual gt To load a Software Upgrade Key file using BootP TFTP take these 6 steps 1 Place the Software Upgrade Key file typically a txt file in the same folder in which the device s cmp file is located 2 Start the BootP TFTP Server utility Version 5 6 235 November 2008 A Ee AudioCodes Mediant 2000 3 3 5 2 3 From the Services menu choose Clients the Client Configuration screen is displayed From the INI File drop down list select the Software Upgrade Key file Note that the device s cmp file must be specified in the Boot File field Configure the initial BootP TFTP parameters as required and then click OK Reset the device the cmp and Software Upgrade Key files are loaded to the device Note To load the Software Upgrade Key using BootP TFTP the extension name of the key file must be ini Software Upgrade Wizard The Software Upgrade Wizard guides you through the process of software upgrade selecting files and loading them to the device The wizard also enables you to upgrade software while maintaining the existing configuration Using the wizard obligates you to load and burn a cmp file to the device You can cho
319. efer to Saving Configuration on page 230 The logo image must be a GIF JPG or JPEG file The logo image must have a fixed height of 30 pixels The width can be up to 199 pixels the default being 141 pixels The size of the image file can be up to 64 Kbytes SIP User s Manual 42 Document LTRT 68808 SIP User s Manual 3 Web Based Management Tip If you encounter any problem during the loading of the file or you want to restore the default image click the Restore Default Images button gt To replace the default logo with a different image using the ini file take these 3 steps 1 Place your corporate logo image file on the TFTP server in the same folder where the device s ini file is located 2 Configure the ini file parameters as described in the table below For a description on using the ini file refer to Modifying an ini File on page 259 3 Load the ini file to the device using BootP TFTP i e not through the Web interface For detailed information on the BootP TFTP application refer to the Product Reference Manual Table 3 2 ini File Parameters for Changing Logo Image Parameter Description LogoFileName The name of the image file for your corporate logo Use a gif jpg or jpeg image file The default is AudioCodes logo file Note The length of the name of the image file is limited to 48 characters LogoWidth Width in pixels of the logo image The range is 0 199 The default value
320. eived from an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on busy when the original call is received from an external line not an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate call forward on no answer when the original call is received from an external line not an internal extension The valid range is a 120 character string 215 November 2008 ca AudioCodes Parameter Forward on Do Not Disturb Digit Pattern External DigitPatternForwardOnDNDExt Forward on No Reason Digit Pattern External DigitPatternForwardNoReasonExt Internal Call Digit Pattern DigitPatternInternalCall External Call Digit Pattern DigitPatternExternalCall Disconnect Call Digit Pattern TelDisconnectCode Digit To Ignore Digit Pattern DigitPatternDigitTolgnore Message Waiting Indication MWI MWI Off Digit Pattern MWIOffCode MWI On Digit Pattern MWIOnCode MWI Suffix Pattern MWISuffixCode MWI Source Number MW ISourceNumber SMDI Enable SMDI SMDI SIP User s Manual Mediant 2000 Description Determines the digit pattern used by the PBX to indicate call forward on do not disturb when the original call is received from an external line not an internal extension The valid range is a 120 character string Determines the di
321. elease IP Tel Unknown Termination Reason Fax Transaction during the Call Number of Incoming Packets Number of Outgoing Packets Local Packet Loss Number of Outgoing Lost Packets unique RTP ID Call Setup Time Call Connect Time Call Release Time 366 Document LTRT 68808 SIP User s Manual Field Name RTPdelay RTPjitter RTPssrc RemoteRTPssrc RedirectReason TON MeteringPulses NPI RedirectPhonNum 7 IP Telephony Capabilities Description RTP Delay RTP Jitter Local RTP SSRC Remote RTP SSRC Redirect Reason Redirection Phone Number Type Number of Generated Metering Pulses Redirection Phone Number Plan Redirection Phone Number 7 12 Trunk to Trunk Routing Example This example describes two devices each interfacing with the PSTN through four E1 spans Device A is configured to route all incoming Tel to IP calls to Device B Device B generates calls to the PSTN on the same E1 trunk on which the call was originally received in Device A E Device A IP address 192 168 3 50 E Device B IP address 192 168 3 51 The ini file parameters configuration for devices A and B include the following 1 At both devices define four trunk groups each with 30 B channels TrunkGroup_1 0 1 31 1000 TrunkGroup_ 2 1 1 31 2000 TrunkGroup_ 3 2 1 31 3000 TrunkGroup_4 3 1 31 4000 2 At Device A add the originating Trunk Group ID as a prefix to the destination number for Tel to IP calls AddTrunkGroupA
322. els Table 3 63 Color Coding Icons for Trunk and Channel Status Trunk Channel Icon Color Description Icon Color Description Gray Disabled aA Light Blue Inactive ye Green Active OK ad Green Active Yellow RAI Alarm ad Purple SS7 Blue AIS Alarm ad Blue ISDN Signaling m Red LOS LOF Alarm A Gray Non Voice ye Orange D Channel Alarm A Yellow CAS Blocked The Trunks amp Channels Status page also allows you to view detailed information regarding a selected trunk channel as described in the procedure below Version 5 6 247 November 2008 A Ee AudioCodes Mediant 2000 gt To view detailed channel information of a trunk s channel take these 2 steps 1 Click a required channel pertaining to a trunk for which you want to view information the Basic Channel Information page appears displaying basic information about the channel Figure 3 117 Basic Channel Information Page SIP Basic RIP RTCP Voice Settings Ww Channel Identifier 126 Status Inactive Call ID 0 Endpoint ID Call Duration sec o Call Type Voice Call Destination 10 8 55 85 Coder Transparent 2 To view additional channel information click the buttons SIP Basic RTP RTCP and Voice Settings located above on the page 3 6 2 Gateway Statistics The Gateway Statistics page allows you to monitor real time activity such as IP connectivity information call details and call statistics incl
323. en not applicable to the currently opened page Q Burn Saves parameter settings to flash memory refer to Saving Configuration on page 230 Device Actions w Device Opens a drop down menu list with frequently needed commands Actions Load Configuration File opens the Configuration File page for loading an ini file refer to Backing Up and Restoring Configuration on page 240 Save Configuration File opens the Configuration File page for saving the ini file to a PC refer to Backing Up and Restoring Configuration on page 240 Reset opens the Maintenance Actions page for resetting the device refer to Resetting the Device on page 228 Software Upgrade Wizard opens the Software Upgrade Wizard page for upgrading the device s software refer to Software Upgrade Wizard on page 236 Home Opens the Home page refer to Using the Home Page on page 46 4 4 E J Help Opens the Online Help topic of the currently opened configuration page in the Work pane refer to Getting Help on page 45 a Log off Logs off a session with the Web interface refer to Logging Off the Web Interface on page 49 If you modify parameters that take effect only after a device reset after you click the Submit button the toolbar displays the word Reset in red color as shown in the figure below This is a reminder to later save burn your settings to flash memory and reset the device Figure 3 3 Res
324. enable the redundant Proxies mechanism set the parameter EnableProxyKeepAlive to 1 or 2 Determines if the device s internal routing table takes precedence over a Proxy for routing calls 0 No Only a Proxy server is used to route calls default 1 Yes The device checks the routing rules in the Tel to IP Routing table or Outbound IP Routing table if EnableSBC is set to 1 for a match with the Tel to IP call Only if a match is not found is a Proxy used Determines whether to use the device s internal routing table to obtain the URI host name and optionally an IP profile per call even if a Proxy server is used 0 Disable Don t use internal routing table default 1 Enable Use the Tel to IP Routing table or Outbound IP Routing table if EnableSBC is set to 1 Notes This parameter appears only if the Use Default Proxy parameter is enabled The domain name is used instead of a Proxy name or IP address in the INVITE SIP URI Determines whether the device sends SIP messages and responses through a Proxy server 0 Disable Use standard SIP routing rules default 1 Enable All SIP messages and responses are sent to a Proxy server Note Applicable only if Proxy server is used i e the parameter IsProxyUsed is set to 1 Determines the type of redundant routing mechanism to implement when a call can t be completed using the main route 0 Disable N
325. encapsulated QSIG CALL PROCEEDING message without waiting fora CALL PROCEEDING message from the Tel side If tunneling is disabled and the incoming INVITE includes a QSIG body a 415 response is sent Mid call communication After the SIP connection is established all QSIG messages are encapsulated in SIP INFO messages Call tear down The SIP connection is terminated once the QSIG call is complete The RELEASE COMPLETE message is encapsulated in the SIP BYE message that terminates the session To enable QSIG tunneling set the parameter EnableQSIiGTunneling to 1 on both the originating and terminating devices and the parameter ISDNDuplicateQ931BuffMode to 128 duplicate all messages both parameters are described in ISDN and CAS Interworking Related Parameters on page 307 SIP User s Manual 406 Document LTRT 68808 SIP User s Manual 11 Supplied SIP Software Package 11 Supplied SIP Software Package The table below lists the standard SIP software package supplied with the SIP device File Name Ram cmp file Mediant_SIP_xxx cmp ini files Mediant_SIP_T1 ini Mediant_SIP_E1 ini Usa_tones_xx dat Usa_tones_xx ini voice_prompts dat Utilities DConvert ACSyslog BootP CAS Protocol Files MIB Files CAS Capture Tool ISDN Capture Tool Version 5 6 Table 11 1 Supplied Software Package Description Image file containing the software for the Mediant 2000 Sample ini file for Mediant 2000 E1 device
326. ended values from 10 to 25 When the same frequency is used for a continuous tone and a cadence tone the Signal On Time parameter of the continuous tone must have a value that is greater than the Signal On Time parameter of the cadence tone Otherwise the continuous tone is detected instead of the cadence tone The tones frequency should differ by at least 40 Hz from one tone to other defined tones For example to configure the dial tone to 440 Hz only enter the following text Dial tone CALL PROGRESS TONE 1 Tone Type 1 Tone Form 1 continuous Low Freq Hz 440 High Freq Hz 0 Low Freq Level dBm 10 10 dBm High Freq Level dBm 32 use 32 only if a single tone is required First Signal On Time 10msec 300 the dial tone is detected after 3 sec First Signal Off Time 10msec 0 Second Signal On Time 10msec 0 Second Signal Off Time 10msec 0 6 2 Prerecorded Tones PRT File The Call Progress Tones CPT mechanism has several limitations such as a limited number of predefined tones and a limited number of frequency integrations in one tone To overcome these limitations and provide tone generation capability that is more flexible the Prerecorded Tones PRT file can be used If a specific prerecorded tone exists in the PRT file it takes precedence over the same tone that exists in the CPT file and is played instead of it Note The Prerecorded tones are used only for generation
327. energy of the output signal to a required level It estimates the energy of the incoming signal calculates the essential gain and performs amplification Feedback ensures that the output signal is not clipped You can define the required Gain Slope in decibels sec and the required energy threshold When the AGC first detects a signal in the input it begins operating in Fast Mode This means that the Gain Slope is 8 dB sec for the first 1 5 seconds After this period the Gain Slope is changed to the user defined value You can disable or enable the feature by using the ini file parameter AGCDisableFastAdaptation After Fast Mode is used the signal should be off for two minutes in order to have the feature turned on again This feature is designed so that AGC can fast adapt when a conversation is started gt To configure the IP media parameters take these 4 steps 1 Open the IPMedia Settings page Configuration tab gt Media Settings menu gt IPmedia Settings page item Figure 3 43 IPMedia Settings Page v IPMedia Settings Enable Answer Detector Answer Detector Activity Delay Answer Detector Silence Time Answer Detector Redirection Answer Detector Sensitivity Answer Machine Detector Sensitivity Enable Energy Detector Disable Energy Detector Quality Factor 4 Energy Detector Threshold 3 Enable Pattern Detector Disable Enable 4GC Disable AGC Sl
328. equired after the burst tone ends and the tone detection is reported For continuous tones this parameter is ignored Second Signal On Time 10 msec Signal On period in 10 msec units for the second cadence on off cycle Can be omitted if there isn t a second cadence Second Signal Off Time 10 msec Signal Off period in 10 msec units for the second cadence on off cycle Can be omitted if there isn t a second cadence Third Signal On Time 10 msec Signal On period in 10 msec units for the third cadence ON OFF cycle Can be omitted if there isn t a third cadence 336 Document LTRT 68808 SIP User s Manual 6 Auxiliary Configuration Files e Third Signal Off Time 10 msec Signal Off period in 10 msec units for the third cadence ON OFF cycle Can be omitted if there isn t a third cadence e Fourth Signal On Time 10 msec Signal On period in 10 msec units for the fourth cadence ON OFF cycle Can be omitted if there isn t a fourth cadence e Fourth Signal Off Time 10 msec Signal Off period in 10 msec units for the fourth cadence ON OFF cycle Can be omitted if there isn t a fourth cadence e Carrier Freq Hz frequency of the carrier signal for AM tones e Modulation Freq Hz frequency of the modulated signal for AM tones valid range from 1 to 128 Hz e Signal Level dBm level of the tone for AM tones e AM Factor steps of 0 02 amplitude modulation factor valid range from 1 to 50 Recomm
329. equired tab e Configuration refer to Configuration Tab on page 50 e Management refer to Management Tab on page 220 e Status amp Diagnostics refer to Status amp Diagnostics Tab on page 241 Version 5 6 23 November 2008 A ge AudioCodes Mediant 2000 3 3 2 1 gt To navigate to a page take these 2 steps 1 Navigate to the required page item by performing the following e Drilling down using the plus signs to expand the menus and submenus e Drilling up using the minus amp signs to collapse the menus and submenus 2 Select the required page item the page opens in the Work pane Displaying Navigation Tree in Basic and Full View You can view an expanded or reduced Navigation tree display regarding the number of listed menus and submenus This is relevant when using the configuration tabs Configuration Management and Status amp Diagnostics on the Navigation bar The Navigation tree menu can be displayed in one of two views m Basic displays only commonly used menus m Full displays all the menus pertaining to a configuration tab The advantage of the Basic view is that it prevents cluttering the Navigation tree with menus that may not be required Therefore a Basic view allows you to easily locate required menus gt To toggle between Full and Basic view take this step m Select the Basic option located below the Navigation bar to display a reduced menu tree select the Full option to disp
330. er 2008 7a J AudioCodes Parameter RegistrarName GWRegistrationName AuthenticationMode OOSOnRegistrationFail RegistrationTime RegistrationTimeDivider RegistrationRetryTime RegisterOnInviteFailure RegistrationTimeThreshold ZeroSDPHandling ForkingHandlingMode Account SIP User s Manual Mediant 2000 Description For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to Proxy amp Registration Parameters on page 132 Determines the device s response to an incoming SDP with an IP address of 0 0 0 0 in the Connection line 0 Sets the IP address of the outgoing SDP Connection line to 0 0 0 0 default 1 Sets the IP address of the outgoing S
331. er Blue Sth Tx OTMF Option RFC 2833 Payload Type Hook flash Opten a a Advanced Parameters jai Tone Durahon set gt r in Lighter Blue Hotine Dial Tone Ouraben sec Enable Special Digits Oefeuk Destination Number Special Digt Representan For ease of identification the basic parameters are displayed with a darker blue color background than the advanced parameters Note When the Navigation tree is in Full mode refer to Navigation Tree on page 23 configuration pages display all their parameters i e the Advanced Parameter List view is displayed Version 5 6 27 November 2008 A gA AudioCodes Mediant 2000 3 3 3 2 2 Showing Hiding Parameter Groups Some pages provide groups of parameters which can be hidden or shown To toggle between hiding and showing a group simply click the group name button that appears above each group The button appears with a down pointing or up pointing arrow indicating that it can be collapsed or expanded when clicked respectively Figure 3 8 Expanding and Collapsing Parameter Groups management somnos SN Bove Parameter Uit a Syslog Setings Syvleg Server IP Address Syslog Server Port Enatte Syslog Anelog Ports F ter v SNMP Seng SAMP Treo Oesinabens SNMP Corntrutety String SNMP V2 Table SNMP Trusted Managers Cradle SNMP Trap Manager Most Name v Actretty Types te Report vie Activity Log Messages Parameters Value Charge a Auriliary Tiles
332. er Description index The row index of the remote file system The valid range is 0 to 4 Host Or IP The domain name or IP address of the NFS server If a domain name is provided a DNS server must be configured Root Path Path to the root of the remote file system in the format path For example audio NFS version used to access the remote file system NFS Version 2 NFS Version 2 8 NFS Version 3 default Authentication method used for accessing the remote file system Authentication Type 0 Auth NULL 1 Auth UNIX default UID User ID used in authentication when using Auth UNIX The valid range is 0 to 65537 The default is 0 GID Group ID used in authentication when using Auth UNIX The valid range is 0 to 65537 The default is 1 The VLAN type for accessing the remote file system 0 OAMP VLAN Type 1 Media default Note This parameter applies only if VLANs are enabled or if Multiple IPs is configured refer to VLANS and Multiple IPs on page 384 Version 5 6 61 November 2008 A EA AudioCodes Mediant 2000 3 4 1 5 Configuring the IP Routing Table The IP Routing Table page allows you to define up to 50 static IP routing rules for the device For example you can define static routing rules for the OAMP and Control networks since a default gateway is supported only for the Media traffic network refer to Configuring the Multiple Interface Table on page 53 Before sending an
333. er name is set to the channel s phone number Enables the use of DNS Naming Authority Pointer NAPTR and Service Record SRV queries to resolve Proxy and Registrar servers and to resolve all domain names that appear in the Contact and Record Route headers 0 A Record A Record default 1 SRV SRV 2 NAPTR NAPTR If set to A Record 0 no NAPTR or SRV queries are performed If set to SRV 1 and the Proxy Registrar IP address parameter Contact Record Route headers or IP address defined in the Routing tables contains a domain name an SRV query is performed The device uses the first host name received from the SRV query The device then performs a DNS A record query for the host name to locate an IP address If set to NAPTR 2 an NAPTR query is performed If it is successful an SRV query is sent according to the information received in the NAPTR response If the NAPTR query fails an SRV query is performed according to the configured transport type If the Proxy Registrar IP address parameter the domain name in the Contact Record Route headers or the IP address defined in the Routing tables contains a domain name with port definition the device performs a regular DNS A record query If a specific Transport Type is defined a NAPTR query is not performed Note To enable NAPTR SRV queries for Proxy servers only use the parameter ProxyDNSQueryType Enables the use of DNS Naming Authority Pointer NAP
334. er s Manual Mediant 2000 Specification Operating Temp 0 to 40 C 32 to 104 F Storage 40 to 70 C 40 to 158 F Humidity 10 to 90 non condensing CPCI blades are full hot swappable Power supplies are redundant but not hot swappable 445 x 44 x 300 mm 17 5 x 1 75 x 12 inches Approx 4 8 kg fully populated 16 spans 4 2 kg for 1 span 1U 19 inch 2 slot CPCI chassis rack shelf or desktop mount options Rack mount using two side brackets 2 additional rear side brackets optional Motorola PowerQUICC 8260 SDRAM 64 128 MB on 60 channel models AudioCodes AC486 VoIP DSP based on TI DSP TMS5541 each core at 133 MHz 33 MHz 32 bit slave mode PICMG 2 0 revision 2 1 6U single cPCI slot PICMG 2 0 R2 1 and R2 16 and R 3 0 CompactPCI blade 480 channels 40 7 W 3A at 5 V 7 8 A at 3 3 V 240 channels 24 W 1 5 A at 5 V 5 A at 3 3 V 120 channels 18 4 W 0 9 A at 5 V 4 2 A at 3 3 V Humidity 10 to 90 non condensing 500 Linear Feet per Minute LFM at 50 C ambient temp supporting 480 ports 400 LFM at 50 C ambient temp supporting 400 ports 300 LFM at 50 C ambient temp supporting 240 ports E1 T1 status LAN status Status of device Fail ACT Power and Swap Ready Supported by Syslog Server per RFC 3164 IETF standard SNMP v2c SNMP v3 Configuration of device using Web browser or ini files SNMP v2c SNMP v3 Syslog RFC 3164 Web Management via HT
335. er s Manual 8 Networking Capabilities a Inthe IP Settings page modify the IP parameters to correspond to the values shown in the figure below Note that the OAM Control and Media Network Settings parameters appear only after you select the options Multiple IP Networks or Dual IP in the field IP Networking Mode Figure 8 4 OAM Control Media IP Configuration in the IP Settings Page IP Settings S IP Networking Mode Multiple IP Networks OAM Network Settings IP Address 10 31 174 50 HS Subnet Mask 255 255 0 0 Default Gateway Address Control Network Settings l IP Address 10 32 174 50 Subnet Mask 255 255 0 0 H Default Gateway Address Media Network Settings IP Address 10 33 174 50 l Subnet Mask 255 255 0 0 H Default Gateway Address 10 33 0 1 Instead of configuring in the IP Settings page you can use the Multiple Interface Table page which is accessed from the IP Settings page by clicking the right arrow um button alongside the label Multiple Interface Table refer to Configuring the Multiple Interface Table on page 53 The Multiple Interface Table page provides greater configuration flexibility whereby you can also assign VLANs to the different interfaces Figure 8 5 Multiple Interface Table Page Index ApplicationTypes 1Pv InterfaceMode IPAddress PrefixLength Gateway VianI
336. er source IP address and UDP port 8 6 Multiple Routers Support Multiple routers support is designed to assist the device when it operates in a multiple routers network The device learns the network topology by responding to Internet Control Message Protocol ICMP redirections and caches them as routing rules with expiration time When a set of routers operating within the same subnet serve as devices to that network and intercommunicate using a dynamic routing protocol the routers can determine the shortest path to a certain destination and signal the remote host the existence of the better route Using multiple router support the device can utilize these router messages to change its next hop and establish the best path Note Multiple Routers support is an integral feature that doesn t require configuration 8 7 Simple Network Time Protocol Support The Simple Network Time Protocol SNTP client functionality generates requests and reacts to the resulting responses using the NTP version 3 protocol definitions according to RFC 1305 Through these requests and responses the NTP client synchronizes the system time to a time source within the network thereby eliminating any potential issues should the local system clock drift during operation By synchronizing time to a network time source traffic handling maintenance and debugging become simplified for the network administrator Version 5 6 383 November 2008 A ge
337. er to the Product Reference Manual 3 Load the voiceprompts dat file to the device using TFTP refer to the Product Reference Manual or Web interface refer to Loading Auxiliary Files on page 231 SIP User s Manual 338 Document LTRT 68808 SIP User s Manual 6 Auxiliary Configuration Files 6 4 CAS Protocol Auxiliary Files The CAS Protocol auxiliary files contain the CAS Protocol definitions that are used for CAS terminated trunks You can either use the supplied files or construct your own files Up to eight files can be loaded and different files can be assigned to different trunks The CAS files can be loaded to the device using the Web interface or ini file refer to Loading Auxiliary Files on page 231 Note All CAS files loaded together must belong to the same Trunk Type i e either E1 or T1 6 5 Dial Plan File The source file for the Dial Plan configuration contains a list of known prefixes e g area codes and international telephone number patterns for the PSTN to which the device is connected The device uses this information to detect end of dialing in certain CAS configurations where the end indicator ST is not used The device supports up to 8 000 distinct prefixes in the dial plan file The CasTrunkDialPlanName ini file parameter determines which Dial Plan in a Dial Plan file to use for a specific trunk refer to Configuring the Trunk Settings on page 82 The Dial Plan can be loaded using the Web
338. ernet redundancy scheme refer to Ethernet Interface Redundancy on page 380 For detailed information on the Ethernet interface configuration refer to Ethernet Interface Configuration on page 379 gt To view Ethernet port information take the following step E Open the Ethernet Port Information page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Ethernet Port Information page item Figure 3 110 Ethernet Port Information Page v Ethernet Information Active Port Port 1 Duplex Mode Port 1 Speed Port 2 Duplex Mode Port 2 Speed Table 3 62 Ethernet Port Information Parameters Parameter Description Active Port Displays the active Ethernet port 1 or 2 Port Duplex Mode Displays the Duplex mode of the Ethernet port Half Duplex or Full Duplex Port Speed Displays the speed in Mbps of the Ethernet port 10 Mbps 100 Mbps Version 5 6 243 November 2008 A Ee AudioCodes Mediant 2000 3 6 1 3 Viewing Active IP Interfaces The Active IP Interfaces page displays the device s IP interfaces which you configured in the Multiple Interface Table page refer to Configuring the Multiple Interface Table on page 53 and that are currently active gt To view the Active IP Interfaces page take this step m Open the Active IP Interfaces page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Active IP Interfaces page item Figure
339. ers page item 171 November 2008 ca AudioCodes Mediant 2000 Figure 3 70 Routing General Parameters Page v General Parameters Add Trunk Group ID as Prefix Add Trunk ID as Prefix Phone Number Enable Alt Routing Tel to IP Alt Routing Tel to IP Mode Replace Empty Destination with B channel Add NPI and TON to Called Number Add NPI and TON to Calling Number No IP to Tel Remove Routing Table Prefix No Alt Routing Tel to IP Connectivity Method Max Allowed Packet Loss for Alt Routing Max Allowed Delay for Alt Routing msec lee a S j Disable Both AIAI 2 Configure the general parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 43 Routing General Parameters Description Parameter Add Trunk Group ID as Prefix AddTrunkGroupAsPrefix Add Trunk ID as Prefix AddPortAsPrefix Replace Empty Destination with B channel Phone Number ReplaceEmptyDstWithPortNumber Add NPI and TON to Calling Number SIP User s Manual Description Determines whether the device s Trunk Group ID is added as a prefix to the destination phone number for Tel to IP Calls 0 No Don t add Trunk Group ID as prefix default 1 Yes Add Trunk Group ID as prefix to called number Notes This option can be
340. erver Certificate Replacement on page 105 Replacing the client certificates refer to Client Certificates on page 108 m Regenerating Self Signed Certificates refer to Self Signed Certificates on page 109 Updating the private key using HTTPSPkeyFileName as described in the Product Reference Manual 3 4 6 4 1 Server Certificate Replacement The device is supplied with a working Secure Socket Layer SSL configuration consisting of a unique self signed server certificate If an organizational Public Key Infrastructure PKI is used you may wish to replace this certificate with one provided by your security administrator Version 5 6 105 November 2008 A EA AudioCodes Mediant 2000 gt To replace the device s self signed certificate take these 8 steps 1 Your network administrator should allocate a unique DNS name for the device e g dns_name corp customer com This DNS name is used to access the device and should therefore be listed in the server certificate 2 Open the Certificates Signing Request page Configuration tab gt Security Settings menu gt Certificates page item Figure 3 54 Certificates Signing Request Page Certificate Signing Request Subject Name Copy the certificate signing request and send it to your Certification Authority for signing Press the button Generate self signed to create a self signed certificate using the subject name provided above Important this is a le
341. ese 4 steps 1 Open the SAS Configuration page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Stand Alone Survivability page item Figure 3 66 SAS Configuration Page v Enable SAS Disie SAS Local SIP UDP Port 5080 SAS Default Gateway IP SAS Registration Time 20 E Short Number Length o o 7 SAS Local SIP TCP Port 5080 SAS Local SIP TLS Port 5081 SAS Proxy Set 0 Redundant SAS Proxy Set 1 2 Configure the parameters according to the table below 3 Click the Submit button to apply your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 37 Stand Alone Survivability Parameters Description Parameter Description Enable SAS Enables the Stand Alone Survivability SAS feature EnableSAS 0 Disable Disabled default 1 Enable SAS is enabled When enabled the device receives the registration requests from different SIP entities in the local network and then forwards them to the defined proxy If the connection to the proxy fails Emergency Mode the device serves as a proxy by allowing calls internal to the local network or outgoing to PSTN SAS Local SIP UDP Port Local UDP port for sending and receiving SIP messages for SAS The SASLocalSIPUDPPort SIP entities in the local network need to send the registration requests to this port When forwarding the
342. ess The time period in 100 msec units that an RTP packet is not received after which a call is disconnected The valid range is 1 to 1 000 The default value is 100 i e 10 seconds Notes Applicable only if DisconnectOnBrokenConnection 1 Currently this feature works only if Silence Suppression is disabled Determines whether calls are disconnected after detection of silence 1 Yes The device disconnects calls in which silence occurs in both call directions for more than a user defined time 0 No Call is not disconnected when silence is detected default The silence duration can be set by the FarEndDisconnectSilencePeriod parameter default 120 Note To activate this feature set EnableSilenceCompression and FarEndDisconnectSilenceMethod to 1 Duration of silence period in seconds prior to call disconnection The range is 10 to 28 800 i e 8 hours The default is 120 seconds Note This parameter is applicable only to devices that use DSP templates 2 and 3 Silence detection method 0 None Silence detection option is disabled 1 Packets Count According to packet count 2 Voice Energy Detectors N A 8 All N A Enables or disables re routing of Tel to IP calls that are identified as fax calls 0 Disable Disabled default 1 Enable Enabled If a CNG tone is detected on the Tel side of a Tel to IP call a FAX prefix is appended to the destination
343. ess of the first incoming packet Notes The NAT mechanism must be enabled for this parameter to take effect DisableNAT set to 0 For information on RTP Multiplexing refer to RTP Multiplexing ThroughPacket on page 360 262 Document LTRT 68808 SIP User s Manual Parameter EnableUDPPortTranslatio n NoOpEnable NoOpInterval RTPNoOpPayloadType EnableDetectRemoteMAC Change StaticNatIP SyslogServerlP SyslogServerPort EnableSyslog Version 5 6 4 ini File Configuration Description 0 Disable UDP port translation default 1 Enable UDP port translation When enabled the device compares the source UDP port of the first incoming packet to the remote UDP port stated in the opening of the channel If the two UDP ports don t match the NAT mechanism is activated Consequently the remote UDP port of the outgoing stream is replaced by the source UDP port of the first incoming packet Note The NAT mechanism and the IP address translation must be enabled for this parameter to take effect DisableNAT 0 EnablelpAddrTranslation 1 Enables or disables the transmission of RTP or T 38 No Op packets 0 Disable default 1 Enable This mechanism ensures that the NAT binding remains open during RTP or T 38 silence periods Defines the time interval in which RTP or T 38 No Op packets are sent in the case of silence no RTP T 38 traffic when No Op packet transmission i
344. et Displayed on Toolbar of Submit Q Bun Reset Device Actions vw A Home Help P Log off Reset Notification SIP User s Manual 22 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 3 2 Navigation Tree The Navigation tree located in the Navigation pane displays the menus pertaining to the menu tab selected on the Navigation bar used for accessing the configuration pages The Navigation tree displays a tree like structure of menus You can easily drill down to the required page item level to open its corresponding page in the Work pane The terminology used throughout this manual for referring to the hierarchical structure of the tree is as follows menu first level highest level m submenu second level contained within a menu mpage item last level lowest level in a menu contained within a menu or submenu Figure 3 4 Terminology for Navigation Tree Levels Management Foeni Scenarios Search O Basic Full network Settings Amedia Settings security Settings Protocol Configuration Protocol Definition SIP General Parameters Proxy amp Registration Proxy Sets Table _iCoders DTMF amp Dialing sIP Advanced Parameters t manipulation Tables routing Tables Profile Definitions Endpoint Settings t Endpoint Number uuntie Group advanced Applications gt To view menus in the Navigation tree take this step m On the Navigation bar select the r
345. et to Out Of Service if the parameter OOSOrRegistrationFail is set to 1 refer to Proxy amp Registration Parameters on page 132 Contact User Defines the AOR user name It appears in REGISTER From To headers as ContactUser HostName and in INVITE 200 OK Contact headers as ContactUser lt device s IP address gt If not configured the Contact User parameter from the IP Group Table page is used instead Note If registration fails then the userpart in the INVITE Contact header contains the source party number 3 4 7 7 Configuring the Digital Gateway Parameters The Digital Gateway Parameters page allows you to configure miscellaneous digital parameters gt To configure the digital gateway parameters take these 4 steps 1 Open the Digital Gateway Parameters page Configuration tab gt Protocol Configuration menu gt Digital Gateway submenu gt Digital Gateway Parameters page item Figure 3 86 Digital Gateway Parameters Page B channel Negotiation Exclusive Swap Redirect and Called Numbers No MFC R2 Category 1 Disconnect Call on Busy Tone Detection CAS Enable Disconnect Call on Busy Tone Detection ISDN Disable 4 Enable TDM Tunneling Disable Send Screening Indicator to IP Not Configured Send Screening Indicator to ISDN Not Configured Add IE in SETUP Trunk Groups to Send IE Enable User to User IE for Tel to IP Disable Enable User to User IE for IP
346. etailed information on Ethernet interface configuration refer to Ethernet Interface Configuration on page 379 For a description of this parameter refer to Configuring the IP Settings on page 50 Determines the DHCP renewal speed 0 Disable 1 Normal default 2 to 10 Fast When set to 0 the DHCP lease renewal is disabled Otherwise the renewal time is divided by this factor Some DHCP enabled routers perform better when set to 4 Enables or disables DHCP renewal support 0 Disable default 1 Enable This parameter is applicable only if DHCPEnable is set to 0 for cases where booting up the device via DHCP is not desirable but renewing DHCP leasing is When the device is powered up it attempts to communicate with a BootP server If there is no response and if DHCP is disabled the device boots from flash It then attempts to communicate with the DHCP server to renew the lease For a description of this parameter refer to General Parameters on page 151 For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 260 Document LTRT 68808 SIP User s Manual 4 ini File Configuration Parameter DNS2IP SRV2IP Version 5 6 Description This ini file table parameter configures the internal DNS table for resolving host names into IP addresses Up to four dif
347. ether a Cisco gateway exists at the remote side 0 No Cisco gateway exists at the remote side default 1 A Cisco gateway exists at the remote side When there is a Cisco gateway at the remote side the device must set the value of the annexb parameter of the fmtp attribute in the SDP to no This logic is used if EnableSilenceCompression 2 enable without adaptation In this case Silence Suppression is used on the channel but not declared in the SDP Note The IsCiscoSCEMode parameter is only relevant when the selected coder is G 729 For a description of this parameter refer to Configuring the Voice Settings on page 66 For a description of this parameter refer to Configuring the General Media Settings on page 78 Defines the echo cancellation Non Linear Processing NLP mode 0 NLP adapts according to echo changes default 1 Disables NLP Enables or disables the Aggressive Non Linear Processor NLP in the first 0 5 second of the call 0 Disabled default 1 Enabled Enables disables the DSP Noise Reduction mechanism 0 Disable default 1 Enable Note When this parameter is enabled the channel capacity might be reduced 328 Document LTRT 68808 SIP User s Manual Parameter EnableStandardSIDPayloadT ype ComfortNoiseNegotiation RTPSIDCoeffNum DTMFVolume DTMFGenerationTwist DTMFinterDigitinterval DTMFDigitLength RxDTMFHangOverTi
348. ettings on page 50 Defines the VLAN ID to which untagged incoming traffic is assigned Outgoing packets sent to this VLAN are sent only with a priority tag VLAN ID 0 When this parameter is equal to one of the VLAN IDs in the Interface Table and VLANs are enabled untagged incoming traffic is considered as an incoming traffic for that interface Outgoing traffic sent from this interface is sent with the priority tag tagged with VLAN ID 0 When this parameter is different from any value in the VLAN ID column in the Interface Table untagged incoming traffic is discarded and all outgoing traffic is tagged Note If this parameter is not set i e default value is 1 but one of the interfaces has a VLAN ID configured to 1 this interface is still considered the Native VLAN If you do not wish to have a Native VLAN ID and want to use VLAN ID 1 set this parameter to a value other than any VLAN ID in the table 56 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 1 3 Configuring the Application Settings The Application Settings page is used for configuring various application parameters such as Telnet gt To configure the Application settings parameters take these 4 steps 1 Open the Application Settings page Configuration tab gt Network Settings menu gt Application Settings page item Figure 3 36 Application Settings Page wv NTP Settings NTP Server IP Address
349. etwork Specific 3 Subscriber 4 or Abbreviated 6 167 November 2008 ca AudioCodes Parameter Presentation _lsPresentationResitricted Notes Mediant 2000 Description This parameter is applicable only to Number Manipulation tables for IP to Tel calls a The default is Unknown Determines whether Caller ID is permitted Not Configured privacy is determined according to the Caller ID table refer to Caller ID Allowed sends Caller ID information when a call is made using these destination source prefixes Restricted restricts Caller ID information for these prefixes Notes Only applicable to Number Manipulation tables for source number manipulation If Presentation is set to Restricted and Asserted Identity Mode is set to P Asserted the From header in the INVITE message includes the following From anonymous lt sip anonymous anonymous invalid gt and privacy id header 3 4 7 3 1 Dialing Plan Notation The dialing plan notation applies to the Number Manipulation tables Tel to IP Routing table refer to Tel to IP Routing Table on page 175 and IP to Trunk Group Routing table refer to IP to Trunk Group Routing on page 181 The dialing notation applies to digits entered for the destination and source prefixes to represent multiple numbers Notation n m n m Pound sign at the end of a number A single asterisk Table
350. evice CDRs are generated at the end and optionally at the beginning of each call determined by the parameter CDRReportLevel and then sent to a Syslog server The destination IP address for CDR logs is determined by the parameter CDRSyslogServerIP For CDR in RADIUS format refer to Supported RADIUS Attributes on page 362 The following table lists the CDR fields that are supported 365 November 2008 ca AudioCodes Field Name ReportType Cid Callld Trunk BChan Conld TG EPTyp Orig Sourcelp Desilp TON NPI SrcPhoneNum SrcNumBeforeMap TON NPI DstPhoneNum DstNumBeforeMap Durat Coder Intrv Rtplp Port TrmSd TrmReason Fax InPackets OutPackets PackLoss RemotePackLoss Uniqueld SetupTime ConnectTime ReleaseTime SIP User s Manual Table 7 4 Supported CDR Fields Description Mediant 2000 Report for either Call Started Call Connected or Call Released Port Number SIP Call Identifier Physical Trunk Number Selected B Channel SIP Conference ID Trunk Group Number Endpoint Type Call Originator IP Tel Source IP Address Destination IP Address Source Phone Number Type Source Phone Number Plan Source Phone Number Source Number Before Manipulation Destination Phone Number Type Destination Phone Number Plan Destination Phone Number Destination Number Before Manipulation Call Duration Selected Coder Packet Interval RTP IP Address Remote RTP Port Initiator of Call R
351. fault 1 NSE enabled Notes This feature can be used only if VxxModemTransportType 2 Bypass f NSE mode is enabled the SDP contains the following line a rtpmap 100 X NSE 8000 To use this feature The Cisco gateway must include the following definition modem passthrough nse payload type 100 codec g711alaw Set the Modem transport type to Bypass mode VxxModemTransportT ype 2 for all modems Configure the gateway parameter NSEPayloadType 100 In NSE bypass mode the device starts using G 711 A Law default or G 711u Law according to the parameter FaxModemBypassCoderType The payload type used with these G 711 coders is a standard one 8 for G 711 A Law and 0 for G 711 u Law The parameters defining payload type for the old AudioCodes Bypass mode FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass The bypass packet interval is selected according to the parameter FaxModemBypassBasicRtpPacketinterval NSE payload type for Cisco Bypass compatible mode The valid range is 96 127 The default value is 105 Note Cisco gateways usually use NSE payload type of 100 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67
352. fault value is assigned to that parameter according to the cmp file loaded to the device and saved to the non volatile memory thereby overriding the value previously defined for that parameter m Restoring all default settings including the device s IP address and Web interface s login user name and password Use the device s hardware Reset button refer to the device s nstallation Manual Version 5 6 333 November 2008 A ge AudioCodes Mediant 2000 Reader s Notes SIP User s Manual 334 Document LTRT 68808 SIP User s Manual 6 Auxiliary Configuration Files 6 6 1 Auxiliary Configuration Files This section describes the auxiliary files with the dat file extension which are loaded in addition to the ini file to the device You can load the auxiliary files to the device using one of the following methods m Web interface refer to Loading Auxiliary Files on page 231 E inifile specify the name of the relevant auxiliary file in the device s ini file and then load the ini file to the device refer to Loading Auxiliary Files on page 231 Configuring the Call Progress Tones File The Call Progress Tones CPT auxiliary file used by the device is a binary file with file extension dat This file contains the definitions of the Call Progress Tones levels and frequencies that are detected generated by the device You can either use one of the supplied device auxiliary dat files or create your own file
353. ference TDMBusLocalReference TDM Bus PSTN Auto Clock TDMBusPSTNAutoCloc kEnable TDM Bus PSTN Auto Clock Reverting TDMBusPSTNAutoCloc kRevertingEnable TDM Bus Clock Source TDMBusClockSource Version 5 6 3 Web Based Management Description Defines the PCM Pattern that is applied to the E1 T1 timeslot B channel when the channel is idle The range is 0 to 255 The default is set internally according to the Law select 1 OxFF for Mu Law 0x55 for A law Defines the ABCD CAS Pattern that is applied to the CAS signaling bus when the channel is idle The valid range is 0x0 to OxF The default is 1 i e default pattern 0000 Note This parameter is applicable only when using PSTN interface with CAS protocols Physical Trunk ID from which the device recovers receives its clock synchronization The range is 0 to the maximum number of Trunks The default is Trunk ID 1 Note This parameter is applicable only if the parameter TDMBusClockSource is set to 4 and the parameter TDMBusPSTNAutoClockEnable is set to 0 Enables or disables the PSTN trunk Auto Fallback Clock feature 0 Disable default Recovers the clock from the E1 T1 line defined by the parameter TDMBusLocalReference 1 Enable Recovers the clock from any connected synchronized slave E1 T1 line If this trunk loses its synchronization the device attempts to recover the clock from the next trunk Note that initially the devi
354. ferent IP addresses in dotted decimal notation can be assigned to a host name The format of this parameter is as follows Dns2Ip FORMAT Dns2lIp_Index Dns2lp_DomainName Dns2Ip_FirstlpAddress Dns2lp_ SecondlpAddress Dns2Ip_ThirdlpAddress Dns2Ip_FourthlpAddress Dns2Ip For example Dns2Ip Dns2lp 0 DnsName 1 1 1 1 2 2 2 2 3 3 3 3 4 4 4 4 Dns2Ip Notes This parameter can include up to 20 indices f the internal DNS table is used the device first attempts to resolve a domain name using this table If the domain name isn t found the device performs a DNS resolution using an external DNS server To configure the internal DNS table using the Web interface and for a description of the parameters in this ini file table parameter refer to Internal DNS Table on page 186 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 This ini file table parameter defines the internal SRV table for resolving host names to DNS A Records Three different A Records can be assigned to a host name Each A Record contains the host name priority weight and port The format of this parameter is as follows SRV2IP FORMAT SRV2IP_Index SRV2IP_InternalDomain SRV2IP_TransportType SRV2IP_Dns1 SRV2IP_Priority1 SRV2IP_Weight1 SRV2IP_Port1 SRV2IP_Dns2 SRV2IP_Priority2 SRV2IP_Weight2 SRV2IP_Port2 SRV2IP_Dns3 SRV2IP_Priority3 SRV2IP_Weight3 SRV
355. figuring the Number Manipulation Tables on page 164 0 Route calls before manipulation IP to Tel calls are routed before the number manipulation rules are applied default 1 Route calls after manipulation IP to Tel calls are routed after the number manipulation rules are applied The request URI host name prefix of the incoming SIP INVITE message If this routing rule is not required leave the field empty Note For notations representing multiple numbers refer to Dialing Plan Notation on page 168 However the asterisk wildcard cannot be used to depict any source host prefix The From URI host name prefix of the incoming SIP INVITE message If this routing rule is not required leave the field empty Notes For notations representing multiple numbers refer to Dialing Plan Notation on page 168 However the asterisk wildcard cannot be used to depict any source host prefix If the P asserted ID header is present in the incoming INVITE message then the parameter Source Host Prefix is compared to the P Asserted ID URI hostname and not to the From header Represents a called telephone number prefix The prefix can be 1 to 49 digits long Note For notations representing multiple numbers refer to Dialing Plan Notation on page 168 Represents a calling telephone number prefix The prefix can be 1 to 49 digits long Note For notations representing multiple numbers refer to Dialing Plan
356. format of this parameter is as follows IPSec_IKEDB_ Table Format IKE_DB_INDEX IKEPolicySharedKey IKEPolicyProposalEncryption_X IKEPolicyProposalAuthentication_X IKEPolicyProposalDHGroup_X IKEPolicyLifelnSec IKEPolicyLifelnKB IkePolicyAuthenticationMethod IPSEC_IKEDB_TABLE For example 278 Document LTRT 68808 SIP User s Manual Parameter 4 ini File Configuration Description IPSec_IKEDB_ Table Format IKE_DB_INDEX IKEPolicySharedKey IKEPolicyProposalEncryption_0 IKEPolicypRoposalAuthentication_0 IKEPolicyProposalDHGroup_0 IKEPolicyProposalEncryption_1 IKEPolicyProposalAuthentication_1 IKEPolicyProposalDHGroup_1 IKEPolicyLifelnSec IkePolicyAuthenticationMethod IPSEC_IKEDB_TABLE 0 123456789 1 2 0 2 2 1 28800 0 IPSEC_IKEDB_TABLE In the example above a single IKE peer is configured and a pre shared key authentication is selected Its pre shared key is 123456789 Two security proposals are configured DES SHA1 786DH and 3DES SHA1 1024DH Notes Each row in the table refers to a different IKE peer To support more than one Encryption Authentication DH Group proposal for each proposal specify the relevant parameters in the Format line The proposal list must be contiguous To configure the IKE table using the Web interface refer to Configuring the IKE Table on page 117 Foran explanation on using ini file table parameters refer to Structure of ini File Ta
357. fresh Time Enable Fallback to Routing Table Prefer Routing Table Always Use Proxy Redundant Routing Mode SIP ReRouting Mode Enable Registration Gateway Name Gateway Registration Name DNS Query Type Proxy DNS Query Type Number of RTX Before Hot Swap Use Gateway Name for OPTIONS User Name Password Cnonce Authentication Mode Challenge Caching Mode Mutual Authentication Mode No 10 33 2 56 Parking 60 Disable No Disable Routing T able Standard Mode Disable 4 Record 4 Record 3 No Default_Passwd Default_Cnonce Per Gateway None Optional Configure the Proxy and Registration parameters according to the following table Click the Submit button to save your changes or click the Register or Un Register buttons to save your changes and register unregister to a Proxy Registrar Version 5 6 133 To save the changes to flash memory refer to Saving Configuration on page 230 November 2008 ca AudioCodes Mediant 2000 Table 3 31 Proxy amp Registration Parameters Parameter Proxy Parameters Use Default Proxy IsProxyUsed Proxy Set Table button Proxy Name ProxyName Redundancy Mode ProxyRedundancyMode Proxy IP List Refresh Time ProxylPListRefreshTime Enable Fallback to Routing Table IsFallbackUsed
358. g 6 steps 1 Access the Management Settings page as described in Configuring the Management Settings on page 220 SIP User s Manual 226 Document LTRT 68808 SIP User s Manual 3 Web Based Management 2 In the SNMP Trusted Managers field click the right pointing arrow we button the SNMP Trusted Managers page appears Figure 3 94 SNMP Trusted Managers Trusted Managers IP Address SNMP Trusted Manager 1 SNMP Trusted Manager 2 SNMP Trusted Manager 3 SNMP Trusted Manager 4 SNMP Trusted Manager 5 3 Select the check box corresponding to the SNMP Trusted Manager that you want to enable and for whom you want to define an IP address 4 Define an IP address in dotted decimal notation 5 Click the Submit button to apply your changes 6 To save the changes refer to Saving Configuration on page 230 3 5 1 2 Configuring the Regional Settings The Regional Settings page allows you to define and view the device s internal date and time gt To configure the device s date and time take these 3 steps 1 Open the Regional Settings page Management tab gt Management Configuration menu gt Regional Settings page item Figure 3 95 Regional Settings Page Minutes Seconds 16 2 Enter the current date and time in the geographical location in which the device is installed 3 Click the Submit button the date and time are automatically
359. g IP Group the username and password for digest authentication defined in this table is used For Tel to IP calls the Serving IP Group is the destination IP Group defined in the Trunk Group Settings table or Tel to IP Routing table refer to Tel to IP Routing Table on page 175 For IP to Tel calls the Serving IP Group is the Source IP Group ID defined in the IP to Trunk Group Routing table refer to IP to Trunk Group Routing on page 181 Note If no match is found in this table for incoming or outgoing calls the username and passwordthe global parameters UserName and Password defined on the Proxy amp Registration page refer to Proxy amp Registration Parameters on page 132 are used Digest MD5 Authentication user name up to 50 characters Digest MD5 Authentication password up to 50 characters Defines the Address of Record AOR host name It appears in REGISTER From To headers as ContactUser HostName For successful registrations this HostName is also included in the INVITE request s From header URI If not configured or if registration fails the SIP Group Name parameter from the IP Group table is used instead This parameter can be up to 49 characters Enables registration No Don t register Yes Register When enabled the device sends REGISTER requests to the Serving IP Group In addition to activate registration you also need to set the parameter Registration Mode to
360. g IP addressing schemes m Single IP address for all traffic i e for Media Control and OAMP Separate IP address for each of the three traffic types The different traffic types are separated into three dedicated networks Instead of a single IP address the device is assigned three IP addresses and subnet masks each relating to a different traffic type This architecture enables you to integrate the device into a three network environment that is focused on security and segregation Each entity in the device e g Web and RTP is mapped to a single traffic type according to the table in IEEE 802 1p Q VLANs and Priority on page 385 in which it operates SIP User s Manual 384 Document LTRT 68808 SIP User s Manual 8 Networking Capabilities 8 9 2 m Dual IP mode The device is assigned two IP addresses for the different traffic types One IP address is assigned to a combination of two traffic types Media and Control OAMP and Control or OAMP and Media while the other IP address is assigned to whichever traffic type not included in this combination For example a typical scenario using this mode includes one IP address assigned to Control and OAMP and another IP address assigned to Media For detailed information on integrating the device into a VLAN and multiple IPs network refer to Getting Started with VLANS and Multiple IPs on page 387 For detailed information on configuring the multiple IP parameters refer to Ne
361. g Method FramingMethod Valid Range and Description 0 0 dB default 1 7 5 dB 2 15dB 8 22 5 dB Note This parameter is not applicable for PRI E1 trunks Defines the trace level 0 No Trace default 1 Full ISDN Trace 2 Layer 3 ISDN Trace 3 Only ISDN Q 931 Messages Trace 4 Layer 3 ISDN No Duplication Trace Determines the physical framing method for the trunk 0 default according to protocol type E1 or T1 E1 default is E1 CRC4 MultiFrame Format extended G 706B as c T1 default is T1 Extended SuperFrame with CRC6 as D 1 T1 SuperFrame Format as B a E1 DoubleFrame Format b E1 CRC4 MultiFrame Format ce E1 CRC4 MultiFrame Format extended G 706B A T1 4 Frame multiframe B T1 12 Frame multiframe D4 C T1 Extended SuperFrame without CRC6 D T1 Extended SuperFrame with CRC6 E T1 72 Frame multiframe SLC96 F J1 Extended SuperFrame with CRC6 Japan ISDN Configuration Parameters ISDN Termination Side TerminationSide NFAS Group Number NFASGroupNumber_x Version 5 6 Selects the ISDN termination side Applicable only to ISDN protocols 0 User side ISDN User Termination Equipment TE side default 1 Network side ISDN Network Termination NT side Note Select User side when the PSTN or PBX side is configured as Network side and vice versa If you don t know the device s IS
362. g Started with VLANS and Multiple IPs on page 387 gt To integrate the device into a multiple IPs network withVLANs using the ini file take these 3 steps 1 Prepare an ini file using the ini file table parameter InterfaceTable with relevant parameters e Ifthe BootP TFTP utility and the OAMP interface are located on the same network the Native VLAN ID VilanNativeVlanid must be equal to the OAMP VLAN ID VlanOamVlanld which in turn must be equal to the PVID of the switch port to which the device is connected Therefore set the PVID of the switch port to 4 in this example e Configure the OAMP parameters only if the OAMP networking parameters are different from the networking parameters used in the Single IP Network mode e The IP Routing table is required to define static routing rules for the OAMP and Control networks since a default Gateway isn t supported for these networks SIP User s Manual 390 Document LTRT 68808 SIP User s Manual 8 Networking Capabilities Below is an example of an ini file containing VLAN and Multiple IPs parameters Interface Table Configuration InterfaceTable FORMAT InterfaceTable Index InterfaceTable ApplicationTypes InterfaceTable IPv InterfaceMode InterfaceTable IPAddress InterfaceTable PrefixLength InterfaceTable Gateway InterfaceTable VlanID InterfaceTable InterfaceName InterfaceTable 0 QO 10 31 174 50 16 0 0 0 0 4 OAM gt InterfaceTable 0 i I
363. ge 18 to 3 corresponds to 18 dBm to 3 dBm in 1 dB steps The default is 6 dBm fax gain control 325 November 2008 ca AudioCodes Parameter FaxBypassOutputGain ModemBypassOutputGain T38MaxDatagram T38FaxMaxBufferSize DetFaxOnAnswerTone NTEMaxDuration EchoCancellerAggressiveNL P FaxModemBypassBasicRTP Packetinterval FaxModemBypassDJBufMin Delay EnableFaxModemIinbandNet workDetection SIP User s Manual Mediant 2000 Description For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67 Defines the maximum size in bytes of a T 38 buffer supported by the device This value is included in the outgoing SDP when T 38 is used for fax relay over IP The valid range is 100 to 1 024 The default value is 1 024 For a description of this parameter refer to SIP General Parameters on page 121 Maximum time for sending Named Telephony Events NTEs to the IP side regardless of the time range when the TDM signal is detected The range is 1 to 200 000 000 msec i e 55 hours The default is 1 i e NTE stops only upon detection of an End event Enables or disables the Aggressive NLP at the first 0 5 second of the call When enabled the echo is rem
364. ge Management tab gt Software Update menu gt Load Auxiliary Files page item Figure 3 99 Load Auxiliary Files Page INI file Co Classe CAS file Ce lease oice Prompts file Call Progress Tones file Browse Load Fie Prerecorded Tones file T Cleese Dial Plan file o eae User Info file Ce re 2 Click the Browse button corresponding to the file type that you want to load navigate to the folder in which the file is located and then click Open the name and path of the file appear in the field next to the Browse button 3 Click the Load File button corresponding to the file you want to load 4 Repeat steps 2 through 3 for each file you want to load 5 To save the loaded auxiliary files to flash memory refer to Saving Configuration on page 230 6 To reset the device if you have loaded a Call Progress Tones file refer to Resetting the Device on page 228 SIP User s Manual 232 Document LTRT 68808 SIP User s Manual 3 Web Based Management Saving an auxiliary file to flash memory may disrupt traffic on the device To avoid this disable all traffic on the device by performing a graceful lock refer to Locking and Unlocking the Device on page 229 You can schedule automatic loading of updated auxiliary files using HTTP HTTPS FTP or NFS refer to the Product Reference Manual You can also load the Auxiliary files u
365. gement Description f EnableSilenceCompression is 2 and IsCiscoSCEMode is 1 annexb no Determines whether echo cancellation is enabled and therefore echo from voice calls is removed 0 Off Echo Canceler is disabled 1 On Echo Canceler is enabled default Note This parameter is used to maintain backward compatibility Determines the DTMF transport type 0 DTMF Mute Erases digits from voice stream and doesn t relay to remote 2 Transparent DTMF Digits remain in voice stream 3 RFC 2833 Relay DTMF Erases digits from voice stream and relays to remote according to RFC 2833 default 7 RFC 2833 Relay Rev Mute DTMFs are sent according to RFC 2833 and muted when received Note This parameter is automatically updated if one of the following parameters is configured TxDTMFOption or RxDTMFOption Not Applicable DTMF gain control value in decibels to the TDM side The valid range is 31 to 0 dB The default value is 11 dB Controls the ABCD signaling transport type over IP 0 CAS Events Only Disable CAS relay default 1 CAS RFC2833 Relay Enable CAS relay mode using RFC 2833 The CAS relay mode can be used with the TDM tunneling feature to enable tunneling over IP for both voice and CAS signaling bearers Defines the range in decibels between the high and low frequency components in the DTMF signal Positive decibel values cause the higher frequency component to be st
366. ges The valid range is 1 to 65534 The default value is 5060 Local TCP port for SIP messages The valid range is 1 to 65534 The default value is 5060 Local TLS port for SIP messages The valid range is 1 to 65534 The default value is 5061 Note The value of must be different than the value of SIP TCP Local Port TCPLocalSIPPort Enables secured SIP SIPS URI connections over multiple hops 0 Disable default 1 Enable When SIP Transport Type is set to TLS SIPTransportType 2 and Enable SIPS is disabled TLS is used for the next network hop only When SIP Transport Type is set to TCP or TLS SIPTransportType 2 or 1 and Enable SIPS is enabled TLS is used through the entire connection over multiple hops Note If this parameter is enabled and SIP Transport Type is set to UDP SIPTransportType 0 the connection fails Enables the reuse of the same TCP connection for all calls to the same destination 0 Disable Use a separate TCP connection for each call default 1 Enable Use the same TCP connection for all calls Defines the Timer B INVITE transaction timeout timer and Timer F non INVITE transaction timeout timer as defined in RFC 3261 when the SIP Transport Type is TCP The valid range is 0 to 40 sec The default value is 64 SIPT1Rtx msec SIP destination port for sending initial SIP requests The valid range is 1 to 65534 The default port is 5060 Note SI
367. git pattern used by the PBX to indicate call forward with no reason when the original call is received from an external line not an internal extension The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate an internal call The valid range is a 120 character string Determines the digit pattern used by the PBX to indicate an external call The valid range is a 120 character string Determines a digit pattern that when received from the Tel side indicates the device to disconnect the call The valid range is a 25 character string A digit pattern that if received as Src S or Redirect R numbers is ignored and not added to that number The valid range is a 25 character string Determines the digit code used by the device to notify the PBX that there aren t any messages waiting for a specific extension This code is added as prefix to the dialed number The valid range is a 25 character string Determines the digit code used by the device to notify the PBX of messages waiting for a specific extension This code is added as prefix to the dialed number The valid range is a 25 character string Determines the digit code used by the device as a suffix for MWI On Digit Pattern and MWI Off Digit Pattern This suffix is added to the generated DTMF string after the extension number The valid range is a 25 character string Determines the calling party s phone number used
368. gle device registration is used i e Authentication Mode is set to Authentication Per gateway The password used for Basic Digest authentication with a Proxy Registrar server A single password is used for all device ports The default is Default_Passwd 139 November 2008 ca AudioCodes Parameter Cnonce Cnonce Authentication Mode AuthenticationMode Set Out Of Service On Registration Failure OOSOnRegistrationFail Challenge Caching Mode SIPChallengeCachingMode Mutual Authentication Mode Mutual AuthenticationMode SIP User s Manual Mediant 2000 Description Cnonce string used by the SIP server and client to provide mutual authentication Free format i e Cnonce 0a4f113b The default is Default_Cnonce Determines the device s registration and authentication method 0 Per Endpoint Registration and Authentication separately for each B channel 1 Per Gateway Single Registration and Authentication for the entire device default Single Registration and Authentication Authentication Mode 1 is usually defined for and digital modules Enables setting a trunk or the entire device i e all endpoints to out of service if registration fails 0 Disable Disabled default 1 Enable Enabled If the registration is per Endpoint i e AuthenticationMode is set to 0 or Account refer to Configuring the Trunk Group Settings on page 197 and a specific endpoin
369. gotiations To refrain from a situation where the SA expires a new SA is negotiated while the old one is still valid As soon as the new SA is created it replaces the old one This procedure occurs whenever an SA is about to expire First to Fourth Proposal Encryption Type IKEPolicyProposalEncryption _X Determines the encryption type used in the main mode negotiation for up to four proposals For the ini file parameter X depicts the proposal number 0 to 3 Version 5 6 119 November 2008 A ge AudioCodes Mediant 2000 Parameter Name Description 1 DES CBC 2 Triple DES CBC 8 AES CBC Not Defined default First to Fourth Proposal Determines the authentication protocol used in the main mode Authentication Type negotiation for up to four proposals For the ini file parameter X IKEPolicyProposalAuthenticat depicts the proposal number 0 to 3 lon_X 2 HMAC SHA1 96 4 HMAC MD5 96 Not Defined default First to Fourth Proposal DH Determines the length of the key created by the DH protocol for Group up to four proposals For the ini file parameter X depicts the IKEPolicyProposalDHGroup _ proposal number 0 to 3 x 0 DH 786 Bit 1 DH 1024 Bit Not Defined default 3 4 7 Protocol Configuration The Protocol Configuration menu allows you to configure the device s SIP parameters and contains the following submenus Protocol Definition refer to Configuring the Protocol Definitio
370. grade Key using one of the following m Web interface BootP TFTP configuration utility refer to Loading via BootP TFTP on page 235 m AudioCodes EMS refer to AudioCodes EMS User s Manual or EMS Product Description Warning Don t modify the contents of the Software Upgrade Key file The Software Upgrade Key is an encrypted key Each TPM utilizes a unique key The Software Upgrade Key is provided only by AudioCodes The procedure below describes how to load a Software Upgrade Key to the device using the Web interface Version 5 6 233 November 2008 A ge AudioCodes Mediant 2000 gt To load a Software Upgrade Key take these 6 steps 1 Open the Software Upgrade Key Status page Management tab gt Software Update menu gt Software Upgrade Key page item Figure 3 100 Software Upgrade Key Status Page Current Key Key features Board Type TrunkPack 1610 IP Media VXAL Security IPSEC MediaEncryption StrongEncryption EncryptControlProtocol ElTrunks 3 TiTrunks 8 DSP Voice features IpmDetector Control Protocols MGCP SIP SASurvivability Coders G723 G729 G728 NETCODER GSM FR GSM EFR AMR EVRC QCELP G727 ILBC EVRC B AMR WB G722 EG711 Channel Type RTP DspCh240 Default features Coders G711 G726 Add a Software Upgrade Key Add Key Send Upgrade Key file from your computer to the device Browse Send File Reset with flash burn is required after file is loaded 2 Backup your curren
371. gurations m lIsFaxUsed 0 E FaxTransportMode 1 Additional configuration parameters e FaxRelayEnhancedRedundancyDepth e FaxRelayRedundancyDepth e FaxRelayECMEnable e FaxRelayMaxRate Fax Modem Bypass Mode In this proprietary mode when fax or modem signals are detected the channel automatically switches from the current voice coder to a high bit rate coder according to the parameter FaxModemBypassCoderType In addition the channel is automatically reconfigured with the following fax modem adaptations m Disables silence suppression m Enables echo cancellation for fax m Disables echo cancellation for modem m Performs certain jitter buffering optimizations The network packets generated and received during the bypass period are regular voice RTP packets per the selected bypass coder but with a different RTP payload type according to the parameters FaxBypassPayloadType and ModemBypassPayloadType During the bypass period the coder uses the packing factor which is defined by the parameter FaxModemBypassM The packing factor determines the number of coder payloads each the size of FaxModemBypassBasicRTPPacketinterval that are used to generate a single fax modem bypass packet When fax modem transmission ends the reverse switching from bypass coder to regular voice coder is performed To configure fax modem bypass mode perform the following configurations E lIsFaxUsed 0 E FaxTransportMode 2 m V21ModemTranspo
372. h the DestinationPrefix SourcePrefix and SourceAddress conditions The manipulation rules are executed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and Suffix2Add Parameters can be skipped by using two dollar signs The Source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all addresses between 10 8 8 10 and 10 8 8 99 The Source IP address can include the asterisk wildcard to represent any number between 0 and 255 For example 10 8 8 represents all the addresses between 10 8 8 0 and 10 8 8 255 To configure manipulation of source numbers for IP to Tel calls using the Web interface refer to Configuring the Number Manipulation Tables on page 164 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 For ETSI ISDN variant the following Number Plan and Type combinations Plan Type are supported in the Destination and Source Manipulation tables 0 0 Unknown Unknown 9 0 Private Unknown 9 1 Private Level 2 Regional 9 2 Private Level 1 Regional 9 3 Private PISN Specific 9 4 Private Level 0 Regional local 1 0 Public ISDN E 164 Unknown Version 5 6 321 November 2008 A ge AudioCodes Mediant 2000 Parameter Description 1 1 Public ISDN E 164 International 1 2 Public ISDN E 164 National 1 3 Public
373. he servername is equal to ProxyName if configured The ProxyName can be any string m Otherwise the servername is equal to ProxylP either FQDN or numerical IP address The parameter GWRegistrationName can be any string This parameter is used only if registration is per device If the parameter is not defined the parameter UserName is used instead If the registration is per endpoint the endpoint phone number is used The sipgatewayname parameter defined in the ini file or Web interface can be any string Some Proxy servers require that the sipgatewayname in REGISTER messages is set equal to the Registrar Proxy IP address or to the Registrar Proxy domain name The sipgatewayname parameter can be overwritten by the TrunkGroupSettings_GatewayName value if the TrunkGroupSettings_RegistrationMode is set to Per Endpoint REGISTER messages are sent to the Registrar s IP address if configured or to the Proxy s IP address A single message is sent once per device or messages are sent per B channel according to the parameter AuthenticationMode There is also an option to configure registration mode per Trunk Group using the TrunkGroupSettings table The registration request is resent according to the parameter RegistrationTimeDivider For example if RegistrationTimeDivider 70 and Registration Expires time 3600 the device resends its registration request after 3600 x 70 2520 sec The default value of Registra
374. he INVITE message is set to the value of the parameter Dest IP Address if not empty otherwise it is set to the value of the parameter SIP Group Name defined in the IP Group table Note To configure Proxy Sets refer to Proxy Sets Table on page 141 The IP Profile ID configured in Configuring the Profile Definitions on page 190 assigned to this routing rule entry for the IP destination A read only field representing the Quality of Service of the destination IP address n a Alternative Routing feature is disabled OK IP route is available Ping Error No ping to IP destination route is not available QoS Low Bad QoS of IP destination route is not available DNS Error No DNS resolution only when domain name is used instead of an IP address 3 4 7 4 3 Outbound IP Routing Table The Outbound IP Routing Table page allows you to configure the device for routing outbound i e sent IP to IP calls This table routes inbound IP calls identified in Inbound IP Routing Table on page 184 received from an IP Group refer to Configuring the IP Groups on page 201 to a specific IP Group destination or IP address SIP User s Manual 178 Document LTRT 68808 SIP User s Manual 3 Web Based Management The Outbound IP Routing Table page appears only if the parameter EnableSBC is set to 1 i e enabled in SBC Configuration on page 163 If this parameter is not enabled default
375. he ISDN Setup or the following INFO messages to signal that no more digits are going to be sent m The inter digit timeout configured by the parameter TimeBetweenDigits expires The default for this timeout is 4 seconds E The maximum allowed number of digits configured by the parameter MaxDigits is reached The default is 30 digits A match is found with the defined digit map configured by the parameter DigitMapping Relevant parameters described in PSTN Parameters on page 303 ISDNRxOverlap ISDNRxOverlap_x TimeBetweenDigits MaxDigits ISDNInCallsBehavior DigitMapping ISDN Non Facility Associated Signaling NFAS In regular T1 ISDN trunks a single 64 kbps channel carries signaling for the other 23 B channels of that particular T1 trunk This channel is called the D channel and usually resides on timeslot 24 The ISDN Non Facility Associated Signaling NFAS feature enables the use of a single D channel to control multiple PRI interfaces With NFAS it is possible to define a group of T1 trunks called an NFAS group in which a single D channel carries ISDN signaling messages for the entire group The NFAS group s B channels are used to carry traffic such as voice or data The NFAS mechanism also enables definition of a backup D channel on a different T1 trunk to be used if the primary D channel fails The NFAS group can comprise up to 10 T1 trunks Each T1 trunk is called an NFAS member The T1 trunk wh
376. he Proxy fails the device re routes the call according to the Standard mode 0 When this parameter is set to 2 and the INVITE fails the device re routes the call according to the Standard mode 0 If DNS resolution fails the device attempts to route the call to the Proxy If routing to the Proxy also fails the Redirect Transfer request is rejected When this parameter is set to 2 the XferPrefix parameter can be used to define different routing rules for redirected calls This parameter is disregarded if the parameter AlwaysSendToProxy is set to 1 Always Use Route Table Determines the Request URI host name in outgoing INVITE messages Disable default Enable The device uses the IP address or domain name defined in the Tel to IP Routing table Tel to IP Routing Table on page 175 as the Request URI host name in outgoing INVITE messages instead of the value entered in the SIP Group Name field 3 4 7 6 4 Configuring the Account Table The Account Table page allows you to define accounts per Trunk Groups referred to as Served Trunk Group or to a Served IP Group for registration and or digest authentication user name and password to a destination IP address Serving IP Group The Account table can be used for example to register to an Internet Telephony Service Provider ITSP on behalf of an IP PBX to which the device is connected The registrations are sent to the Proxy Set ID refer to Prox
377. he device supports T 38 as well as VBD codecs i e G 711 A law and G 711 u law The selection of capabilities is performed using the coders table refer to Coders on page 144 When in VBD mode for V 152 implementation support is negotiated between the device and the remote endpoint at the establishment of the call During this time initial exchange of call capabilities is exchanged in the outgoing SDP These capabilities include whether VBD is supported and associated RTP payload types gpmd SDP attribute supported codecs and packetization periods for all codec payload types ptime SDP attribute After this initial negotiation no Re INVITE messages are necessary as both endpoints are synchronized in terms of the other side s capabilities If negotiation fails i e no match was achieved for any of the transport capabilities fallback to existing logic occurs according to the parameter IsFaxUsed Version 5 6 357 November 2008 A EA AudioCodes Mediant 2000 Below is an example of media descriptions of an SDP indicating support for V 152 1 oO O 0 IN IPV4 lt IPAdressA gt tnos S 0 p 1 c IN IP4 lt IPAddressA m audio lt udpPort A gt RTP AVP 18 0 a ptime 10 a rtpmap 96 PCMU 8000 a gpmd 96 vbd yes In the example above V 152 implementation is supported using the dynamic payload type 96 and G 711 u law as the VBD codec as well as the voice codecs G 711 p law and G 729 Instead of using V
378. he main IP route above any alternative route When an appropriate entry destination number matches one of the prefixes is found the prefix s corresponding destination IP address is verified If the destination IP address is disallowed or if the original call fails and the device has made two additional attempts to establish the call without success an alternative route is searched in the table after which an alternative route is used Version 5 6 361 November 2008 A ge AudioCodes Mediant 2000 Destination IP address is disallowed if no ping to the destination is available ping is continuously initiated every seven seconds when an inappropriate level of QoS was detected or when a DNS host name is not resolved The QoS level is calculated according to delay or packet loss of previously ended calls If no call statistics are received for two minutes the QoS information is reset 7 9 2 Determining the Availability of Destination IP Addresses To determine the availability of each destination IP address or host name in the routing table one or all of the following configurable methods are applied m Connectivity The destination IP address is queried periodically currently only by ping mM QoS The QoS of an IP connection is determined according to RTOP statistics of previous calls Network delay in msec and network packet loss in percentage are separately quantified and compared to a certain configurable threshold If
379. he required string you can use the Search History drop down list to select the string saved from a previous search Click Search a list of located parameters based on your search appears in the Navigation pane Each searched result displays the following e inifile parameter name e Link in green to its location page in the Web interface e Brief description of the parameter In the searched list click the required parameter link in green to open the page in which the parameter appears the relevant page opens in the Work pane and the searched parameter is highlighted for easy identification as shown in the figure below Figure 3 13 Searched Result Screen fS AudioCodes Congestion Management soreness Somas eeiiearchies Basic Full Media Pren Search History vian v Gold Priority VLANURONZESERVICECLASSPRIOR A ok VLANS OINI Sets the prionty for the Bronze servic or TROLYLANIO LAN CECLASSPRIORIT Searched Results Note Ifthe searched parameter is not located a notification message is displayed 33 November 2008 A E al AudioCodes Mediant 2000 3 3 5 Working with Scenarios The Web interface allows you to create your own menu with up to 20 pages selected from the menus in the Navigation tree i e pertaining to the Configuration Management and Status amp Diagnostics tabs The menu is a set of configuration pages grouped into a logical entity referred to as a Scenario Ea
380. he switchover In addition it s recommended to set the physical secondary link prior to resetting the device since the MAC configuration cannot be changed thereafter Note that since the two Ethernet ports use the same MAC address the external switches connected to the device can in some cases create a noticeable switchover delay due to their internal switching logic though at the device level the switchover delay is minimal milliseconds NAT Network Address Translation Support Network Address Translation NAT is a mechanism that maps a set of internal IP addresses used within a private network to global IP addresses providing transparent routing to end hosts The primary advantages of NAT include 1 Reduction in the number of global IP addresses required in a private network global IP addresses are only used to connect to the Internet 2 Better network security by hiding its internal architecture The following figure illustrates the device s supported NAT architecture Figure 8 1 NAT Architecture Mediant SIP User s Manual 380 Document LTRT 68808 SIP User s Manual 8 Networking Capabilities 8 3 1 The design of SIP creates a problem for VoIP traffic to pass through NAT SIP uses IP addresses and port numbers in its message body and the NAT server can t modify SIP messages and therefore can t change local to global addresses Two different streams traverse through NAT signaling and media A device located
381. hen forwarding the requests to the proxy Normal Mode this port serves as the source port The valid range is 1 to 65 534 The default value is 5080 Local TLS port used to send receive SIP messages for the SAS application The SIP entities in the local network need to send the registration requests to this port When forwarding the requests to the proxy Normal Mode this port serves as the source port The valid range is 1 to 65 534 The default value is 5081 Determines the Proxy Set index number used in SAS Normal mode to forward REGISTER and INVITE requests from the users that are served by the SAS application The valid range is 0 to 5 The default value is 0 i e default Proxy Set Determines the Proxy Set index number used in SAS Emergency mode for fallback when the user is not found in the Registered Users database Each time a new SIP request arrives the SAS application checks whether the user is listed in the registration database If the user is located in the database the request is sent to the user If the user is not found the request is forwarded to the next redundant SAS defined in the Redundant SAS Proxy Set If that SAS Proxy IP appears in the Via header of the request it is not forwarded so that loops are prevented in the request s course If no such redundant SAS exists the SAS sends the request to its default gateway configured by the parameter SASDefaultGatewayIP The valid range is 1 to
382. ic parameters The Web interface provides a user friendly graphical user interface GUI which can be accessed using any standard Web browser e g Microsoft Internet Explorer Access to the Web interface is controlled by various security mechanisms such as login user name and password read write privileges and limiting access to specific IP addresses The Web interface allows you to configure most of the device s parameters Those parameters that are not available in the Web interface can be configured using the ini file Certain Web interface pages are feature key dependant and therefore only appear if your device s feature key supports the features relating to these pages refer to Upgrading the Software Upgrade Key on page 233 Throughout this section parameters enclosed in square brackets depict the ini file parameters for configuring the device using the ini file Computer Requirements To use the device s Web interface the following is required m A connection to the Internet network World Wide Web m A network connection to the device s Web interface m One of the following Web browsers e Microsoft Internet Explorer version 6 0 or later e Netscape Navigator version 7 2 or later e Mozilla Firefox version 1 5 0 10 or later m Recommended screen resolution of 1024 x 768 pixels or 1280 x 1024 pixels Note Your Web browser must be JavaScript enabled in order to access the Web
383. icable only to ISDN protocols For some ISDN variants when Any 2 is selected the SETUP message does not include the Channel Identification IE The Any 2 option is applicable only if the parameter ISDN Termination Side is set to Use side refer to Configuring the Trunk Settings on page 82 0 No Don t change numbers default 1 Yes Incoming ISDN call that includes a redirect number sometimes referred to as original called number uses the redirect number instead of the called number Determines the tone for MFC R2 Calling Party Category CPC The parameter provides information on the calling party such as National or International call Operator or Subscriber and Subscriber priority The value range is 1 to 15 defining one of the MFC R2 tones The default value is 1 Determines whether a call is disconnected upon detection of a busy tone 0 Disable Do not disconnect call on detection of busy tone 1 Enable Disconnect call on detection of busy tone default Note This parameter is applicable only to CAS protocols Determines whether a call is disconnected upon detection of a busy tone 0 Do not disconnect call upon detection of busy tone 1 Disconnect call upon detection of busy tone default Note This parameter is applicable only to ISDN protocols Enables TDM tunneling 0 Disable Disabled default 1 Enable TDM Tunneling is enabled
384. ication service default 1 Enable RAI service If RAI is enabled an SNMP acBoardCallResourcesAlarm Alarm Trap is sent if device s busy endpoints exceed a predefined configurable threshold High threshold percentage of total calls that are active busy endpoints When the percentage of the device s busy endpoints exceeds this High Threshold the device sends the SNMP acBoardCallResourcesAlarm Alarm Trap with a major Alarm Status The range is 0 to 100 The default value is 90 Note The percentage of busy endpoints is calculated by dividing the number of busy endpoints by the total number of enabled endpoints trunks are physically connected and synchronized with no alarms and endpoints are defined in the Trunk Group table Low threshold percentage of total calls that are active busy endpoints When the percentage of the device s busy endpoints falls below this Low Threshold the device sends an SNMP acBoardCallResourcesAlarm Alarm Trap with a cleared Alarm Status The range is 0 to 100 The default value is 90 269 November 2008 A Ee AudioCodes Mediant 2000 Parameter RAILoopTime Description Time interval in seconds that the device periodically checks call resource availability The valid range is 1 to 200 The default is 10 Disconnect Supervision Parameters TelConnectCode DisconnectOnBrokenCon nection BrokenConnectionEventTi meout EnableSilenceDisconnect FarEndDisconnectSile
385. ication of the sequence number Note Receipt of No Op packets is always supported SIP User s Manual 382 Document LTRT 68808 SIP User s Manual 8 Networking Capabilities 8 4 IP Multicasting The device supports IP Multicasting level 1 according to RFC 2236 i e IGMP version 2 for RTP channels The device is capable of transmitting and receiving Multicast packets 8 5 Robust Reception of RTP Streams This mechanism filters out unwanted RTP streams that are sent to the same port number on the device These multiple RTP streams can result from traces of previous calls call control errors and deliberate attacks When more than one RTP stream reaches the device on the same port number the device accepts only one of the RTP streams and rejects the rest of the streams The RTP stream is selected according to the following The first packet arriving on a newly opened channel sets the source IP address and UDP port from which further packets are received Thus the source IP address and UDP port identify the currently accepted stream If a new packet arrives whose source IP address or UDP port are different to the currently accepted RTP stream one of the following occurs E The device reverts to the new RTP stream when the new packet has a source IP address and UDP port that are the same as the remote IP address and UDP port that were stated during the opening of the channel m The packet is dropped when the new packet has any oth
386. icyLocallPAddressType IPSecPolicyRemoteTunnellPAddress IPsecPolicyRemoteSubnetMask IPSEC_SPD_TABLE For example IPSEC_SPD_TABLE Format SPD_INDEX IPSecMode IPSecPolicyRemotelPAddress lpsecPolicySrcPort IPSecPolicyDStPort IPSecPolicyProtocol IPSecPolicyLifelnSec IPSecPolicyProposalEncryption_0 IPSecPolicyProposalAuthentication_0 IPSecPolicyProposalEncryption_1 IPSecPolicyProposalAuthentication_1 IPSecPolicyKeyExchangeMethodIndex IPSecPolicyLocallPAddressType IPSEC_SPD_TABLE 0 0 10 11 2 21 0 0 17 900 1 2 2 2 1 0 IPSEC_SPD_TABLE In the example above all packets designated to IP address 10 11 2 21 that originate from the OAMP interface regardless of destination and source ports and whose protocol is UDP are encrypted The IPSec SPD also defines an SA lifetime of 900 seconds and two security proposals DES SHA1 and 3DES SHA1 IPsec is performed using the Transport mode Notes Each row in the table refers to a different IP destination To support more than one Encryption Authentication proposal for each proposal specify the relevant parameters in the Format line The proposal list must be contiguous To configure the IKE table using the Web interface refer to Configuring the IPSec Table on page 114 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 This ini file table parameter configures the IKE table The
387. ificates to 1 5 Save the configuration refer to Saving Configuration on page 230 and then restart the device When a user connects to the secured Web server E Ifthe user has a client certificate from a CA that is listed in the Trusted Root Certificate file the connection is accepted and the user is prompted for the system password m If both the CA certificate and the client certificate appear in the Trusted Root Certificate file the user is not prompted for a password thus providing a single sign on experience the authentication is performed using the X 509 digital signature m If the user doesn t have a client certificate from a listed CA or doesn t have a client certificate at all the connection is rejected The process of installing a client certificate on your PC is beyond the scope of this document For more information refer to your Web browser or operating system documentation and or consult your security administrator The root certificate can also be loaded via ini file using the parameter HTTPSRootFileName You can enable Online Certificate Status Protocol OCSP on the device to check whether a peer s certificate has been revoked by an OCSP server For further information refer to the Product Reference Manual SIP User s Manual 108 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 6 4 3 Self Signed Certificates The device is shipped with an operational self signed ser
388. ifies the interface network type to which the RoutingTablelInterfacesColumn routing rule is applied 0 OAMP default 1 Media 2 Control For detailed information on the network types refer to Configuring the Multiple Interface Table on page 53 3 4 1 6 Configuring the QoS Settings The QoS Settings page is used for configuring the Quality of Service QoS parameters This page allows you to assign VLAN priorities IEEE 802 1p and Differentiated Services DiffServ for the supported Class of Service CoS Version 5 6 63 November 2008 ca AudioCodes Mediant 2000 gt Toconfigure QoS take these 4 steps 1 Open the QoS Settings page Configuration tab gt Network Settings menu gt QoS Settings page item Figure 3 39 QoS Settings Page w Priority Settings Network Priority Media Premium Priority Control Premium Priority Gold Priority Bronze Priority v Differential Services Network QoS le Media Premium QoS le Control Premium QoS o Gold QoS s Bronze QoS 2 Configure the QoS parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 12 QoS Settings Parameters Parameter Priority Settings Network Priority VLANNetworkServiceClassPriority Media Premium Priority VLANPremiumServiceClassMediaPriority
389. igure the General Security parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 110 Document LTRT 68808 SIP User s Manual 3 Web Based Management Table 3 25 General Security Parameters Parameter HTTP Authentication Mode WebAuthMode Secured Web Connection HTTPS HTTPSOnly General RADIUS Settings Enable RADIUS Access Control EnableRADIUS Use RADIUS for Web Telnet Login WebRADIUSLogin RADIUS Authentication Server IP Address RADIUSAuthServerlIP RADIUS Authentication Server Port RADIUSAuthPort Version 5 6 Description Determines the authentication mode for the Web interface 0 Basic Mode Basic authentication clear text is used default 1 Digest When Possible Digest authentication MD5 is used 2 Basic if HTTPS Digest if HTTP Digest authentication MD5 is used for HTTP and basic authentication is used for HTTPS Note When RADIUS login is enabled i e the parameter WebRADIUSLogin is set to 1 basic authentication is forced Determines the protocol types used to access the Web interface 0 Disable HTTP and HTTPS default 1 Enable Unencrypted HTTP packets are blocked Determines whether the RADIUS application is enabled 0 Disable RADIUS application is disabled default 1 Enable RADIUS application is en
390. iguring the Certificates on page 105 General Security Settings refer to Configuring the General Security Settings on page 109 IPSec Table refer to Configuring the IPSec Table on page 114 m IKE Table refer to Configuring the IKE Table on page 117 3 4 6 1 Configuring the Web User Accounts To prevent unauthorized access to the Web interface two Web user accounts are available primary and secondary with assigned user name password and access level When you login to the Web interface you are requested to provide the user name and password of one of these Web user accounts If the Web session is idle i e no actions are performed for more than five minutes the Web session expires and you are once again requested to login with your user name and password Up to five Web users can simultaneously open log in to a session on the device s Web interface Each Web user account is composed of three attributes m User name and password enables access login to the Web interface m Access level determines the extent of the access i e availability of pages and read write privileges The available access levels and their corresponding privileges are listed in the table below Version 5 6 99 November 2008 A ge AudioCodes Mediant 2000 Table 3 22 Web User Accounts Access Levels and Privileges Numeric Representation PENG Be Access Level Security Administrator 200 Read write privileges for all pages
391. iguring the IP Settings The IP Settings page is used for configuring basic IP networking parameters such as the device s IP address However from this page you can also access the Multiple Interface Table page for configuring multiple interfaces Note Once you configure multiple interfaces in the Multiple Interface Table page accessed by clicking the 4 button when clicking the IP Settings page item in the Navigation tree the Multiple Interface Table page is accessed instead of the IP Settings page SIP User s Manual 50 Document LTRT 68808 SIP User s Manual 3 Web Based Management gt To configure the IP settings parameters take these 4 steps 1 Open the IP Settings page Configuration tab gt Network Settings menu gt IP Settings page item Figure 3 33 IP Settings Page IP Settings l IP Networking Mode Single IP Network Single IP Settings lS IP Address 10 13 4 13 l Subnet Mask 255 255 0 0 Default Gateway Address 10 13 01 wv Multiple Interface Settings Multiple Interface Table gt v VLAN Mode v VLAN ID Settings Native VLAN ID OAM VLAN ID Control VLAN ID Media VLAN ID sw NAT Settings NAT IP Address 2 Configure the IP parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory
392. in seconds for registering to a Proxy server The value is used in the Expires header In addition this parameter defines the time interval between Keep Alive messages when the parameter EnableProxyKeepAlive is set to 2 REGISTER Typically the device registers every 3 600 sec i e one hour The device resumes registration according to the parameter RegistrationTimeDivider The valid range is 10 to 2 000 000 The default value is 180 Defines the re registration timing in percentage The timing is a percentage of the re register timing set by the Registrar server The valid range is 50 to 100 The default value is 50 For example If RegistrationTimeDivider is 70 and Registration Expires time is 3600 the device re sends its registration request after 3600 x 70 2520 sec Note This parameter may be overriden if the parameter RegistrationTimeThreshold is greater than 0 refer to the description of RegistrationTimeThreshold Defines the time interval in seconds after which a Registration request is resent if registration fails with a 4xx response or if there is no response from the Proxy Registrar server The default is 30 seconds The range is 10 to 3600 Defines a threshold in seconds for re registration timing If this parameter is greater than 0 but lower than the computed re registration timing according to the parameter RegistrationTimeDivider the re registration timing is set to the following timing set by the
393. in the Q 931 MWI SETUP message to PSTN If not configured the channel s phone number is used as the calling number Enables Simplified Message Desk Interface SMDI interface on the device 0 Disable Normal serial default 1 Enable Bellcore 2 Ericsson MD 110 3 NEC ICS Note When the RS 232 connection is used for SMDI messages Serial SMDI it cannot be used for other 216 Document LTRT 68808 SIP User s Manual SMDI Timeout SMDITimeOut Parameter 3 Web Based Management Description applications for example to access the Command Line Interface CLI Determines the time in msec that the device waits for an SMDI Call Status message before or after a SETUP message is received This parameter synchronizes the SMDI and analog CAS interfaces If the timeout expires and only an SMDI message is received the SMDI message is dropped If the timeout expires and only a SETUP message is received the call is established The valid range is 0 to 10000 i e 10 seconds The default value is 2000 3 4 8 2 Configuring RADIUS Accounting Parameters The RADIUS Parameters page is used for configuring the Remote Authentication Dial In User Service RADIUS accounting parameters gt To configure the RADIUS parameters take these 4 steps 1 Open the RADIUS Parameters page Configuration tab gt Advanced Applications menu gt RADIUS Parameters page item Figure 3 88 RADIUS
394. ined by the following according to priority a PDiffServ value in the selected IP Profile a PremiumServiceClassMediaDiffServ Control Premium QoS Defines the DiffServ value for Premium Control PremiumServiceClassControlDiffServ CoS content only if ControllPDiffserv is not set in the selected IP Profile The valid range is 0 to 63 The default value is 40 Note The value for the Premium Control DiffServ is determined by the following according to priority a ControlPDiffserv value in the selected IP Profile PremiumServiceClassControlDiffServ Gold QoS Defines the DiffServ value for the Gold CoS GoldServiceClassDiffServ content The valid range is 0 to 63 The default value is 26 Bronze QoS Defines the DiffServ value for the Bronze CoS BronzeServiceClassDiffServ content The valid range is 0 to 63 The default value is 10 3 4 2 Media Settings The Media Settings menu allows you to configure the device s channel parameters These parameters are applied to all the device s channels This menu contains the following page items m Voice Settings refer to Configuring the Voice Settings on page 66 E Fax Modem CID Settings refer to Configuring the Fax Modem CID Settings on page 67 RTP RTCP Settings refer to Configuring the RTP RTCP Settings on page 71 IPmedia Settings refer to Configuring the IPmedia Settings on page 76 General Media Settings refer to Configuring the General Media Setti
395. ing IP Group For Tel to IP calls the Served Trunk Group is the source Trunk Group from where the call initiated For IP to Tel calls the Served Trunk Group is the Trunk Group ID defined in the IP to Trunk Group Routing table refer to IP to Trunk Group Routing on page 181 For defining Trunk Groups refer to Configuring the Trunk Group Table on page 195 Note For IP to IP call routing this parameter must be set to 1 i e no trunk Served IP Group The Source IP Group e g IP PBX for which registration and or authentication is performed Serving IP Group The destination IP Group ID defined in Configuring the IP Groups on page 201 to where the REGISTER requests if enabled are sent or Authentication is performed The actual destination to where the REGISTER requests are sent is the IP address defined for the Proxy Set ID refer to Proxy Sets Table on page 141 associated with this IP Group This occurs only in the following conditions The parameter Registration Mode is set to Per Account in the Trunk Group Settings table refer to Configuring the Trunk Group Settings on page 197 Version 5 6 205 November 2008 A Ee AudioCodes Mediant 2000 Parameter Username Password HostName Register SIP User s Manual Description The parameter Register in this table is set to 1 In addition for a SIP call that is identified by both the Served Trunk Group Served IP Group and Servin
396. ing SIP messages 1 Use the Reason header value for Release Reason mapping default Determines the device s behavior upon receipt of SIP Re INVITE messages that include the silencesupp off attribute 290 Document LTRT 68808 SIP User s Manual Parameter EnableRport DSPVersionTemplateNumber VBRCoderHeaderFormat VBRCoderHangover AMRFECRedundancyDepth AMRFECNumberOfCodecMod es AMRFECDelayThreshhold AMRFECDelayHysteresis Version 5 6 4 ini File Configuration Description 0 Disregard the silecesupp attribute default 1 Handle incoming Re INVITE messages that include the silencesupp off attribute in the SDP as a request to switch to the Voice Band Data VBD mode Enables disables the usage of the rport parameter in the Via header 0 Enabled 1 Disabled default The device adds an rport parameter to the Via header of each outgoing SIP message The first Proxy that receives this message sets the rport value of the response to the actual port from which the request was received This method is used for example to enable the device to identify its port mapping outside a NAT If the Via doesn t include rport tag the destination port of the response is taken from the host part of the Via header If the Via includes rport tag without a port value the destination port of the response is the source port of the incoming request If the Via includes rport
397. ing the Account table using the Web interface and for a description of the items in this ini file table refer to Configuring the Account Table on page 204 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 This ini file table parameter configures the IP Group table The format of this parameter is as follows IPGroup FORMAT IPGroup_Index IPGroup_ Type IPGroup_Description PGroup_ProxySetld IPGroup_SIPGroupName PGroup_ContactUser IPGroup_EnableSurvivability PGroup_ServinglPGroup IPGroup_SIPReRoutingMode IPGroup_AlwaysUseRouteTable IPGroup For example IPGroup FORMAT IPGroup_Index IPGroup_Type IPGroup_Description IPGroup_ProxySetld PGroup_SIPGroupName IPGroup_ContactUser PGroup_EnableSurvivability IPGroup_Serving PGroup IPGroup_SIPReRoutingMode IPGroup_AlwaysUseRouteT able IPGroup 1 0 acme gateway 1 firstIPgroup 0 1 O 0 IPGroup 2 0 abc server 2 second Pgroup 0 1 0 0 IPGroup 3 0 IP phones 1 thirdIPGroup 0 1 0 0 IPGroup Notes This table parameter can include up to 9 indices 1 9 For configuring the IP Group table using the Web interface and for a description of the items in this ini file table refer to Configuring the IP Groups on page 201 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 Define
398. ing the Trunk Group Table on page 195 Fora description of ini file table parameters refer to Structure of ini File Table Parameters on page 257 DefaultNumber For a description of this parameter refer to DTMF amp Dialing Parameters on page 147 ChannelSelectMode For a description of this parameter refer to SIP General Parameters on page 121 TrunkGroupSettings This ini file table parameter defines rules for port allocation per Trunk Group If no rule exists the global rule defined by the parameter ChannelSelectMode takes effect The format of this parameter is as follows TrunkGroupSettings FORMAT TrunkGroupSettings_ Index TrunkGroupSettings_TrunkGroupld TrunkGroupSettings_ChannelSelectMode TrunkGroupSettings_ RegistrationMode TrunkGroupSettings_GatewayName TrunkGroupSettings Cont actUser TrunkGroupSettings_Serving IPGroup TrunkGroupSettings For example SIP User s Manual 314 Document LTRT 68808 SIP User s Manual Parameter AddTrunkGroupAsPrefix AddPortAsPrefix ReplaceEmptyDstWithPortNum ber CopyDestOnEmptySource AddNPlandTON2CallingNumbe r AddNPlandTON2CalledNumber UseSourceNumberAsDisplayN ame UseDisplayNameAsSourceNum ber AlwaysUseRouteTable Prefix Version 5 6 4 ini File Configuration Description TrunkGroupSettings TrunkGroupSettings 0 1 0 5 audiocodes user 1 TrunkGroupSettings 1 2 1 0 localname user1 2 TrunkGroupSettings Notes
399. ing to packet loss and delay of the IP address Unknown Recent quality information isn t available OK Poor Notes This parameter is applicable only if the parameter Alt Routing Tel to IP Mode is set to QoS or Both AltRoutingTel2IPMode 2 or 3 This parameter is reset if no QoS information is received for 2 minutes Displays QoS information delay and packet loss calculated according to previous calls Notes This parameter is applicable only if the parameter Alt Routing Tel to IP Mode is set to QoS or Both AltRoutingTel2IPMode 2 or 3 This parameter is reset if no QoS information is received for 2 minutes DNS status can be one of the following DNS Disable DNS Resolved DNS Unresolved 253 November 2008 A ge AudioCodes Mediant 2000 Reader s Notes SIP User s Manual 254 Document LTRT 68808 SIP User s Manual 4 ini File Configuration 4 ini File Configuration As an alternative to configuring the device using the Web interface as described in Web Based Management on page 19 you can configure the device by loading an ini file containing user defined parameters The ini file can be loaded using the following methods m AudioCodes BootP TFTP utility refer to the Product Reference Manual m Any standard TFTP server m Web interface refer to Backing Up and Restoring Configuration on page 240 The ini file configuration parameters are saved in the device s non
400. ini File Field Name Web Parameter Name Valid Range and Description simultaneously with the parameter PlayRBTone2Trunk The D in the ini file parameter depicts the trunk number where 0 is the first trunk Progress Indicator to ISDN Progress Indicator PI to ISDN The D in the ini file parameter ProgressIndicator2ISDN_ depicts the trunk number where 0 is the first trunk D 1 Not Configured The PI in ISDN messages is set according to the parameter PlayRBTone2Tel default 0 No PI PI is not sent to ISDN 1 Pl 1 8 PI 8 The PI value is sent to PSTN in Q 931 Proceeding and Alerting messages Typically the PSTN PBX cuts through the audio channel without playing local Ringback tone enabling the originating party to hear remote Call Progress Tones or network announcements Set PI in Rx Disconnect Defines the device s behavior when a Disconnect message is Message received from the ISDN before a Connect message is received The PIForDisconnectMsg_ ID ID in the ini file parameter depicts the trunk number where 0 is the first trunk 1 Not Configured Sends a 183 SIP response according to the received progress indicator PI in the ISDN Disconnect message If Pl 1 or 8 the device sends a 183 response enabling the PSTN to play a voice announcement to the IP side If there isn t a PI in the Disconnect message the call is released default 0 No PI Doesn t send a 183 response to IP The call is
401. interface refer to Loading Auxiliary Files on page 231 The following is an example of an ini file that includes these definitions This ini file is converted using the TrunkPack Conversion Utility refer to the Product Reference Manual to a binary file and loaded to the device Example of dial plan configuration This file contains two dial plans you may specify which one to use in CAS configuration PLAN1 Define the area codes 02 03 04 In these area codes phone numbers have 7 digits 02r OBa 7 OA n T Define the cellular VoIP area codes 052 054 050 and 077 In these area codes phone numbers have 8 digits 052 8 054 8 050 8 077 8 Define the international prefixes 00 012 014 The number following these prefixes may be 7 to 14 digits in length 00 y 1A O12 7 14 014 7 14 Define the emergency number 911 No additional digits are expected 911 0 PLAN2 Define the area codes 02 03 04 In these area codes phone numbers have 7 digits 0 2 4 7 Operator services starting with a star 41 42 43 Version 5 6 339 November 2008 A ge AudioCodes Mediant 2000 No additional digits are expected 4 1 3 0 The list must be prepared in a textual ini file with the following syntax Mm Every line in the file defines a known dialing prefix and the number of digits expected to follow that prefix The prefix must be separated from th
402. ion Detects whether voice or an answering machine is answering the call Note When implementing Answer Machine Detector channel capacity may be reduced 32 tones single tone dual tones or AM tones programmable frequency amp amplitude 64 frequencies in the range 300 to 1980 Hz 1 to 4 cadences per tone up to 4 sets of ON OFF periods 32 dB to 31 dB in steps of 1 dB 32 dB to 31 dB in steps of 1 dB Fax and Modem Transport Modes Real time Fax Relay Fax Transparency Modem Transparency Protocols VoIP Signaling Protocol Communication Protocols Telephony Protocols In Band Signaling Interfaces Telephony Interface SIP User s Manual Group 3 real time fax relay up to 14400 bps with automatic fallback Tolerant network delay up to 9 seconds round trip delay 1 30 PSTN and T 38 IP compliant real time fax CNG tone detection amp Relay per T 38 Answer tone CED or AnsAm detection amp Relay per T 38 Automatic fax bypass pass through to G 711 ADPCM or NSE bypass mode Automatic switching pass through to PCM ADPCM or NSE bypass mode for modem signals V 34 or V 90 modem detection SIP RFC 3261 RTP RTCP packetization IP stack UDP TCP RTP Remote Software load TFTP HTTP and HTTPS PRI ETSI Euro ISDN ANSI NI2 4 5ESS DMS 100 QSIG Japan INS1500 Australian Telecom New Zealand Telecom Hong Kong Variant Korean MIC E1 T1 CAS protocols MFC R2 E amp M wink
403. ion menu gt SIP Advanced Parameters submenu gt Advanced Parameters page item Figure 3 64 Advanced Parameters Page wv General IP Security Filter Calls to IP Enable Digit Delivery to Tel Enable Digit Delivery to IP RTP Only Mode PSTN Alert Timeout Disable Don t Filter Disable Disable Disable 180 wv Disconnect and Answer Supervision Disconnect on Broken Connection Broken Connection Timeout 100 msec Disconnect Call on Silence Detection i Silence Detection Period sec Silence Detection Method Enable Fax Re Routing Yes 100 No 120 Packets Count Disable CDR and Debug CDR Server IP Address CDR Report Level Debug Level w Misc Parameters Progress Indicator to IP Enable Channel Header Enable Busy Out Default Release Cause Max Number of Active Calls Max Call Duration min Enable LAN Watchdog Enable User Information Usage Delay After Reset sec Not Configured Disable Disable 3 2016 0 Enable Disable 7 2 Configure the parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 SIP
404. ions are sent by endpoints pertaining to the Trunk Group For example if the device is configured globally to register all its endpoints using the parameter ChannelSelectMode you can exclude some endpoints from being registered by assigning them to a Trunk Group and configuring the Trunk Group registration mode to Don t Register 5 Per Account Registrations are sent or not to an IP Group according to the settings in the Account table refer to Configuring the Account Table on page 204 Notes To enable Trunk Group registrations configure the global parameter IsRegisterNeeded to 1 This is unnecessary for Per Account registration mode If no mode is selected the registration is performed according to the global registration parameter ChannelSelectMode refer to Proxy amp Registration Parameters on page 132 If the device is configured globally ChannelSelectMode to register Per Endpoint and a Trunk Group comprising four channels is configured to register Per Gateway the device registers all channels except the first four channels The Trunk Group of these four channels sends a single registration request 199 November 2008 A Ee AudioCodes Mediant 2000 Parameter Description Serving IP Group ID The Serving IP Group ID to where INVITE messages initiated by this TrunkGroupSettings Servi Trunk Group s endpoints are sent The actual destination to where ngIPGroup these INVITE messages are sent is to the
405. ires a new SA is negotiated while the old one is still valid As soon as the new SA is created it replaces the old one This procedure occurs whenever an SA is about to expire First to Fourth Proposal Encryption Type Determines the encryption type used in the quick mode IPSecPolicyProposalEncryption_X negotiation for up to four proposals For the ini file parameter X depicts the proposal number 0 to 3 The valid encryption values are 0 None No encryption 1 DES CBC 2 Triple DES CBC 3 AES CBC Not Defined default First to Fourth Proposal Authentication Determines the authentication protocol used in the quick Type mode negotiation for up to four proposals For the ini file IPSecPolicyProposalAuthentication_X _ parameter X depicts the proposal number 0 to 3 The valid authentication values are 2 HMAC SHA 1 96 4 HMAC MD5 96 Not Defined default 3 4 6 7 Configuring the IKE Table The IKE Table page is used to configure the Internet Key Exchange IKE parameters Note You can also configure the IKE table using the ini file table parameter IPSec_IKEDB_Table refer to Security Parameters on page 276 Version 5 6 117 November 2008 ca AudioCodes gt To configure the IKE table take these 5 steps Mediant 2000 1 Open the IKE Table page Configuration tab gt Security Settings menu gt IKE Table page item Figure 3 58 IKE Table Page v Polic
406. irst Proxy Registrar server If there is no response from the first Proxy Registrar server after a specific number of retransmissions configured by the parameter HotSwapRix the INVITE REGISTER message is resent to the next redundant Proxy Registrar server The Coders page allows you to configure up to five coders and their attributes for the device The first coder in the list is the highest priority coder and is used by the device whenever possible If the far end device cannot use the first coder the device attempts to use the next coder in the list and so forth SIP User s Manual 144 Document LTRT 68808 SIP User s Manual 3 Web Based Management The device always uses the packetization time requested by the remote side for sending RTP packets For an explanation on V 152 support and implementation of T 38 and VBD coders refer to Supporting V 152 Implementation on page 357 You can also configure the Coders table using the ini file table parameter CoderName refer to SIP Configuration Parameters on page 284 The coders supported by the device are listed in the table below Coder Name G 711 A law g711 Alaw64k G 711 U law g711Ulaw64k EG 711 A law eg711Alaw EG 711 U law eg711Ulaw G 729 9729 G 723 1 97231 G 726 9726 GSM FR gsmFullRate GSM EFR gsmEnhancedFullRate AMR Amr Version 5 6 Table 3 33 Supported Coders Packetization Time 10 20 defa
407. irst match Phone Context is a unique as it doesn t appear in the Request URI as a Phone Context parameter Instead it s added as a prefix to the phone number The isn t removed from the phone number in the IP to Tel direction To configure the Phone Context table using the Web interface refer to Mapping NPI TON to Phone Context on page 170 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 324 Document LTRT 68808 SIP User s Manual 4 4 13 4 ini File Configuration Channel Parameters The channel related ini file configuration parameters are described in the table below The channel parameters define the DTMF fax and modem transfer modes Table 4 13 Channel ini File Parameters Parameter DJBufMinDelay DJBufOpitFactor FaxTransportMode FaxRelayEnhancedRedunda ncyDepth FaxRelayRedundancyDepth FaxRelayMaxRate FaxRelayECMEnable FaxModemBypassCoderType CNGDetectorMode FaxCNGMode FaxModemBypassM FaxModemNTEMode FaxBypassPayloadType CalleriDTransportType ModemBypassPayloadType FaxModemRelayVolume Version 5 6 Description For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 67
408. is defines the IP Group configured in the Configuring the IP Groups on page 201 from where the SIP INVITE message is received This IP Group can later be used in the Outbound IP Routing table and as the Serving IP Group in the Account table for obtaining authentication user name password for this call 3 4 7 4 6 Internal DNS Table The Internal DNS Table page similar to a DNS resolution is used to translate up to 20 host domain names into IP addresses e g when using the Tel to IP Routing table or Outbound IP Routing table if EnableSBC is enabled Up to four different IP addresses can be assigned to the same host name typically used for alternative routing for Tel to IP call routing The device initially attempts to resolve a domain name using the Internal DNS table If the domain name isn t listed in the table the device performs a DNS resolution using an external DNS server You can also configure the DNS table using the ini file table parameter DNS2IP refer to Networking Parameters on page 260 gt To configure the internal DNS table take these 6 steps 1 Open the Internal DNS Table page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Internal DNS Table page item Figure 3 75 Internal DNS Table Page Domain Name First IP Address Second IP Address Third IP Address Fourth IP Address 1 DomainName com 10 8 2 15 10 8 4 20 10 8 6 17 10 8 6 18
409. is number can later be used for manipulation and routing Determines whether NPI and TON are added to the Called Number for Tel to IP calls 0 No Do not change the Called Number default 1 Yes Add NPI and TON to the Called Number of ISDN Tel to IP call For example After receiving a Called Number of 555 NPI of 1 and TON of 3 the modified number becomes 13555 This number can later be used for manipulation and routing Determines whether the device removes the prefix from the destination number for IP to Tel calls 0 No Don t remove prefix default 1 Yes Remove the prefix defined in the IP to Trunk Group Routing table refer to IP to Trunk Group Routing on page 181 from a telephone number for an P to Tel call before forwarding it to Tel For example To route an incoming IP to Tel call with destination number 21100 the IP to Trunk Group Routing table is scanned for a matching prefix If such a prefix is found e g 21 then before the call is routed to the corresponding Trunk Group the prefix 21 is removed from the original number and therefore only 100 remains Notes Applicable only if number manipulation is performed after call routing for IP to Tel calls i e RouteModelP2Tel parameter is set to 0 Similar operation of removing the prefix is also achieved by using the usual number manipulation rules Determines the IP address that the device uses to determine
410. it Interface ID even if the B channel is on the same trunk as the D channel Note Applicable to 4 5ESS DMS and NI 2 variants 32768 ACCEPT MU LAW Mu Law is also accepted in ETSI 65536 EXPLICIT PRES SCREENING The calling party number octet 3a is always present even when presentation and screening are at their default Note Applicable only to ETSI NI 2 and 5ESS 131072 STATUS INCOMPATIBLE STATE Clears the call on receipt of Q 931 Status with incompatible state Otherwise no action is taken default 262144 STATUS ERROR CAUSE Clear call on receipt of STATUS according to cause value 524288 ACCEPT A LAW A Law is also accepted in 5ESS 2097152 RESTART INDICATION acEV_PSTN_RESTART_CONFIRM is generated on receipt of a RESTART message 4194304 FORCED RESTART On data link re initialization send RESTART if there is no call 1073741824 NS QSI ENCODE INTEGER If this bit is set INTEGER ASN 1 type is used in operator coding compliant to new ECMA standards otherwise OBJECT IDENTIFIER ASN 1 type is used Note Only applicable only to QSIG 90 Document LTRT 68808 SIP User s Manual ini File Field Name Web Parameter Name Outgoing Calls Behavior ISDNOutCallsBehavior Version 5 6 3 Web Based Management Valid Range and Description 2147483648 NS 5ESS NATIONAL Use the National mode of AT amp T 5ESS for B channel maintenance Note To configure the device to support several I
411. itching of modules SIP User s Manual 48 Document LTRT 68808 SIP User s Manual 3 Web Based Management Figure 3 30 Confirmation Message Box for Switching Modules ra a Microsoft Internet Explorer 2 j re you sure you want to switch modules 2 Click OK the Enter Network Password screen pertaining to the Web interface of the switched module appears 3 Enter the login user name and password and then click OK 3 3 9 Logging Off the Web Interface You can log off the Web interface and re access it with a different user account For detailed information on the Web User Accounts refer to User Accounts gt To log off the Web interface take these 2 steps 1 On the toolbar click the Log Off 7 button the Log Off confirmation message box appears Figure 3 31 Log Off Confirmation Box Microsoft Internet Explorer 2 j Logoff 2 Click OK the Web session is logged off and the Log In button appears Figure 3 32 Web Session Logged Off F http 10 13 4 13 HiddenPressLog0Off Microsoft Internet E EBR A File Edit View Favorites Tools Help Grk Or a amp Address El http 10 13 4 13 HiddenPressLogOff Web session is logged off Internet Version 5 6 49 November 2008 A Ee AudioCodes Mediant 2000 3 4 3 4 1 3 4 1 1 To log in again simply click the Log In button and then in the Enter Network Password dialog box enter your user name and password refer to Accessing the W
412. iversal CFU 408 Request Timeout Call Forward No Answer CFNA 480 Temporarily Unavailable 487 Request Terminated 486 Busy Here Call Forward Busy CFB 600 Busy Everywhere If history reason is a Q 850 reason it is translated to the SIP reason according to the SIP ISDN tables and then to ISDN Redirect reason according to the table above User Agent Server UAS Behavior The History Info header is sent only in the final response Upon receiving a request with History Info the UAS checks the policy in the request If session header or history policy tag is found the final response is sent without History Info otherwise it is copied from the request Determines the use of Tel Source Number and Display Name for Tel to IP calls 0 No Ifa Tel Display Name is received the Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name If no Display Name is received from the Tel side the IP Display Name remains empty default 1 Yes If a Tel Display Name is received the Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name If no Display Name is received from the Tel side the Tel Source Number is used as the IP Source Number and also as the IP Display Name 2 Overwrite The Tel Source Number is used as the IP Source Number and also as the IP Display Name even if the received Tel Disp
413. ks field and in the Trunk Settings page select the required Trunk number icon and then click Apply Trunk Seitings 5 Click Submit 6 Reset the device refer to Resetting the Device on page 228 It s strongly recommended that you don t modify the default values unless you fully understand the implications of the changes and know the default values Every change affects the configuration of the state machine parameters and the call process related to the trunk you are using with this state machine You can modify CAS state machine parameters only if the following conditions are met 1 Trunks are inactive stopped i e the Related Trunks field displays the trunk number in green 2 State machine is not in use or is in reset or when it is not related to any trunk If it is related to a trunk you must delete the trunk or de activate Stop the trunk Field values displaying 1 indicate CAS default values In other words CAS state machine values are used The modification of the CAS state machine occurs at the CAS application initialization only for non default values 1 For a detailed description of the CAS Protocol table refer to the Product Reference Manual Version 5 6 97 November 2008 ca AudioCodes Mediant 2000 Table 3 21 CAS State Machine Parameters Description Parameter Generate Digit On Time CasStateMachineGenerateDigitOnTime Generate Inter Digit Time CasStateMachineGenerate
414. l 4 If you selected E 164 Public as the NPI you can select Unknown 0 International 1 National 2 Network Specific 3 Subscriber 4 or Abbreviated 6 Phone Context The Phone Context SIP URI parameter 3 4 7 4 Configuring the Routing Tables The Routing Tables submenu allows you to configure the device s call routing This submenu includes the following page items Routing General Parameters refer to Routing General Parameters on page 171 Tel to IP Routing refer to Tel to IP Routing Table on page 175 Outbound IP Routing refer to Outbound IP Routing Table on page 178 IP to Trunk Group Routing refer to IP to Trunk Group Routing on page 181 Inbound IP Routing refer to Inbound IP Routing Table on page 184 Internal DNS Table refer to Internal DNS Table on page 186 Internal SRV Table refer to Internal SRV Table on page 187 Reasons for Alternative Routing refer to Reasons for Alternative Routing on page 188 Release Cause Mapping refer to Release Cause Mapping on page 189 3 4 7 4 1_ Routing General Parameters The Routing General Parameters page allows you to configure the device s IP to Tel and Tel to IP routing parameters gt 1 Version 5 6 To configure the general routing parameters take these 4 steps Open the Routing General Parameters page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Routing General Paramet
415. l IsVoiceOn 1 IsT38On 1 IsVbdOn 0 L 1 OpenChannel on Trunk 1 BChannel 1 CID 1 with Voici 1 OpenChanne l VoiceVolume 0 DTMFVolume 11 Inpu OpenChannel CoderType 15 Interval 4 N 1 1i FaxXTransportType 1 1 ContigFaxNodemChannelParams NSEMode 0 CNGDetBode Detectors Amd 0 Ans 0 En 0 BScmd Oxal 1 PS50SBoardinterface StopPlayTone Called recy lt OFF_HOOK Ch 1 1 OF F_HOOK_EV SiL OFF_HOOR_EV UpdateChanne Params Channel 1 1 ConfigFaxModerChannelParamns NSENode 0 CNGDetHode ActivateDigitMap for channel 1 MaxDialStringLength 242 Document LTRT 68808 SIP User s Manual 3 Web Based Management The displayed logged messages are color coded as follows e Yellow fatal error message e Blue recoverable error message i e non fatal error e Black notice message 3 To clear the page of Syslog messages in the Navigation tree click the page item Message Log again the page is cleared and new messages begin appearing gt To stop the Message Log take this step m Close the page by accessing any another page in the Web interface 3 6 1 2 Viewing the Ethernet Port Information The Ethernet Port Information page displays read only information on the Ethernet connection used by the device This includes indicating the active port duplex mode and speed You can also access this page from the Home page refer to Using the Home Page on page 46 For detailed information on the Eth
416. l only the coders common to both are used The order of the coders is determined by the preference 5 Configure the Profile s parameters according to your requirements For detailed information on each parameter refer to its description on the page in which it is configured as an individual parameter 6 From the Coder Group drop down list select the Coder Group refer to Coder Group Settings on page 190 or the device s default coder refer to Coders on page 144 to which you want to assign the Profile 7 Repeat steps 2 through 6 to configure additional Tel Profiles optional 8 Click the Submit button to save your changes 9 To save the changes to flash memory refer to Saving Configuration on page 230 3 4 7 5 3 IP Profile Settings The IP Profile Settings page allows you to define up to nine different IP Profiles You can then assign these IP Profiles to routing rules in the Tel to IP Routing page refer to Tel to IP Routing Table on page 175 or Outbound IP Routing Table if EnableSBC is set to 1 refer to Outbound IP Routing Table on page 178 and IP to Trunk Group Routing page refer to IP to Trunk Group Routing on page 181 or Inbound IP Routing Table if EnableSBC is set to 1 refer to Inbound IP Routing Table on page 184 IP Profiles can also be used when working with a Proxy server set AlwaysUseRouteTable to 1 Note You can also configure the IP Profiles using the ini file ta
417. l refer to the Product Reference Manual Note You can also configure the firewall settings using the ini file table parameter AccessList refer to Security Parameters on page 276 gt To add firewall rules take these 5 steps 1 Open the Firewall Settings page Configuration tab gt Security Settings menu gt Firewall Settings page item Figure 3 53 Firewall Settings Page Add Apply DeActivate Delete Duplicate Rule Is Rule Local Port Packet Active Source IP Subnet Mask Range Protocol Size Byte rate Action Match mogmt customer com 255 255 255 255 0 80 TCP 0 0 0 ALLOW O 192 0 0 0 255 0 0 0 0 65535 Any 0 40000 0000 BLOCK O0 o hosiaa 2557552550 Any o o o eos 4 O Yes 104 00 255 255 0 0 4000 9000 Any 0 0 0 BLOCK O0 2 In the Add field enter the index of the access rule that you want to add and then click Add a new firewall rule index appears in the table 3 Configure the firewall rule s parameters according to the table below 4 Click one of the following buttons e Apply saves the new rule without activating it e Duplicate Rule adds a new rule by copying a selected rule e Activate saves the new rule and activates it e Delete deletes the selected rule 5 To save the changes to flash memory refer to Saving Configuration on page 230 Version 5 6 103 November 2008 A ge AudioCodes Mediant 2000 gt To edit a rule take these 4 steps 1 In the Edit Rule column select
418. l RATPTxRx RFC 2833 TX Payload Type 96 RFC 2833 RX Payload Type 96 RFC 2198 Payload Type 104 Fax Bypass Payload Type 102 Enable RFC 3389 CN Payload Type Enable Analog Signal Transport Type Disable Remote RTP Base UDP Port 0 Remote RTP Base UDP Port 0 RTP Multiplexing Local UDP Port 0 l RTP Multiplexing Remote UDP Port 0 wv RTCP XR Settings Enable RTCP XR Disable Burst Threshold Delay Threshold R Value Delay Threshold Minimum Gap Size 2 Configure the RTP RTCP parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 SIP User s Manual 72 Document LTRT 68808 SIP User s Manual 3 Web Based Management Table 3 15 Media Settings RTP RTCP Parameters Parameter Dynamic Jitter Buffer Minimum Delay DJBufMinDelay Dynamic Jitter Buffer Optimization Factor DJBufOptFactor RTP Redundancy Depth RTPRedundancyDepth Packing Factor RTPPackingFactor Basic RTP Packet Interval BasicRTPPacketinterval RTP Directional Control RTPDirectionControl RFC 2833 TX Payload Type RFC2833TxPayloadType RFC 2833 RX Payload Type RFC2833RxPayloadType RFC 2198 Payload Type RFC2198PayloadType Fax Bypass Payload Type FaxBypassPayloadType Enable RFC 3389
419. l rights reserved This document is subject to change without notice Date Published November 18 2008 When viewing this manual on CD Web site or on any other electronic copy all cross references are hyperlinked Click on the page or section numbers shown in blue to reach the individual cross referenced item directly To return back to the point from where you accessed the cross reference press the ALT and keys Trademarks AC logo Ardito AudioCoded AudioCodes AudioCodes logo CTI CTI Squared InTouch IPmedia Mediant MediaPack MP MLQ NetCoder Netrake Nuera Open Solutions Network OSN Stretto 3GX TrunkPack VoicePacketizer VolPerfect What s Inside Matters Your Gateway To VoIP are trademarks or registered trademarks of AudioCodes Limited All other products or trademarks are property of their respective owners WEEE EU Directive Pursuant to the WEEE EU Directive electronic and electrical waste must not be disposed of with unsorted waste Please contact your local recycling authority for disposal of this product Customer Support Customer technical support and service are provided by AudioCodes Distributors Partners and Resellers from whom the product was purchased For Customer support for products purchased directly from AudioCodes contact support audiocodes com Abbreviations and Terminology Each abbreviation unless widely used is spelled out in full when first used Only industry standard te
420. lay Name is not empty 127 November 2008 ca AudioCodes Parameter Use Display Name as Source Number UseDisplayNameAsS ourceNumber Enable Contact Restriction EnableContactRestri ction Play Ringback Tone to IP PlayRBTone2IP Play Ringback Tone to Tel PlayRBTone2Tel SIP User s Manual Mediant 2000 Description Determines the use of Source Number and Display Name for IP to Tel calls 0 No If IP Display Name is received the IP Source Number is used as the Tel Source Number and the IP Display Name is used as the Tel Display Name If no Display Name is received from IP the Tel Display Name remains empty default 1 Yes If an IP Display Name is received it is used as the Tel Source Number and also as the Tel Display Name and Presentation is set to Allowed 0 If no Display Name is received from IP the IP Source Number is used as the Tel Source Number and Presentation is set to Restricted 1 For example When from 100 lt sip 200 201 202 203 204 gt is received the outgoing Source Number and Display Name are set to 100 and the Presentation is set to Allowed 0 When from lt sip 100 101 102 103 104 gt is received the outgoing Source Number is set to 100 and the Presentation is set to Restricted 1 Determines whether the device sets the Contact header of outgoing INVITE requests to anonymous for restricted calls 0 Disabled default 1 Enabled Dete
421. lay all the menus By default the Basic option is selected Figure 3 5 Navigation Tree in Basic and Full View Status as Status Contiguration Management 3 Diagnostics Contiguration Management 3 Diagnostics Scenarios Search Scenarios Search Basic O Full O Basic Full P Network Settings network Settings Full Navigation EMedia Settings PMedia Settings s Tree View H Protocol Configuration t Security Setting f Option P advance Applications t protocol Configuration ine H Advance Applications Only Basic Menus All Menus Note When in Scenario mode refer to Scenarios on page 34 the Navigation tree is displayed in Full view i e all menus are displayed in the Navigation tree SIP User s Manual 24 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 3 2 2 Showing Hiding the Navigation Pane The Navigation pane can be hidden to provide more space for elements displayed in the Work pane This is especially useful when the Work pane displays a page with a table that s wider than the Work pane and to view the all the columns you need to use scroll bars The arrow button located just below the Navigation bar is used to hide and show the Navigation pane E To hide the Navigation pane click the left pointing arrow S the pane is hidden and the button is replaced by the right pointing arrow button m To show the Navigation pane click the right pointing arrow the pane is di
422. le or Outbound IP Routing table if EnableSBC is set to 1 The port is the same port as the local RTP port configured by the parameter BaseUDPPort and the channel on which the call is received 1 Not Configured Use the per device parameter RTPOnlyMode value default 0 Disable Disabled 1 Transmit amp Receive send and receive RTP packets 2 Transmit Only send RTP packets only 3 Receive Only receive RTP packets only Note The D in the ini file parameter depicts the trunk number where 0 is the first trunk 95 November 2008 A Ee AudioCodes Mediant 2000 ini File Field Name Web Parameter Name Digital Out Of Service Behavior DigitalOOSBehavior Transfer Mode TrunkTransferMode Enable TBCT TrunkTransferMode Enable RLT TrunkTransferMode Enable Single Step Transfer TrunkTransferMode Enable ECT TrunkTransferMode SIP User s Manual Valid Range and Description Determines the method for setting digital trunks to Out Of Service state per device 0 Default Uses default behavior for each trunk see note below default 1 Service Sends ISDN In or Out of Service only for ISDN protocols that support Service message 2 D Channel Takes D Channel down or up ISDN only 3 Alarm Sends or clears PSTN AIS Alarm ISDN and CAS 4 Block Blocks trunk CAS only Notes The default behavior value 0 is as follows ISDN Use Ser
423. le Parameters PEPE T aah cas meh E AN NEE I eee NE T 274 Table 4 4 Securty ini File Parameters oncesinde eein ine meee 276 Table 4 5 RADIUS ini File Parameters iiccsccsxecssssescsesesestersanesetunsessddeesaveseivareisvtevaadeivnnavennssesmeeke Table 4 6 SNMP ini File Parameters 282 Tables SIP mi File Pardmelai S aeniea aan a adenine 284 Table 4 8 Media Server ini File Parameters Table 4 9 Voice Mail ini File Parameters i 30 Table 4 10 PSTN ini File E N E E N A O on Table 4 11 ISDN and CAS Interworking Related ini File Parameters EENE A O E IA Table 4 12 Number Manipulation and Routing ini File Parameters reiii tee Table 4 13 Channelini File Parameters vc eusutandaessasaiseeeaean a a aa ni a Table 4 14 Auxiliary Configuration ini File Parameters 0 cccccecceeeeeeeeeeeeeeeeeeeeeeeeeeeeseeeeeeeeeaees 331 Table 8212 User riionrriattary WEIS sien access verses as viecuvqaigeiesianieaiss dei 340 Table 7 1 Approximate AMD Detection aa on North American aN Aimtinensauns 344 Table 7 2 Supported X Detect Event Meat AA T PT EE er AT EE S Eo Table 7 3 Supported RADIUS Attributes E ETIE EA E ANE AAA E A E RE OU Table 7A Sports ODR Fie Senasa aiara a n aiaa ia 366 Table 8 1 Traffic Network Types and Priority dni breath O EA TE A ENA E E AT 386 Table 8 2 Example of VLAN and Multiple IPs Configuration ccccceeeeeeeeeeeeeeeeeeeeeeeeeeneeeeeeeaees 387 Table 9 1 Mappi
424. le used by IPSec to encrypt the IP stream Stand Alone Survivability Feature Session Description Protocol Session Initiation Protocol Simplified Message Desk Interface Small and Medium sized Enterprise Simple Network Management Protocol Secure Real Time Transport Protocol Service Record Secure Shell Secure Socket Layer also known as Transport Layer Security TLS Simple Traversal of UDP through NATs 1 544 Mbps USA Digital Transmission System see E1 and DS1 Transmission Control Protocol Transmission Control Protocol Internet Protocol Terminal Equipment ISDN Time Division Multiplexing Trivial File Transfer Protocol Transport Layer Security Type of Numbering SIP User Agent User Datagram Protocol SIP Uniform Resource Indicators Voice band data Virtual Local Area Network Voice over Internet Protocol Voice over Packet s Voice Prompts File Virtual Private Network A companding algorithm used in the digital telecommunication systems 417 November 2008 7a wi AudioCodes Mediant Media Gateways SIP Mediant 2000 User s Manual Version 5 6 7 VT lar wt AudioCodes KY www audiocodes com
425. les on page 48 view port settings refer to Viewing Trunk Settings on page 48 and assign a name to a port refer to Assigning a Name or Brief Description to a Port on page 47 5 Dual Ethernet Link icons gray No link green Active link You can also view detailed Ethernet port information in the Ethernet Port Information page refer to Viewing the Active Alarms Table on page 245 by clicking this icon 6 Dual Ethernet activity icons gray No Ethernet activity orange Transmit receive activity T1 E1 Trunk Status icons for trunks 9 through 16 Refer to Item 4 for a description Power status icon green Power received by blade red No power received by blade 9 Slot status of installed blade in the chassis SWAP Ready icon 3 3 8 1 Assigning a Name to a Port The Home page allows you to assign an arbitrary name or a brief description to each port This description appears as a tooltip when you move your mouse over the port gt To add a port description take these 3 steps 1 Click the required port icon a shortcut menu appears as shown below Figure 3 27 Shortcut Menu for Assigning a Port Name 2 o0000000 O wo amp Bs T 0o O Port Settings Update Port Info Version 5 6 47 November 2008 A Ee AudioCodes Mediant 2000 2 From the shortcut menu choose Update Port Info a text box appears Figure 3 28 Entering the
426. les in the table Version 5 6 181 November 2008 A Ee AudioCodes Mediant 2000 Note The IP to Trunk Group Routing Table page appears only if the parameter EnableSBC is set to 0 default in SBC Configuration on page 163 If this parameter is enabled the Inbound IP Routing Table page appears instead refer to Inbound IP Routing Table on page 184 for a description of this page The IP to Tel calls are routed to Trunk Groups according to any one of the following or a combination thereof criteria m Destination and source host prefix m Destination and source phone prefix m Source IP address Once the call is routed to the specific Trunk Group the call is sent to the device s channels pertaining to that Trunk Group The specific channel within the Trunk Group to which the call is sent is determined according to the Trunk Group s channel selection mode This channel selection mode can be defined per Trunk Group refer to Configuring the Trunk Group Settings on page 197 or for all Trunk Groups using the global parameter ChannelSelectMode refer to SIP General Parameters on page 121 When a call release reason defined in Reasons for Alternative Routing on page 188 is received for a specific IP to Tel call an alternative Trunk Group for that call can be configured This is performed by assigning the call to an additional routing rule in the table i e repeat the same routing rule but with a different Trunk
427. lete and then click OK at the prompt If no IKE methods are defined Encryption Authentication DH Group the default settings shown in the following table are applied Table 3 28 Default IKE First Phase Proposals Proposal Encryption Proposal 0 3DES Proposal 1 3DES Proposal 2 3DES Proposal 3 3DES SIP User s Manual Authentication SHA1 MD5 SHA1 MD5 118 DH Group 1024 1024 786 786 Document LTRT 68808 SIP User s Manual 3 Web Based Management The parameters described in the following table are used to configure the first phase main mode of the IKE negotiation for a specific peer A different set of parameters can be configured for each of the 20 available peers Table 3 29 IKE Table Configuration Parameters Parameter Name Description Authentication Method Determines the authentication method for IKE IkePolicyAuthenticationMetho 0 Pre shared Key default d 1 RSA Signature Notes For pre shared key authentication peers participating in an IKE exchange must have a prior out of band knowledge of the common key see IKEPolicySharedKey parameter For RSA signature authentication peers must be loaded with a certificate signed by a common CA For additional information on certificates refer to Server Certificate Replacement on page 105 Shared Key Determines the pre shared key in textual format Both peers IKEPolicySharedKey must register the same pre shared key for the
428. ling on page 398 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Supplementary Services on page 159 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Supplementary Services on page 159 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 MLPP normalized service domain string If the device receives an MLPP ISDN incoming call it uses the parameter if different from FFFFFF as a Service domain in the SIP Resource Priority header in outgoing INVITE messages If the parameter is FFFFFF the Resource Priority header is set to the MLPP Service Domain obtained from the Precedence IE The valid value is a 6 hexadecimal digits The default is 000000 Note This parameter is applicable only to device s using the MLPP NI 2 ISDN variant with CallPriorityMode set to 1 MLPP default service domain string If the device receives a non MLPP ISDN incoming call without a Precedence IE it uses the parameter as a Service domain in the SIP Resource Priority header in outgoing Tel to IP calls INVITE messages This parameter is used in conjunction with the
429. lls For IP to IP calls this parameter is not required i e leave the field empty To denote any Trunk Group leave this field empty Note For defining Trunk Groups refer to Configuring the Trunk Group Table on page 195 Represents a called telephone number prefix The prefix can be 1 to 19 digits long An asterisk represents all numbers Represents a calling telephone number prefix The prefix can be 1 to 19 digits long An asterisk represents all numbers Outbound IP calls matching all or any combination of the above routing rules are subsequently sent to the destination IP address or IP Group defined below Notes For alternative routing additional entries of the same prefixes can be configured For notations representing multiple numbers refer to Dialing Plan Notation on page 168 Dest IP Address PREFIX_DestAddress Port PREFIX_DestPort SIP User s Manual The destination IP address to where the outbound call is sent Domain names e g domain com can be used instead of IP addresses Notes If you select a destination IP Group in the Dest IP Group ID field below then the IP address you define in this Dest IP Address field is not used for routing and therefore not required When using domain names you must enter a DNS server IP address or alternatively define these names in the Internal DNS Table refer to Internal DNS Table on page 186 To discard outgoing I
430. lnterDigitTime DTMF Max Detection Time CasStateMachineDTMFMaxOnDetectionTime DTMF Min Detection Time CasStateMachineDTMFMinOnDetectionTime MAX Incoming Address Digits CasStateMachineMaxNumOflncomingAddressDigi ts MAX Incoming ANI Digits CasStateMachineMaxNumOflincomingANIDigits Collet ANI CasStateMachineCollectANl Digit Signaling System CasStateMachineDigitSignalingSystem Description Generates digit on time in msec The value must be a positive value The default value is 1 Generates digit off time in msec The value must be a positive value The default value is 1 Detects digit maximum on time according to DSP detection information event in msec units The value must be a positive value The default value is 1 Detects digit minimum on time according to DSP detection information event in msec units The digit time length must be longer than this value to receive a detection Any number may be used but the value must be less than CasStateMachineDTMFMaxOnDetectionTi me The value must be a positive value The default value is 1 Defines the limitation for the maximum address digits that need to be collected After reaching this number of digits the collection of address digits is stopped The value must be an integer The default value is 1 Defines the limitation for the maximum ANI digits that need to be collected After reaching this number of digits the collection of ANI
431. local routing enter the device s IP address When the device s IP address is unknown e g when DHCP is used enter the IP address 127 0 0 1 177 November 2008 A ge AudioCodes Mediant 2000 Parameter Port PREFIX_DestPort Transport Type PREFIX_TransportType Dest IP Group ID PREFIX_DestIPGroupID IP Profile ID PREFIX_Profileld Status Description When using domain names you must enter a DNS server IP address or alternatively define these names in the Internal DNS Table refer to Internal DNS Table on page 186 The destination port to where you want to route the Tel to IP call The transport layer type for sending the Tel to IP calls 1 Not Configured 0 UDP 1 TCP 2 TLS Note When Not Configured is selected the transport type defined by the parameter SIPTransportType refer to SIP General Parameters on page 121 is used The IP Group 1 9 to where you want to route the Tel to IP call The SIP INVITE messages are sent to the IP address es of the Proxy Set that is associated with the selected IP Group If you select an IP Group it is unnecessary to configure a destination IP address in the Dest IP Address field However if both parameters are configured the INVITE message is sent only to the IP Group If the parameter AlwaysUseRouteTable is set to 1 in the IP Group table refer to Configuring the IP Groups on page 201 the request URI host name in t
432. ls Max Call Duration min MaxCallDuration Enable LAN Watchdog EnableLanWatchDog Enable User Information Usage EnableUserlInfoUsage First Call Ringback Tone ID FirstCallRBTId SIP User s Manual Mediant 2000 Description Release Reason Mapping on page 394 Fora list of SIP responses Q 931 release cause mapping refer to Release Cause Mapping on page 189 Defines the time interval in seconds that the device s operation is delayed after a reset The valid range is 0 to 45 The default value is 7 seconds Note This feature helps to overcome connection problems caused by some LAN routers or IP configuration parameters modifications by a DHCP server Defines the maximum number of simultaneous active calls supported by the device If the maximum number of calls is reached new calls are not established The default value is the maximum available channels no restriction on the maximum number of calls The valid range is 1 to 240 Defines the maximum call duration in minutes If this time expires both sides of the call are released IP and Tel The valid range is 0 to 35 791 The default is 0 i e no limitation Determines whether the LAN Watch Dog feature is enabled 0 Disable Disable LAN Watch Dog default 1 Enable Enable LAN Watch Dog When LAN Watch Dog is enabled the device s overall communication integrity is checked periodically If no communication for about 3 minutes is
433. ls mechanism is disabled Schedules an automatic update to a predefined time of the day The range is HH MM 24 hour format For example 20 18 Note The actual update time is randomized by five minutes to reduce the load on the Web servers Invokes an immediate restart of the device This option can be used to activate offline i e not on the fly parameters that are loaded via IniFileUrl 0 The immediate restart mechanism is disabled default 1 The device immediately restarts after an ini file with this parameter set to 1 is loaded 271 November 2008 A Ee AudioCodes Mediant 2000 Parameter BootP and TFTP Parameters Description The BootP parameters are special Hidden parameters Once defined and saved in the flash memory they are used even if they don t appear in the ini file BootPRetries BootPSelectiveEnable BootPDelay SIP User s Manual Note This parameter only takes effect from the next reset of the device This parameter is used to Set the number of BootP requests the Set the number of DHCP device sends during start up The packets the device sends device stops sending BootP requests After all packets were sent when either BootP reply is received or if there s still no reply the number of retries is reached device loads from flash 1 1 BootP retry 1 sec 1 4 DHCP packets 2 2 BootP retries 3 sec 2 5 DHCP packets 3 3 BootP retries 6 sec
434. m length of the entire digit pattern is 152 characters Available notations n m Range of numbers not letters single dot Repeat digits until next notation e g T x Any single digit T Dial timeout configured by the parameter TimeBetweenDigits Immediately applies a specific rule that is part of a general rule For example if your digit map includes a general rule x T and a specific rule 11x for the specific rule to take precedence over the general rule append S to the specific rule i e 11xS An example of a digit map is shown below 11xS OOT 1 7 Xxx 8xxxXxXXx XXXXXXX Xx 9 1XXXXXXXXXX 901 1x T In the example above the last rule can apply to International numbers 9 for dialing tone 011 Country Code and then any number of digits for the local number x Note For PRI interfaces the digitmap mechanism is applicable only when ISDN Overlap dialing is used ISDNRxOverlap is set to 1 Duration in seconds that the dial tone is played to an ISDN terminal This parameter is applicable for overlap dialing when ISDNInCallsBehavior 65536 The dial tone is played if the ISDN SETUP message doesn t include the called number The valid range is 0 to 60 The default is 5 150 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Description Default Destination Number Defines the default destination phone number used if the received DefaultNumber messag
435. m the Rate drop down list select the bit rate in kbps for the selected coder SIP User s Manual 146 Document LTRT 68808 SIP User s Manual 3 Web Based Management 5 In the Payload Type field if the payload type for the selected coder is dynamic enter a value from 0 to 120 payload types of well known coders cannot be modified The payload type identifies the format of the RTP payload 6 From the Silence Suppression drop down list enable or disable the silence suppression option for the selected coder 7 Repeat steps 2 through 6 for the second to fifth optional coders 8 Click the Submit button to save your changes 9 To save the changes to flash memory refer to Saving Configuration on page 230 Each coder i e Coder Name can appear only once If packetization time and or rate are not specified the default value is applied Only the packetization time of the first coder in the coder list is declared in INVITE 200 OK SDP even if multiple coders are defined For G 729 it s also possible to select silence suppression without adaptations If the coder G 729 is selected and silence suppression is disabled for this coder the device includes the string annexb no in the SDP of the relevant SIP messages If silence suppression is enabled or set to Enable w o Adaptations annexb yes is included An exception to this logic is when the remote gateway is a Cisco device IsCisco
436. mber this number is overridden by the new called number if this parameter is set to 1 or 2 This parameter can also be configured for IP Profiles refer to IP Profile Settings on page 193 Determines whether Calling Party Category CPC is mapped between SIP and PRI 0 Disable Don t relay the CPC between SIP and PRI default 1 Enable The CPC is relayed between SIP and PRI If enabled the CPC received in the Originating Line Information OLI IE of an incoming ISDN SETUP message is relayed to the From P Asserted Identity headers using the cpc parameter in the outgoing INVITE message and vice versa For example calling party is a payphone From lt sip 2000 cpc payphone 10 8 23 70 gt tag 1c1806157451 Note This feature is supported only when using the NI 2 PRI variant Determines the method for setting digital trunks to Out Of Service state per device 0 Default Uses default behavior for each trunk see note below default 1 Service Sends ISDN In or Out of Service only for ISDN protocols that support Service message 2 D Channel Takes D Channel down or up ISDN only 3 Alarm Sends or clears PSTN AIS Alarm ISDN and CAS 4 Block Blocks trunk CAS only Notes The default behavior value 0 is as follows ISDN Use Service messages on supporting variants and use Alarm on non supporting variants CAS Use Alarm When updating this parameter value a
437. me TxDTMFHangOverTime DTMFTransportType RFC2833PayloadType R1DetectionStandard UserDefinedToneDetectorEn able UDTDetectorFrequencyDevia tion CPTDetectorFrequencyDevia tion Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 Determines the number of spectral coefficients added to an SID packet being sent according to RFC 3389 Valid only if EnableStandardS IDPayloadType is set to 1 The valid values are 0 default 4 6 8 and 10 For a description of this parameter refer to Configuring the Voice Settings on page 66 For a description of this parameter refer to Configuring the Voice Settings on page 66 Time in msec between generated DTMF digits to PSTN side if TxDTMFOption 1 2 or 3 The default value is 100 msec The valid range is 0 to 32767 Time in msec for generating DTMF tones to the PSTN side if TxDTMFOption 1 2 or 3 It also configures the duration that is sent in INFO Cisco messages The valid range is 0 to 32767 The default value is 100 Defines the Voice Silence time in msec units after playing DTMF or MF digits to the Tel PSTN side that arrive as Relay from the IP side Valid range is 0 to 2 000 msec The default is 1 000 msec Defines the Voice Silence time in msec
438. me 4 NAS IP Address 6 Service Type H323 26 Incoming Conf Id H323 26 Remote Address H323 Conf 26 ID 26 H323 Setu p Time 26 H323 Cal l Origin 26 H323 Call Type H323 26 Connect Time 26 H323 Version 5 6 Table 7 3 Supported RADIUS Attributes VSA No 23 24 25 26 27 28 29 Purpose Account number or calling party number or blank IP address of the requesting device Type of service requested SIP call identifier IP address of the remote gateway H 323 SIP call identifier Setup time in NTP format 1 The call s originator Answering IP or Originator PSTN Protocol type or family used on this leg of the call Connect time in NTP format Disconnect time in NTP 363 Value Format String up to 15 digits long Numeric Numeric Up to 32 octets Numeric Up to 32 octets String String String String String Example 5421385747 192 168 14 43 1 login Answer Originate etc VoIP 7 IP Telephony Capabilities AAA Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Stop Acc Stop November 2008 A Ee AudioCodes Mediant 2000 Attribute Attribute VSA Value 1 Number Name No Purpose Format Sxaniple on Disconnect format Acc Time H323 Q 931
439. me usa_tones dat Coder Table File Name codertable test dat Internet 10 Click End Process to close the wizard and then in the Enter Network Password dialog box enter your login user name and password described in Accessing the Web Interface on page 20 and click OK a message box appears informing you of the new CMP file Figure 3 107 Message Box Informing of Upgraded CMP File The board detected a new label 5 404 000 4 new CMP has been loaded into the device 11 Click OK the Web interface now becomes active and reflecting the upgraded device 3 5 2 4 Backing Up and Restoring Configuration The Configuration File page allows you to save a copy of the device s current configuration file modifications as an ini file to a PC This is useful for backing up your configuration to protect your device configuration The saved ini file includes only those parameters that were modified as well as parameters with other than default values In addition this page allows you to load an ini file to the device If the device has lost its configuration you can restore the device s configuration by loading the previously saved ini file or by simply loading a newly created ini file SIP User s Manual 240 Document LTRT 68808 SIP User s Manual 3 Web Based Management gt To save and restore the ini file take these 3 steps 1 Open the Configuration File page Management tab gt Software Update menu gt Co
440. meter is not required in the routing rule leave the field empty To denote any Trunk Group you can enter the asterisk symbol Represents a called telephone number prefix The prefix can be 1 to 19 digits long An asterisk represents all numbers Represents a calling telephone number prefix The prefix can be 1 to 19 digits long An asterisk represents all numbers All Tel calls matching all or any combination of the above routing rules are subsequently sent to the destination IP address defined below Notes For alternative routing additional entries of the same prefixes can be configured For notations representing multiple numbers refer to Dialing Plan Notation on page 168 Dest IP Address PREFIX_DestAddress Version 5 6 The destination IP address in dotted decimal notation to where these calls must be sent Domain names e g domain com can be used instead of IP addresses Notes If you select a destination IP Group in the Dest IP Group ID field below then the IP address you define in this Dest IP Address field is not used for routing and therefore not required To discard outgoing IP calls of a specific Tel to IP routing rule enter 0 0 0 0 For example if you want to prohibit dialing of international calls then in the Dest Phone Prefix field enter 00 and in the Dest IP Address field enter 0 0 0 0 For routing calls between phones connected to the device i e
441. method type is REGISTER e Using SIP protocol sip e Proxy IP from ini file is 10 2 2 222 e The equation to be evaluated is REGISTER sip 10 2 2 222 e The MD5 algorithm is run on this equation and stored for future usage e The result is a9a031cfddcb10d91c8e7b4926086f7e SIP User s Manual 372 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 6 Final stage e The A1 result The nonce from the proxy response is 11432d6bce58ddf02e3b5e1c77c010d2 e The A2 result The equation to be evaluated is A1 11432d6bce58ddf02e3b5e1c77c010d2 A2 e The MD65 algorithm is run on this equation The outcome of the calculation is the response needed by the device to register with the Proxy e The response is b9c45d0234a5abf5ddf5c704029b38c At this time a new REGISTER request is issued with the following response REGHS PERISH OOM 22222 SIP 2 0 Via SIP 2 0 UDP 10 1 1 200 From lt Sip 122 10 1 1 200 gt tag 1c23940 TOSSI p I22 10 1 i1 200 Call ID 654982194 10 1 1 200 Server Audiocodes Sip Gateway Mediant 2000 v 5 40 010 006 CSeq 1 REGISTER Comicacies sujosl22 iL 1 1 200 Expires 3600 Authorization Digest username 122 realm audiocodes com nonce 11432d6bce58ddf02e3b5e1c77c010d2 UiEt 10 4 2 ALA response b9c45d0234a5abf 5dd 5c704029b3 8c 7 Upon receiving this request and if accepted by the Proxy the proxy returns a 200 O
442. mmonly implemented when there is no response to an INVITE message after INVITE retransmissions The device then issues an internal 408 No Response implicit release reason If this reason is included in the Reasons for Alternative Routing table the device immediately initiates a call to the redundant destination using the next matched entry in the Tel to IP Routing table Note that if a domain name in this table is resolved into two IP addresses the timeout for INVITE retransmissions can be reduced by using the parameter Number of RTX Before Hotswap If the alternative routing destination is the device itself the call can be configured to be routed back to the PSTN This feature is referred to as PSTN Fallback meaning that if poor voice quality occurs over the IP network the call is routed through the legacy telephony system PSTN Tel to IP routing can be performed before or after applying the number manipulation rules To control when number manipulation is performed use the Tel to IP Routing Mode or RouteModeTel2IP ini file parameter described in the table below You can also configure the Tel to IP Routing table using the ini file table parameter Prefix refer to Number Manipulation and Routing Parameters on page 313 gt To configure the Tel to IP Routing table take these 5 steps 1 Open the Tel to IP Routing page Configuration tab gt Protocol Configuration menu gt Routing Tables
443. mon parameters used in both IP and Tel profiles TelPreference determines the priority of the Profile 1 to 20 where 20 is the highest preference If both IP and Tel profiles apply to the same call the coders and other common parameters indicated with an asterisk of the preferred Profile are applied to that call If the preference of the Tel and IP profiles is identical the Tel Profile parameters are applied For example TelProfile TelProfile 1 FaxProfile 1 1 1 40 13 22 33 0 0 0 1 0 0 0 TelProfile 2 ModemProfile 2 2 0 40 13 5 0 0 0 0 5 TelProfile Notes This parameter can appear up to 9 times i e indices 1 9 Two adjacent dollar signs indicates that the parameter s default value is used The TelProfile index can be used in the Trunk Group table TrunkGroup parameter The Profile Name assigned to a Profile index must enable users to identify it intuitively and easily To configure the Tel Profile table using the Web interface refer to Tel Profile Settings on page 192 Fora description of using ini file table parameters refer to Structure of ini File Table Parameters on page 257 SIP User s Manual 300 Document LTRT 68808 SIP User s Manual 4 4 8 4 ini File Configuration Media Server Parameters The media processing related ini file configuration parameters are described in the table below Tabl
444. mum length of the URL address is 255 characters Specifies the name of the ini file and the location of the server IP address or FQDN from which the device loads the ini file The ini file can be loaded using HTTP HTTPS FTP FTPS or NFS For example http 192 168 0 1 filename http 192 8 77 13 config lt MAC gt https lt username gt lt password gt lt IP address gt lt file name gt Notes When using HTTP or HTTPS the date and time of the ini file are validated Only more recently dated ini files are loaded The optional string lt MAC gt is replaced with the device s MAC address Therefore the device requests an ini file name that contains its MAC address This option enables loading different configurations for specific devices 270 Document LTRT 68808 SIP User s Manual Parameter PrtFileURL CptFileURL VpFileURL CasFileURL TLSRootFileUrl TLSCertFileUrl UserInfoFileURL AutoUpdateCmpFile AutoUpdateFrequency AutoUpdatePredefinedTim e ResetNow Version 5 6 4 ini File Configuration Description The maximum length of the URL address is 99 characters Specifies the name of the Prerecorded Tones file and the location of the server IP address or FQDN from which it is loaded For example http server_name file https server_nameffile Note The maximum length of the URL address is 99 characters Specifies the name of the CPT file and the location of the server
445. n Digit Mapping Rules DigitMapping Dial Tone Duration sec TimeForDialTone SIP User s Manual Description The RFC 2833 DTMF relay dynamic payload type The valid range is 96 to 99 and 106 to 127 The default is 96 The 100 102 to 105 range is allocated for proprietary usage Notes Certain vendors e g Cisco use payload type 101 for RFC 2833 When RFC 2833 payload type PT negotiation is used the parameter TxDTMFOption is set to 4 this payload type is used for the received DTMF packets If negotiation isn t used this payload type is used for receive and for transmit Determines the supported hook flash Transport Type i e method by which hook flash is sent and received 0 Not Supported Hook Flash indication isn t sent default 1 INFO Send proprietary INFO message with Hook Flash indication 4 RFC 2833 5 INFO Lucent Send proprietary INFO message with Hook Flash indication Notes The RFC 2833 4 option is currently not supported The DTMF HookFlashCode is send to IP according to the parameter HookFlashOption Digit map pattern used to reduce the dialing period when Overlap dialing is used If the digit string i e dialed number matches one of the patterns in the digit map the device stops collecting digits and establishes a call with the collected number The digit map pattern can contain up to 52 options each separated by a vertical bar The maximu
446. n access the device Trusted managers can work with all community strings The port to which the keep alive traps are sent The valid range is 0 65534 The default is port 162 When enabled this parameter invokes the keep alive trap and sends it every 9 10 of the time defined in the parameter defining NAT Binding Default Timeout 0 Disable 1 Enable Defines the base product system OID Default is eSNMP_AC_PRODUCT_BASE_OID_D Defines a Trap Enterprise OID Default is eSNMP_AC_ENTERPRISE_OID The inner shift of the trap in the AcTrap subtree is added to the end of the OID in this parameter Defines the description of the input alarm Defines the severity of the input alarm Determines the maximum number of rows in the Alarm History table The parameter can be controlled by the Config Global Entry Limit MIB located in the Notification Log MIB The valid range is 50 to 1000 The default value is 500 282 Document LTRT 68808 SIP User s Manual Parameter SNMP Trap Parameters SNMPManagerTablelP_x SNMPManagerTrapPort_x SNMPManagerTrapUser_x SNMPManagerlsUsed_x SNMPManagerTrapSendingE nable_x SNMPTrapManagerHostNam e 4 ini File Configuration Description For a description of this parameter refer to Configuring the SNMP Managers Table on page 222 For a description of this parameter refer to Configuring the SNMP Managers Table on page 222 This parameter can be set to the
447. n Parameters on page 120 SIP Advanced Parameters refer to Configuring the SIP Advanced Parameters on page 151 Manipulation Tables refer to Configuring the Number Manipulation Tables on page 164 Routing Tables refer to Configuring the Routing Tables on page 171 Profile Definitions refer to Configuring the Profile Definitions on page 190 Trunk IP Group refer to Configuring the Trunk and IP Groups on page 195 Digital Gateway refer to Configuring the Digital Gateway Parameters on page 207 3 4 7 1 Configuring the Protocol Definition Parameters The Protocol Definition submenu allows you to configure the main SIP protocol parameters This submenu contains the following page items SIP General Parameters refer to SIP General Parameters on page 121 Proxy amp Registration refer to Proxy amp Registration Parameters on page 132 Proxy Sets Table refer to Proxy Sets Table on page 141 Coders refer to Coders on page 144 DTMF amp Dialing refer to DTMF amp Dialing Parameters on page 147 SIP User s Manual 120 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 7 1 1 SIP General Parameters The SIP General Parameters page is used to configure general SIP parameters gt To configure the general SIP protocol parameters take these 4 steps 1 Open the SIP General Parameters page Configuration tab gt Protocol Configuration menu gt Protocol Definition subme
448. n create a Scenario Version 5 6 35 November 2008 A ge AudioCodes Mediant 2000 3 3 5 2 Accessing a Scenario Once you have created the Scenario you can access it at anytime by following the procedure below gt To access the Scenario take these 2 steps 1 On the Navigation bar select the Scenario tab a message box appears requesting you to confirm the loading of the Scenario Figure 3 16 Scenario Loading Message Box Microsoft Internet Explorer dd Loading Scenario PBX Interoperability 2 Click OK the Scenario and its Steps appear in the Navigation tree as shown in the example figure below Figure 3 17 Scenario Example Contiguraten Management Poua ai Seanas Search OBasic Orult Scenario Max Digits In Phone Num Scenario Name PBX Inter Digit Timeout for Overlap Dialing sec SRST Declare RFC 2633 in SOP i ine Coders z C oefe Mas as LAER 3 Definie Voice Mail tep 3rd Tx OTHF Option 4th Tx OTMF Option Sth Tx DTMF Option RFC 2833 Payload Type Hook Flash Option Digit Mapping Rules Dial Tome Duration sec Hotline Dial Tone Duration sec Enable Special Digits Default Destination Number Special Digit Representation Butions n x L Na for Scenario Pages When you select a Scenario Step the corresponding page is displayed in the Work pane In each page the available parameters are indicated by a dark blue background the unavailable parameters are indicated by a gray
449. n factor to 13 and to enable echo cancellation for fax and disable it for modem To configure fax modem transparent mode use the following parameters IsFaxUsed 0 FaxTransportMode 0 V21ModemTransportType 0 V22ModemtTransportType 0 V23ModemTransportType 0 V32ModemTransportType 0 V34ModemTransportType 0 BellModemTransportType 0 Additional configuration parameters e CoderName e DJBufOptFactor e EnableSilenceCompression EnableEchoCanceller Note This mode can be used for fax but is not recommended for modem transmission Instead use the modes Bypass refer to Fax Modem Bypass Mode on page 352 or Transparent with Events refer to Fax Modem Transparent with Events Mode on page 355 for modem SIP User s Manual 354 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 7 5 2 5 7 5 2 6 Fax Modem Transparent with Events Mode In this mode fax and modem signals are transferred using the current voice coder with the following automatic adaptations Echo Canceller on or off for modems m Echo Canceller Non Linear Processor Mode off m Jitter buffering optimizations To configure fax modem transparent with events mode perform the following configurations IsFaxUsed 0 FaxTransportMode 3 V21ModemTransportType 3 V22ModemtTransportType 3 V23ModemtTransportType 3 V32ModemtTransportType 3 V34ModemtTransportType 3 BellModemTransportType 3
450. n seconds between SIP OPTIONS Keep Alive messages used for the IP Connectivity application The valid range is 5 to 2 000 000 The default value is 60 Packet loss percentage at which the IP connection is considered a failure and Alternative Routing mechanism is activated The range is 1 to 20 The default value is 20 174 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Description Max Allowed Delay for Alt Routing Transmission delay in msec at which the IP connection is msec considered a failure and Alternative Routing mechanism is IPConnQoSMaxAllowedDelay activated The range is 100 to 1000 The default value is 250 3 4 7 4 2 Tel to IP Routing Table The Tel to IP Routing page provides a table for configuring up to up to 50 routing rules for Tel to IP calls where Tel calls are routed to destinations based on IP address or IP Group The Tel to IP Routing page appears only if the parameter EnableSBC is set to 0 default in SBC Configuration on page 163 If this parameter is enabled the Outbound IP Routing Table page appears instead refer to Outbound IP Routing Table on page 178 for a description of this page This routing table associates called and or calling telephone number prefixes originating from a specific Trunk Group with a destination IP address or Fully Qualified Domain Name FQDN or IP Group When a call is routed by the device i e a Proxy serve
451. n this section are only used as an example Since a default Gateway is available only for the Media network for the device to be able to communicate with an external device network on its OAMP and Control networks IP routing rules must be used 8 9 3 1 Integrating Using the Web Interface The procedure below describes how to integrate the device into a multiple IPs network withVLANs using the Web interface gt To integrate the device into a multiple IPs network withVLANs using the Web interface take these 6 steps 1 Access the Web interface refer to Accessing the Web Interface on page 20 2 Use the Software Upgrade Wizard refer to Software Upgrade Wizard on page 236 to load and burn the firmware version to the device VLANs and multiple IPs support is available only when the firmware is burned to flash 3 Configure the VLAN parameters by completing the following steps a Open the IP Settings page refer to Configuring the IP Settings on page 50 b Modify the VLAN parameters to correspond to the values shown in the following figure Figure 8 3 VLAN Configuration in the IP Settings Page vw kd VLAN Mode Enable w VLAN ID Settings Native VLAN ID OAM VLAN ID Control VLAN ID Media VLAN ID c Click the Submit button to save your changes 4 Configure the multiple IP parameters by completing the following steps SIP User s Manual 388 Document LTRT 68808 SIP Us
452. n working with certain application servers Such as BroadSoft s BroadWorks in client server mode the application server controls all supplementary services and keypad features by itself the device s supplementary services must be disabled Call Hold and Retrieve Hold and Retrieve The party that initiates the hold is called the holding party the other party is called the held party The device can t initiate Call Hold but it can respond to hold requests and as such it s a help party After a successful Hold the holding party hears a Dial tone HELD_TONE defined in the device s Call Progress Tones file After a successful retrieve the voice is connected again The hold and retrieve functionalities are implemented by Re INVITE messages The IP address 0 0 0 0 as the connection IP address or the string a inactive in the received Re INVITE SDP cause the device to enter Hold state and to play the Held tone configured in the device to the PBX PSTN If the string a sendonly is received in the SDP message the device stops sending RTP packets but continues to listen to the incoming RTP packets Usually the remote party plays in this scenario Music on Hold MOH and the device forwards the MOH to the held party You can also configure the device to keep a call on hold for a user defined time after which the call is disconnected using the ini file parameter HeldTimeout refer to Supplementary Services on page 159
453. nagement Configuration menu allows you to configure the device s management parameters This menu contains the following page items m Management Settings refer to Configuring the Management Settings on page 220 m Regional settings refer to Configuring the Regional Settings on page 227 m Maintenance Actions refer to Maintenance Actions on page 228 3 5 1 1 Configuring the Management Settings The Management Settings page allows you to configure the device s management parameters gt To configure the Management parameters take these 4 steps 1 Open the Management Settings page Management tab gt Management Configuration menu gt Management Settings page item Figure 3 90 Management Settings Page wv Syslog Settings Syslog Server IP Address 10 33 2 20 Syslog Server Port 514 Enable Syslog Enable Trunks Filter E v SNMP Settings SNMP Trap Destinations SNMP Community String SNMP V3 Table SNMP Trusted Managers Disable SNMP N Trap Manager Host Name wv Activity Types to Report via Activity Log Messages Parameters Value Change Auxiliary Files Loading s Device Reset Flash Memory Burning Device Software Update Access to Restricted Domains Non Authorized Access Sensitive Parameters Value Change 2 Configure the Management Settings according to the table below SIP User s Manual 220 Document LTRT 68808 SIP User s Manual 3 Web Based Management
454. name or password SPC Sensitive Parameters Value Change Changes made to sensitive parameters 1 IP Address 2 Subnet Mask 3 Default Gateway IP Address 268 Document LTRT 68808 SIP User s Manual Parameter ECHybridLoss GwDebugLevel CDRReportLevel CDRSyslogServerIP HeartBeatDestIP HeartBeatDestPort HeartBeatintervalmsec EnableRAIl RAIlHighThreshold RAILowThreshold Version 5 6 4 ini File Configuration Description 4 ActivityListToLog For example ActivityListToLog pvc afl dr fb swu ard naa spc Sets the four wire to two wire worst case Hybrid loss the ratio between the signal level sent to the hybrid and the echo level returning from the hybrid 0 6 dB default 1 N A 2 0dB 3 3dB For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Advanced Parameters on page 151 Destination IP address in dotted format notation to which the device sends proprietary UDP ping packets The default IP address is 0 0 0 0 Destination UDP port to which the heartbeat packets are sent The range is 0 to 64000 The default is 0 Delay in msec between consecutive heartbeat packets 10 100000 1 disabled default 0 Disable RAI Resource Available Ind
455. nce Period FarEndDisconnectSilence Method FarEndDisconnectSilence Threshold For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Advanced Parameters on page 151 Threshold of the packet count in percentages below which is considered silence by the device The valid range is 1 to 100 The default is 8 Note Applicable only if silence is detected according to packet count FarEndDisconnectSilenceMethod 1 Automatic Update Parameters CmpFileURL IniFileURL SIP User s Manual Specifies the name of the cmp file and the location of the server IP address or FQDN from which the device loads a new cmp file and updates itself The cmp file can be loaded using HTTP HTTPS FTP FTPS or NFS For example http 192 168 0 1 filename Notes When this parameter is set in the ini file the device always loads the cmp file after it is reset The cmp file is validated before it s burned to flash The checksum of the cmp file is also compared to the previously burnt checksum to avoid unnecessary resets The maxi
456. ndex entry 6 is re assigned index number 2 Figure 3 12 Compacting a Web Interface Table PrefixLength Gateway VianID InterfaceName m Ouplicate Compact Index ApplicationTypes IPv6InterfaceMode IPAddress PrefixLength Gateway VianIO InterfaceName gt To delete an existing index table entry take these 3 steps 1 In the Index column select the index corresponding to the table row that you want to delete 2 Click Delete the table row is removed from the table 3 3 4 Searching for Configuration Parameters The Web interface provides a search engine that allows you to search any ini file parameter that is configurable by the Web interface i e has a corresponding Web parameter You can search for a specific parameter e g EnablelPSec or a sub string of that parameter e g sec If you search for a sub string all parameters that contain the searched sub string in their names are listed SIP User s Manual 32 Document LTRT 68808 SIP User s Manual 3 Web Based Management gt To search for ini file parameters configurable in the Web interface 1 2 3 4 Version 5 6 take these 4 steps On the Navigation bar click the Search tab the Search engine appears in the Navigation pane In the Search field enter the parameter name or sub string of the parameter name that you want to search If you have performed a previous search for such a parameter instead of entering t
457. nel number in the Trunk Group is always first selected and if that channel is unavailable the next lowest channel is selected 5 Dest Number Cyclic Ascending The channel is first selected according to the called number If the called number isn t found the next available channel in ascending cyclic order is selected Note that if the called number is found but the channel associated with the number is busy the call is released 6 By Source Phone Number Selects the channel according to the calling number 7 Trunk Cyclic Ascending The first channel of the next Trunk i e next to the Trunk from which the previous channel was allocated is selected Registration mode per Trunk Group 1 Per Gateway Single registration for the entire device default This mode is applicable only if a default Proxy or Registrar IP are configured and Registration is enabled i e parameter IsRegisterUsed is set to 1 In this mode the URI userpart in the From To and Contact headers is set to the value of the global registration parameter GWRegistrationName refer to Proxy amp Registration Parameters on page 132 or username if GWRegistrationName is not configured 0 Per Endpoint Each channel in the Trunk Group registers individually The registrations are sent to the ServinglPGroupID if defined in the table otherwise to the default Proxy and if no default Proxy then to the Registrar IP 4 Don t Register No registrat
458. nelHeader Enable Busy Out EnableBusyOut Default Release Cause DefaultReleaseCause Version 5 6 3 Web Based Management Description it sends a 183 Session Progress message with SDP in response to only the first received ISDN Proceeding or Progress message after a call is placed to PBX PSTN over the trunk Determines whether the x channel header is added to SIP messages for trunk B channel information 0 Disable x channel header is not used default 1 Enable x channel header is generated with trunk B channel and IP address information The header provides information on the E1 T1 physical trunk B channel on which the call is received or placed For example x channel DS DS1 5 22 IP 192 168 13 1 where DS DS 1 is a constant string 5 is the trunk number 22 is the B channel and in addition the device s IP address is added to the header This header is generated by the device and is sent in INVITE messages and 183 180 2000K responses Determines whether the Busy Out feature is enabled 0 Disable Busy out feature is not used default 1 Enable Busy out feature is enabled When Busy Out is enabled and certain scenarios exist the device performs the following All E1 T1 trunks are automatically taken out of service by taking down the D Channel or by sending a Service Out message for T1 PRI trunks supporting these messages NI 2 4 5 ESS DMS 100 and Meridian These behaviors ar
459. nes whether Silence Indicator SID packets are sent according to RFC 3389 0 Disable G 711 SID packets are sent in a proprietary method default 1 Enable SID comfort noise packets are sent with the RTP SID payload type according to RFC 3389 Applicable to G 711 and G 726 coders Enables negotiation and usage of Comfort Noise CN 0 Disable Disable default 1 Enable Enable The use of CN is indicated by including a payload type for CN 73 November 2008 ca AudioCodes Parameter RTP Base UDP Port BaseUDPPort Remote RTP Base UDP Port RemoteBaseUDPPort RTP Multiplexing Local UDP Port L1L1ComplexTxUDPPort RTP Multiplexing Remote UDP Port L1L1ComplexRxUDPPort SIP User s Manual Mediant 2000 Description on the media description line of the SDP The device can use CN with a codec whose RTP timestamp clock rate is 8 000 Hz G 711 G 726 The static payload type 13 is used The use of CN is negotiated between sides Therefore if the remote side doesn t support CN it is not used Note Silence Suppression must be enabled to generate CN Lower boundary of UDP port used for RTP RTCP RTP port 1 and T 38 RTP port 2 The upper boundary is the Base UDP Port 10 number of device s channels The range of possible UDP ports is 6 000 to 64 000 The default base UDP port is 6000 For example If the Base UDP Port is set to 6000 default then 1 The first channel uses
460. nfiguration page allows you to configure the device s Stand Alone Survivability SAS feature This feature is useful for providing a local backup via the PSTN in Small or Medium Enterprises SME that are serviced by IP Centrex services In such environments the enterprise s incoming and outgoing telephone calls external and internal are controlled by the Proxy which communicates with the enterprise through the WAN interface SAS ensures that incoming outgoing and internal calls service is maintained in case of a WAN or Proxy failure using a PSTN or an alternate VoIP backup connection and the device s built in internal routing To utilize the SAS feature the VolP CPEs such as IP phones or residential gateways need to be defined so that their Proxy and Registrar destination addresses and UDP port equal the SAS feature s IP address and SAS local SIP UDP port The SAS Configuration page is Feature Key dependant and therefore is available only if included in the device s Feature Key refer to Upgrading the Software Upgrade Key on page 233 For a detailed explanation on SAS and for configuring various SAS setups refer to Stand Alone Survivability SAS Feature on page 346 For additional SAS parameters configurable only using the ini file refer to SIP Configuration Parameters on page 284 Version 5 6 161 November 2008 A ge AudioCodes Mediant 2000 gt To configure the Stand Alone Survivability parameters take th
461. nfiguration File Figure 3 108 Configuration File Page Save the INI file to the PC Save INI File Load the INI file to the device Browse _ Load INI File The device will perform a reset after sending the INI file 2 To save the ini file to a PC perform the following a Click the Save INI File button the File Download dialog box opens b Click the Save button navigate to the folder in which you want to save the ini file on your PC and then click Save the device copies the ini file to the selected folder 3 To load an ini file to the device perform the following a Click the Browse button navigate to the folder in which the ini file is located select the file and then click Open the name and path of the file appear in the field beside the Browse button b Click the Load INI File button and then at the prompt click OK the device uploads the ini file and then resets from the cmp version stored on the flash memory Once complete the Enter Network Password dialog box appears requesting you to enter your user name and password 3 6 Status amp Diagnostics Tab The Status amp Diagnostics tab on the Navigation bar displays all menus related to the operating status of the device and device diagnostics These menus appear in the Navigation tree and include the following m Status amp Diagnostics refer to Status amp Diagnostics on page 242 m Gateway Statistics refer to Gateway
462. nfigure the fax Modem and CID parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 14 Media Settings Fax Modem CID Parameters Parameter Fax Transport Mode FaxTransportMode SIP User s Manual Description Fax transport mode used by the device 0 Disable transparent mode 1 T 38 Relay default 2 Bypass 3 Events Only Note This parameter is overridden by the parameter IsFaxUsed refer to SIP General Parameters on page 68 Document LTRT 68808 SIP User s Manual Parameter Caller ID Transport Type CallerIDTransportType V 21 Modem Transport Type V21ModemTransportType V 22 Modem Transport Type V22ModemTransportType V 23 Modem Transport Type V23ModemTransportType V 32 Modem Transport Type V32ModemTransportType V 34 Modem Transport Type V34ModemTransportType Version 5 6 3 Web Based Management Description 121 If the parameter IsFaxUsed is set to 1 T 38 Relay or 3 Fax Fallback then FaxTransportMode is always set to 1 T 38 relay Determines the device s behavior for Caller ID detection 0 Disable Caller ID is not detected DTMF digits remain in the voice stream 1 Relay Caller ID is detected DTMF digits are erased from the voice stream 3 Mute Caller ID is detected DTMF digits are erased
463. ng Parameters on page 147 E MGCPDTMFDetectionPoint DTMFVolume DTMFTransportType DTMFDigitLength and DTMFinterDigitInterval refer to Channel Parameters on page 324 7 5 Fax and Modem Capabilities 7 5 1 Fax Modem Operating Modes The device supports two modes of operations mM Fax modem negotiation that isn t performed during the establishment of the call E VBD mode for V 152 implementation refer to Supporting V 152 Implementation on page 357 fax modem capabilities are negotiated between the device and the remote endpoint at the establishment of the call During a call when a fax modem signal is detected transition from voice to VBD or T 38 is automatically performed and no additional SIP signaling is required If negotiation fails i e no match is achieved for any of the transport capabilities fallback to existing logic occurs according to the parameter IsFaxUsed SIP User s Manual 350 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 7 5 2 Fax Modem Transport Modes The device supports the following transport modes for fax per modem type V 22 V 23 Bell V 32 V 34 m T 38 fax relay refer to Fax Relay Mode on page 351 m Fax and modem bypass a proprietary method that uses a high bit rate coder refer to Fax Modem Bypass Mode on page 352 m NSE Cisco s Pass through bypass mode for fax and modem refer to Fax Modem NSE Mode on page 353 m Transparent passing the fax mo
464. ng device automatically initiates SIP calls from all enabled B channels belonging to the E1 T1 J1 spans that are configured with the protocol type Transparent for ISDN trunks or Raw CAS for CAS trunks The called number of each call is the internal phone number of the B channel from where the call originates The IP to Trunk Group Routing table refer to IP to Trunk Group Routing on page 181 is used to define the destination IP address of the terminating device The terminating device automatically answers these calls if its E1 T1 protocol type is set to Transparent ProtocolType 5 or Raw CAS ProtocolType 3 for T1 and 9 for E1 and the parameter ChannelSelectMode is set to 0 By Phone Number It s possible to configure both devices to also operate in symmetric mode To do so set EnableTDMOverlP to 1 and configure the Tel to IP Routing tables in both devices In this mode each device after it s reset initiates calls to the second device The first call for each B channel is answered by the second device The device continuously monitors the established connections If for some reason one or more calls are released the device automatically re establishes these broken connections In addition when a failure in a physical trunk or in the IP network occurs the device re establishes the tunneling connections when the network is restored Note It s recommended to use the keep alive mechanism f
465. ng numbers that start with 551 Numbering Plans and Type of Number Numbers are classified by their Numbering Plan Indication NPI and their Type of Number TON The device supports all NPI TON classifications used in the standard The list of ISDN ETSI NPI TON values is shown in the following table Table 3 41 NPI TON Values for ISDN ETSI NPI TON Description Unknown 0 Unknown 0 A valid classification but one that has no information about the numbering plan E 164 Public Unknown 0 A public number in E 164 format but no information 1 on what kind of E 164 number International 1 A public number in complete international E 164 format e g 16135551234 National 2 A public number in complete national E 164 format e g 6135551234 Subscriber 4 A public number in complete E 164 format representing a local subscriber e g 5551234 Private 9 Unknown 0 A private number but with no further information about the numbering plan Level 2 Regional 1 Level 1 Regional 2 A private number with a location e g 3932200 PISN Specific 3 Level 0 Regional local 4 A private local extension number e g 2200 For NI 2 and DMS 100 ISDN variants the valid combinations of TON and NPI for calling and called numbers include Plan Type m 0 0 Unknown Unknown m 1 1 International number in ISDN Telephony numbering plan m 1 2 National number in ISDN Telephony numbering plan m 1 4 Subscriber local number in ISDN Telepho
466. ng of ISDN Release Reason to SIP RESPONSE cccccccssccsssessscessseeceseeecseecseeeaee 394 Table 9 2 Mapping of SIP Response to ISDN Release Reason c cccccsccessesssecseecseeceeeecseeeseeeaes 396 Table 9 32 Calling Name DISplay ta rcsasctcsessccoacadiscesned inen anaa bi iiA aE 401 Table 9 4 Redirect Number EPEE E E E E E EE E caveman Table 11 1 Supplied Software Package ees j Table 12 1 Mediant 2000 Functional al Specifications EE EA A EREA E E AA E A T ain 409 Table 13 1 Glossary of Terms er rere a e aa NG SIP User s Manual 12 Document LTRT 68808 SIP User s Manual Notices Notice This document describes the AudioCodes Mediant 2000 SIP gateway Information contained in this document is believed to be accurate and reliable at the time of printing However due to ongoing product improvements and revisions AudioCodes cannot guarantee accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions Before consulting this document check the corresponding Release Notes regarding feature preconditions and or specific support in this release In cases where there are discrepancies between this document and the Release Notes the information in the Release Notes supersedes that in this document Updates to this document and other documents can be viewed by registered customers at http www audiocodes com support Copyright 2008 AudioCodes Lid Al
467. ng to the parameters DTMFTransportType and RFC2833PayloadType default 1 INFO Nortel Sends DTMF digits according to IETF lt draft choudhuri sip info digit O0 gt 2 NOTIFY Sends DTMF digits according to lt draft mahy sipping signaled digits 01 gt 3 INFO Cisco Sends DTMF digits according to Cisco format 4 RFC 2833 5 INFO Korea Sends DTMF digits according to Korea Telecom format Notes DTMF negotiation methods are prioritized according to the order of their appearance When out of band DTMF transfer is used 1 2 3 or 5 the parameter DTMFTransportType is automatically set to 0 DTMF digits are erased from the RTP stream When RFC 2833 4 is selected the device 1 Negotiates RFC 2833 Payload Type PT using local and remote SDPs 2 Sends DTMF packets using RFC 2833 PT according to the PT in the received SDP 3 Expects to receive RFC 2833 packets with the same PT as configured by the parameter RFC2833PayloadType 4 Sends DTMF digits in transparent mode as part of the voice stream When TxDTMF Option is set to 0 the RFC 2833 PT is set according to the parameter RFC2833PayloadType for both transmit and receive The ini file table parameter TxDTMFOption can be repeated 5 times for configuring the DTMF transmit methods Version 5 6 149 November 2008 A ge AudioCodes Mediant 2000 Parameter RFC 2833 Payload Type RFC2833PayloadType Hook Flash Option HookFlashOptio
468. ngs on page 78 DSP Templates refer to Configuring the DSP Templates on page 79 Media Security refer to Configuring Media Security on page 80 Channel parameters can be modified on the fly Changes take effect from the next call Some channel parameters can be configured per channel or call routing using profiles refer to Configuring the Profile Definitions on page 190 Version 5 6 65 November 2008 A C al AudioCodes Mediant 2000 3 4 2 1 Configuring the Voice Settings The Voice Settings page is used for configuring various voice parameters such as voice volume gt To configure the Voice parameters take these 4 steps 1 Open the Voice Settings page Configuration tab gt Media Settings menu gt Voice Settings page item Figure 3 40 Voice Settings Page Voice Volume 32 to 31 dB 0 Input Gain 32 to 31 dB 0 Silence Suppression Disable DTMF Transport Type RFC2833 Relay DTMF MF Transport Type RFC2833 Relay MF DTMF Volume 31 to 0 dB 11 NTE Max Duration J CAS Transport Type CASEventsOnly l DTMF Generation Twist 0 Echo Canceller Enable 2 Configure the Voice parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 13 Media Settings Voice Settings Parameters Parameter Description Voice Volume
469. ngthy operation during this time the device will be out of service After the operation complete save configuration and reset the dence Certificate Files Send Server Certificate file from your computer to the device Browse Send File Send Trusted Root Certificate Store file from your computer to the device Browse Send file Send Private Key file from your computer to the device Browse Send file Note Replacing the private key is not recommended but if it s done it should be over a physically secure network link 3 In the Subject Name field enter the DNS name and then click Generate CSR A textual certificate signing request that contains the SSL device identifier is displayed 4 Copy this text and send it to your security provider The security provider also known as Certification Authority or CA signs this request and then sends you a server certificate for the device 5 Save the certificate to a file e g cert txt Ensure that the file is a plain text file containing the BEGIN CERTIFICATE header as shown in the example of a Base64 Encoded X 509 Certificate below BEGIN CERTIFICATE MI IDkzCCAnugAwI BAgI EAGAAADANBgkqhkiG9w0 BAQQFADA MQswCQYDVQQGEwJGUj ETMBEGA1UEChMKQ2VydGlwb3N0ZTEbMBkGA1UEAxXMSQ2VydGlwb3N0ZSBTZXJ2ZXVy MB4XDTk4MDYyNDA4MDAWMFOXDTE4MDYyNDA4MDAWMFowPZELMAkKGA1UEBhMCRIIXEZ ARBGNVBAOTCKN1 cnRpcG9 zdGUxGzZAZBgNVBAMTEKN1 cnRpcG9 zdGUgU2VydmV1cjCC ASEwDQYJKoZ IhvcNAQEBBQ
470. nly mode parameters can t be modified In addition the following pages can t be accessed Web User Accounts Certificates Regional Settings Maintenance Actions and all file loading pages Load Auxiliary Files Software Upgrade Wizard and Configuration File Note To return to read write after you have applied read only using this parameter set to 1 you need to reboot your device with an ini file that doesn t include this parameter using the BootP TFTP Server utility refer to the Product Reference Manual HTTP port used for Web management default is 80 Defines the file name of the Scenario file to be loaded to the device The file name must have the dat extension and can be up to 47 characters For loading a Scenario using the Web interface refer to Loading a Scenario to the Device on page 39 For a description of this parameter refer to Configuring the Application Settings on page 57 For a description of this parameter refer to Configuring the Application Settings on page 57 For a description of this parameter refer to Configuring the Application Settings on page 57 For a description of this parameter refer to Configuring the Application Settings on page 57 For a description of this parameter refer to Configuring the Application Settings on page 57 Customizing the Web Appearance Parameters For detailed information on customizing the Web interface interface
471. nstead of creating a Scenario you can load a Scenario file data file from your PC to the device gt To load a Scenario to the device take these 4 steps 1 Onthe Navigation bar click the Scenarios tab the Scenario appears in the Navigation tree 2 Click the Get Send Scenario File button located at the bottom of the Navigation tree the Scenario File page appears refer to Saving a Scenario to a PC on page 38 3 Click the Browse button and then navigate to the Scenario file stored on your PC 4 Click the Send File button Version 5 6 You can only load a Scenario file to a device that has an identical hardware configuration setup to the device in which it was created For example if the Scenario was created in a device with FXS interfaces the Scenario cannot be loaded to a device that does not have FXS interfaces The loaded Scenario replaces any existing Scenario You can also load a Scenario file using BootP by loading an ini file that contains the ini file parameter ScenarioFileName refer to Web and Telnet Parameters on page 273 The Scenario dat file must be located in the same folder as the ini file For a detailed description on BootP refer to the Product Reference Manual 39 November 2008 A Ee AudioCodes Mediant 2000 3 3 5 6 Deleting a Scenario You can delete the Scenario by using the Delete Scenario File button as described in the procedure below gt To delete the Scenario take these
472. nsure that you enable VLANs using the VLAN Mode VIANMode parameter When booting using BootP DHCP protocols refer to the Product Reference Manual an IP address is obtained from the server This address is used as the OAMP address for this session overriding the IP address you configured in the Multiple Interface Table page The address specified in this table takes effect only after you save the configuration to the device s flash memory This enables the device to use a temporary IP address for initial management and configuration while retaining the address defined in this table for deployment For a detailed description on multiple IP interfaces and VLANs refer to VLANS and Multiple IPs on page 384 For a description of the Web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 30 Table 3 8 Multiple Interface Table Parameters Description Parameter Description Table parameters Index of each interface Index The range is 0 3 Note Each interface index must be unique Types of applications that are allowed on the specific interface 0 OAM Only Operations Administration Maintenance and Provisioning OAMP applications e g Web Telnet SSH and SNMP are allowed on the interface 1 Media Only Media i e RTP streams of voice video is allowed on the interface 2 Control Only Call Control applications e g SIP are allowe
473. nt to the default Proxy or according to the Outbound IP Routing table Determines whether Survivability mode is enabled for USER type IP Groups Disable default Enable Survivability mode is enabled The device records in its local database the registration messages sent by the clients belonging to the USER type IP Group If communication with the Serving IP Group e g IP PBX fails the USER type IP Group enters into Survivability mode in which the device uses its database for routing calls between the clients e g IP phones of the USER type IP Group The RTP packets between the IP phones in Survivability mode always traverse through the device In Survivability mode the device is capable of receiving new registrations When the Serving IP Group is available again the device returns to normal mode sending INVITE and REGISTER messages to the Serving IP Group Notes This field is available only if EnableSBC is set to 1 refer to SBC Configuration on page 163 This parameter is applicable only to USER type IP Groups Defines the routing mode for outgoing SIP INVITE messages 0 Not Configured The routing is done according to the selected Serving IP Group If no Serving IP Group is selected the device routes the call according to the Outbound IP Routing table refer to Outbound IP Routing Table on page 178 1 Routing Table The device routes the call according to the Outbound IP
474. ntact ra Sao OO GOs Siar Ss X Detect Request CPT FAX SIP 2 0 200 OK Via SIP 2 0 UDP 10 33 2 53 branch z9hG4bKac5906 From anonymous lt sip anonymous anonymous invalid gt tag 1c25298 To lt sip 101 10 33 2 53 user phone gt tag 1c19282 Calil mp3 LiIG23 10 33 2 53 CSeq 1 INVITE Contact lt sip 101 10 33 2 53 gt X Detect Response CPT FAX INFO sip 101 10 33 2 53 user phone SIP 2 0 Via SIP 2 0 UDP 10 33 2 53 branch z9hG4bKac5906 Max Forwards 70 From anonymous lt sip anonymous anonymous invalid gt tag 1c25298 To lt Sip 101 10 33 2 53 user phone gt Call 1De II923 10 33 2 53 CSeq 1 INVITE Contact lt sip 100 10 33 2 53 gt X Detect Response CPT FAX Content Type Application xX Detect Content Length xxx Type CPT Subtype SIT Version 5 6 359 November 2008 7 7 A Ee AudioCodes Mediant 2000 RTP Multiplexing ThroughPacket The device supports a proprietary method to aggregate RTP streams from several channels to reduce the bandwidth overhead caused by the attached Ethernet IP UDP and RTP headers and to reduce the packet data transmission rate This option reduces the load on network routers and can typically save 50 e g for G 723 on IP bandwidth RTP Multiplexing ThroughPacket is accomplished by aggregating payloads from several channels that are sent to the same destination IP address into a single IP packet RTP multiplexing can be applied to the entire device refer
475. nterface refer to Backing Up and Restoring Configuration on page 240 2 Open the ini file using a text file editor such as Microsoft Notepad and then modify the ini file parameters according to your requirements 3 Save the modified ini file and then close the file 4 Load the modified ini file to the device using either the BootP TFTP utility or the Web interface refer to Backing Up and Restoring Configuration on page 240 Tip Before loading the ini file to the device verify that the file extension of the ini file saved on your PC is correct i e ini Version 5 6 259 November 2008 A ge AudioCodes Mediant 2000 4 4 Reference for ini File Parameters This subsection lists all the ini file parameters References to their descriptions in the Web interface are provided except for those ini file parameters that can only be configured using the ini file 4 4 1 Networking Parameters The networking related ini file configuration parameters are described in the table below Parameter EthernetPhyConfiguration DHCPEnable DHCPSpeedFactor EnableDHCPLeaseRenew al EnableLANWatchDog DNSPriServerIP DNSSecServerlP SIP User s Manual Table 4 1 Networking ini File Parameters Description Defines the Ethernet connection mode type 0 10Base T half duplex 1 10Base T full duplex 2 100Base TX half duplex 3 100Base TX full duplex 4 Auto negotiate default For d
476. nu gt SIP General Parameters page item Figure 3 59 SIP General Parameters Page Version 5 6 v SIP General PRACK Mode Channel Select Mode Enable Early Media 183 Message Behavior Session Expires Time Minimum Session Expires Session Expires Method Asserted Identity Mode Fax Signaling Method Detect Fax on Answer Tone SIP Transport Type SIP UDP Local Port SIP TCP Local Port SIP TLS Local Port Enable SIPS Enable TCP Connection Reuse TCP Timeout SIP Destination Port Use user phone in SIP URL Use user phone in From Header Use Tel URI for Asserted Identity Tel to IP No Answer Timeout Enable Remote Party ID Add Number Plan and Type to RPI Header Enable History Info Header Use Source Number as Display Name Use Display Name as Source Number Enable Contact Restriction Play Ringback Tone to IP Play Ringback Tone to Tel Use Tarp information Enable GRUU User Agent Information SDP Session Owner Play Busy Tone to Tel Subject Multiple Packetization Time Format Enable Semi Attended Transfer 3xx Behavior Enable P Charging Vector Enable VoiceMail URI Retry After Time Enable P Associated URI Header Source Number Preference Forking Handling Mode Enable Reason Header Supported Cyclic Ascending Disable Alert 0 30 Re INVITE Disabled No Fax Initiate T 38 on Preamble UDP 5060 5060
477. number before routing and manipulations An entry of FAX as destination number in the Tel to IP Routing table is then used to route the call and the destination number 155 November 2008 A Ee AudioCodes Mediant 2000 Parameter CDR and Debug CDR Server IP Address CDRSyslogServerlIP CDR Report Level CDRReportLevel Debug Level GwDebugLevel Misc Parameters Progress Indicator to IP ProgresslIndicator2IP SIP User s Manual Description manipulation mechanism is used to remove the FAX prefix if required If the initial INVITE used to establish the voice call not fax was already sent a CANCEL if not connected yet or a BYE if already connected is sent to tear down the voice call Notes To enable this feature set CNGDetectorMode to 2 and IsFaxUsed to 1 2 or 3 The FAX prefix in routing and manipulation tables is case sensitive Defines the destination IP address to where CDR logs are sent The default value is a null string which causes CDR messages to be sent with all Syslog messages to the Syslog server Note The CDR messages are sent to UDP port 514 default Syslog port Determines whether Call Detail Records CDR are sent to the Syslog server and when they are sent 0 None CDRs are not used default 1 End Call CDR is sent to the Syslog server at the end of each call 2 Start amp End Call CDR report is sent to Syslog at the start and end of each
478. ny numbering plan m 9 4 Subscriber local number in Private numbering plan Version 5 6 169 November 2008 A Ee AudioCodes Mediant 2000 3 4 7 3 3 Mapping NPI TON to Phone Context The Phone Context Table page is used to map NPI and TON to the Phone Context SIP parameter When a call is received from the ISDN the NPI and TON are compared against the table and the Phone Context value is used in the outgoing SIP INVITE message The same mapping occurs when an INVITE with a Phone Context attribute is received The Phone Context parameter appears in the standard SIP headers where a phone number is used Request URI To From Diversion gt To configure the Phone Context tables take these 4 steps 1 Open the Phone Context Table page Configuration tab gt Protocol Configuration menu gt Manipulation Tables submenu gt Phone Context Table page item Figure 3 69 Phone Context Table Page v dd Phone Context As Prefix Enable v Phone Context Index 1 10 v NPI TON Phone Context 1 Unknown vi Unknown v unkn own com 2 Private vjl Level 2 Regional v host com g 3 E164 Public vi National v na e164 hostcom ell a a 2 Configure the Phone Context table according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Several rows with the same NPI TON or Phone Context
479. o 5 indices For defining the Reasons for Alternative Routing table using the Web interface refer to Reasons for Alternative Routing on page 188 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 AltRouteCauselP2Tel This ini file table parameter configures call failure reason values received from the PSTN side in Q 931 presentation If a call is released as a result of one of these reasons the device attempts to locate an alternative Trunk Group for the call in the IP to Trunk Group Routing table The format of this parameter is as follows AltRouteCauselP2Tel FORMAT AltRouteCauselP2Tel_Index SIP User s Manual 322 Document LTRT 68808 SIP User s Manual Parameter EnableETS Diversion CopyDest2RedirectNumber FilterCalls2IP Alternative Routing Parameters RedundantRoutingMode AltRoutingTel2IPEnable AltRoutingTel2IPMode AltRoutingTel2IPConnMethod AltRoutingTel2IPKeepAliveTim e IPConnQoSMaxAllowedPL IPConnQoSMaxAllowedDelay Version 5 6 4 ini File Configuration Description AltRouteCauselP2Tel_ ReleaseCause AltRouteCauselP2Tel For example AltRouteCauselP2Tel AltRouteCauselP2Tel 0 3 No Route to Destination AltRouteCauselP2Tel 1 1 Unallocated Number AltRouteCauselP2Tel 2 17 Busy Here AltRouteCauselP2Tel Notes This parameter can include up to 5 indices If the device fails to establish a c
480. o Tel calls an alternative Trunk Group is provided Refer to Tel to IP Routing Table on page 175 for information on defining an alternative IP address refer to IP to Trunk Group Routing on page 181 for information on defining an alternative Trunk Group You can use the Reasons for Alternative Routing page for the following example scenarios Tel to IP calls when there is no response to an INVITE message after INVITE retransmissions the device issues an internal 408 No Response implicit release reason m IP to Tel calls when the destination is busy and release reason 17 is issued or for other call releases that issue the default release reason 3 Refer to DefaultReleaseCause in Advanced Parameters on page 151 The reasons for alternative routing for Tel to IP calls only apply when a Proxy isn t used For Tel to IP calls the device sends the call to an alternative route only after the call has failed and the device has subsequently attempted twice to establish the call unsuccessfully You can also configure alternative routing using the ini file table parameters AltRouteCauseTel2IP and AltRouteCauselP2Tel refer to Number Manipulation and Routing Parameters on page 313 gt To configure the reasons for alternative routing take these 5 steps 1 Open the Reasons for Alternative Routing page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Reasons for Alternati
481. o redundant routing is used If the call can t be completed using the main route using the active Proxy or the first matching rule in the internal routing table the call is disconnected 1 Routing Table Internal routing table is used to locate a redundant route default 2 Proxy Proxy list is used to locate a redundant route Determines the routing mode after a call redirection i e a 3xx SIP response is received or transfer i e a SIP REFER request is received 0 Standard INVITE messages that are generated as a result of Transfer or Redirect are sent directly to the URI according to the Refer To header in the REFER message or Contact header in the 3xx response default 1 Proxy Sends a new INVITE to the Proxy Note Applicable only if a Proxy server is used and the parameter AlwaysSendtoProxy is set to 0 135 November 2008 ca AudioCodes Parameter Mediant 2000 Description 2 Routing Table Uses the Routing table to locate the destination and then sends a new INVITE to this destination Notes When this parameter is set to 1 and the INVITE sent to the Proxy fails the device re routes the call according to the Standard mode 0 When this parameter is set to 2 and the INVITE fails the device re routes the call according to the Standard mode 0 If DNS resolution fails the device attempts to route the call to the Proxy If routing to the Proxy also fails the Redirect
482. ocument LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 7 2 2 Supplementary Services The Supplementary Services page is used to configure parameters that are associated with supplementary services For detailed information on supplementary services refer to Working with Supplementary Services on page 377 gt To configure the supplementary services parameters take these 4 steps Open the Supplementary Services page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Supplementary Services page item Figure 3 65 Supplementary Services Page Enable Hold Enable Hold Format 0 0 0 0 Enable Transfer Enable Transfer Prefix Enable Call Forward Enable Enable Call Waiting Enable Hook Flash Code 2 Configure the supplementary services parameters according to the table below 3 Click the Submit button to save your changes or click the Subscribe to MWI or Unsubscribe to MWI buttons to save your changes and to subscribe unsubscribe to the MWI server 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 36 Supplementary Services Parameters Parameter Description Enable Hold Enables interworking of the Hold Retrieve supplementary service from PRI EnableHold to SIP 0 Disable Disables 1 Enable Enables default Notes This capability is only supported by the Euro ISDN varian
483. of tones Detection of tones is performed according to the CPT file Version 5 6 gor November 2008 A Ee AudioCodes Mediant 2000 The PRT is a dat file containing a set of prerecorded tones that can be played by the device Up to 40 tones totaling approximately 10 minutes can be stored in a single PRT file on the device s flash memory The prerecorded tones are prepared offline using standard recording utilities such as CoolEdit and combined into a single file using AudioCodes TrunkPack Downloadable Conversion utility refer to the Product Reference Manual The raw data files must be recorded with the following characteristics m Coders G 711 A law or G 711 p law m Rate 8 kHz Resolution 8 bit m Channels mono The generated PRT file can then be loaded to the device using AudioCodes BootP TFTP utility or the Web interface refer to Loading Auxiliary Files on page 231 The prerecorded tones are played repeatedly This allows you to record only part of the tone and then play the tone for the full duration For example if a tone has a cadence of 2 seconds on and 4 seconds off the recorded file should contain only these 6 seconds The PRT module repeatedly plays this cadence for the configured duration Similarly a continuous tone can be played by repeating only part of it 6 3 Voice Prompts File The voice announcement file contains a set of Voice Prompts that can be played by the device during operation The
484. ofile EnableDigitDelivery TelProfile EnableEC TelProfile MWIAnalog TelProfile MWIDisplay TelProfile FlashHookPeriod TelProfile EnableEarlyMedia Fecal L00017 31L 2000 L5 3L 3000 15 1 4000 1 16 7000 2 16 7001 2 16 7002 27 16 16 7003 2 T SS fee yea Pot E fl flan fea fla LS e WNHROWNHFO WNROWNEFO Q OOOO SIP User s Manual 404 Document LTRT 68808 SIP User s Manual 10 Tunneling Applications TelProfile ProgressIndicator2IP TelProfile 1 voice 1 88 88 88 98 88 TelProfile 2 data 2 8 8 8 Tel Profile Originating Side E1 TRANSPARENT 3 ProtocolType 0 ProtocolType 1 ProtocolType 2 ProtocolType 3 Channel selection by Phone number ChannelSelectMode 0 TrunkGroup FORMAT TrunkGroup Index TrunkGroup TrunkGroupNum TrunkGroup FirstTrunkId TrunkGroup LastTrunkId TrunkGroup FirstBChannel TrunkGroup LastBChannel TrunkGroup FirstPhoneNumber TrunkGroup Profileld TrunkGroup Module TrunkGroup 0 TrunkGroup TrunkGroup TrunkGroup TrunkGroup TrunkGroup TrunkGroup TrunkGroup TrunkGroup CoderName FORMAT CoderName Index CoderName Type CoderName PacketInterval CoderName rate CoderName PayloadType CoderName Sce CoderName E972 onal CoderName 2 Transparent CoderName TelProfile FORMAT TelProfile Index TelProfile ProfileName Telerofile TelpPreference TelBrofile CodersGroupiD TelProfile IsFaxUse
485. ol 0 None default 1 Hardware 273 November 2008 A EA AudioCodes Mediant 2000 4 4 3 Web and Telnet Parameters The Web and Telnet related ini file configuration parameters are described in the table below Parameter WebAccessList_x WebRADIUSLogin DisableWebTask ResetWebPassword WelcomeMessage SIP User s Manual Table 4 3 Web and Telnet ini File Parameters Description Defines up to ten IP addresses that are permitted to access the device s Web interface and Telnet interfaces Access from an undefined IP address is denied This security feature is inactive i e the device can be accessed from any IP address when the table is empty For example WebAccessList_0 10 13 2 66 WebAccessList_1 10 13 77 7 The default value is 0 0 0 0 i e the device can be accessed from any IP address For defining the Web and Telnet Access list using the Web interface refer to Configuring the Web and Telnet Access List on page 102 For a description of this parameter refer to Configuring the General Security Settings on page 109 0 Enable Web management default 1 Disable Web management Resets the username and password of the primary and secondary accounts to their defaults 0 Password and username retain their values default 1 Password and username are reset for the default username and password refer to User Accounts Note The username and password
486. on menu gt Routing Tables submenu gt Internal SRV Table page item Figure 3 76 Internal SRV Table Screen sper wp v 2 In the Domain Name field enter the host name to be translated You can enter a string of up to 31 characters long 3 From the Transport Type drop down list select a transport type 4 In the DNS Name 1 field enter the first DNS A Record to which the host name is translated In the Priority Weight and Port fields enter the relevant values Repeat steps 4 through 5 for the second and third DNS names if required 5 6 7 Repeat steps 2 through 6 for each entry 8 Click the Submit button to save your changes 9 To save the changes so they are available after a hardware reset or power fail refer to Saving Configuration on page 230 Version 5 6 187 November 2008 A ge AudioCodes Mediant 2000 3 4 7 4 8 Reasons for Alternative Routing The Reasons for Alternative Routing page includes two groups IP to Tel Reasons and Tel to IP Reasons Each group allows you to define up to four different release reasons If a call is released as a result of one of these reasons the device tries to find an alternative route for that call The release reason for IP to Tel calls is provided in Q 931 notation The release reason for Tel to IP calls is provided in SIP 4xx 5xx and 6xx response codes For Tel to IP calls an alternative IP address is provided for IP t
487. ones file must include a Call Waiting Ringback tone The EnableHold parameter must be enabled on the called side For information on the Call Waiting feature refer to Call Waiting For information on the Call Progress Tones file refer to Configuring the Call Progress Tones File 160 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Description Hook Flash Code Determines the digit pattern used by the PBX to indicate a Hook Flash HookFlashCode event When this pattern is detected from the Tel side the device responds as if a Hook Flash event occurs and sends a SIP INFO message if HookFlashOption is set to 1 indicating Hook Flash If configured and a Hook Flash indication is received from the IP side the device generates this pattern to the Tel side The valid range is a 25 character string The default is a null string MLPP Multilevel Precedence and Preemption Note For additional MLPP parameters refer to Configuring the Digital Gateway Parameters on page 207 Call Priority Mode Enables Priority Calls handling CallPriorityMode 0 Disable Disable default 1 MLPP Priority Calls handling is enabled MLPP DiffServ Defines the DiffServ value differentiated services code point MLPPDiffserv DSCP used in IP packets containing SIP messages that are related to MLPP calls The valid range is 0 to 63 The default value is 50 3 4 7 2 3 Stand Alone Survivability The SAS Co
488. onfiguring the Number Manipulation Tables The device provides four Number Manipulation tables for incoming IP to Tel and outgoing Tel to IP calls These tables are used to modify the destination and source telephone numbers so that the calls can be routed correctly For example telephone number manipulation can be implemented for the following E Strip or add dialing plan digits from or to the number For example a user may need to first dial 9 before dialing the phone number to indicate an external line This number 9 can then be removed by the Manipulation table before the call is setup m Allow or disallow Caller ID information to be sent according to destination or source prefixes m Assign NPI TON to IP to Tel calls The device can use a single global setting for NPI TON classification or it can use the setting in this table on a call by call basis The number manipulation is configured in the following tables For Tel to IP calls e Destination Phone Number Manipulation Table for Tel to IP Calls NumberMapTel2IP ini file parameter e Source Phone Number Manipulation Table for Tel to IP Calls SourceNumberMapTel2IP ini file parameter SIP User s Manual 164 Document LTRT 68808 SIP User s Manual 3 Web Based Management For IP to Tel calls e Destination Phone Number Manipulation Table for IP to Tel Calls NumberMapIP2Tel ini file parameter e Source Phone Number Manipulation Table for IP to Tel Calls SourceN
489. onnection type suitable for field wiring applications connecting DC Power connector MSTB2 5 2 STF 5 08 mm from Phoenix Contact Bonding and earthing 6 32 UNC screw is provided correct ring terminal and 16 AWG wire minimum must be used Or crimp connection refer to note below Note To meet UL approval customers must fulfill the criteria below 2 pin terminal block crimp connection type comprising a Phoenix Contact Adaptor Shroud MSTBC2 5 2 STZF 5 08 Contacts MSTBC MTO 5 1 0 Cable 18 AWG x 1 5 m length Universal 90 to 260 VAC 1A max 47 63 Hz Dual redundant power supply optional 1 or2 span 39 7 W 4 spans 42 1 W approx 8 spans 45 3 W 36 to 72 VDC nominal 48 VDC 4A max floating input 1 or2 span 28 8 W 4spans 32 8 W 8 spans 36 4 W Operating Temp 0 to 40 C 32 to 104 F Short Term Operating Temp per NEBS 0 to 55 C 32 to 131 F Storage 40 to 70 C 40 to 158 F Humidity 10 to 90 non condensing 411 November 2008 ca AudioCodes Function Environmental AC Hot Swap Enclosure Dimensions Weight Installation cPCI Blade Control Processor Control Processor Memory Signal Processors PCI Bus Interface Physical Supply Voltages and Power Consumption typical Environmental Cooling Diagnostics Front panel Status LEDs Syslog events SNMP MIBs and Traps Management Configuration Management and Maintenance SIP Us
490. ope 3 AGC Redirection 0 AGC Target Energy 19 lS Active Speakers Min Interval 20 Configure Audio Playback Playback Audio Format fas Configure Audio Recording End Of Record Trim Record Audio Format 2 Configure the IP media parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 SIP User s Manual 76 Document LTRT 68808 SIP User s Manual Parameter Enable Answer Detector EnableAnswerDetector Answer Detector Activity Delay AnswerDetectorActivityDelay Answer Detector Silence Time AnswerDetectorSilenceTime Answer Detector Redirection AnswerDetectorRedirection Answer Detector Sensitivity AnswerDetectorSensitivity Answer Machine Detector Sensitivity AMDDetectionSensitivity Enable AGC EnableAGC AGC Slope AGCGainSlope Version 5 6 3 Web Based Management Table 3 16 IPMedia Parameters Description N A N A N A N A Determines the Answer Detector sensitivity The range is 0 most sensitive to 2 least sensitive The default is 0 Determines the Answer Machine Detector AMD detection sensitivity AMD can be useful in automatic dialing applications In some of these applications it is important to detect if a human voice or an answering machine is answering the call AMD can be activated and
491. or NTP services VLAN Determines the traffic type for NTP services 1 OAMP default 0 Control Specify whether to send non tagged packets on the native VLAN 0 Sends priority tag packets default 1 Sends regular packets with no VLAN tag For a description of this parameter refer to Configuring the IP Settings on page 50 Note This parameter is not applicable when configuring multiple interfaces using the ini file table parameter InterfaceTable For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 265 November 2008 A Ee AudioCodes Mediant 2000 Parameter LocalControllPAddress LocalControlSubnetMask LocalControlIDefaultGW LocalOAMIPAddress LocalOAMSubnetMask LocalOAMDefaultGW Multiple Interface Table InterfaceTable SIP User s Manual Description For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the
492. or each connection by activating the session expires timeout and using Re INVITE messages By utilizing the Profiles mechanism refer to Configuring the Profile Definitions on page 190 you can configure the TDM Tunneling feature to choose different settings based on a timeslot or groups of timeslots For example you can use low bit rate vocoders to transport voice and Transparent coder to transport data e g for D channel You can also use Profiles to assign ToS for DiffServ per source a timeslot carrying data or signaling is assigned a higher priority value than a timeslot carrying voice For tunneling of E1 T1 CAS trunks set the protocol type to Raw CAS ProtocolType 3 9 and enable RFC 2833 CAS relay mode CAS Transport Type parameter is set to CAS RFC2833 Relay refer to Configuring the Voice Settings on page 66 Version 5 6 403 November 2008 7 ge AudioCodes Mediant 2000 Note For TDM over IP the Caller ID Transport Type parameter must be set to Disable i e transparent refer to Configuring the Fax Modem CID Settings on page 67 Below is an example of ini files for two devices implementing TDM Tunneling for four E1 spans Note that in this example both devices are dedicated to TDM tunneling Terminating Side EnableTDMOverIP 1 E1 TRANSPARENT 3 ProtocolType_0 ProtocolType 1 ProtocolType 2 PIEOEOQEOIIVI NS _ 3 PREFIX FORMAT PREFIX Index
493. ort channel allocation algorithm for IP to Tel calls 0 By Dest Phone Number Selects the device s channel according to the called number default 1 Cyclic Ascending Selects the next available channel in an ascending cyclic order Always selects the next higher channel number in the trunk group When the device reaches the highest channel number in the trunk group it selects the lowest channel number in the trunk group and then starts ascending again 2 Ascending Selects the lowest available channel It always starts at the lowest channel number in the trunk group and if that channel is not available selects the next higher channel 3 Cyclic Descending Selects the next available channel in descending cyclic order Always selects the next lower channel number in the trunk group When the device reaches the lowest channel number in the trunk group it selects the highest channel number in the trunk group and then starts descending again 4 Descending Selects the highest available channel Always starts at the highest channel number in the trunk group and if that channel is not available selects the next lower channel 5 Dest Number Cyclic Ascending First selects the device s port according to the called number If the called number isn t found it then selects the next available channel in ascending cyclic order Note that if the called number is found but the port associated with this number is bu
494. ose D channel is used for signaling is called the Primary NFAS Trunk The T1 trunk whose D channel is used for backup signaling is called the Backup NFAS Trunk The primary and backup trunks each carry 23 B channels while all other NFAS trunks each carry 24 B channels The device supports up to 9 NFAS groups Each group must contain different T1 trunks SIP User s Manual 398 Document LTRT 68808 SIP User s Manual 9 Advanced PSTN Configuration The NFAS group is identified by an NFAS GroupID number possible values are 1 to 9 To assign a number of T1 trunks to the same NFAS group use the ini file parameter NFASGroupNumber_x groupID where x is the physical trunk ID 0 to the maximum number of trunks or the Web interface refer to Configuring the Trunk Settings on page 82 The parameter DchConfig_x Trunk_type defines the type of NFAS trunk Trunk_type is set to 0 for the primary trunk to 1 for the backup trunk and to 2 for an ordinary NFAS trunk x depicts the physical trunk ID 0 to the maximum number of trunks You can also use the Web interface refer to Configuring the Trunk Settings on page 82 For example to assign the first four T1 trunks to NFAS group 1 in which trunk 0 is the primary trunk and trunk 1 is the backup trunk use the following configuration NFASGroupNumber 0 1 NFASGroupNumber 1 1 NFASGroupNumber 2 1 NFASGroupNumber 3 1 DchConfig 0 0 Primary Tl trunk
495. ose to also use the wizard to load the ini and auxiliary files e g Call Progress Tones but this option cannot be pursued without loading the cmp file For the ini and each auxiliary file type you can choose to reload an existing file load a new file or not load a file at all The Software Upgrade Wizard allows you to load the following files cmp mandatory compressed firmware file ini configuration file Auxiliary files CPT Call Progress Tone VP Voice Prompts PRT Prerecorded Tones CAS and USRINF User Info Warnings e Before upgrading the device to a new major software version e g from version 5 4 to 5 6 save a copy of the device s configuration settings i e ini file to your PC refer to Backing Up and Restoring Configuration on page 240 and ensure that you have all the original auxiliary files e g CPT file currently being used by the device After you have upgraded the device upload these files to the device The Software Upgrade Wizard requires the device to be reset at the end of the process which may disrupt its traffic To avoid this disable all traffic on the device before initiating the wizard by performing a graceful lock refer to Locking and Unlocking the Device on page 229 Before you can load an ini or any auxiliary file you must first load a cmp file When you activate the wizard the rest of the Web interface is unavailable After you load the desired files access to the full
496. ot require any special behavior from them For detailed information on STUN refer to STUN on page 381 Notes For defining the STUN server domain name use the ini file parameter STUNServerDomainName refer to Networking Parameters on page 260 This parameter cannot be changed on the fly and requires a device reset Defines the IP address of the primary STUN server The valid range is the legal IP addresses The default value is 0 0 0 0 Defines the IP address of the secondary STUN server The valid range is the legal IP addresses The default value is 0 0 0 0 For detailed information on configuring the NFS table refer to Configuring the NFS Settings on page 60 Determines whether Dynamic Host Control Protocol DHCP is enabled 0 Disable Disable DHCP support on the device default 1 Enable Enable DHCP support on the device After the device powers up it attempts to communicate with a BootP server If a BootP server does not respond and if DHCP is enabled then the device attempts to obtain its IP address and other networking parameters from the DHCP server Notes After you enable the DHCP server perform the following procedure 1 Click the Submit button and then save the configuration refer to Saving Configuration on page 230 2 Perform a cold reset using the device s hardware reset button soft reset via Web interface doesn t trigger the BootP DHCP procedure and this parameter rever
497. oveFromRight LeaveFromRight Prefix2Add and Suffix2Add Parameters can be skipped by using two dollar signs Number Plan and Type can optionally be used in Remote Party ID RPID header by using the EnableRPlHeader and AddTON2RPI parameters To configure manipulation of destination numbers for Tel to IP calls using the Web interface refer to Configuring the Number Manipulation Tables on page 164 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 318 Document LTRT 68808 SIP User s Manual 4 ini File Configuration Parameter NumberMapIP2Tel Version 5 6 Description This ini file table parameter manipulates the destination number of IP to Tel calls The format of this parameter is as follows NumberMaplp2Tel FORMAT NumberMaplp2Tel_Index NumberMaplp2Tel_DestinationPrefix NumberMaplp2Tel_SourcePrefix NumberMaplp2Tel_SourceAddress NumberMaplp2Tel_NumberType NumberMaplp2Tel_NumberPlan NumberMaplp2Tel_RemoveFromLett NumberMaplp2Tel_ RemoveFromRight NumberMaplp2Tel_LeaveFromRight NumberMaplp2Tel_Prefix2Add NumberMapIp2Tel_ Suffix2Add NumberMaplp2Tel_IsPresentationRestricted NumberMaplp2Tel For example NumberMaplp2Tel NumberMaplp2Tel 0 01 034 10 13 77 8 0 2 667 NumberMaplp2Tel 1 10 10 1 1 1 1 255 255 3 0 5 100 255 NumberMaplp2Tel Notes This table parameter can include up to 100
498. oved only in the first half a second of the incoming IP signal 0 Disable default 1 Enable Determines the basic frame size that is used during fax modem bypass sessions 0 Determined internally default 1 5 msec not recommended 2 10 msec 3 20 msec Note When set for 5 msec 1 the maximum number of simultaneous channels supported is 120 Determines the Jitter Buffer delay in milliseconds during fax and modem bypass session The range is 0 to 150 msec The default is 40 Enables or disables in band network detection related to fax modem 0 Disable default 1 Enable When this parameter is enabled on Bypass mode VxxTransportType 2 a detection of an Answer Tone from the network triggers a switch to bypass mode in addition to the local Fax Modem tone detections However only a high bit rate coder voice session effectively detects the Answer Tone sent by a remote Endpoint This can be useful when for example the payload of voice and bypass is the same allowing the originator to switch to bypass mode as well 326 Document LTRT 68808 SIP User s Manual Parameter NSEMode NSEPayloadType V21ModemTransportType V22ModemTransportType V23ModemTransportType V32ModemTransportType V34ModemTransportType V34FaxTransportType Version 5 6 4 ini File Configuration Description Cisco compatible fax and modem bypass mode 0 NSE disabled de
499. p Client Verify Subject Name when acting as a server or client for the TLS connection When a remote certificate is received and this parameter is not disabled the SubjectAltName value is compared with the list of available Proxies If a match is found for any of the configured Proxies the TLS connection is established The comparison is performed if the SubjectAltName is either a 113 November 2008 A Ee AudioCodes Mediant 2000 Parameter Description DNS name DNSName or an IP address If no match is found and the SubjectAltName is marked as critical the TLS connection is not established If DNSName is used the certificate can also use wildcards to replace parts of the domain name If the SubjectAltName is not marked as critical and there is no match the CN value of the SubjectName field is compared with the parameter TLSRemoteSubjectName If a match is found the connection is established Otherwise the connection is terminated TLS Client Verify Server Certificate Determines whether the device when acting as client for TLS VerifyServerCertificate connections verifies the Server certificate The certificate is verified with the Root CA information 0 Disable default 1 Enable Note If Subject Name verification is necessary the parameter PeerHostNamevVerificationMode must be used as well TLS Remote Subject Name Defines the Subject Name that is compared with the name TLSRemoteSu
500. p to 6 indices 0 5 For configuring the Proxy Sets refer to the ini file parameter ProxylP For configuring the Proxy Set ID table using the Web interface and for a description of the parameters of this ini file table refer to Proxy Sets Table on page 141 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 Determines whether the device removes the to header tag from final SIP failure responses to INVITE transactions 0 Do not remove tag default 1 Remove tag For a description of this parameter refer to Proxy amp Registration Parameters on page 132 For a description of this parameter refer to SIP General Parameters on page 121 Enables or disables the use of the rtcp attribute in the outgoing SDP 0 Disable 1 Enable
501. page 175 or Outbound IP Routing Table page if EnableSBC is set to 1 refer to Outbound IP Routing Table on page 178 m IP to Trunk Group Routing page refer to IP to Trunk Group Routing on page 181 or Inbound IP Routing Table page refer to Inbound IP Routing Table on page 184 if EnableSBC is set to 1 In addition you can associate different Profiles per the device s channels Each Profile contains a set of parameters such as coders T 38 Relay Voice and DTMF Gain Silence Suppression Echo Canceler RTP DiffServ Current Disconnect and more The Profiles feature allows you to customize these parameters or turn them on or off per source or destination routing and or per the device s trunks channels For example specific E1 T1 spans can be assigned a Profile that always uses G 711 Each call can be associated with one or two Profiles Tel Profile and or IP Profile If both IP and Tel profiles apply to the same call the coders and other common parameters of the preferred Profile determined by the Preference option are applied to that call If the Preference of the Tel and IP Profiles is identical the Tel Profile parameters take precedence The default values of the parameters in the Tel Profile Settings and IP Profile Settings pages are identical to their default values in their respective primary configuration page If you modify a parameter in its primary configuration page orini file that al
502. parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 Same as the description for parameter FramingMethod but for a specific trunk ID x 0 7 For a description of this parameter refer to Configuring the Trunk Settings on page 82 Same as the description for parameter TerminationSide but for a specific trunk ID x 0 7 For a description of this parameter refer to Configuring the Trunk Settings on page 82 Same as the description for parameter ClockMaster but for a specific trunk ID x 0 7 For a description of this parameter refer to Configuring the TDM Bus Settings on page 218 For a description of this parameter refer to Configuring the TDM Bus Settings on page 218 For a description of this parameter refer to Configuring the TDM Bus Settings on page 218 For a description of this parameter refer to Configuring the Trunk Settings on page 82 303 November 2008 ca AudioCodes Parameter TDMBusPSTNAutoClockR evertingEnable TDMBusEnableFallback TDMBusFallbackClock TDMBusNetrefSpeed LineCode LineCode_x EnableCallingPartyCatego ry BChannelNegotiation NFASGroupNumber_x DChConfig_x ISDNNFASInterfacelD_x CASTablelndex_x CASFileName_0 CASFileName_1 CASFileName_7 CASTablesNum IdleABCDPattern IdlePCMPattern SIP User s Manual Mediant
503. parameter SipDefaultCallPriority The valid value is a 6 hexadecimal digits The default is 000000 Note This parameter is applicable only to device s using the MLPP NI 2 ISDN variant with CallPriorityMode set to 1 305 November 2008 ca AudioCodes Parameter TrunkAdministrativeState Mediant 2000 Description Defines the administrative state of a trunk 0 Lock the trunk stops trunk traffic to configure the trunk protocol type 2 Unlock the trunk default enables trunk traffic ISDN Flexible Behavior Parameters ISDN protocol is implemented in different Switches PBXs by different vendors Several implementations vary a little from the specification Therefore to provide a flexible interface that supports these ISDN variants the ISDN behavior parameters are used ISDNInCallsBehavior ISDNIBehavior ISDNGeneralCCBehavior ISDNOutCallsBehavior ISDNIBehavior_x ISDNInCallsBehavior_x ISDNOutCallsBehavior_x PlayRBTone2Tel PlayRBTone2IP ProgressIndicator2IP TimeForReorderTone ISDNDisconnectOnBusyT one DisconnectOnBusyTone EnableVoiceDetection SIP User s Manual For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parame
504. play a busy or reorder tone to the PSTN after a Tel to IP call is released 0 Don t Play Immediately sends an ISDN Disconnect message default 1 Play when Disconnecting Sends an ISDN Disconnect message with PI 8 and plays a busy or reorder tone to the PSTN depending on the release cause 2 Play before Disconnect Delays the sending of an ISDN Disconnect message for a user defined time configured by the TimeForReorderTone parameter and plays a busy or reorder tone to the PSTN Applicable only if the call is released from the IP Busy Here 486 or Not Found 404 before it reaches the Connect state otherwise the Disconnect message is sent immediately and no tones are played Defines the value of the Subject header in outgoing INVITE messages If not specified the Subject header isn t included default The maximum length is up to 50 characters Determines whether the mptime attribute is included in the outgoing SDP 0 None Disabled default 1 PacketCable includes the mptime attribute in the outgoing SDP PacketCable defined format The mptime attribute enables the device to define a separate Packetization period for each negotiated coder in the SDP The mptime attribute is only included if this parameter is enabled even if the remote side includes it in the SDP offer Upon receipt each coder receives its ptime value in the following precedence from mptime attribute from p
505. ps 1 Open the General Security Settings page Configuration tab gt Security Settings menu gt General Security Settings page item Figure 3 56 General Security Settings Page Yv HTTP Authentication Mode l Secured Web Connection HTTPS Digest When Possible HTTP and HTTPS v General RADIUS Setting Enable RADIUS Access Control Use RADIUS for Web Telnet Login RADIUS Authentication Server IP Address RADIUS Authentication Server Port RADIUS Shared Secret Disable Disable 0 0 0 0 1645 General RADIUS Authentication Default Access Level Device Behavior Upon RADIUS Timeout Local RADIUS Password Cache Mode Local RADIUS Password Cache Timeout sec RADIUS YSA Vendor ID RADIUS YSA Access Level Attribute 200 Verify Access Locally Reset Timer Upon Access 300 5003 35 v EtherDiscover Setting EtherDiscover Operation Mode Unconfigured Device Only wv IPSec Setting Enable IP Security Dead Peer Detection Mode Disable v Disabled v v TLS Settings TLS version TLS Client Re Handshake Interval TLS Mutual Authentication Peer Host Name Verification Mode TLS Client Verify Server Certificate TLS Remote Subject Name SSL 2 0 3 0 and TLS 1 0 0 Disable Disable Disable 2 Conf
506. r configuration for Proxy Set ID 2 but using different IP addresses Figure 7 4 Configuring Proxy Set ID 1 in the Proxy Sets Table Page vw Proxy Set ID Proxy Address Transport Type 10 33 37 77 uD 10 33 37 79 TCP v v v Enable Proxy Keep Alive Using Options Proxy Keep Alive Time 60 Proxy Load Balancing Method Round Robin Is Proxy Hot Swap No 3 In the IP Group Table page refer to Configuring the IP Groups on page 201 configure the two IP Groups 1 and 2 Assign Proxy Sets 1 and 2 to IP Groups 1 and 2 respectively Figure 7 5 Configuring IP Groups 1 and 2 in the IP Group Table Page Send Always Description Proxy Set ID SIP Group Name Invite To Use Route Proxy Table ITSP_1 1 Disable Y Disable W ITSP_2 2 Disable Y Disable v Version 5 6 375 November 2008 A ge AudioCodes Mediant 2000 4 Inthe Trunk Group Table page refer to Configuring the Trunk Group Table on page 195 enable the Trunks connected between the Enterprise s PBX and the device Trunk Group ID 1 and between the local PSTN and the device Trunk Group ID 2 Figure 7 6 Assign the Trunk to Trunk Group ID 1 in the Trunk Group Table Page Group From To Trunk Index
507. r refer to the Product Reference Manual gt To activate the Message Log take these 3 steps 1 In the Advanced Parameters page refer Advanced Parameter on page 151 set the parameter Debug Level or ini file parameter GwDebugLevel to 6 This parameter determines the Syslog logging level in the range 0 to 6 where 6 is the highest level Open the Message Log page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Message Log page item the Message Log page is displayed and the log is activated Figure 3 109 Message Log Screen Log is Activated 11d 14h 43m 9s 11d 14h 43m 95s iid i4h 43m 99s 11d 14h 43m 95s iid 14h 43m 9s 31d 14h 43m 9s did 14h 43 99 11d 14h 43m 95s Lid 14h 43m 93 1id 14h 43m 9s id 14h 43m 99 1id 14h 43m 9s did 14h 43m 9s iid 14h 43m 9s 3id 14h 43m 9s Lid 14h 43 98 1id 14h 43m 95s 2id 14h 43m 92 11d 14h 43m 9s SIP User s Manual igr_pabrdex 2662 igr_flow 2663 igr_flow 2664 igr_pabrdit 2665 igr_pabrdit 2666 igr_psbrdit 2667 igr_pabrdit 2668 igr_psbrdit 2669 igr_pabrdif 2670 igr_pabrdif 2671 igr_pabrdit 2672 igr_pabrdif 2673 igr_pabrdit 2674 igr_pabrdex 2675 igr_tiow 2676 igr_flow 2677 igr_pabrdif 2678 igr_pabrdit 2679 igr_psabrdit 2680 recy lt ON_HOOK Ch 1 1 08_HOOK_EV 1 08_HOOK_EV i cpDigitMapHndir Stop Stoped 0 1i CloseChannel ChannelNum 1 Open channe
508. r Enable SSHServerEnable SSH Server Port SSHServerPort DNS Settings DNS Primary Server IP DNSPriServerIP DNS Secondary Server IP DNSSecServerlIP SIP User s Manual IP address in dotted decimal notation of the NTP server The default IP address is 0 0 0 0 i e internal NTP client is disabled Defines the Universal Time Coordinate UTC offset in seconds from the NTP server The default offset is 0 The offset range is 43200 to 43200 Defines the time interval in seconds that the NTP client requests for a time update The default interval is 86400 i e 24 hours The range is 0 to 214783647 Note AudioCodes does not recommend setting this parameter to beyond one month i e 2592000 seconds Enables or disables the device s embedded Telnet server Telnet is disabled by default for security reasons 0 Disable default 1 Enable Unsecured 2 Enable Secured SSL Note Only the primary Web User Account which has Security Administration access level can access the device using Telnet refer to Configuring the Web User Accounts on page 99 Defines the port number for the embedded Telnet server The valid range is all valid port numbers The default port is 23 Defines the timeout in minutes for disconnection of an idle Telnet session When set to zero idle sessions are not disconnected The valid range is any value The default value is 0 Enables or disables the embedded Secure
509. r Tel gt IP Calls page If this Source IP Group has a Serving IP Group then all calls originating from this Source IP Group is sent to the Serving IP Group In this scenario this table is used only if the parameter PreferRouteTable is set to 1 Destination called telephone number prefix An asterisk represents any number 166 Document LTRT 68808 SIP User s Manual 3 Web Based Management Parameter Source Prefix _SourcePrefix Source IP _ SourceAddress Stripped Digits From Left L RemoveFromLeft Stripped Digits From Right L RemoveFromRight Prefix to Add _Prefix2Add Suffix to Add Suffix2Add Number of Digits to Leave _LeaveFromRight NPI _NumberPlan TON _NumberType Version 5 6 Description Source calling telephone number prefix An asterisk represents any number Source IP address of the caller obtained from the Contact header in the INVITE message Notes This parameter is applicable only to the Number Manipulation tables for IP to Tel calls The source IP address can include the x wildcard to represent single digits For example 10 8 8 xx represents all IP addresses between 10 8 8 10 to 10 8 8 99 The source IP address can include the asterisk wildcard to represent any number between 0 and 255 For example 10 8 8 represents all IP addresses between 10 8 8 0 and 10 8 8 255 Number of digits to remove from the left of the telephone number
510. r is set to the appropriate SIP response m Receiving Reason header If a call is disconnected from the IP side and the SIP message includes the Reason header it is sent to the Tel side according to the following logic e If the Reason header includes a Q 850 cause it is sent as is e If the Reason header includes a SIP response Ifthe message is a final response the response status code is translated to Q 850 format and passed to ISDN Ifthe message isn t a final response it is translated to a Q 850 cause e When the Reason header is received twice i e SIP Reason and Q 850 the Q 850 takes precedence over the SIP reason and is sent to the Tel side Fixed Mapping of ISDN Release Reason to SIP Response The following table describes the mapping of ISDN release reason to SIP response Table 9 1 Mapping of ISDN Release Reason to SIP Response SIP Response Description Unallocated number 404 Not found No route to network 404 Not found No route to destination 404 Not found Channel unacceptable 406 Not acceptable SIP User s Manual 394 Document LTRT 68808 SIP User s Manual 9 Advanced PSTN Configuration ae Description elle Description Beason Response 7 EE AE n Helvetet iman 500 Server internal error 16 Normal call clearing BYE 17 User busy 486 Busy here 18 No user responding 408 Request timeout 19 No answer from the user 480 Temporarily unavailable 21 Call rejected 403 Forbidden 22 Number chang
511. r isn t used the called and calling numbers are compared to the list of prefixes in this table Calls that match these prefixes are sent to the corresponding IP address If the number dialed does not match these prefixes the call is not made When using a Proxy server you do not need to configure this table unless you require one of the following m Fallback routing when communication with Proxy servers is lost E Implement the Filter Calls to IP and IP Security features m Obtain different SIP URI host names per called number m Assign IP profiles Note that for this table to take precedence over a Proxy for routing calls set the parameter PreferRouteTable to 1 The device checks the Destination IP Address field in this table for a match with the outgoing call A Proxy is used only if a match is not found Possible uses for Tel to IP routing include the following m Fallback to internal routing table if there is no communication with the Proxy servers E Call Restriction when Proxy isn t used rejects all outgoing Tel to IP calls that are associated with the destination IP address 0 0 0 0 m IP Security When the IP Security feature is enabled SecureCallFromlIP 1 the device accepts only those IP to Tel calls with a source IP address defined in the Tel to IP Routing table m Filter Calls to IP When a Proxy is used the device checks the Tel to IP Routing table before a telephone number is routed to the Prox
512. r of Digits to to Add Leave Presentation 1001 5 23 Restricted 1234510014 0 8 Not Configured 30 40 px Not Configured 2001 5 Not Configured Not Configured The figure above shows an example of the use of manipulation rules in the Source Phone Number Manipulation Table for Tel gt IP Calls When the destination number is 035000 and source number is 20155 the source number is changed to 97120155 e When the source number is 1001876 it is changed to 587623 e When the source number is 1234510012001 it is changed to 20018 e When the source number is 3122 it is changed to 2312 Version 5 6 165 November 2008 A C al AudioCodes Mediant 2000 2 From the Table Index drop down list select the range of entries that you want to edit up to 20 entries can be configured for Source Number IP to Tel Manipulation up to 120 entries can be configured for Source Number Tel to IP Manipulation and up to 100 entries for Destination Number Manipulation 3 Configure the Number Manipulation table according to the table below 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 230 The manipulation rules are executed in the following order 1 Number of stripped digits 2 Number of digits to leave 3 Prefix suffix to add The manipulation rules can be applied to any incoming call whose source IP address if applicable source Tr
513. r the Called Number in the Request URI and or the Calling Number in the From header these values are mapped to the outgoing ISDN SETUP message If the incoming ISDN SETUP message includes subaddress values for the Called Number and or the Calling Number these values are mapped to the outgoing SIP INVITE message s isub parameter in accordance with RFC 4715 For a description of this parameter refer to Supplementary Services on page 159 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 Determines whether the Redirect Number is retrieved from the Facility IE 0 Not supported default 1 Supports partial retrieval of Redirect Number number only from the Facility IE in ISDN SETUP messages Applicable to Redirect Number according to ECMA 173 Call Diversion Supplementary Services Note To enable this feature ISDNDuplicateQ931BuffMode must be set to 1 Determines whether Carrier Identification Code CIC is relayed to ISDN 0 Do not relay the Carrier Identification Code CIC to ISDN default 1 CIC is relayed to the ISDN in Transit Network Selection TNS IE If enabled the CIC code received in an INVITE Request URI is
514. rameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 Controls the activation deactivation of delivering raw Q 931 messages 0 ISDN messages aren t duplicated default 128 All ISDN messages are duplicated Note This parameter is not updated on the fly and requires a device reset For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 307 November 2008 ca AudioCodes Parameter CauseMapSIP2ISDN CauseMapISDN2SIP SIP User s Manual Mediant 2000 Description This ini file table parameter maps SIP Responses to Q 850 Release Causes The format of this parameter is as follows CauseMapSIP2ISDN FORMAT CauseMapSIP2ISDN_Index CauseMapSIP2ISDN_SipResponse CauseMapSIP2ISDN_IsdnReleaseCause CauseMapSIP2ISDN Where SipResponse SIP Response IsdnReleaseCause Q 850 Release Cause
515. range D channel alarm ISDN only 2 Select the trunk that you want to configure by clicking the desired Trunk number icon The bar initially displays the first eight trunk number icons i e trunks 1 through 8 To scroll through the trunk number icons i e view the next last or previous first group of eight trunks refer to the figure below Figure 3 48 Trunk Scroll Bar Number of Next Groups i of 8 Trun k 5 O18 Irunks View the First8 Trunks If the Trunk scroll bar displays all the available trunks the scroll bar buttons are unavailable After you have selected a trunk the following is displayed e The read only Trunk ID field displays the selected trunk number e The read only Trunk Configuration State displays the state of the trunk e g Active or Inactive e The parameters displayed in the page pertain to the selected trunk only 1 Click the Stop Trunk m button located at the bottom of the page to de activate the trunk so that you can configure currently grayed out unavailable parameters Skip this step if you want to configure parameters that are also available when the trunk is active The stopped trunk is indicated by the following e The Trunk Configuration State field displays Inactive A e The Stop Trunk button is replaced by the Apply Trunk Settings button When all trunks are stopped the Apply to All Trunks button also appears e All the par
516. rate analog or digital gateway For PSTN fallback the local VoIP gateway should be equipped with analog FXO lines or digital E1 T1 trunk s for PSTN connectivity In this way the Enterprise preserves its capability for internal and outgoing calls SIP User s Manual 346 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities The SAS agent continuously attempts to communicate with the Proxy using the regular keep alive method After the connection is re established the SAS agent switches to pre Normal mode In this mode the SAS agent maintains all terminations of existing calls while any new SIP call signaling issued by new INVITE sessions is transacted to from the Proxy server This is accomplished using the SAS agent s database of current active calls After releasing all calls established during Emergency mode the SAS agent resumes operating in Normal mode For SAS implementation the primary Proxy server for the VoIP CPE s e g IP phones is the SAS agent i e the device itself while the IP Centrex or IP PBX is defined as the secondary Proxy server For SAS configuration the device is composed of two different applications SAS and Gateway where each application has its own SIP interface UDP TCP TLS ports m Configuring the device to use and operate with the SAS capabilities refer to Configuring SAS on page 347 H Configuring SAS emergency call routing refer to Configuring Emergency Calls on page
517. re defined For G 729 you can also select silence suppression without adaptations If silence suppression is enabled for G 729 the device includes the string annexb no in the SDP of the relevant SIP messages If silence suppression is set to Enable w o Adaptations annexb yes is included An exception is when the remote device is a Cisco gateway IsCiscoSCEMode You can also configure the coder groups using the ini file table parameter CoderName refer to SIP Configuration Parameters on page 284 gt To configure coder groups take these 11 steps 1 Open the Coder Group Settings page Configuration tab gt Protocol Configuration menu gt Profile Definitions submenu gt Coder Group Settings page item Figure 3 79 Coder Group Settings Page Yv l Coder Group ID Silence Coder Name Packetization Time Payload Type Suppression 30 F Ws i Disabled v 2 From the Coder Group ID drop down list select a coder group ID 3 From the Coder Name drop down list select the first coder for the coder group 4 From the Packetization Time drop down list select the packetization time in msec for the coder The packetization time determines how many coder payloads are combined into a single RTP packet 5 From the Rate drop down list select the bit rate in kbps for the coder you selected 6 In the Payload Type field if the payload type for
518. re used to present the originating party s Caller ID The Caller ID is composed of a Calling Number and optionally a Calling Name P Asserted ldentity or P Preferred Identity headers are used together with the Privacy header If Caller ID is restricted P Asserted Identity is not sent the Privacy header includes the value id Privacy id Otherwise for allowed Caller ID Privacy none is used If Caller ID is restricted received from PSTN the From header is set to lt anonymous anonymous invalid gt The logic for filling the calling party parameters is as follows the SIP header is selected first from which the calling party parameters are 123 November 2008 A Ee AudioCodes Mediant 2000 Parameter Fax Signaling Method IsFaxUsed Detect Fax on Answer Tone DetFaxOnAnswerTon e SIP User s Manual Description obtained first priority is P Asserted Identity second is Remote Party ID and third is the From header Once a URL is selected all the calling party parameters are set from this header If P Asserted Identity is selected the Privacy header is checked and if the Privacy is set to id the calling number is assumed restricted Determines the SIP signaling method for establishing and transmitting a fax session after a fax is detected 0 No Fax No fax negotiation using SIP signaling Fax transport method is according to the parameter FaxTransportMode default 1 T 38 Relay Initiates T 38 fax
519. read write privileges for all pages except Administtalor ne security related pages which are read only No access to security related and file loading User Monitor 50 pages read only access to the other pages This read only access level is typically applied to the secondary Web user account No Access 0 No access to any page The numeric representation of the access level is used only to define accounts in a RADIUS server the access level ranges from 1 to 255 The default attributes for the two Web user accounts are shown in the following table Table 3 23 Default Attributes for the Web User Accounts Account Attribute User Name Password Access Level Case Sensitive Case Sensitive Primary Account Admin Admin Security Administrator Note The Access Level cannot be changed for this account type Secondary Account User User User Monitor gt To change the Web user accounts attributes take these 4 steps 1 Open the Web User Accounts page Configuration tab gt Security Settings menu gt Web User Accounts page item Figure 3 50 Web User Accounts Page for Users with Security Administrator Privileges Current Logged User Admin Account Data for User Admin User Name Change User Name Access Level w Fill in the following 3 fields to change the password Current Password New Password Confirm New Password Change Password wv Account Data for User User 2 User Name User 2 Ch
520. refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 330 Document LTRT 68808 SIP User s Manual Parameter DisableRTCPRandomize RTCPXRESCTransportType RTCPXREscIP RTCPXRReportMode 4 ini File Configuration Description For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 Determines the transport layer used for outgoing SIP dialogs initiated by the device to the RTCP XR Collection Server 1 Not Configured default 0 UDP 1 TCP 2 TLS Note When set to Not Configured the value of the parameter SIPTransportType is used For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 71 4 4 14 Auxiliary Configuration Files Parameters The configuration files i e auxiliary files can be loaded to the
521. refreshed according to ProxylPListRefreshTime If a change in the order of the entries in the list occurs all load statistics are erased and balancing starts over again When the Random Weights algorithm is used the outgoing requests are not distributed equally among the Proxies The weights are received from the DNS server by using SRV records The device sends the requests in such a fashion that each Proxy receives a percentage of the requests according to its assigned weight A single FQDN should be configured as a Proxy IP address The Random Weights Load Balancing is not used in the following scenarios The Proxy Set includes more than one Proxy IP address The only Proxy defined is an IP address and not an FQDN SRV is not enabled DNSQueryType The SRV response includes several records with a different Priority value 143 November 2008 ca AudioCodes Parameter Enable Proxy Keep Alive EnableProxyKeepAlive Proxy Keep Alive Time ProxyKeepAliveTime Is Proxy Hot Swap IsProxyHotSwap 3 4 7 1 4 Coders Mediant 2000 Description Determines whether Keep Alive with the Proxy is enabled or disabled This parameter is configured per Proxy Set 0 Disable Disable default 1 Using OPTIONS Enables Keep Alive with Proxy using OPTIONS 2 Using REGISTER Enable Keep Alive with Proxy using REGISTER If set to Using OPTIONS the SIP OPTIONS message is sent every user defined interval
522. ring Machine Detection Address of Record American Wire Gauge Bits per second AudioCodes Proprietary Bootstrap Loader Utility Channel Associated Signaling Class of Service Compressed File device Firmware Compact PCI Industry Standard Call Progress Tones Decibels Dynamic Host Control Protocol Direct Inward Dial Differentiated Services Domain Name System or Server Debug Recording 1 544 Mbps USA Digital Transmission System see E1 and T1 Digital Signal Processor or Processing Dual Tone Multiple Frequency Touch Tone 2 048 Mbps European Digital Transmission System see T1 European Telecommunications Standards Institute Fully Qualified Domain Name Globally Routable User Agent URIs Internet Control Message Protocol Information Element ISDN layer 3 protocol basic building block Internet Engineering Task Force Internet Key Exchange for IPSec Internet Protocol IP Security 415 November 2008 A ge AudioCodes Mediant 2000 Term ISDN ISO ITU ITU T IVR Jitter kbps LAPD LFA LOF Mbps MIB MLPP ms or msec MSCML NT MWI NAPTR NAT NFAS NFS NPI NTP OAMP OSI PBX PCI PCM P PKI POTS PRT PRI PSTN PVID QoS SIP User s Manual Meaning Integrated Services Digital Network International Standards Organization International Telecommunications Union Telecommunications section of the ITU Interactive Voice Response Variation of interpacket timing interval Kilobit per second 1 000
523. rmation Tone SIT detection according to the ITU T recommendation E 180 Q 35 0 Disable default 1 Enable SourcelPAddressInput For a description of this parameter refer to Routing General Parameters on page 171 EnableSBC For a description of this parameter refer to SBC Configuration on page 163 SBCRegisirationTime For a description of this parameter refer to SBC Configuration on page 163 Stand Alone Survivability SAS Parameters EnableSAS For a description of this parameter refer to Stand Alone Survivability on page 161 SASLocalSIPUDPPort For a description of this parameter refer to Stand Alone Survivability on page 161 SASDefaultGatewayIP For a description of this parameter refer to Stand Alone Survivability on page 161 SASRegisirationTime For a description of this parameter refer to Stand Alone Survivability on page 161 SASLocalSIPTCPPort For a description of this parameter refer to Stand Alone Survivability on page 161 SASLocalSIPTLSPort For a description of this parameter refer to Stand Alone Survivability on page 161 SASProxySet For a description of this parameter refer to Stand Alone Survivability on page 161 RedundantSASProxySet For a description of this parameter refer to Stand Alone Survivability on page 161 SASSurvivabilityMode Determines the Survivability mode used by the SAS application 0 Standard All in
524. rmines whether or not the device plays a ringback tone RBT to the IP side of the call IP to Tel calls 0 Don t Play Ringback tone isn t played default 1 Play Ringback tone is played after SIP 183 session progress response is sent If configured to 1 Play and EnableEarlyMedia 1 the device plays a ringback tone according to the following For CAS interfaces the device opens a voice channel sends a 183 SDP response and then plays a ringback tone to IP For ISDN interfaces if a Progress or an Alert message with PI 1 or 8 is received from the ISDN the device opens a voice channel sends a 183 SDP or 180 SDP response but doesn t play a ringback tone to IP If Pl 1 or 8 is received from the ISDN the device assumes that ringback tone is played by the ISDN switch Otherwise the device plays a ringback tone to IP after receiving an Alert message from the ISDN It sends a 180 SDP response signaling to the calling party to opena voice channel to hear the played ringback tone Notes To enable the device to send a 183 180 SDP responses set EnableEarlyMedia to 1 If EnableDigitDelivery 1 the device doesn t play a ringback tone to IP and doesn t send 183 or 180 SDP responses Determines the method used to play a ringback tone to the Tel side It applies to all trunks that are not configured by the parameter PlayRBTone2Trunk Similar description as the parameter PlayRBTone2Trunk 0 Don t
525. rms are used throughout this manual Hexadecimal notation is indicated by Ox preceding the number Version 5 6 13 November 2008 A EA AudioCodes Mediant 2000 Related Documentation Document Manual Name LTRT 523xx where xxis the Product Reference Manual document version LTRT 690xx Mediant 3000 amp Mediant 2000 amp TP Series SIP Release Notes LTRT 701xx Mediant 2000 amp IPmedia 2000 SIP MGCP MEGACO Installation Manual LTRT 665xx CPE Configuration Guide for IP Voice Mail LTRT 400xx IP to IP SIP Call Routing Application Note Warning The device is supplied as a sealed unit and must only be serviced by qualified service personnel The term device used throughout this manual refers to the Mediant 2000 media gateway unless otherwise specified Where network appears in this manual it means Local Area Network LAN Wide Area Network WAN etc accessed via the device s Ethernet interface The terms P to Tel and Tel to P refer to the direction of the call relative to the AudioCodes device P to Tel refers to calls received from the IP network and destined to the PSTN PBxX i e telephone connected directly or indirectly to the device Te l to P refers to calls received from the PSTN PBX and destined for the IP network SIP User s Manual 14 Document LTRT 68808 SIP User s Manual 1 Overview 1 Overview This manual provides you with the information for installing configuring and operating the
526. ronger than the lower one Negative values cause the opposite effect For any parameter value both components change so that their average is constant The valid range is 10 to 10 dB The default value is 0 dB 67 November 2008 ca AudioCodes Mediant 2000 3 4 2 2 Configuring the Fax Modem CID Settings The Fax Modem CID Settings page is used for configuring fax modem and Caller ID CID parameters gt To configure the fax modem and CID parameters take these 4 steps 1 Open the Fax Modem CID Settings page Configuration tab gt Media Settings menu gt Fax Modem CID Settings page item Figure 3 41 Fax Modem CID Settings Page Fax Transport Mode Caller ID Transport Type Caller ID Type 21 Modem Transport Type 22 Modem Transport Type 23 Modem Transport Type 32 Modem Transport Type 34 Modem Transport Type Fax Relay ECM Enable Fax Relay Max Rate bps Fax Bypass Output Gain Modem Bypass Output Gain Fax CNG Mode CNG Detector Mode Fax Relay Redundancy Depth Fax Relay Enhanced Redundancy Depth 4 Fax Modem Bypass Coder Type Fax Modem Bypass Packing Factor RelayEnable Mute Standard Bellcore Disable Enable Bypass Enable Bypass Enable Bypass SNS S SSIS INS Enable Bypass 0 Enable 14400bps G711Alaw_64 1 0 0 Disable Disable 2 Co
527. roup_FirstBChannel the channel numbers You can enter a range of channels by TrunkGroup_LastBChannel using the format n m where n represents the lower channel number and m the higher channel number e g 1 24 specifies channels 1 through 24 Notes The number of defined channels must not exceed the number of the Trunk s B channels To represent all channels enter a single asterisk SIP User s Manual 196 Document LTRT 68808 SIP User s Manual Parameter Phone Number TrunkGroup_FirstPhoneNumber Trunk Group ID TrunkGroup_TrunkGroupNum Profile ID TrunkGroup_Profileld 3 Web Based Management Description Enter the first telephone number that you want to assign to the first channel defined in the Channels field Subsequent channels are assigned the next consecutive phone number Notes Ifthe Phone Number field includes alphabetical characters and the phone number is defined for a range of channels e g 1 4 then the phone number must end with a number e g usert This field is optional The logical numbers defined in this field are used when an incoming PSTN PBX call doesn t contain the calling number or called number the latter being determined by the parameter ReplaceEmptyDstWithPortNumber these numbers are used to replace them These logical numbers are also used for channel allocation for P to Tel calls if the Trunk Group s Channel Select Mode is set to By Dest
528. rtType 2 m V22ModemTransportType 2 SIP User s Manual 352 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities V23ModemTransportType 2 V32ModemtTransportType 2 V34ModemtTransportType 2 BellModemTransportType 2 Additional configuration parameters e FaxModemBypassCoderType e FaxBypassPayloadType e ModemBypassPayloadType e FaxModemBypassBasicRTPPacketinterval e FaxModemBypassDJBufMinDelay Note When the device is configured for modem bypass and T 38 fax V 21 low speed modems are not supported and fail as a result When the remote non AudioCodes gateway uses G711 coder for voice and doesn t change the coder payload type for fax or modem transmission it is recommended to use the Bypass mode with the following configuration EnableFaxModemInbandNetworkDetection 1 FaxModemBypassCoderType same coder used for voice FaxModemBypassM same interval as voice ModemBypassPayloadType 8 if voice coder is A Law 0 if voice coder is Mu Law 7 5 2 3 Fax Modem NSE Mode In this mode fax and modem signals are transferred using Cisco compatible Pass through bypass mode Upon detection of fax or modem answering tone signal the terminating device sends three to six special NSE RTP packets using NSEpayloadType usually 100 These packets signal the remote device to switch to G 711 coder according to the parameter FaxModemBypassCoderType After a few NSE packets are exchanged between th
529. rver the following INFO sip Sipp 172 22 2 9 5060 SIP 2 0 Via SIP 2 0 UDP 172 22 168 249 branch z9hG4bKac482466515 Max Forwards 70 From sut lt sip 3000 172 22 168 249 5060 gt tag 1c419779142 To sipp lt Sip Sipp 172 22 2 9 5060 gt tag 1 Call ID 1 29753 172 22 2 9 CSeq 1 INFO Contact lt sip 56700 172 22 168 249 gt Supported em timer replaces path resource priority Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO SUB SCRIBE UPDATE User Agent Audiocodes Sip Gateway IPmedia 260 UN v 5 20A 040 004 Content Type application x detect Content Length 34 Type PIT SubType SPEECH END 7 3 4 The Application server now sends its message to the answering message If the device detects voice and not an answering machine the SIP INFO message includes Type AMD SubType VOICE If the device detects silence the SIP INFO message includes the SubType SILENT Stand Alone Survivability SAS Feature The device s Stand Alone Survivability SAS feature ensures telephony communication continuity survivability for enterprises using hosted IP services such as IP Centrex or IP PBX in cases of failure of these entities In case of failure of the IP Centrex IP PBX servers or even WAN connection and access Internet modem the enterprise typically loses its internal telephony service at any branch between its offices as well as with the external environment In addition typicall
530. s TE E pennar Log Off Confirmation Box said Web Session Logged Off IP Settings Paga 205 sicrsaaiccniecnstnusceteasioctsentadas Confirmation Message 3 for Accessing the Multipl e Interfa e Table Interface Table Page Application Settings Page aonn A E E E ere E ee A AA NFS Sef ngs Page soosse snn SEA aa ED EAAS Eae O I ROUINO Tae PAE oroen A Taa a EE a EA a A AEE AN E A N A E A AT E o vore S0 PAIE a EA E a a 3 FarModem CID Settings PAJE asiriene Ea AE SEa ERA 68 RTP RTOP Settings Page i i i i IPNEda Seug PAGE nEaN AI EATS SIEEN AEEA General Media Settings Page i i i PBT rie Pae aaa AAE 80 Meda ese ING PIJE drnnorirni snno aane AADA EEEREN re Tank TOMI PEG aaea EEEa EEEREN 8 Tonk Soor E AV aiir EANA E EAA une Eee CAS State Machine a a a i dies 97 Web User Accounts Page for Users with Security Administrator Privileges arasi TOO Web amp Telnet Access List Page Add New a AENEA sine 102 Web amp Telnet Access List Table siete Me Firewall Settings Page sage 109 Certificates Signing Request Page 106 IKE Table Listing Loaded Certificate Files OE genni AOA A O E N T General Security Settings A E E E A te Document LTRT 68808 SIP User s Manual Figure Figure Figure Figure 3 Figure Figure 3 69 Figure 3 71 Figure 3 7 Figure Figure 3 73 Figure 3 74 Fi 3 75 k Cor firmation N Figure 3 112 Figure 3 113 i Figure 3 114 A
531. s refer to Structure of ini File Table Parameters on page 257 This ini file table parameter configures the IP profiles table The format of this parameter is as follows IPProfile FORMAT IPProfile_Index IPProfile_ProfileName IPProfile_lpPreference IPProfile_ CodersGroupID IPProfile_IsFaxUsed IPProfile_JitterBufMinDelay IPProfile_JitterBufOptFactor IPProfile_IPDiffServ IPProfile_SigIPDiffServ N A IPProfile_ RTPRedundancyDepth IPProfile_ RemoteBaseUDPPort IPProfile CNGmode IPProfile_VxxTransportType IPProfile_NSEMode N A IPProfile_ PlayRBTone2IP PProfile_EnableEarlyMedia IPProfile_ProgressIndicator2IP IPProfile_ EnableEchoCanceller IPProfile_CopyDest2RedirectNumber IPProfile_MediaSecurityBehaviour IPProfile_CallLimit IPProfile_ DisconnectOnBrokenConnection IPProfile For example IPProfile IPProfile_1 name1 2 1 0 10 13 15 44 1 1 6000 0 2 0 0 0 1 0 1 0 0 1 1 IPProfile_2 name2 55 55 55 5 5 5 5 5 40 IPProfile Notes This parameter can appear up to 9 times i e indices 1 9 ndicates common parameters used in both IP and Tel profiles IpPreference determines the priority of the Profile 1 to 20 where 20 is the highest preference If both IP and Tel profiles apply to the same call the coders and other common parameters indicated with an asterisk of the preferred Profile are applied to that call If the Tel an
532. s Phone Call Ca Niel PAM Onc ORS icSO m audio 4000 RTP AVP 8 96 a rtpmap 8 pcma 8000 a rtpmap 96 telephone event 8000 ES MEOI DS CO iS a ptime 20 F2 10 8 201 10 gt gt 10 8 201 108 TRYING Version 5 6 369 November 2008 Ao ei AudioCodes Mediant 2000 SIP 2 0 100 Trying Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacsiJkDGd From lt sip 8000 10 8 201 108 gt tag 1c5354 TOR Sape LOOOC LO 8 20i 10 gt Call ID 534366556655skKw 8000 1000 10 8 201 108 Server Audiocodes Sip Gateway Mediant 2000 v 5 40 010 006 CSeq 18153 INVITE Content Length 0 F3 10 8 201 10 gt gt 10 8 201 108 180 RINGING SIP 2 0 180 Ringing Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacsiJkDGd From lt sip 8000 10 8 201 108 gt tag 1c5354 To lt Sip 1000 10 8 201 10 gt tag 1c7345 Call ID 534366556655skKw 8000 1000 10 8 201 108 Server Audiocodes Sip Gateway Mediant 2000 v 5 40 010 006 CSeq 18153 INVITE Supported 100rel em Content Length 0 Note Phone 1000 answers the call and then sends a 200 OK message to device 10 8 201 108 m F4 10 8 201 10 gt gt 10 8 201 108 200 OK STP 2 0 200 Ox Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacsiJkDGd From lt Sip 8000 10 8 201 108 gt tag 1 c5354 To lt Sip 1000 10 8 201 10 gt tag 1 c7345 Call ID 534366556655skKw 8000 1000 10 8 201 108 CSeq 18153 INVITE Contact lt sip 1000 10 8 201 10 user phone gt Server A
533. s and not only with the active one Determines the device s mode of operation when Authentication and Key Agreement AKA Digest Authentication is used 0 Optional Incoming requests that don t include AKA authentication information are accepted default 1 Mandatory Incoming requests that don t include AKA authentication information are rejected 140 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 7 1 3 Proxy Sets Table The Proxy Sets Table page allows you to define Proxy Sets A Proxy Set is a group of Proxy servers defined by IP address or fully qualified domain name FQDN You can define up to six Proxy Sets each having a unique ID number and each containing up to five Proxy server addresses For each Proxy server address you can define the transport type i e UDP TCP or TLS In addition Proxy load balancing and redundancy mechanisms can be applied per Proxy Set if a Proxy Set contains more than one Proxy address Proxy Sets can later be assigned to IP Groups of type SERVER only refer to Configuring the IP Groups on page 201 When the device sends an INVITE message to an IP Group it is sent to the IP address domain name defined for the Proxy Set that is associated with the specific IP Group In other words the Proxy Set represents the destination of the call Typically for IP to IP call routing at least two Proxy Sets are defined for call destination one for each leg IP Group
534. s are included in the Remote Party ID RPID header 0 No 1 Yes default If RPID header is enabled EnableRPlHeader 1 and AddTON2RPI 1 it s possible to configure the calling and called number type and number plan using the Number Manipulation tables for Tel to IP calls 126 Document LTRT 68808 SIP User s Manual Parameter Enable History Info Header EnableHistorylnfo Use Source Number as Display Name UseSourceNumberA sDisplayName Version 5 6 3 Web Based Management Description Enables usage of the History Info header 0 Disable Disable default 1 Enable Enable User Agent Client UAC Behavior Initial request The History Info header is equal to the Request URI If a PSTN Redirect number is received it is added as an additional History Info header with an appropriate reason Upon receiving the final failure response the device copies the History Info as is adds the reason of the failure response to the last entry and concatenates a new destination to it if an additional request is sent The order of the reasons is as follows 1 Q 850 Reason 2 SIP Reason 3 SIP Response code Upon receiving the final response success or failure the device searches for a Redirect reason in the History Info i e 3xx 4xx SIP reason If found it is passed to ISDN according to the following table SIP Reason Code ISDN Redirecting Reason 302 Moved Temporarily Call Forward Un
535. s enabled The valid range is 20 to 65 000 msec The default is 10 000 Note To enable No Op packet transmission use the NoOpEnable parameter Determines the payload type of No Op packets The valid range is 96 to 127 for the range of Dynamic RTP Payload Type for all types of non hard coded RTP Payload types refer to RFC 3551 The default value is 120 Note When defining this parameter ensure that it doesn t cause collision with other payload types Changes the RTP packets according to the MAC address of received RTP packets and according to Gratuitous Address Resolution Protocol GARP messages 0 nothing is changed 1 If the device receives RTP packets with a different source MAC address than the MAC address of the transmitted RTP packets then it sends RTP packets to this MAC address and removes this IP entry from the device s ARP cache table 2 The device uses the received GARP packets to change the MAC address of the transmitted RTP packets 3 both 1 and 2 options above are used default For a description of this parameter refer to Configuring the IP Settings on page 50 For a description of this parameter refer to Configuring the Management Settings on page 220 For a description of this parameter refer to Configuring the Management Settings on page 220 For a description of this parameter refer to Configuring the Management Settings on page 220 263 November 2008 c
536. s the maximum number of active SIP dialogs that are not call related i e REGISTER and SUBSCRIBE This parameter is used to control the Registration Subscription rate The valid range is 1 to 20 The default value is 20 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 Defines a representative number up to 50 characters that is used as the User Part of the Request URI in the P Asserted Identity header of an outgoing INVITE for Tel to IP calls The default value is NULL 289 November 2008 ca AudioCodes Parameter UseAORInReferToHeader UseTelURIForAssertedID EnableRPlheader IsUserPhone IsUserPhonelnFrom IsUseToHeaderAsCalledNumb er EnableHistoryInfo SIPSubject MultiPtimeFormat EnableReasonHeader EnableSemiAttendedTransfer SIP183Behavior EnablePtime EnableUserInfoUsage HandleReasonHeader EnableSilenceSuppInSDP SIP User s Manual Mediant 2000 Description Defines the source for the SIP URI set in the Refer To header of outgoing REFER messages 0 Use SIP URI from Contact header of the initial call default 1 Use SIP URI from To From header of the initial call For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this p
537. sPrefix 1 3 At Device A route all incoming PSTN calls starting with prefixes 1 2 3 and 4 to the IP address of Device B Prefix 1 192 168 3 51 Prefix 2 192 168 3 51 Prefix 3 192 168 3 51 Prefix 4 192 168 3 51 Note You can also define Prefix 192 168 3 51 instead of the four lines above 4 At Device B route IP to PSTN calls to Trunk Group ID according to the first digit of the called number Version 5 6 PSTNPrefix 1 1 PSTNPrefix 2 2 367 November 2008 A ge AudioCodes Mediant 2000 7 13 e PSTNPrefix 3 4 e PSTNPrefix 4 4 5 At Device B remove the first digit from each IP to PSTN number before it is used in an outgoing call NumberMapIP2Tel 1 Proxy or Registrar Registration Example Below is an example of Proxy and Registrar Registration REGISTER sip servername SIP 2 0 VIA SIP 2 0 UDP 212 179 22 229 branch z 9hG4bRaC7AU234 From lt sip GWRegistrationName sipgatewayname gt tag 1c29347 To lt sip GWRegistrationName sipgatewayname gt Calil mpDe 10455021217922 229 Seq 1 REGISTER Expires 3600 Contact sip GWRegistrationName 212 179 22 229 Content Length 0 The servername string is defined according to the following rules m The servername is equal to RegistrarName if configured The RegistrarName can be any string m Otherwise the servername is equal to RegistrarlP either FQDN or numerical IP address if configured m Otherwise t
538. se sensitive If silence suppression is not defined for a specific coder the value defined by the parameter EnableSilenceCompression is used The value of several fields is hard coded according to common standards e g payload type of G 711 U law is always 0 Other values can be set dynamically If no value is specified for a dynamic field a default value is assigned If a value is specified for a hard coded field the value is ignored Only the ptime of the first coder in the defined coder list is declared in INVITE 200 OK SDP even if multiple coders are defined lf the coder G 729 is selected and silence suppression is enabled for this coder the device includes the string annexb no in the SDP of the relevant SIP messages If silence suppression is set to Enable w o Adaptations annexb yes is included An exception is when the remote device is a Cisco gateway IsCiscoSCEMode Both GSM FR and MS GSM coders use Payload Type 3 When using SDP it isn t possible to differentiate between the 298 Document LTRT 68808 SIP User s IPProfile Version 5 6 Manual Parameter 4 ini File Configuration Description two Therefore it is recommended not to select both coders simultaneously Fora list of supported coders refer to Coders on page 144 To configure the Coders table in the Web interface refer to Coders on page 144 Fora description of using ini file table parameter
539. service RTP traffic Media Premium media RTCP traffic Media Premium media T 38 traffic Media Premium media SIP Control Premium control SIP over TLS SIPS Control Premium control Syslog Management Bronze ICMP Management Determined by the initiator of the request ARP listener Determined by the initiator of the Network request SNMP Traps Management Bronze DNS client DNS EnableDNSasOAM Network Depends on traffic type NTP NTP EnableNTPasOAM Control Premium control Management Bronze NFSServers_VlanType in the ahs NFSServers table Geld 8 9 3 Getting Started with VLANS and Multiple IPs By default the device operates without VLANs and multiple IPs using a single IP address subnet mask and default Gateway IP address This section provides an example of the configuration required to integrate the device into a multiple IPs network withVLANs using the Web interface refer to Integrating Using the Web Interface on page 388 and ini file refer to Integrating Using the ini File on page 390 The following table shows an example configuration used in this subsection Table 8 2 Example of VLAN and Multiple IPs Configuration Network Subnet Default Gateway External Routing Type _ PAddress ask IP Address VEAP Rule OAMP 10 31 174 50 255 255 0 0 0 0 0 0 83 4 87 X Control 10 32 174 50 255 255 0 0 0 0 0 0 130 33 4 6 Media 10 33 174 50 255 255 0 0 10 33 0 1 Version 5 6 387 November 2008 A ge AudioCodes Mediant 2000 The values provided i
540. sing the ini file Before you load the files to the device in the ini file you need to include certain ini file parameters associated with these files These ini file parameters specify the files that you want loaded and whether they must be stored in the non volatile memory For a description of the ini file parameters associated with the auxiliary files refer to Configuration Files Parameters on page 331 gt To load the auxiliary files via the ini file take these 3 steps 1 Inthe inifile define the auxiliary files to be loaded to the device You can also define in the ini file whether the loaded files must be stored in the non volatile memory so that the TFTP process is not required every time the device boots up 2 Save the auxiliary files you want to load and the ini file in the same directory on your PC 3 Invoke a BootP TFTP session the ini and auxiliary files are loaded to the device 3 5 2 2 Upgrading the Software Upgrade Key The device is supplied with a Software Upgrade Key for each of its TrunkPack Modules TPM You can upgrade the device s features capabilities and quantity of available resources by by purchasing a new key to match your requirements The Software Upgrade Key is provided in string format in a text file which is loaded to the device s non volatile flash memory The string defines the device s allowed features and capabilities A new key overwrites a previously installed key You can load a Software Up
541. so appears in the profile pages the parameter s new value is automatically updated in the profile pages However once you modify any parameter in the profile pages modifications to parameters in the primary configuration pages orini file no longer impact that profile 3 4 7 5 1 pages Coder Group Settings The Coder Group Settings page provides a table for defining up to four different coder groups These coder groups are used in the Tel Profile Settings and IP Profile Settings pages to assign different coders to Profiles For each coder group you can define up to five coders where the first coder and its attributes in the table takes precedence over the second coder and so on The first coder is the highest priority coder and is used by the device whenever possible If the far end device cannot use the coder assigned as the first coder the device attempts to use the next coder and so on For a list of coders supported by the device refer to Coders on page 144 SIP User s Manual 190 Document LTRT 68808 SIP User s Manual 3 Web Based Management Each coder type can appear only once per Coder Group The device always uses the packetization time requested by the remote side for sending RTP packets If not specified the packetization time ptime is assigned the default value Only the packetization time of the first coder in the defined coder list is declared in INVITE 200 OK SDP even if multiple coders a
542. splayed and the button is replaced by the left pointing arrow button Figure 3 6 Showing and Hiding Navigation Pane A ome map Show Hide Button Displayed Navigation QoS Settings Delete Selected Ermes _Pane _ nad a nom tabio entry me WB adeance Appicanons Destination IP Address Oevtination Mask Gateway IP Address Hep Count Are Oot Sina s s a Delete Destination 1P Row A d ress Delate Selected Entes Derteaton iP Address Destination Mask Gateway iP Address Ho Add New Ermy 3 3 3 Working with Configuration Pages The configuration pages contain the parameters for configuring the device The configuration pages are displayed in the Work pane which is located to the right of the Navigation pane Version 5 6 25 November 2008 A Ee AudioCodes Mediant 2000 3 3 3 1 3 3 3 2 Accessing Pages The configuration pages are accessed by clicking the required page item in the Navigation tree gt To open a configuration page in the Work pane take these 2 steps 1 On the Navigation bar click the required tab e Configuration refer to Configuration Tab on page 50 e Management refer to Management Tab on page 220 e Status amp Diagnostics refer to Status amp Diagnostics Tab on page 241 The menus of the selected tab appears in the Navigation tree 2 In the Navigation tree drill down to the required page item the page opens in the Work pane
543. ss to WEB amp Telnet Adda new authorized IP addresss Add New Entry 3 To delete authorized IP addresses select the Delete Row check boxes corresponding to the IP addresses that you want to delete and then click Delete Selected Addresses the IP addresses are removed from the table and these IP addresses can no longer access the Web and Telnet interfaces 4 To save the changes to flash memory refer to Saving Configuration on page 230 SIP User s Manual 102 Document LTRT 68808 SIP User s Manual 3 Web Based Management The first authorized IP address in the list must be your PC s terminal IP address otherwise access from your PC is denied Only delete your PC s IP address last from the Web amp Telnet Access List page If it s deleted before the last access from your PC is denied after it s deleted 3 4 6 3 Configuring the Firewall Settings The device provides an internal firewall allowing you the security administrator to define network traffic filtering rules You can add up to 50 ordered firewall rules For each packet received on the network interface the table is scanned from the top down until a matching rule is found This rule can either deny block or permit allow the packet Once a rule in the table is located subsequent rules further down the table are ignored If the end of the table is reached without a match the packet is accepted For detailed information on the internal firewal
544. ssList 22 10 4 0 0 255 255 0 0 4000 9000 any 0 0 0 block ACCESSLIST In the example above Rule 10 allows traffic from the host mgmt customer com destined to TCP ports 0 to 80 Rule 22 blocks traffic from the subnet 10 4 xxx yyy destined to ports 4000 to 9000 Notes This parameter can include up to 50 indices f the end of the table is reached without a match the packet is accepted To configure the firewall using the Web interface and for a description of the parameters of this ini file table parameter refer to Configuring the Firewall Settings on page 103 Fora description of configuring with ini file table parameters 280 Document LTRT 68808 SIP User s Manual Parameter AccessList_MatchCount 4 4 5 4 ini File Configuration Description refer to Structure of ini File Table Parameters on page 257 For a description of this parameter refer to Configuring the Firewall Settings on page 103 RADIUS Parameters The RADIUS related ini file configuration parameters are described in the table below For detailed information on the supported RADIUS attributes refer to Supported RADIUS Attributes on page 362 Table 4 5 RADIUS ini File Parameters Parameter EnableRADIUS AAAlndications BehaviorUponRadiusTimeout MaxRADIUSSessions SharedSecret RADIUSRetransmission RadiusTO RADIUSAuthServerlP RADIUSAuthPort RADIUSAccServerlP RADIUSAccPort Radius
545. sseeeene ets 394 9 2 3 Fixed Mapping of SIP Response to ISDN Release Reason 396 9 3 ISDN Overlap Dialing here eee rene ene ne ener EE E 9 4 ISDN Non Facility Associated dgl NFAS er E T R 398 g4 1 NEARS Intenace ID eenen aan ai aaa kaaa Eai aieka 399 9 4 2 Working with DMS 100 Switches uisissnnisinins rnein naiiai aaie ananda naaa 400 SIP User s Manual 6 Document LTRT 68808 SIP User s Manual Contents 9 4 3 Creating an NF 400 401 96 Automatic Gain Control AGC wivsicnisnianceminminmiiniminioniioiannamyOe 10 Tunneling Applications 6 ciianicmninimniomimmionnninumemmnamencannanns AS 10 1 TDM Tunne 10 2 QSIG Tunneling 11 Supplied SI 12 Selected Technical Specifications scecececeeeseeseeeeeseeseeseeseseeseeeeeeers 409 13 Clos a rasonas le Version 5 6 7 November 2008 ed AudioCodes Figure 1 1 Figure 3 1 Figure 3 2 Figure 3 3 Figure 3 4 Figure 3 5 Figure 3 6 Figure 3 7 T Figure 3 8 Figure 3 9 Figure 3 10 Figure 3 11 Figure 3 12 Figure 3 13 Figure 3 14 Figure 3 15 Figure 3 16 Figure 3 17 Figure 3 18 Figure 3 19 Figure 3 20 Figure 3 21 Figure 3 22 Figure 3 23 Figure 3 24 Figure 3 25 Figure 3 26 Figure 3 27 Figure 3 28 Figure 3 29 Figure 3 30 Figure 3 31 Figure 3 32 Figure 3 33 Figure 3 34 Figure 3 35 Figure 3 36 Figure 3 37 Figure 3 38 Figure 3 39 Figure 3 40 Figure 3 41 Figure 3 42
546. ssor in evenly spaced intervals The device uses a dynamic jitter buffer that can be configured using the following two parameters Minimum delay DJBufMinDelay 0 msec to 150 msec Defines the starting jitter capacity of the buffer For example at 0 msec there is no buffering at the start At the default level of 10 msec the device always buffers incoming packets by at least 10 msec worth of voice frames Optimization Factor DJBufOptFactor 0 to 12 13 Defines how the jitter buffer tracks to changing network conditions When set at its maximum value of 12 the dynamic buffer aggressively tracks changes in delay based on packet loss statistics to increase the size of the buffer and doesn t decay back down This results in the best packet error performance but at the cost of extra delay At the minimum value of 0 the buffer tracks delays only to compensate for clock drift and quickly decays back to the minimum level This optimizes the delay performance but at the expense of a higher error rate SIP User s Manual 360 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 7 9 The default settings of 10 msec Minimum delay and 10 Optimization Factor should provide a good compromise between delay and error rate The jitter buffer holds incoming packets for 10 msec before making them available for decoding into voice The coder polls frames from the buffer at regular intervals in order to produce continuous spee
547. start Immediate start delay start loop start ground start Feature Group B D for E1 T1 DTMF TIA 464A MF R1 MFC R2 User defined Call Progress Tones 1 2 4 8 or 16 E1 T1 J1 Balanced 120 100 Ohm or 75 Ohm using a BNC to RJ 45 dual E1 T1 G 703 Balun adapter Note The following Balun adaptors were tested and certified by AudioCodes Manufacture Name AC amp E Part Number B04040072 Manufacture Name RIT Part Number R3712271 410 Document LTRT 68808 SIP User s Manual Function Network Interface RS 232 Interface LED Indicators LED Indications on Front Panel Connectors amp Switches Rear Panel Trunks 1 to 8 and 9 to 16 Ethernet 1 and 2 RS 232 AC Power DC Power Physical AC Power Supply AC Power Consumption DC Power Supply optional DC Power Consumption Environmental DC Version 5 6 12 Selected Technical Specifications Specification Two 10 100Base TX half or full duplex with auto negotiation RS 232 terminal interface provided by DB 9 connector on rear panel available only on the 1 2 and 4 span configurations Power ACT Fail T1 E1 status LAN status Swap ready indication Two 50 pin female Telco connectors DDK57AE 40500 21D or 8 RJ 48c connectors for trunks 1 to 8 only Two 10 100Base TX RJ 45 shielded connectors DB 9 Console port Standard IEC320 Appliance inlet Dual fully redundant power supply optional 2 pin terminal block screw c
548. sual If not the start up application tries the secondary Ethernet interface If this interface is connected the whole start up procedure is performed using it If both interfaces are not connected the start up procedure commences using the parameters tables and software residing on the device s non volatile memory Note that Ethernet switchover occurs only once during the start up procedure at the beginning If the Ethernet interface fails after the selection is made the device does not switch over to the second port After start up is complete and the operational software is running the device continues to use the Ethernet port used for software upload The device switches over from one Ethernet port to the other each time an Ethernet link carrier loss is detected on the active Ethernet port and if the Ethernet link of the other port is operational Switchover occurs only once per link loss i e the secondary interface stays the active one even if the primary interface has returned to life After start up the device generates a gratuitous ARP message each time a switchover occurs For correct functionality of the redundancy mechanism it s recommended to configure both links to the same mode It is essential that both link partners primary and secondary have the same capabilities This ensures that whenever a switchover occurs the device is able to provide at least the same Ethernet services as were provided prior to t
549. submenu gt Tel to IP Routing page item Figure 3 71 Tel to IP Routing Page Routing Index 110 4 Tel To IP Routing Mode Rote cals betore marqatation Dest Phone Prefix Source Phone Prefrx gt Dest IP Address Dest IP Group 1D IP Profile 10 10 33 45 63 10 33 45 60 1033 45 64 SIP User s Manual 176 Document LTRT 68808 SIP User s Manual 3 Web Based Management 2 From the Routing Index drop down list select the range of entries that you want to add 3 Configure the Tel to IP Routing table according to the table below 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 44 Tel to IP Routing Table Parameters Description Parameter Tel to IP Routing Mode RouteModeTel2IP Src Trunk Group ID PREFIX_SrcTrunkGroupID Dest Phone Prefix PREFIX_DestinationPrefix Source Phone Prefix PREFIX_SourcePrefix Description Determines whether to route Tel calls to IP before or after manipulation of destination number 0 Route calls before manipulation Tel to IP calls are routed before the number manipulation rules are applied default 1 Route calls after manipulation Tel to IP calls are routed after the number manipulation rules are applied Notes Not applicable if outbound Proxy routing is used The source Trunk Group for Tel to IP calls The range is 1 99 Notes If this para
550. swapped numbers Applicable for Tel to IP calls 0 Disabled default 1 Swap calling and called numbers For a description of this parameter refer to SIP General Parameters on page 121 This ini file table parameter manipulates the destination number of Tel to IP calls The format of this parameter is as follows NumberMapTel2Ip FORMAT NumberMapTel2lp_Index NumberMapTel2lp_ DestinationPrefix NumberMapTel2Ip_ SourcePrefix NumberMapTel2Ilp SourceAddress NumberMapTel2lp_ NumberType NumberMapTel2lp_NumberPlan NumberMapTel2lp_ RemoveFromLeft NumberMapTel2lp_ RemoveFromRight NumberMapTel2 lp LeaveFromRight NumberMapTel2 p_Prefix2Add NumberMapTel2lp_Suffix2Add NumberMapTel2Ip_IsPresentationRestricted NumberMapTel2Ip_SreTrunkGroupID NumberMapTel2Ip__ SrcIPGroupID NumberMapTel2Ip For example NumberMapTel2Ip NumberMapTel2Ip 0 01 0 0 2 971 9 NumberMapTel2Ip 1 10 10 255 255 3 0 5 100 255 NumberMapTel2Ip Notes This table parameter can include up to 100 indices The parameters SourceAddress and IsPresentationRestricted are not applicable Set these to The parameter RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add LeaveFromRight NumberType and NumberPlan are applied if the called and calling numbers match the DestinationPrefix and SourcePrefix conditions The manipulation rules are executed in the following order RemoveFromLeft Rem
551. sy the call is released 6 By Source Phone Number Selects the device s channel according to the calling number 7 Trunk Cyclic Ascending Selects the device s port from the first channel of the next trunk next to the trunk from which the previous channel was allocated Notes The internal numbers of the device s B channels are defined by the TrunkGroup parameter For defining the channel select mode per Trunk Group refer to Configuring the Trunk Group Settings on page 197 122 Document LTRT 68808 SIP User s Manual Parameter Enable Early Media EnableEarlyMedia 183 Message Behavior SIP183Behaviour Session Expires Time SIPSessionExpires Minimum Session Expires MinSE Session Expires Method SessionExpiresMeth od Asserted Identity Mode AssertedidMode Version 5 6 3 Web Based Management Description Enables the device to send a 183 Session Progress response with SDP instead of 180 Ringing allowing the media stream to be established prior to the answering of the call 0 Disable Early Media is disabled default 1 Enable Enables Early Media Sending a 183 response depends on the Progress Indicator PI It is sent only if Pl is set to 1 or 8 are received in Proceeding or Alert PRI messages For CAS devices see the ProgressIndicator2IP parameter Defines the ISDN message that is sent when the 183 Session Progress message is received for IP to Tel calls
552. t 20 default 40 60 80 100 120 20 default 40 60 80 100 120 10 20 default 30 40 50 60 80 100 120 10 20 default 30 40 50 60 80 100 120 N A Rate Variable 0 default 1 8 1 1 2 3 Full 4 15 default 13 Always 13 Always 13 Always 64 Always 64 Always 64 N A Payload Type Dynamic 0 120 Dynamic 0 120 Always 3 Always 12 Dynamic 0 120 Dynamic 0 120 Dynamic 0 120 N A Mediant 2000 Silence Suppression Disable 0 Enable 1 Disable 0 Enable 1 Disable 0 Enable 1 Disable 0 Enable 1 Disable 0 Enable 1 N A N A N A gt To configure the device s coders take these 9 steps 1 Open the Coders page Configuration tab gt Protocol Configuration menu gt Protocol Definition submenu gt Coders page item Coder Name Figure 3 62 Coders Page Packetization Time Payload Type Silence Suppression 30 Disabled 2 From the Coder Name drop down list select the coder you want to use For the full list of available coders and their corresponding attributes refer to the table below 3 From the Packetization Time drop down list select the packetization time in msec for the selected coder The packetization time determines how many coder payloads are combined into a single RTP packet 4 Fro
553. t LTRT 68808 SIP User s Manual 3 Web Based Management 3 4 3 2 Configuring the CAS State Machines The CAS State Machine page allows you to modify various timers and other basic parameters to define the initialization of the CAS state machine without changing the state machine itself no compilation is required The change doesn t affect the state machine itself but rather the configuration gt To modify the CAS state machine parameters take these 6 steps 1 Open the CAS State Machine page Configuration tab gt PSTN Settings menu gt CAS State Machines page item Figure 3 49 CAS State Machine Page Gi te Digit t T CAS Table Neme enerate Digit Lewerete later OTM Mex DTMF Min Mex lecaming Man Incoming Collect ANI Time Detection Time Detection Thae Address Digits ANI Digits Digit Sigasling inir aen Related Trunks 2 Ensure that the trunk is inactive The trunk number displayed in the Related Trunks field must be green If it is red indicating that the trunk is active click the trunk number to open the Trunk Settings page refer to Configuring the Trunk Settings on page 82 select the required Trunk number icon and then click Stop Trunk 3 In the CAS State Machine page modify the required parameters according to the table below 4 Once you have completed the configuration activate the trunk if required in the Trunk Settings page by clicking the trunk number in the Related Trun
554. t Account registration fails SIP 4xx or no response then that endpoint is set to out of service until a success response is received in a subsequent registration request When the registration is per the entire device i e AuthenticationMode is set to 1 and registration fails all endpoints are set to out of service If all the Accounts of a specific Trunk Group fail registration and if the Trunk Group comprises a complete trunk then the trunk is set to out of service Determines the mode for Challenge Caching which reduces the number of SIP messages transmitted through the network The first request to the Proxy is sent without authorization The Proxy sends a 401 407 response with a challenge This response is saved for further uses A new request is resent with the appropriate credentials Subsequent requests to the Proxy are automatically sent with credentials calculated from the saved challenge If the Proxy doesn t accept the new request and sends another challenge the old challenge is replaced with the new one 0 None Challenges are not cached Every new request is sent without preliminary authorization If the request is challenged a new request with authorization data is sent default 1 INVITE Only Challenges issued for INVITE requests are cached This prevents a mixture of REGISTER and INVITE authorizations 2 Full Caches all challenges from the proxies Note Challenge Caching is used with all proxie
555. t Software Upgrade Key as a precaution so that you can re load this backup key to restore the device s original capabilities if the new key doesn t comply with your requirements a Inthe Current Key field copy the string of text and paste it in any standard text file b Save the text file to a folder on your PC with a name of your choosing 3 Open the new Software Upgrade Key file and ensure that the first line displays LicenseKeys and that it contains one or more lines in the following format S N lt serial number of the first or second module gt lt long Software Upgrade Key gt For example S N370604 jCx6r5tovCIKaBBbhPtT53Yj One S N must match the serial number of your device The device s serial number can be viewed in the Device Information page refer to Device Information on page 244 4 Follow one of the following procedures depending on whether you are loading a single or multiple key S N lines e Single key S N line a Open the Software Upgrade Key text file using for example Microsoft Notepad b Select and copy the key string of the device s S N and paste it into the field Add a Software Upgrade Key c Click the Add Key button SIP User s Manual 234 Document LTRT 68808 SIP User s Manual 3 Web Based Management e Multiple S N lines as shown below Figure 3 101 Software Upgrade Key with Multiple S N Lines ij sampleins Notepad OF WMPDE yenso4PbBF 8eOZAby
556. t and only from TE user to NT network To support interworking of the Hold Retrieve supplementary service from SIP to ISDN set EnableHold2ISDN to 1 Enable Hold to ISDN Enables interworking of the Hold Retrieve supplementary service from SIP EnableHold2ISDN to PRI 0 Disabled default 1 Enabled Notes This capability is supported only for QSIG and Euro ISDN variants To support interworking of the Hold Retrieve supplementary service from ISDN to SIP set the parameter EnableHold to 1 Version 5 6 159 November 2008 ca AudioCodes Parameter Hold Format HoldFormat Held Timeout HeldTimeout Enable Transfer EnableTransfer Transfer Prefix xferPrefix Enable Call Forward EnableForward Enable Call Waiting EnableCallWaiting SIP User s Manual Mediant 2000 Description Determines the format of the call hold request 0 0 0 0 0 The connection IP address in SDP is 0 0 0 0 default 1 Send Only The SDP contains the attribute a sendonly Note This parameter is applicable only to QSIG and Euro ISDN protocols Determines the time interval that the device can allow a call to remain on hold If a Resume un hold Re INVITE message is received before the timer expires the call is renewed If this timer expires the call is released 1 The call is placed on hold indefinitely until the initiator of on hold retrieves the call again default 0 2400 Time to wait in s
557. t run time you must stop the trunk and then restart it for the update to take effect To determine the method for setting Out Of Service state per trunk use the DigitalOOSBehaviorFor Trunk_ID parameter refer to Trunk Settings on page 82 212 Document LTRT 68808 SIP User s Manual Parameter 3 Web Based Management Description MLPP Multilevel Precedence and Preemption Note For additional MLPP parameters refer to Supplementary Services on page 159 MLPP Default Namespace MLPPDefaultNamespace Default Call Priority SIPDefaultCallPriority Preemption Tone Duration PreemptionToneDuration Version 5 6 Determines the Namespace used for MLPP calls received from the ISDN side and destined for the Application server The Namespace value is not present in the Precedence IE of the PRI SETUP message Therefore the value is used in the Resource Priority header of the outgoing SIP INVITE request 1 DSN DSN default 2 DOD DOD 3 DRSN DRSN Defines the default call priority for MLPP calls 0 0 ROUTINE default 2 2 PRIORITY 6 6 IMMEDIATE 8 8 FLASH OVERRIDE 9 9 FLASH OVERRIDE OVERRIDE If the incoming SIP INVITE request doesn t contain a valid priority value in the SIP Resource Priority header the default value is used in the Precedence IE after translation to the relevant ISDN Precedence value of the outgoing PRI SETUP message If the incoming PRI
558. t string is trapuser 3 5 1 1 3 Configuring SNMP V3 Users The SNMP V3 Settings page allows you to configure authentication and privacy for up to 10 SNMP v3 users gt To configure the SNMP v3 users take the following 6 steps 1 Access the Management Settings page as described in Configuring the Management Settings on page 220 2 In the SNMP V3 Table field click the right pointing arrow u button the SNMP V3 Settings page appears Figure 3 93 SNMP V3 Setting Page Add Apply User Name Authentication Protocol Privacy Protocol Authentication Key Privacy Key Group L None v None Read VWrite 3 To add an SNMP v3 user in the Add field enter the desired row index and then click Add A new row appears 4 Configure the SNMP V3 Setting parameters according to the table below 5 Click the Apply button to save your changes 6 To save the changes refer to Saving Configuration on page 230 For a description of the web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 30 You can also configure SNMP v3 users using the ini file table parameter SNMPUsers refer to SNMP Parameters on page 282 Version 5 6 225 November 2008 A Ee AudioCodes Mediant 2000 Table 3 60 SNMP V3 Users Parameters Parameter Index SNMPUsers_Index User Name SNMPUsers_Username Authentication Protocol SNMPUsers_AuthProtocol Privac
559. t to define the default NAT binding lifetime in seconds STUN is used to refresh the binding information after this time expires Version 5 6 381 November 2008 A Ee AudioCodes Mediant 2000 STUN only applies to UDP doesn t support TCP and TLS STUN can t be used when the device is located behind a symmetric NAT Use either the STUN server IP address STUNServerPrimaryIP or domain name STUNServerDomainName method with priority to the 8 3 2 8 3 3 first one First Incoming Packet Mechanism If the remote device resides behind a NAT device it s possible that the device can activate the RTP RTCP T 38 streams to an invalid IP address UDP port To avoid such cases the device automatically compares the source address of the incoming RTP RTCP T 38 stream with the IP address and UDP port of the remote device If the two are not identical the transmitter modifies the sending address to correspond with the address of the incoming stream The RTP RTCP and T 38 can thus have independent destination IP addresses and UDP ports You can disable the NAT mechanism by setting the ini file parameter DisableNAT to 1 The two parameters EnablelpAddrTranslation and EnableUdpPortTranslation allow you to specify the type of compare operation that occurs on the first incoming packet To compare only the IP address set EnablelpAddrTranslation to 1 and EnableUdpPortTranslation to 0 In this case if the first incoming packet arrives with
560. table parameter InterfaceT able refer to Networking Parameters on page 260 gt To configure the multiple IP interface table take these 7 steps 1 Open the IP Settings page refer to Configuring the IP Settings on page 50 2 Under the Multiple Interface Settings group click the right arrow gt button alongside Multiple Interface Table a confirmation message box appears Figure 3 34 Confirmation Message for Accessing the Multiple Interface Table rc Microsoft Internet Explorer 2 If switching to the advanced interface configuration mode the current page wil no longer be available Are you sure you want to continue Cee 3 Click OK to confirm the Multiple Interface Table page appears Figure 3 35 Interface Table Page Index Application Type IP Address Prefix Length Gateway VLAN ID Interface Name 1 1 ar v 10 13 413 6 i0301 a Al interfaces v VLAN Mode Native VLAN ID 4 in the Add field enter the desired index number for the new interface and then click Add the index row is added to the table 5 Configure the interface according to the table below 6 Click the Apply button the interface is immediately applied to the device 7 To save the changes to flash memory refer to Saving Configuration on page 230 SIP User s Manual 54 Document LTRT 68808 SIP User s Manual 3 Web Based Management When adding more than one interface to the table e
561. tact header includes a tgrp parameter it is copied to the corresponding outgoing messages in that dialog Determines whether the Globally Routable User Agent URIs GRUU mechanism is used 0 Disable Disable default 1 Enable Enable The device obtains a GRUU by generating a normal REGISTER request This request contains a Supported header with the value gruu The device includes a sip instance Contact header parameter for each contact for which the GRUU is desired This Contact parameter contains a globally unique ID that identifies the device instance The global unique ID is as follows If registration is per endpoint AuthenticationMode 0 it is the MAC address of the device concatenated with the phone number of the endpoint Ifthe registration is per device AuthenticationMode 1 it is only the MAC address When the User Information mechanism is used the globally unique ID is the MAC address concatenated with the phone number of the endpoint defined in the User Info file If the Registrar Proxy supports GRUU the REGISTER responses contain the gruu parameter in each Contact header field The Registrar Proxy provides the same GRUU for the same AOR and instance id in case of sending REGISTER again after expiration of the registration The device places the GRUU in any header field which contains a URI It uses the GRUU in the following messages INVITE requests 2xx responses to INVITE SUBSCRIBE req
562. ter refer to Configuring the Trunk Settings on page 82 Same as the description for parameter ISDNIBehavior but for a specific trunk ID Same as the description for parameter ISDNInCallsBehavior for a specific trunk ID Same as the description for parameter ISDNOutCallsBehavior but for a specific trunk ID For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to SIP General Parameters on page 121 For a description of this parameter refer to Advanced Parameters on page 151 Busy or Reorder Tone duration that the device when configured to protocol type CAS plays before releasing the line The valid range is 0 to 15 The default value is 10 seconds Applicable also to ISDN if PlayBusyTone2ISDN 2 Selection of Busy or Reorder tone is done according to release cause received from IP For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 1 The device sends 200 OK to INVITE messages when speech fax modem is detected from the Tel side 0 The device sends 200 OK messages immediately after the device finishes dialing to the Tel side default Usually this feature is used only when early media EnableEarlyMedia is used to establish voice path before the call is answered No
563. tes To activate this feature set EnableDSPIPMDetectors to 1 This feature is applicable only when the protocol type is CAS 306 Document LTRT 68808 SIP User s Manual Parameter DigitMapping TimeBetweenDigits MaxDigits TimeForDialTone RegretTime 4 4 11 ISDNand CAS 4 ini File Configuration Description For a description of this parameter refer to DTMF amp Dialing Parameters on page 147 For a description of this parameter refer to DTMF amp Dialing Parameters on page 147 For a description of this parameter refer to DTMF amp Dialing Parameters on page 147 For a description of this parameter refer to DTMF amp Dialing Parameters on page 147 For a description of this parameter refer to Advanced Parameters on page 151 Interworking Related Parameters The ISDN and CAS related ini file configuration parameters are described in the table below Table 4 11 ISDN and CAS Interworking Related ini File Parameters Parameter EnableTDMoverlIP EnablelSDNTunnelingTel2IP EnablelSDNTunnelingIP2Tel ISDNDuplicateQ931 BuffMode EnableQSIGTunneling PlayRBTone2Trunk_ID DigitalOOSBehaviorFor Trunk_ID DigitalOOSBehavior RemoveCallingName DefaultCauseMapISDN2IP Version 5 6 Description For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Pa
564. the Channel Identifier parameter also for the Primary trunk The parameter ISDNNFASInterfacelD_x ID can define the Interface ID for any Primary T1 trunk even if the T1 trunk is not a part of an NFAS group However to include the Interface Identifier in Q 931 Setup Channel Identification IE configure ISDNIBehavior_x 2048 in the ini file Version 5 6 399 November 2008 A gA AudioCodes Mediant 2000 9 4 2 Working with DMS 100 Switches The DMS 100 switch requires the following NFAS Interface ID definitions InterfacelD 0 for the Primary trunk InterfacelD 1 for the Backup trunk InterfacelD 2 for a 24 B channel T1 trunk InterfacelD 3 for a 24 B channel T1 trunk and so on for subsequent T1 trunks For example if four T1 trunks on a device are configured as a single NFAS group with Primary and Backup T1 trunks that is used with a DMS 100 switch the following parameters should be used NFASGroupNumber 0 1 NFASGroupNumber 1 1 NFASGroupNumber 2 1 NFASGroupNumber 3 1 DchConfig 0 0 Primary T1 trunk Decon igii S E Backup Tl trunk DchConfig 2 2 B Channel NFAS trunk Dencontigi o rai B channel NFAS trunk If there is no NFAS Backup trunk the following configuration should be used ISDNNFASInterfaceID 0 0 ISDNNFASInterfaceID 1 2 ISDNNFASInterfaceID 2 3 ISDNNFASInterfaceID 3 4 ISDNIBehavior 512 This parameter should be added because of ISDNNFASInterfaceID coniguration
565. the Management VLAN ID and Bronze CoS Each of these parameters can be configured with a 802 1p Q value traffic type to VLAN ID and CoS to 802 1p priority Figure 8 2 Multiple Network Interfaces and VLANs i Router AudioCodes Media Gateway ok en ee Network Internet Separated Networks Scheme For security the VLAN mechanism is activated only when the device is loaded from the flash memory Therefore when using BootP Load an inifile with VianMode set to 1 and SaveConfiguration set to 1 Then after the device is active reset the device with TFTP disabled or by using any method except for BootP For information on how to configure VLAN parameters refer to Configuring the IP Settings on page 50 The device must be connected to a VLAN aware switch and the switch s PVID must be equal to the device s native VLAN ID The mapping of an application to its CoS and traffic type is shown in the table below Table 8 1 Traffic Network Types and Priority Application Traffic Network Types Class of Service Priority Debugging interface Management Bronze Telnet Management Bronze SIP User s Manual 386 Document LTRT 68808 SIP User s Manual 8 Networking Capabilities Application Traffic Network Types Class of Service Priority DHCP Management Network Web server HTTP Management Bronze SNMP GET SET Management Bronze Web server HTTPS Management Bronze IPSec IKE Determined by the service Determined by the
566. the Navigation bar click the Configuration or Management tab to display their respective menus in the Navigation tree 5 In the Navigation tree select the required page item for the Step and then in the page itself select the required parameters by selecting the check boxes corresponding to the parameters 6 Inthe Step Name field enter a name for the Step SIP User s Manual 34 Document LTRT 68808 SIP User s Manual 3 Web Based Management 7 Click the Next button located at the bottom of the page the Step is added to the Scenario and appears in the Scenario Step list Figure 3 15 Creating a Scenario Status Conteqantee Menegemert 3 Disyortics Scenarios Search Selected Parameter Seals Parenatertle Basic Full Max Depts In Phone Num Dig Teneout for Ov p ing MB network Settings Inter Digit Teneout for Overlap Dialing sec Ul Medes Setongs Declare AFC 2033 in SOP t Wsecunty Setungs ist Tx OTMF Option i Protocol Configuration 2nd Tx OTMF Option wd Protocol Oeafirebon 3rd Tx DTMF Option SIP General Parameters 4th Tx OTMF Option Proxy Registraben E9 m T Sth Tx OTMF Option Feed Selected eases DTMF Dialing 3 RFC 2833 Payload Type PUBSIP Advanced Parameters Page Mook Flash Option B Marcpulation Tables Cigt Mapping fules routing Tables ul Profile Onfrebons ublendooen Setinas Scenario Name PBX Enadle Special Digits Interoperability i Define Coders Deal Tone D
567. the call duration is zero Indicates the number of calls that were terminated due to a call forward The counter is incremented as a result of the following release reason RELEASE_BECAUSE_FORWARD Indicates the number of calls whose destinations weren t found It is incremented as a result of one of the following release reasons GWAPP_UNASSIGNED_NUMBER 1 GWAPP_NO_ROUTE_TO_DESTINATION 3 Indicates the number of calls that failed due to mismatched device capabilities It is incremented as a result of an internal identification of capability mismatch This mismatch is reflected to CDR via the value of the parameter DefaultReleaseReason default is GWAPP_NO_ROUTE_TO_DESTINATION 3 or by the GWAPP_SERVICE_NOT_IMPLEMENTED_UNSPECIFIED 79 reason Indicates the number of calls that failed due to unavailable resources or a device lock The counter is incremented as a result of one of the following release reasons GWAPP_RESOURCE_UNAVAILABLE_UNSPECIFIED RELEASE_BECAUSE_GW_LOCKED This counter is incremented as a result of calls that failed due to reasons not covered by the other counters The average call duration ACD in seconds of established calls The ACD value is refreshed every 15 minutes and therefore this value reflects the average duration of all established calls made within a 15 minute period Indicates the number of attempted fax calls Indicates the number of successful fax calls 3 6 2 2 Call Routing St
568. the coder you selected is dynamic enter a value from 0 to 120 payload types of well known coders cannot be modified The payload type identifies the format of the RTP payload Version 5 6 191 November 2008 A ge AudioCodes Mediant 2000 7 From the Silence Suppression drop down list enable or disable the silence suppression option for the coder you selected 8 Repeat steps 3 through 7 for the second to fifth coders optional 9 Repeat steps 2 through 8 for the second to fourth coder groups optional 10 Click the Submit button to save your changes 11 To save the changes to flash memory refer to Saving Configuration on page 230 3 4 7 5 2 Tel Profile Settings The Tel Profile Settings page allows you to define up to nine different Tel Profiles You can then assign these Tel Profiles to the device s channels in the Trunk Group Table page thereby applying different behaviors to different channels Note You can also configure Tel Profiles using the ini file table parameter TelProfile refer to SIP Configuration Parameters on page 284 gt To configure Tel Profiles take these 9 steps 1 Open the Tel Profile Settings page Configuration tab gt Protocol Configuration menu gt Profile Definitions submenu gt Tel Profile Settings page item Figure 3 80 Tel Profile Settings Page v Profile ID Profile Name w Profile Parameters Profile Preference Fax
569. the following ports RTP 6000 RTCP 6001 and T 38 6002 2 the second channel uses RTP 6010 RTCP 6011 and T 38 6012 etc Note If RTP Base UDP Port is not a factor of 10 the following message is generated invalid local RTP port For detailed information on the default RTP RTCP T 38 port allocation refer to the Product Reference Manual Determines the lower boundary of UDP ports used for RTP RTCP and T 38 by a remote device If this parameter is set to a non zero value ThroughPacket RTP multiplexing is enabled The device uses this parameter and BaseUDPPort to identify and distribute the payloads from the received multiplexed IP packet to the relevant channels The valid range is the range of possible UDP ports 6 000 to 64 000 The default value is 0 i e RTP multiplexing is disabled For detailed information on RTP multiplexing refer to RTP Multiplexing ThroughPacket on page 360 Notes The value of this parameter on the local device must equal the value of BaseUDPPort on the remote device To enable RTP multiplexing the parameters L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort must be set to a non zero value When VLANs are implemented RTP multiplexing is not supported Determines the local UDP port used for outgoing multiplexed RTP packets applies to RTP multiplexing The valid range is the range of possible UDP ports 6 000 to 64 000 The default value is 0 i e RTP multiplexing is disabl
570. the parameters IP address and port SNMPManagerisUsed_x number of the corresponding SNMP Manager used to receive SNMP IP Address traps 0 Check box cleared Disabled default 1 Check box selected Enabled IP address of the remote host used as an SNMP Manager The device SNMPManagerTablelP_x sends SNMP traps to these IP addresses Trap Port Enter the IP address in dotted decimal notation e g 108 10 1 255 Defines the port number of the remote SNMP Manager The device SNMPManagerTrapPort_x sends SNMP traps to these ports Version 5 6 The valid SNMP trap port range is 100 to 4000 The default port is 162 223 November 2008 A ge AudioCodes Mediant 2000 Parameter Description Trap Enable Activates or de activates the sending of traps to the corresponding SNMPManagerTrapSendi SNMP Manager ngEnable_x 0 Disable Sending is disabled 1 Enable Sending is enabled default 3 5 1 1 2 Configuring the SNMP Community Strings The SNMP Community String page allows you to configure up to five read only and up to five read write SNMP community strings and to configure the community string that is used for sending traps For detailed information on SNMP community strings refer to the Product Reference Manual gt Toconfigure the SNMP community strings take these 5 steps 1 Access the Management Settings page as described in Configuring the Management Settings on page 22
571. the play of announcements from IP to PSTN without the need to answer the Tel to IP call It can be used with PSTN networks that don t support the opening of a TDM channel before an ISDN Connect message is received 0 Connect message isn t sent after SIP 183 Session Progress message is received default 1 Connect message is sent after SIP 183 Session Progress message is received For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Trunk Settings on page 82 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 Determines the format of the subaddress value for ISDN Calling and Called numbers 0 ASCII default 309 November 2008 ca AudioCodes Parameter EnableHold2ISDN EnableUUITel2IP EnableUUIIP2Tel ScreeningInd2IP SupportRedirectinFacility EnableCiC EnableAOC SIP User s Manual Mediant 2000 Description 1 BCD Binary Coded Decimal 2 User Specified For IP to Tel calls if the incoming SIP INVITE message includes subaddress values in the isub parameter fo
572. the rule that you want to edit 2 Modify the fields as desired 3 Click the Apply button to save the changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 gt To activate a de activated rule take these 2 steps 1 Inthe Edit Rule column select the de activated rule that you want to activate 2 Click the Activate button the rule is activated gt To de activate an activated rule take these 2 steps 1 In the Edit Rule column select the activated rule that you want to de activate 2 Click the DeActivate button the rule is de activated gt To delete a rule take these 3 steps 1 Select the radio button of the entry you want to activate 2 Click the Delete Rule button the rule is deleted 3 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 24 Internal Firewall Parameters Parameter Description Is Rule Active Source IP AccessList_Source_IP Subnet Mask A read only field indicating whether the rule is active or not Note After device reset all rules are active IP address or DNS name of source network or a specific host IP network mask 255 255 255 255 for a single host or the AccessList_Net_Mask appropriate value for the source IP addresses The IP address of the sender of the incoming packet is bitwise ANDed with this mask and then compared to the field Source IP Local Port Range The destination UDP TCP ports on this
573. the source of incoming INVITE messages for IP to Tel routing 1 Not configured default 0 SIP Contact Header Use the IP address received in the Contact header of the incoming INVITE message 1 Layer 3 Source IP Use the actual IP address Layer 3 from which the SIP packet was received Note If the IP to IP feature is enabled i e supported by the Feature Key and EnableSBC is set to 1 refer to SBC Configuration on page 163 this parameter is automatically set to 1 If the IP to IP feature is disabled this parameter is automatically set to 0 173 November 2008 ca AudioCodes Parameter Enable Alt Routing Tel to IP AltRoutingTel2IPEnable Alt Routing Tel to IP Mode AltRoutingTel2IPMode Alt Routing Tel to IP Connectivity Method AltRoutingTel2IPConnMethod Alt Routing Tel to IP Keep Alive Time AltRoutingTel2IPKeepAliveTime Max Allowed Packet Loss for Alt Routing IPConnQoSMaxAllowedPL SIP User s Manual Mediant 2000 Description Enables the Alternative Routing feature for Tel to IP calls 0 Disable Disables the Alternative Routing feature default 1 Enable Enables the Alternative Routing feature 2 Status Only The Alternative Routing feature is disabled but read only information on the Quality of Service of the destination IP addresses is provided For information on the Alternative Routing feature refer to Configuring Alternative Routing B
574. time attribute and then from default value Determines the device behavior when Transfer is initiated while in Alerting state 0 Disable Send REFER with Replaces default 1 Enable Send CANCEL and after a 487 response is received send REFER without Replaces 130 Document LTRT 68808 SIP User s Manual Parameter 3xx Behavior 8xxBehavior Enable P Charging Vector EnablePChargingVec tor Enable VoiceMail URI EnableVMURI Retry After Time RetryAfterTime Enable P Associated URI Header EnablePAssociatedU RiHeader Source Number Preference SourceNumberPrefer ence Version 5 6 3 Web Based Management Description Determines the device s behavior regarding call identifiers when a 3xx response is received for an outgoing INVITE request The device can either use the same call identifiers Call ID Branch To and From tags or change them in the new initiated INVITE 0 Forward Use different call identifiers for a redirected INVITE message default 1 Redirect Use the same call identifiers Enables the addition of a P Charging Vector header to all outgoing INVITE messages 0 Disable Disable default 1 Enable Enable Enables or disables the interworking of target and cause for redirection from Tel to IP and vice versa according to RFC 4468 0 Disable Disable default 1 Enable Enable Upon receipt of an ISDN SETUP message with redirect v
575. tion Proxy Set ID SIP Group Name Contact User SIP User s Manual Table 3 51 IP Group Parameters Description Description The IP Group can be defined as one of the following types SERVER used when the destination address configured by the Proxy Set of the IP Group e g ITSP Proxy IP PBX or Application server is known USER represents a group of users such as IP phones and softphones where their location is dynamically obtained by the device when REGISTER requests and responses traverse or are terminated by the device These users are considered remote far end users Typically this IP Group is configured with a Serving IP Group that represents an IP PBX Application or Proxy server that serves this USER type IP Group Each SIP request sent by a user of this IP Group is proxied to the Serving IP Group For registrations the device updates its internal database with the AOR and contacts of the users Digest authentication using SIP 401 407 responses if needed is performed by the Serving IP Group The device forwards these responses directly to the SIP users To route a call to a registered user a rule must be configured in the Outbound IP Routing table refer to Outbound IP Routing Table on page 178 The device searches the dynamic database by using the request URI for an entry that matches a registered AOR or Contact Once an entry is found the IP destination is obtained from this entry and
576. tion The device s Ethernet connection can be configured using the ini file parameter EthernetPhyConfiguration for one of the following modes Manual mode e 10Base T Half Duplex or 10Base T Full Duplex e 100Base TX Half Duplex or 100Base TX Full Duplex m Auto Negotiation chooses common transmission parameters such as speed and duplex mode The Ethernet connection should be configured according to the following recommended guidelines m When the device s Ethernet port is configured for Auto Negotiation the opposite port must also operate in Auto Negotiation Auto Negotiation falls back to Half Duplex mode when the opposite port is not in Auto Negotiation mode but the speed i e 10 100Base T or 1000Base TX in this mode is always configured correctly Configuring the device to Auto Negotiation mode while the opposite port is set manually to Full Duplex is invalid as it causes the device to fall back to Half Duplex mode while the opposite port is Full Duplex Any mismatch configuration can yield unexpected functioning of the Ethernet connection m When configuring the device s Ethernet port manually the same mode i e Half Duplex or Full Duplex and speed must be configured on the remote Ethernet port In addition when the device s Ethernet port is configured manually it is invalid to set the remote port to Auto Negotiation Any mismatch configuration can yield unexpected functioning of the Ethernet connection m It s recommende
577. tion IP to IP calls are routed after the number manipulation rules are applied Note Not applicable if outbound Proxy routing is used Version 5 6 179 November 2008 ca AudioCodes Parameter Src IPGroupID PREFIX_SrclPGroupID Src Host Prefix PREFIX_SrcHostPrefix Dest Host Prefix PREFIX_DestHostPrefix Src Trunk Group ID PREFIX_SrcTrunkGroupID Dest Phone Prefix PREFIX_DestinationPrefix Source Phone Prefix PREFIX_SourcePrefix Mediant 2000 Description The IP Group ID from where the IP to IP call originated Typically the IP Group of an incoming INVITE is determined according to the Inbound IP Routing table To denote all IP Groups leave the field empty Notes If this Source IP Group has a Serving IP Group then all calls originating from this Source IP Group are sent to the Serving IP Group In this scenario this table is used only if the parameter PreferRouteTable is set to 1 For defining IP Groups refer to Configuring the IP Groups on page 201 The prefix of the SIP URI host name in the From header of the incoming SIP INVITE message If this routing rule is not required leave the field empty To denote any prefix use the asterisk symbol The request SIP URI host name prefix of the incoming SIP INVITE message If this routing rule is not required leave the field empty To denote any prefix use the asterisk symbol The source Trunk Group 1 99 for Tel to IP ca
578. tion The CIDR style presentation is the latest method for interpretation of IP addresses Specifically instead of using eight bit address blocks it uses the variable length subnet masking technique to allow allocation on arbitrary length prefixes refer to http en wikipedia org wiki Classless_Inter Domain_Routing for more information The prefix length values range from 0 to 31 Defines the IP address of the default gateway used by the device Notes Only one default gateway can be configured for the device and it must be configured on an interface for Media traffic All other table entries for this column must have the value 0 0 0 0 The default gateway s IP address must be in the same subnet as the interface address For configuring additional routing rules for other interfaces refer to Configuring the IP Routing Table on page 62 Defines the VLAN ID for each interface When using VLANs the VLAN ID must be unique for each interface Incoming traffic tagged with this VLAN ID is routed to the corresponding interface and outgoing traffic from that interface is tagged with this VLAN ID Defines a string up to 16 characters to name this interface This name is displayed in management interfaces Web CLI and SNMP for better readability and has no functional use Note The interface name is a mandatory parameter and must be unique for each interface For a description of this parameter refer to Configuring the IP S
579. tionTimeDivider is 50 If registration per B channel is selected on device startup the device sends REGISTER requests according to the maximum number of allowed SIP dialogs configured by the parameter NumberOfActiveDialogs After each received response the subsequent REGISTER request is sent SIP User s Manual 368 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 7 14 7 14 1 Configuration Examples SIP Call Flow The SIP call flow shown in the following figure describes SIP messages exchanged between two devices during a simple call In this call flow example device 10 8 201 158 with phone number 6000 dials device 10 8 201 161 with phone number 2000 Figure 7 2 SIP Call Flow Jz 10 8 201 10 10 8 201 108 pI Phone 1000 INVITE F1 Ringing F3 200 OK F4 m F1 10 8 201 108 gt gt 10 8 201 10 INVITE INVITE sip 1000 10 8 201 10 user phone SIP 2 0 Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacsiJkDGd From lt Sip 8000 10 8 201 108 gt tag 1 c5354 To lt sip 1000 10 8 201 10 gt Call ID 534366556655skKw 8000 1000 10 8 201 108 CSeq 18153 INVITE Contact lt sip 8000 10 8 201 108 user phone gt User Agent Audiocodes Sip Gateway Mediant 2000 v 5 40 010 006 Supported 100rel em Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO Content Type application sdp Content Length 208 v 0 o AudiocodesGW 18132 74003 IN IP4 10 8 201 108
580. to Configuring the CAS State Machines on page 97 For a description of this parameter refer to Configuring the CAS State Machines on page 97 For a description of this parameter refer to Configuring the CAS State Machines on page 97 For a description of this parameter refer to Configuring the CAS State Machines on page 97 Enables or disables the device s DSP detectors 0 Disable default 1 Enable Notes The device s Feature Key should contain the IPMDetector DSP option When enabled 1 the number of available channels is reduced by a factor of 5 6 For example a device with 8 E1 spans capacity is reduced to 6 spans 180 channels while a device with 8 T1 spans capacity remains the same 192 channels For a description of this parameter refer to Advanced Parameters on page 151 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 For a description of this parameter refer to Configuring the Digital Gateway Parameters on page 207 Overrides the T310 timer for the DMS 100 ISDN variant T310 defines the timeout between the reception of a PROCEEDING message and the reception of an ALERTING CONNECT message The valid range is 10 to 30 The default value is 10 seconds Note Applicable only to Nortel DMS and Nortel MERIDIAN PRI variants ProtocolType 14 and 35 312 Document LTRT 68808 SIP User s Manual Parameter
581. to Tel Disable Enable ISDN Tunneling Tel to IP Disable Enable QSIG Tunneling Disable Enable ISDN Tunneling IP to Tel Disable ISDN Transfer on Connect Alert Remove CLI when Restricted No HCAS Ka Remove Calling Name Disable E Default Cause Mapping From ISDN to SIP 0 Default Cause Mapping From ISDN to SIP 0 Add Prefix to Redirect Number Copy Destination Number to Redirect Number Don t copy Enable Calling Party Category Disable w MLPP MLPP Default Namespace Default Call Priority Preemption tone Duration Version 5 6 207 November 2008 ca AudioCodes Mediant 2000 2 Configure the Digital Gateway parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 53 Digital Gateway Parameters Description Parameter B channel Negotiation BchannelNegotiation Swap Redirect and Called Numbers SwapRedirectNumber MFC R2 Category R2Category Disconnect Call on Busy Tone Detection CAS DisconnectOnBusyTone Disconnect Call on Busy Tone Detection ISDN ISDNDisconnectOnBusyTone Enable TDM Tunneling EnableTDMoverlP SIP User s Manual Description Determines the ISDN B Channel negotiation mode 0 Preferred 1 Exclusive default 2 Any Notes Appl
582. to replace AudioCodes default Web logo with a user defined logo The file name can be up to 47 characters Security Parameters The security related ini file configuration parameters are described in the table below Table 4 4 Security ini File Parameters Parameter EnableMediaSecurity MediaSecurityBehaviour SRTPTxPacketMkKISize RTPAuthenticationDisableTx RTPEncryptionDisableTx RTCPEncryptionDisableTx EnableSIPS TLSLocalSIPPort TLSVersion TLSReHandshakelnterval SIPSRequireClientCertificate SIP User s Manual Description For a description of this parameter refer to Configuring Media Security on page 80 For a description of this parameter refer to Configuring Media Security on page 80 For a description of this parameter refer to Configuring Media Security on page 80 For a description of this parameter refer to Configuring Media Security on page 80 For a description of this parameter refer to Configuring Media Security on page 80 For a description of this parameter refer to Configuring Media Security on page 80 For a description of this parameter refer to General Parameters on page 151 For a description of this parameter refer to General Parameters on page 151 For a description of this parameter refer to Configuring the General Security Settings on page 109 For a description of this parameter refer to Configuring the General Security Settings on pag
583. troubleshooting procedures 1 Delete all cookies in the Temporary Internet Files folder If this does not resolve the problem the security settings may need to be altered continue with Step 2 In Internet Explorer navigate to Tools menu gt Internet Options gt Security tab gt Custom Level and then scroll down to the Logon options and select Prompt for username and password Select the Advanced tab and then scroll down until the HTTP 1 1 Settings are displayed and verify that Use HTTP 1 1 is selected 3 Quit and start the Web browser again SIP User s Manual 20 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 3 Getting Acquainted with the Web Interface The figure below displays the general layout of the Graphical User Interface GUI of the Web interface Figure 3 2 Main Areas of the Web Interface GUI F AudioCodes Microsoft Internet Explorer Fie Edt View Favortes Took heb GO tak jf E hetp 20 13 4 13 Title Bar _ 3 A Cod wee v Sarat Q Bun Device Actions 4 Home Hap a ro Status Contguaten Management SO cctes Soananes Seach Bore PoramoterUst a v Syslog Settings Basic O Full Syslog Server IP Address ull Management Settings Syslog Server Port Management Configuration Enable Syslog Regione Sethogs Memtenance Achons tu software Update a Activity Types to Report vie Activity Log Messages The Web GUI is composed of the following main
584. ts Therefore to achieve better performance during modem and fax calls the Optimization Factor should be set to 13 In this special mode the clock drift correction is performed less frequently only when the Jitter Buffer is completely empty or completely full When such condition occurs the correction is performed by dropping several voice packets simultaneously or by adding several BFI packets simultaneously so that the Jitter Buffer returns to its normal condition Configuring Alternative Routing Based on Connectivity and QoS The Alternative Routing feature enables reliable routing of Tel to IP calls when a Proxy isn t used The device periodically checks the availability of connectivity and suitable Quality of Service QoS before routing If the expected quality cannot be achieved an alternative IP route for the prefix phone number is selected Note If the alternative routing destination is the device itself the call can be configured to be routed back to one of the device s trunk groups and thus back into the PSTN PSTN Fallback 7 9 1 Alternative Routing Mechanism When a Tel to IP call is routed through the device the call s destination number is compared to the list of prefixes defined in the Tel to IP Routing table described in Tel to IP Routing Table on page 175 The Tel to IP Routing table is scanned for the destination number s prefix starting at the top of the table For this reason enter t
585. ts to Disable Throughout the DHCP procedure the BootP TFTP application 59 November 2008 A ge AudioCodes Mediant 2000 Parameter Description must be deactivated otherwise the device receives a response from the BootP server instead of from the DHCP server For additional information on DHCP refer to the Product Reference Manual DHCPEnable is a special Hidden parameter Once defined and saved in flash memory its assigned value doesn t revert to its default even if the parameter doesn t appear in the ini file 3 4 1 4 Configuring the NFS Settings Network File System NFS enables the device to access a remote server s shared files and directories and to handle them as if they re located locally You can configure up to five different NFS file systems As a file system the NFS is independent of machine types OSs and network architectures NFS is used by the device to load the cmp ini and auxiliary files using the Automatic Update mechanism refer to Automatic Update Mechanism Note that an NFS file server can share multiple file systems There must be a separate row for each remote file system shared by the NFS file server that needs to be accessed by the device gt To add remote NFS file systems take these 6 steps 1 Open the Application Settings page refer to Configuring the Application Settings on page 57 2 Under the NFS Settings group click the right arrow u button alongside NFS Table
586. tton to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 17 Media Settings Parameters Parameter Description Max Echo Canceller Length Determines the maximum Echo Canceler Length in msec MaxEchoCancellerLength which is the maximum echo path delay tail length for which the echo canceller is designed to operate 0 Default based on various internal device settings to attain maximum channel capacity default 11 64 msec 22 128 msec Notes Using 128 msec reduces the channel capacity to 200 channels Reset the device after modifying this parameter Itisn t necessary to configure the parameter EchoCancellerLength as it automatically acquires its value from this parameter Enable Continuity Tones N A Version 5 6 79 November 2008 A ge AudioCodes Mediant 2000 3 4 2 6 Configuring the DSP Templates The DSP Templates page allows you to assign up to two DSP templates to the device In addition you can define the percentage of DSP resources allocated per DSP template gt To select DSP templates take these 5 steps 1 Open the DSP Templates page Configuration tab gt Media Settings menu gt DSP Templates page item Figure 3 45 DSP Templates Page Index DSP Template Number DSP Resources Percentage o O a1 3 2 Select an index row by clicking the corresponding Index radio button 3 Click Edit
587. tworking Parameters on page 260 A default Gateway is supported only for the Media traffic type for Control and OAM traffic use the IP Routing table refer to Configuring the IP Routing Table on page 62 The IP address and subnet mask used in the Single IP Network mode are used for the OAM traffic type in the Multiple IP Network mode IEEE 802 1p Q VLANs and Priority The Virtual Local Area Network VLAN mechanism enables the device to be integrated into a VLAN aware environment that includes switches routers and endpoints When in VLAN enabled mode each packet is tagged with values that specify its priority class of service IEEE 802 1p and the identifier traffic type of the VLAN to which it belongs Media Control or OAMP IEEE 802 1Q The class of service CoS mechanism can be utilized to accomplish Ethernet Quality of Service QoS Packets sent by the device to the Ethernet network are divided into five different priority classes Network Premium Media Premium Control Gold and Bronze The priority of each class is determined by a corresponding ini file parameter Traffic type tagging can be used to implement Layer 2 VLAN security By discriminating traffic into separate and independent domains the information is preserved within the VLAN Incoming packets received from an incorrect VLAN are discarded The traffic tagging mechanism is as follows Outgoing packets from the device to the switch All
588. uage on which the device s AMD mechanism can base its voice detector algorithms for detecting these voices The data needed for an accurate calibration should be recorded under the following guidelines e Statistical accuracy The number of recordings should be large i e about 100 and varied The calls must be made to different people at different times The calls must be made in the specific location in which the device s AMD mechanism is to operate Real life recording The recordings should simulate real life answering of a person picking up the phone without the caller speaking until the AMD decision Normal environment interferences The environment should almost simulate real life scenarios i e not sterile but not too noisy either Interferences for example could include background noises of other people talking spikes and car noises SIP User s Manual 344 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities The SIP call flows below show an example of implementing the device s AMD feature This scenario example allows a third party Application server to play a recorded voice message to an answering machine 1 Upon detection by the device of the answering machine the device sends a SIP INFO message to the Application server INFO sip sipp 172 22 2 9 5060 SIP 2 0 Via SIP 2 0 UDP 172 22 168 249 branch z9hG4bKac1566945480 Max Forwards 70 From sut lt sip 3000 172 22 168 249 5060 gt 5 tag
589. uding the number of call attempts failed calls fax calls etc This menu includes the following page items m P to Tel Calls Count and Tel to IP Calls Count refer to Call Counters on page 248 Call Routing Status refer to Call Routing Status on page 250 m SAS SBC Registered Users refer to SAS SBC Registered Users on page 251 m P Connectivity refer to IP Connectivity on page 252 Note The Gateway Statistics pages don t refresh automatically To view updated information re access the required page 3 6 2 1 Call Counters The IP to Tel Calls Count and Tel to IP Calls Count pages provide you with statistical information on incoming IP to Tel and outgoing Tel to IP calls The statistical information is updated according to the release reason that is received after a call is terminated during the same time as the end of call Call Detail Record or CDR message is sent The release reason can be viewed in the Termination Reason field in the CDR message You can reset the statistical data displayed on the page i e refresh the display by clicking the Reset Counters button located on the page SIP User s Manual 248 Document LTRT 68808 SIP User s Manual 3 Web Based Management gt To view the IP to Tel and Tel to IP Call Counters pages take this step m Open the Call Counters page that you want to view Status amp Diagnostics tab gt Gateway Statistics menu gt IP to Tel Calls Count
590. udiocodes Sip Gateway Mediant 2000 v 5 40 010 006 Supported 100rel em Allow REGISTER OPTIONS INVITE ACK CANCEL BYE NOTIFY PRACK REFER INFO Content Type application sdp Content Length 206 0 o AudiocodesGW 30221 87035 IN IP4 10 8 201 10 s Phone Call GSEN m24 0 8 20 110 ic 0 0 m audio 7210 RTP AVP 8 96 a rtpmap 8 pcma 8000 a ptime 20 a rtpmap 96 telephone event 8000 GSemco g9G O 115 SIP User s Manual 370 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities m F5 10 8 201 108 gt gt 10 8 201 10 ACK ACK sip 1000 10 8 201 10 user phone SIP 2 0 Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacZYpJWxZ From lt Sip 8000 10 8 201 108 gt tag 1c5354 To lt Sip 1000 10 8 201 10 gt tag 1c7345 Call ID 534366556655skKw 8000 1000 10 8 201 108 User Agent Audiocodes Sip Gateway Mediant 2000 v 5 40 010 006 cseg less PACK Supported 100rel em Content Length 0 Note Phone 8000 goes on hook and device 10 8 201 108 sends a BYE to device 10 8 201 10 Voice path is established m F6 10 8 201 108 gt gt 10 8 201 10 BYE BYE sip 1000 10 8 201 10 user phone SIP 2 0 Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacRKCVBud From lt Sip 8000 10 8 201 108 gt tag 1c5354 To lt Sip 1000 10 8 201 10 gt tag 1 c7345 Call ID 534366556655skKw 8000 1000 10 8 201 108 User Agent Audiocodes Sip Gateway Mediant 2000 v 5 40 010 006 CSeq 18154 BYE Supported 1
591. uests 2xx responses to SUBSCRIBE NOTIFY requests REFER requests and 2xx responses to REFER Note If the GRUU contains the opaque URI parameter the device obtains the AOR for the user by stripping the parameter The resulting URI is the AOR 129 November 2008 ca AudioCodes Parameter User Agent Information UserAgentDisplayInf o SDP Session Owner SIPSDPSessionOwn er Play Busy Tone to Tel PlayBusyTone2ISDN Subject SIPSubject Multiple Packetization Time Format MultiPtimeFormat Enable Semi Attended Transfer EnableSemiAttended Transfer SIP User s Manual Mediant 2000 Description For example AOR sip alice example com GRUU sip alice example com opaque kjh29x97us97d Defines the string that is used in the SIP request header User Agent and SIP response header Server If not configured the default string AudioCodes product name s w version is used e g User Agent Audiocodes Sip Gateway Mediant 2000 v 5 40 010 006 When configured the string UserAgentDisplaylInfo s w version is used e g User Agent MyNewOEM v 5 40 010 006 Note that the version number can t be modified The maximum string length is 50 characters Determines the value of the Owner line o field in outgoing SDP messages The valid range is a string of up to 39 characters The default value is AudiocodesGW For example o AudiocodesGW 1145023829 1145023705 IN IP4 10 33 4 126 Enables the device to
592. ult 1 Transmit amp Receive Send and receive RTP 2 Transmit Only Send RTP only 8 Receive Only Receive RTP only Notes To configure the RTP Only mode per trunk use the RTPOnlyModeForTrunk_ID refer to Configuring the Trunk Settings on page 82 f per trunk configuration using RTPOnlyModeForTrunk is set to other than default the RTPOnlyMode parameter value is overridden Alert Timeout in seconds ISDN T301 timer for calls to PSTN This timer is used between the time a SETUP message is sent to the Tel side IP to Tel call establishment and a CONNECT message is received If an ALERTING message is received the timer is restarted The default is 180 seconds The range is 1 to 600 Note If per trunk configuration using TrunkPSTNAlertTimeout is set to other than default refer to Configuring the Trunk Settings on page 82 the PSTNAlertTimeout parameter value is overridden Determines the time period the device waits for an MFC R2 Resume Reanswer signal once a Suspend Clear back signal is received from the PBX If this timer expires the call is released Note Applicable only for MFC R2 CAS Brazil variant The valid range is 0 to 255 in seconds The default value is 0 Disconnect and Answer Supervision Send Digit Pattern on Connect TelConnectCode Disconnect on Broken Connection DisconnectOnBrokenCo SIP User s Manual Defines a digit pattern to send to the Tel side after
593. ult 30 40 50 60 80 100 120 10 20 default 30 40 50 60 80 100 120 10 default 20 30 10 default 20 30 10 20 default 30 40 50 60 80 100 30 default 60 90 120 10 20 default 30 40 50 60 80 100 120 20 default 40 60 80 0 20 default 30 40 50 60 80 100 20 default Rate Always 64 Always 64 Always 64 Always 64 Always 8 5 3 0 6 3 1 default 16 0 24 1 32 2 default 40 3 Always 13 12 2 4 75 0 5 15 1 5 90 2 6 70 3 7 40 4 7 95 5 10 2 6 12 2 7 default 145 Payload Type Always 8 Always 0 Dynamic 0 120 Dynamic 0 120 Always 18 Always 4 Dynamic 0 120 Always 3 Dynamic 0 120 Dynamic 0 120 Silence Suppression Disable 0 Enable 1 Disable 0 Enable 1 N A N A Disable 0 Enable 1 Enable w o Adaptations 2 Disable 0 Enable 1 Disable 0 Enable 1 Disable 0 Enable 1 Disable 0 Enable 1 Disable 0 Enable 1 November 2008 ca AudioCodes Coder Name EVRC Evrc iLBC iLBC MS GSM gsmMS QCELP QCELP Transparent Transparent G 711A law_VBD g711AlawVbd G 711U law_VBD g711UlawVbd T 38 t38fax Packetization Time 20 default 40 60 80 100 20 default 40 60 80 100 120 30 default 60 90 120 40 defaul
594. umber further using the Number Manipulation tables refer to Number Manipulation and Routing Parameters on page 313 to leave only the last 3 digits for example for sending to a PBX 0 Disabled default 1 Enabled Defines the prefix that is added to the destination number received in the SIP Refer to header in IP to Tel calls This parameter is applicable for CAS Blind Transfer modes TrunkTransferMode 3 The valid range is a string of up to 9 characters The default is an empty string 293 November 2008 r Cad AudioCodes Parameter EnableHold HoldFormat HeldTimeout EnableForward EnableCallWaiting Send180ForCallWaiting HookFlashCode UseSIPURIForDiversionHeade r RTPOnlyModeForTrunk_ID RTPOnlyMode TimeoutBetween100And18x TransparentCoderPresentatio n RxDTMFOption TxDTMFOption SIP User s Manual Mediant 2000 Description For a description of this parameter refer to Supplementary Services on page 159 For a description of this parameter refer to Supplementary Services on page 159 For a description of this parameter refer to Supplementary Services on page 159 For a description of this parameter refer to Supplementary Services on page 159 For a description of this parameter refer to Supplementary Services on page 159 Determines the SIP response code for indicating call waiting 0 Use 182 Queued response to indicate call waiting default
595. umberMapIP2Tel ini file parameter Number manipulation can occur before or after a routing decision is made For example you can route a call to a specific Trunk Group according to its original number and then you can remove or add a prefix to that number before it is routed To determine when number manipulation is performed configure the IP to Tel Routing Mode parameter RouteModelP2Tel described in IP to Trunk Group Routing on page 181 and Tel to IP Routing Mode parameter RouteModeTelZ2IP described in Tel to IP Routing Table on page 175 or Outbound IP Routing Table on page 178 For configuring number manipulation using ini file table parameters NumberMapIP2Tel NumberMapTel2IP SourceNumberMapIP2Tel and SourceNumberMapTelZ2IP refer to Number Manipulation and Routing Parameters on page 313 gt To configure the Number Manipulation tables take these 5 steps 1 Open the required Number Manipulation page Configuration tab gt Protocol Configuration menu gt Manipulation Tables submenu gt Dest Number IP gt Tel Dest Number Tel gt IP Source Number IP gt Tel or Source Number Tel gt IP page item the relevant Manipulation table page is displayed e g Source Phone Number Manipulation Table for Tel gt IP Calls page Figure 3 68 Source Phone Number Manipulation Table for Tel to IP Calls Stripped Destination Prefix Source Prefix Digits Number 03 201 971 Allowed Prefix Suffix Numbe
596. unk Group if applicable source IP Group if applicable destination number prefix and source number prefix matches the values defined in the Source IP Address Source Trunk Group Source IP Group Destination Prefix and Source Prefix fields respectively The number manipulation can be performed using a combination of each of the above criteria or using each criterion independently For available notations that represent multiple numbers refer to Dialing Plan Notation on page 168 Table 3 39 Number Manipulation Parameters Description Parameter Source Trunk Group _SrcTrunkGroupID Source IP Group _SrclPGroupID Destination Prefix _DestinationPrefix SIP User s Manual Description The source Trunk Group 1 99 for Tel to IP calls To denote any Trunk Group leave this field empty Notes This parameter is available only in the Source Phone Number Manipulation Table for Tel gt IP Calls and Destination Phone Number Manipulation Table for Tel gt IP Calls pages For IP to IP call routing this parameter is not required i e leave the field empty The IP Group from where the IP to IP call originated Typically this IP Group of an incoming INVITE is determined classified using the Inbound IP Routing table If not used i e any IP Group simply leave the field empty Notes This parameter is available only in the Source Phone Number Manipulation Table fo
597. unk Groups to selected Serving IP Group IDs if defined You can add up to 24 entries in this table Note You can also configure the Trunk Group Settings table using the ini file table parameter TrunkGroupSettings refer to Number Manipulation and Routing Parameters on page 313 Version 5 6 197 November 2008 A EA AudioCodes Mediant 2000 gt To configure the Trunk Group Settings table take these 5 steps 1 Open the Trunk Group Settings page Configuration tab gt Protocol Configuration menu gt Trunk IP Group submenu gt Trunk Group Settings page item Figure 3 83 Trunk Group Settings Page v Routing Index Serving IP Group 1D 1 Cyclic Ascending Per Gateway a a Trunk Group ID Channel Select Mode Registration Mode Gateway Name Contact User 2 From the Routing Index drop down list select the range of entries that you want to edit up to 24 entries can be configured 3 Configure the Trunk Group according to the table below 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 50 Trunk Group Settings Parameters Description Parameter Description Trunk Group ID The Trunk Group ID that you want to configure TrunkGroupSettings_ Trun The valid range is 1 99 kGroupld Trunks are assigned to Trunk Groups in the Trunk Group Table page refer to Configuring the Trunk Group Table on page
598. unt Table on page 204 Proxy Address The IP address and optionally port number of the Proxy server Up to five IP addresses can be configured per Proxy Set Enter the IP address as an FQDN or in dotted decimal notation e g 201 10 8 1 You can also specify the selected port in the format lt IP address gt lt port gt If you enable Proxy Redundancy by setting the parameter EnableProxyKeepaAlive to 1 or 2 the device can operate with multiple Proxy servers If there is no response from the first primary Proxy defined in the list the device attempts to communicate with the other redundant Proxies in the list When a redundant Proxy is located the device either continues operating with it until the next failure occurs or reverts to the primary Proxy refer to the parameter ProxyRedundancyMode If none of the Proxy servers respond the device goes over the list again The device also provides real time switching Hot Swap mode between the primary and redundant proxies refer to the parameter IsProxyHotSwap If the first Proxy doesn t respond to the INVITE message the same INVITE message is immediately sent to the next Proxy in the list The same logic applies to REGISTER messages if RegistrarlP is not defined SIP User s Manual 142 Document LTRT 68808 SIP User s Manual Parameter Transport Type Proxy Load Balancing Method ProxyLoadBalancingMethod Version 5 6 3 Web Based Management Description Notes
599. up SNMPUsers 1 v3admin1 1 0 myauthkey 1 SNMPUsers The example above configures user v3admin1 with security level authNoPriv 2 authentication protocol MD5 authentication text password myauthkey and ReadWriteGroup2 Notes This parameter can include up to 10 indices To configure SNMP v3 users through the Web interface and for a description of the parameters of this nifile table refer to Configuring SNMP V3 Users on page 225 283 November 2008 ca AudioCodes Parameter 4 4 7 Mediant 2000 Description Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 257 SIP Configuration Parameters The SIP related ini file configuration parameters are described in the table below Parameter ReliableConnectionPersistent Mode SIPTransportType TCPLocalSIPPort S IPDestinationPort EnableTCPConnectionReuse SIPTCPTimeout LocalSIPPort EnableFaxReRouting SIPGatewayName IsProxyUsed ProxyName SIP User s Manual Table 4 7 SIP ini File Parameters Description Determines whether all TCP TLS connections are set as persistent and therefore not released 0 Disable default all TCP connections except those that are set to a proxy IP are released if not used by any SIP dialog transaction 1 Enable TCP connections to all destinations are persistent and not released unless the device reaches 70 of its
600. uration sec Hotline Dial Tone Duration sec Default Destination Number Special Digit Representaton Added Scenario Step Scenario Mame PEX Iriesoperabd y Defining Scenario Name Next Button gt Step Name SIPPDDIME Defining Step Name pica Save 8 Firtsh Cancel Scenarios Get Send Scenario Fie 8 Repeat steps 5 through 8 to add additional Steps i e pages 9 When you have added all the required Steps for your Scenario click the Save amp Finish button located at the bottom of the Navigation tree a message box appears informing you that the Scenario has been successfully created 10 Click OK the Scenario mode is quit and the menu tree of the Configuration tab appears in the Navigation tree You can add up to 20 Steps to a Scenario where each Step can contain up to 25 parameters When in Scenario mode the Navigation tree is in Full display i e all menus are displayed in the Navigation tree and the configuration pages are in Advanced Parameter List display i e all parameters are shown in the pages This ensures accessibility to all parameters when creating a Scenario For a description on the Navigation tree views refer to Navigation Tree on page 23 If you previously created a Scenario and you click the Create Scenario button the previously created Scenario is deleted and replaced with the one you are creating Only users with access level of Security Administrator ca
601. urceNumberMapTel2lp FORMAT SourceNumberMapTel2Ip_Index SourceNumberMapTel2lp_DestinationPrefix SourceNumberMapTel2lp_SourcePrefix SourceNumberMapTel2Ip SourceAddress SourceNumberMapTel2lp_NumberType SourceNumberMapTel2lp_NumberPlan SourceNumberMapTel2lp_RemoveFromLeft SourceNumberMapTel2lp_RemoveFromRight SourceNumberMapTel2lp_LeaveFromRight SourceNumberMapTel2lp_Prefix2Add SourceNumberMapTel2lp_Suffix2Add SourceNumberMapTel2Ip_IsPresentationRestricted NumberMaptTel2Ip_ SrceTrunkGroupID NumberMapTel2Ip_SrelPGroupID SourceNumberMapTel2Ip For example SourceNumberMapTel2lp SourceNumberMapTel2Ip 0 22 03 0 0 2 667 0 SourceNumberMapTel2Ip 0 10 10 255 255 3 0 5 100 255 SourceNumberMapTel2Ip Notes This table parameter can include up to 120 indices RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add LeaveFromRight NumberType NumberPlan and IsPresentationRestricted are applied if the called and calling numbers match the DestinationPrefix and SourcePrefix conditions The manipulation rules are executed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and Suffix2Add Parameters can be skipped by using two dollar signs An asterisk represents all IP addresses IsPresentationRestricted is set to Restricted only if Asserted Identity Mode is set to P Asserted Number Plan and Type can optionally be
602. urrent state LOCKED or UNLOCKED gt To unlock the device take these 2 steps 1 Open the Maintenance Actions page refer to Maintenance Actions on page 228 2 Under the LOCK UNLOCK group click the UNLOCK button Unlock starts immediately and the device accepts new incoming calls 3 5 1 3 3 Saving Configuration The Maintenance Actions page allows you to save burn the current parameter configuration including loaded auxiliary files to the device s non volatile memory i e flash The parameter modifications that you make throughout the Web interface s pages are temporarily saved to the volatile memory RAM when you click the Submit button on these pages Parameter settings that are only saved to the device s RAM revert to their previous settings after a hardware software reset or power failure Therefore to ensure that your configuration changes are retained you must save them to the device s flash memory using the burn option described below gt To save the changes to the non volatile flash memory take these 2 steps 1 Open the Maintenance Actions page refer to Maintenance Actions on page 228 2 Under the Save Configuration group click the BURN button a confirmation message appears when the configuration successfully saves SIP User s Manual 230 Document LTRT 68808 SIP User s Manual 3 Web Based Management Saving configuration to the non volatile memory may disrupt traffic on
603. vailable Contact SIP URI that can be used to contact that specific instance of the User Agent for subsequent requests 3 6 2 4 IP Connectivity The IP Connectivity page displays online read only network diagnostic connectivity information on all destination IP addresses configured in the Tel to IP Routing page refer to Tel to IP Routing Table on page 175 or Outbound IP Routing Table page if EnableSBC is set to 1 refer to Outbound IP Routing Table on page 178 This information is available only if the parameter Enable Alt Routing Tel to IP refer to Routing General Parameters on page 171 is set to 1 Enable or 2 Status Only The information in columns Quality Status and Quality Info per IP address is reset if two minutes elapse without a call to that destination gt To view the IP connectivity information take these 2 steps 1 In the Routing General Parameters page set the parameter Enable Alt Routing Tel to IP or ini file parameter AltRoutingTel2IPEnable to Enable 1 or Status Only 2 2 Open the IP Connectivity page Status amp Diagnostics tab gt Gateway Statistics menu gt IP Connectivity page item Figure 3 121 IP Connectivity Page Connectivity Connectivity Quality IP Address Host Name Method Status Status Quality Info DNS Status Unused Unused Unused Unused Unused Unused Unused Unused 9 Unused 10 Unused la 2 3 A 5 6
604. ve Routing page item Figure 3 77 Reasons for Alternative Routing Page IP to Tel Reasons Reason 1 Reason 2 Reason 3 Reason 4 Tel to IP Reasons Reason 1 Reason 2 Reason 3 Reason 4 2 In the IP to Tel Reasons group select up to four different call failure reasons that invoke an alternative IP to Tel routing SIP User s Manual 188 Document LTRT 68808 SIP User s Manual 3 Web Based Management In the Tel to IP Reasons group select up to four different call failure reasons that invoke an alternative Tel to IP routing Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 230 3 4 7 4 9 Release Cause Mapping The Release Cause Mapping page consists of two groups that allow the device to map up to 12 different SIP Responses to Q 850 Release Causes and vice versa thereby overriding the hard coded mapping mechanism described in Release Reason Mapping on page 394 You can also configure SIP Responses Q 850 Release Causes mapping using the ini file table parameters CauseMapISDN2SIP and CauseMapSIP2ISDN refer to ISDN and CAS _Interworking Related Version 5 6 Parameters on page 307 To configure Release Cause Mapping take these 5 steps Open the Release Cause Mapping page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Release Cause Mapping page item Figure 3 78
605. ve this domain name to an IP address and port sort the server list and use the servers according to the sorted list Note Use either the STUNServerPrimarylP or the STUNServerDomainName parameter with priority to the first one Defines the default NAT binding lifetime in seconds STUN is used to refresh the binding information after this time expires The valid range is 0 to 2 592 000 The default value is 30 Enables disables the Network Address Translation NAT mechanism 0 Enabled 1 Disabled default Note The compare operation that is performed on the IP address is enabled by default and is controlled by the parameter EnablelPAddrTranslation The compare operation that is performed on the UDP port is disabled by default and is controlled by the parameter EnableUDPPortTranslation Enables IP address translation 0 Disable IP address translation 1 Enable IP address translation for RTP RTCP and T 38 packets default 2 Enable IP address translation for RTP Multiplexing ThroughPacket 3 Enable IP address translation for all protocols RTP RTCP T 38 and RTP Multiplexing When enabled the device compares the source IP address of the first incoming packet to the remote IP address stated in the opening of the channel If the two IP addresses don t match the NAT mechanism is activated Consequently the remote IP address of the outgoing stream is replaced by the source IP addr
606. ver When a port number is specified DNS NAPTR SRV queries aren t performed even if DNSQueryType is set to 1 or 2 If the RegistrarlP is set to an FQDN and is resolved to multiple addresses the device also provides real time switching hotswap mode between different Registrar IP addresses IsProxyHotSwap is set to 1 If the first Registrar doesn t respond to the REGISTER message the same REGISTER message is sent immediately to the next Proxy EnableProxyKeepAlive must be set to 0 for this logic to apply When a specific Transport Type is defined using RegistrarTransportType a DNS NAPTR query is not performed even if DNSQueryType is set to 2 136 Document LTRT 68808 SIP User s Manual Parameter Registrar Transport Type RegistrarTransportType Registration Time RegistrationTime Re registration Timing RegistrationTimeDivider Registration Retry Time RegistrationRetryTime Registration Time Threshold RegistrationTimeThreshold Re register On INVITE Failure RegisterOnInviteFailure ReRegister On Connection Failure ReRegisterOnConnectionFai lure Version 5 6 3 Web Based Management Description Determines the transport layer used for outgoing SIP dialogs initiated by the device to the Registrar 1 Not Configured default 0 UDP 1 TCP 2 TLS Note When set to Not Configured the value of the parameter SIPTransportType is used Defines the time interval
607. ver certificate The subject name for this default certificate is ACL_nnnnnnn where nnnnnnn denotes the serial number of the device However this subject name may not be appropriate for production and can be changed while still using self signed certificates gt To change the subject name and regenerate the self signed certificate take these 4 steps 1 Before you begin ensure the following e You have a unique DNS name for the device e g dns_name corp customer com This name is used to access the device and should therefore be listed in the server certificate e No traffic is running on the device The certificate generation process is disruptive to traffic and should be executed during maintenance time 2 Open the Certificates page refer to Server Certificate Replacement on page 105 3 In the Subject Name field enter the fully qualified DNS name FQDN as the certificate subject and then click Generate Self signed after a few seconds a message appears displaying the new subject name 4 Save configuration refer to Saving Configuration on page 230 and then restart the device for the new certificate to take effect Version 5 6 109 November 2008 SIP User s Manual ca AudioCodes Mediant 2000 3 4 6 5 Configuring the General Security Settings The General Security Settings page is used to configure various security features gt To configure the general security parameters take these 4 ste
608. vers_Index NFSServers_HostoOrlP NFSServers_RootPath NFSServers_NfsVersion NFSServers_AuthType NFSServers_UID NFSServers_ GID NFSServers_VlanType NFSServers 1 101 1 13 audio1 3 1 0 1 1 NFSServers Notes You can configure up to five NFS file systems 0 4 The combination of Host IP and Root Path must be unique for each index in the table For example the table must include only one index entry with a Host IP of 192 168 1 1 and Root Path of faudio This parameter is applicable only if VLANs are enabled or if Multiple IPs is configured To configure NFS using the Web interface and for a description of the parameters of this ini file table parameter refer to Configuring the NFS Settings on page 60 Fora description of configuring ini file table parameters refer to Structure of ini File Table Parameters on page 257 267 November 2008 A EA AudioCodes Mediant 2000 44 2 System Parameters The system related ini file configuration parameters are described in the table below Parameter EnableDiagnostics GWaAppDelayTime ActivityListToLog SIP User s Manual Table 4 2 System ini File Parameters Description Checks the correct functionality of the different hardware components on the device On completion of the check if the test fails the device sends information on the test results of each hardware component to the Syslog server 0 Rapid and Enhanced self test mod
609. vice messages on supporting variants and use Alarm on non supporting variants CAS Use Alarm When updating this parameter value at run time you must stop the trunk and then restart it for the update to take effect To determine the method for setting Out Of Service state per trunk use the DigitalOOSBehaviorFor Trunk_ID parameter refer to Trunk Settings on page 82 Enables the trunk Transfer Mode Refer to TrunkTransferMode 0 1 or 3 in ISDN and CAS Interworking Related Parameters on page 307 Note This parameter is only available for Protocol Type T1 CAS Enables the Two B Channel Transfer TBCT trunk transfer mode Refer to TrunkTransferMode 0 and 2 in ISDN and CAS Interworking Related Parameters on page 307 Note This parameter is only available for Protocol Type T1 N12 ISDN Enables the Release Link Trunk RLT trunk transfer mode Refer to TrunkTransferMode 0 and 2 in ISDN and CAS Interworking Related Parameters on page 307 Note This parameter is only available for Protocol Type T1 DMS100 ISDN Enables the Single Step Transfer Trunk transfer mode Refer to TrunkTransferMode 0 and 4 in ISDN and CAS Interworking Related Parameters on page 307 Enables the Explicit Call Transfer ECT trunk transfer mode Refer to TrunkTransferMode 0 and 2 in ISDN and CAS Interworking Related Parameters on page 307 Note This parameter is only available for Protocol Type E1 EURO ISDN 96 Documen
610. voice announcements are prepared offline using standard recording utilities and combined into a single file using the TrunkPack Downloadable Conversion Utility The generated announcement file can then be loaded to the device using the BootP TFTP utility refer to the Product Reference Manual If the size of the combined Voice Prompts file is less than 1 MB it can permanently be stored on flash memory Larger files up to 10 MB are stored in RAM and should be loaded again using BootP TFTP utility after the device is reset The Voice Prompts integrated file is a collection of raw voice recordings and or wav files These recordings can be prepared using standard utilities such as CoolEdit Goldwave and others The raw voice recordings must be sampled at 8000 kHz mono 8 bit The wav files must be recorded with G 711p Law A Law Linear When the list of recorded files is converted to a single voiceprompts dat file every Voice Prompt is tagged with an ID number starting with 1 This ID is used later by the device to start playing the correct announcement Up to 1 000 Voice Prompts can be used AudioCodes provides a professionally recorded English U S Voice Prompts file gt To generate and load the Voice Prompts file take these 3 steps 1 Prepare one or more voice files using standard utilities 2 Use the TrunkPack Downloadable Conversion Utility to generate the voiceprompts dat file from the pre recorded voice messages ref
611. volatile memory after the file is loaded to the device When a parameter is absent from the ini file the default value is assigned to that parameter according to the cmp file loaded to the device and stored in the non volatile memory thereby overriding the value previously defined for that parameter Some of the device s parameters are configurable only through the ini file and not the Web interface These parameters usually determine a low level functionality and are seldom changed for a specific application For a list of the ini file parameters refer to The ini File Parameter Reference on page 260 The ini file parameters that are configurable in the Web interface are described in Web Based Management on page 19 The ini parameters that can t be configured using the Web interface are described in this section To define or restore default settings using the ini file refer to Default Settings on page 333 4 1 Secured Encoded ini File The ini file contains sensitive information that is required for the functioning of the device Typically it is loaded to or retrieved from the device using TFTP or HTTP These protocols are not secure and vulnerable to potential hackers To overcome this security threat the AudioCodes TrunkPack Downloadable Conversion Utility DConvert allows you to binary encode the ini file before loading it to the device refer to the Product Reference Manual If you retrieve an ini file from the
612. with the pre defined default values for each configured line m The order of the fields in the Format line isn t significant as opposed to the Index fields The fields in the Data lines are interpreted according to the order specified in the Format line m The double dollar sign in a Data line indicates the default value for the parameter The order of the Data lines is insignificant m Data lines must match the Format line i e it must contain exactly the same number of Indices and Data fields and must be in exactly the same order m Aline in a table is identified by its table name and Index fields Each such line may appear only once in the ini file m Table dependencies Certain tables may depend on other tables For example one table may include a field that specifies an entry in another table This method is used to specify additional attributes of an entity or to specify that a given entity is part of a larger entity The tables must appear in the order of their dependency i e if Table X is referred to by Table Y Table X must appear in the ini file before Table Y The table below displays an example of an ini file table parameter PREFIX FORMAT PREFIX Index PREFIX DestinationPrefix PREFIX DestAddress PREFIX SourcePrefix PREFIX Profileld PREFIX MeteringCode PREFIX DestPort PREFIX 0 AO TOMT 6 255 PREFIX 1 20 IL 13 O 255 PREFIX 2 KOTT lt 255 PREFIX 3 TOMBE O ASB PREFIX 0 Or
613. word for this call 3 4 7 4 5 Inbound IP Routing Table The Inbound IP Routing Table page allows you to identify received calls as inbound IP to IP calls and assign them to an IP Group defined in Configuring the IP Groups on page 201 termed the Source IP Group This table identifies these IP calls based on any combination of the following criteria rules m Destination and source host prefixes m Destination and source telephone number prefixes m Source IP address Assigning these IP calls to Trunk Group ID 1 identifies them as inbound IP to IP calls These calls now pertaining to an IP Group can later be routed to an outbound destination IP Group refer to Outbound IP Routing Table on page 178 The Inbound IP Routing Table page appears only if the parameter EnableSBC is set to 1 i e enabled in SBC Configuration on page 163 If this parameter is not enabled default the IP to Trunk Group Routing Table page appears instead refer to IP to Trunk Group Routing Table on page 181 for a description of this page gt To configure Inbound Routing take these 5 steps 1 Open the Inbound IP Routing Table page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt IP to Trunk Group Routing page item Figure 3 74 Inbound IP Routing Table Matching Rules Destination Rules SIP User s Manual 184 Document LTRT 68808 SIP User s Manual 3 Web Based Management
614. xy Set ID s can be configured The Proxy Set ID 0 is used as the default Proxy Set and if defined is backward compatible to the list of Proxies from earlier releases Note Although not recommended you can use both default Proxy Set ID 0 and IP Groups for call routing For example on the Trunk Group Settings page refer to Configuring the Trunk Group Settings on page 197 you can configure a Serving IP Group to where you want to route specific Trunk Group s channels while all other device channels use the default Proxy Set At the same you can also use IP Groups in the Tel to IP Routing table refer to Tel to IP Routing Table on page 175 or Outbound IP Routing table if EnableSBC is set to 1 refer to Outbound IP Routing Table on page 178 to configure the default Proxy Set if the parameter PreferRouteTable is set to 1 To summarize if the default Proxy Set is used the INVITE message is sent according to the following preferences To the Trunk Group s Serving IP Group ID as defined in the Trunk Group Settings table According to the Tel to IP Routing table or Outbound IP Routing table if EnableSBC is set to 1 if the parameter PreferRouteTable is set to 1 To the default Proxy Typically when IP Groups are used there is no need to use the default Proxy and all routing and registration rules can be configured using IP Groups and the Account tables refer to Configuring the Acco
615. y Read Only Mode is displayed at the bottom of the page SIP User s Manual 26 Document LTRT 68808 SIP User s Manual 3 Web Based Management 3 3 3 2 1 Displaying Basic and Advanced Parameters Some pages provide you with an Advanced Parameter List Basic Parameter List toggle button that allows you to show or hide advanced parameters in addition to displaying the basic parameters This button is located on the top right corner of the page and has two states m Advanced Parameter List button with down pointing arrow click this button to display all parameters E Basic Parameter List button with up pointing arrow click this button to show only common basic parameters The figure below shows an example of a page displaying basic parameters only and then showing advanced parameters as well using the Advanced Parameter List button Figure 3 7 Toggling between Basic and Advanced Page View Toggle Button Click to View All Parameters Atverced Pamen Un v Declare RFC 2833 in SOP No ist Tx OTMF Option RFC 2833 and Tx OTMF Opmon Jrd Tx OTMF Option 4th Tx OTMF Option Sth Tx DTMF Option RFC 2033 Payload Type Oefauk Destinabon Number Spean Dept Represertahon Bone PorameterList a Max Oigts In Phone Num 5 Inter Digt Timesut fer Overlap Dising y sec Declare RFC 2633 in SOP No ist Tx OTM Option PIC 2 2ed Tx DTMF Option Parar TS Pa Sed Tx OTME Option nie 4th Tx DTMF Option in Dark
616. y a Differential Services Entering Phone Numbers in Various Tables Phone numbers or prefixes that you enter in various tables throughout the Web interface such as the Tel to IP Routing table must only be entered as digits without any other characters For example if you wish to enter the phone number 555 1212 it must be entered as 5551212 without the hyphen If the hyphen is entered the entry is invalid Working with Tables The Web interface includes many configuration pages that provide tables for configuring the device Some of these tables provide the following command buttons m Add adds an index entry to the table Duplicate duplicates a selected existing index entry Compact organizes the index entries in ascending consecutive order Delete deletes a selected index entry Apply saves the configuration SIP User s Manual 30 Document LTRT 68808 SIP User s Manual 3 Web Based Management gt To add an entry to a table take these 2 steps 1 In the Add field enter the desired index entry number and then click Add an index entry row appears in the table Figure 3 11 Adding an Index Entry to a Table Entered Index Number Add Button Duplicate Compact Delete Apply Index ApplicabonTypes IPy6InterfaceMode IPAddress PrefixLength Gateway VianID InterfaceName 1 l6 lo 10 13 413 16 10 13 0 1 lo aLL Added Table Index Entry 2 Click Apply to save the index entry Before you can add
617. y If the number is not allowed number isn t listed or a Call Restriction routing rule is applied the call is released m Always Use Routing Table When this feature is enabled AlwaysUseRouteTable 1 even if a Proxy server is used the SIP URI host name in the sent INVITE message is obtained from this table Using this feature you can assign a different SIP URI host name for different called and or calling numbers Version 5 6 175 November 2008 A Ee AudioCodes Mediant 2000 Assign Profiles to destination addresses also when a Proxy is used E Alternative Routing when a Proxy isn t used an alternative IP destination for telephone number prefixes is available To associate an alternative IP address to a called telephone number prefix assign it with an additional entry with a different IP address or use an FQDN that resolves into two IP addresses The call is sent to the alternative destination when one of the following occurs e No ping to the initial destination is available poor QoS delay or packet loss calculated according to previous calls is detected or a DNS host name is not resolved For detailed information on Alternative Routing refer to Configuring Alternative Routing Based on Connectivity and QoS on page 361 e A release reason defined in the Reasons for Alternative Tel to IP Routing table is received refer to Reasons for Alternative Routing on page 188 Alternative routing using this table is co
618. y Initially the device sends requests according to the maximum number of allowed SIP dialogs configured by the parameter NumberOfActiveDialogs After each received response the subsequent request is sent Therefore no more than NumberOfActiveDialogs dialogs are active simultaneously The user name and password are used for SIP Authentication when required The calling number of outgoing Tel to IP calls is first translated to an IP number and then if defined the manipulation rules are performed The Display Name is used in the From header in addition to the IP number The called number of incoming IP to Tel calls is translated to a PBX extension only after manipulation rules if defined are performed Version 5 6 341 November 2008 A ge AudioCodes Mediant 2000 Reader s Notes SIP User s Manual 342 Document LTRT 68808 SIP User s Manual 7 IP Telephony Capabilities 7 IP Telephony Capabilities This section describes the device s IP telephony capabilities 7 1 IP to IP Routing SIP Trunking The AudioCodes device supports IP to IP VoIP call routing or SIP trunking The device enables Enterprises to seamlessly connect their IP PBX to a SIP trunk provided by an Internet Telephony Service Provider ITSP The Enterprise can communicate with the PSTN through the ITSP which interfaces directly with PSTN Alternatively the device can also provide the interface with the PSTN At the same time the device can also provide an
619. y be set to the incorrect values m Parameter string values that denote file names e g CallProgressTonesFileName must be enclosed with inverted commas e g CallProgressTonesFileName cpt_usa dat m The parameter name is not case sensitive The parameter value is not case sensitive except for coder names E The ini file must end with at least one carriage return Structure of Individual ini File Parameters The structure of individual ini file parameters in an ini file is shown below Subsection Name Parameter Name Parameter Value Parameter Name Parameter Value REMARK An example of an ini file containing individual ini file parameters is shown below SYSTEM Params Syo llogserver IP LOFILE T9 EnableSyslog 1 These are a few of the system related parameters WEB Params LogoWidth 339 WebLogoText My Device UseWeblogo 1 These are a few of the Web related parameters Files CallProgressTonesFileName cpusa dat SIP User s Manual 256 Document LTRT 68808 SIP User s Manual 4 ini File Configuration 4 2 3 Structure of ini File Table Parameters You can use anini file to configure table parameters which include several parameters table columns grouped according to the applications they configure e g NFS and IPSec When loading an ini file to the device it s recommended to include only tables that belong to applications that are to be configure
620. y Index 0 State Does not exist Internet Key Exchange table row does not exist 7 Authentication Method Shared Key IKE S LifeTime sec IKE SA LifeTime KB First Proposal Encryption Type First Proposal Authentication Type First Proposal DH Group Second Proposal Encryption Type Second Proposal Authentication Type Second Proposal DH Group Third Proposal Encryption Type Third Proposal Authentication Type Third Proposal DH Group Fourth Proposal Encryption Type Fourth Proposal Authentication Type Fourth Proposal DH Group Pre shared Key 28800 0 Not Defined Not Defined E Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined 2 From the Policy Index drop down list select the peer you want to edit up to 20 peers can be configured Configure the IKE parameters according to the table below Up to two IKE main mode proposals Encryption Authentication DH group combinations can be defined The same proposals must be configured for all peers 4 Click Create a row is created in the IKE table 5 To save the changes to flash memory refer to Saving Configuration on page 230 To delete a peer from the IKE table select it from the Policy Index drop down list click the button De
621. y Protocol SNMPUsers _PrivProtocol Authentication Key SNMPUsers_AuthKey Privacy Key SNMPUsers_PrivKey Group SNMPUsers_ Group Description The table index The valid range is 0 to 9 Name of the SNMP v3 user This name must be unique Authentication protocol of the SNMP v3 user 0 None default 1 MD5 2 SHA 1 Privacy protocol of the SNMP v3 user 0 None default 1 DES 2 3DES 8 AES 128 4 AES 192 5 AES 256 Authentication key Keys can be entered in the form of a text password or long hex string Keys are always persisted as long hex strings and keys are localized Privacy key Keys can be entered in the form of a text password or long hex string Keys are always persisted as long hex strings and keys are localized The group with which the SNMP v3 user is associated 0 Read Only default 1 Read Write 2 Trap Note All groups can be used to send traps 3 5 1 1 4 Configuring SNMP Trusted Managers The SNMP Trusted Managers page allows you to configure up to five SNMP Trusted Managers based on IP addresses By default the SNMP agent accepts SNMP Get and Set requests from any IP address as long as the correct community string is used in the request Security can be enhanced by using Trusted Managers which is an IP address from which the SNMP agent accepts and processes SNMP requests gt To configure the SNMP Trusted Managers take the followin
622. y Sets Table on page 141 associated with these Serving IP Groups A Trunk Group can register to more than one Serving IP Group e g ITSP s by configuring multiple entries in this Account table with the same Served Trunk Group but with different Serving IP Groups user name password Host Name and Contact User parameters SIP User s Manual 204 Document LTRT 68808 SIP User s Manual 3 Web Based Management Note You can also configure the Account table using the ini file table parameter Account refer to SIP Configuration Parameters on page 284 gt To configure Accounts take these 5 steps 1 Open the Account Table page Configuration tab gt Protocol Configuration menu gt Trunk IP Group submenu gt Account Table page item Figure 3 85 Account Table Page 2 To add an Account in the Add field enter the desired table row index and then click Add A new row appears 3 Configure the Account parameters according to the table below 4 Click the Apply button to save your changes 5 To save the changes refer to Saving Configuration on page 230 Note For a description of the Web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 30 Table 3 52 Account Parameters Description Parameter Description Served Trunk The Trunk Group ID for which the device performs registration and or Group authentication to a destination IP Group i e Serv
623. y these failures lead to the inability to make emergency calls e g 911 in North America Despite these possible point of failures the device s SAS feature ensures that the Enterprise s telephony services e g SIP IP phones or soft phones are maintained by routing calls to the PSTN i e providing PSTN fallback The SAS feature operates in one of two modes m Normal Initially the device s SAS agent serves as a registrar and outbound Proxy server to which every VoIP CPE e g IP phones within the Enterprise s LAN registers The SAS agent at the same time sends all these registration requests to the Proxy server e g IP Centrex or IP PBX This ensures registration redundancy by the SAS agent for all telephony devices Therefore SAS agent functions as a stateful proxy passing all SIP requests received from the Enterprise to the Proxy and vice versa In parallel the SAS agent continuously maintains a keep alive handshake with the Proxy server using SIP OPTIONS or re INVITE messages m Emergency The SAS agent switches to Emergency mode if it detects from the keep alive responses that the connection with the Proxy is lost This can occur due to Proxy server failure or WAN problems In this mode when the connection with the Proxy server is down the SAS agent controls all internal calls within the Enterprise In the case of outgoing calls the SAS agent forwards them to a local VoIP gateway this can be the device itself or a sepa
624. yption On Transmitted RTP Packets D Disable Encryption On Transmitted RTCP Packets 0 v SRTP Setting Master Key Identifier MKI Size 2 Configure the media security parameters according to the table below 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 230 Table 3 19 Media Security Parameters Parameter Description Media Security Enables Secure Real Time Transport Protocol SRTP EnableMediaSecurity 0 Disable SRTP is disabled default 1 Enable SRTP is enabled Media Security Behavior Determines the device s mode of operation when SRTP is used MediaSecurityBehaviour EnableMediaSecurity 1 0 Preferable The device initiates encrypted calls If negotiation of the cipher suite fails an unencrypted call is established Incoming calls that don t include encryption information are accepted 1 Mandatory The device initiates encrypted calls but if negotiation of the cipher suite fails the call is terminated Incoming calls that don t include encryption information are rejected default Disable Authentication On On a secured RTP session this parameter determines whether Transmitted RTP Packets to enable Authentication on transmitted RTP packets RTPAuthenticationDisableTx 0 Enable default 1 Disable Version 5 6 81 November 2008 A ge AudioCodes Mediant 2000 Parameter Description Disa

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