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Audiocodes MediaPack 114

1. w Versions Version ID 5 304 015 DSP Type 0 DSP Software Version 54012 DSP Software Name 204IM Flash Version 195 w Loaded Files Loaded Call Progress Tones Default Progress Tones Loaded Coder Table Default CODERTABLE Version 5 6 221 November 2008 7a E tal AudioCodes MediaPack Series The Board Type field number depicts the following devices m MP 118 56 m MP 114 57 mM MP 112 58 m MP 124 FXS 3 gt To delete any of the loaded files take this step m 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 207 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 3 112 Performance Statistics Page Performance Statistics o Statisti Statistics for 2811 seconds Active TDM channels Active DSP resources Active analog channels Active G 711 channels Average voi
2. v SIP General PRACK Mode Supported v Channel Select Mode Cyclic Ascending v Enable Early Media Disable v 183 Message Behavior Alert v Session Expires Time 0 Minimum Session Expires 30 Session Expires Method Re INYITE v Asserted Identity Mode Disabled v Fax Signaling Method No Fax v Detect Fax on Answer Tone Initiate T 38 on Preamble v SIP Transport Type UDP v SIP UDP Local Port 5060 SIP TCP Local Port 5060 SIP TLS Local Port 5061 Enable SIPS Disable v Enable TCP Connection Reuse Enable v TCP Timeout 0 SIP Destination Port 5060 Use user phone in SIP URL Yes v Use user phone in From Header No v Use Tel URI for Asserted Identity Disable v Tel to IP Mo Answer Timeout 180 Enable Remote Party ID Disable v Add Number Plan and Type to RPI Header Yes v Enable History Info Header Disable v Use Source Number as Display Name No v Use Display Name as Source Number No v Enable Contact Restriction Disable v Play Ringback Tone to IP Don t Play v Play Ringback Tone to Tel Play According to Early Media v Use Tarp information Disable v Enable GRUU Disable w User Agent Information SDP Session Owner AudiocodesGW Subject Multiple Packetization Time Format None v Enable Semi Attended Transfer Disable v 3xx Behavior Forward v Enable P Charging Vector Disable v Enable VoiceMail URI Disable Retry After Time 0 Enable P Associated URI Header Disable v Source Number Preference Forking Handling Mode Parallel handling v Enable Rea
3. O d3 94 e INVITE sendrecv 200 OK sendrecv INVITE Hold inactive INVITE Retrieve sendrecv I 200 OK inactive 200 OK inactive i V 4 5 Conversation gt l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l l gt l 1 l l l l l l l l l l l l l l l l l l l l l l cans S i Yyse y INVITE Retrieve sendrecv 200 OK sendrecv INVITE Hold inactive 200 OK inactive I I f 1 I I 1 l Conversation gt I I I I I I I I I SIP User s Manual 358 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities The previous flowchart describes the following double call hold scenario 1 A calls B and establishes a voice path A places B on hold A hears a Dial tone and B hears a Held tone A calls C and establishes a voice path B places A on hold B hears a Dial tone B calls D and establishes a voice path A ends call with C A hears a Held tone B ends call with D ON O SF PN B retrieves call with A If a party that is placed on hold e g B in the above figure is called by another party e g D then the on hold party receives a Call Waiting tone instead of the Held tone While in a Double Hold state placing the phone on hook disconnects both calls i e call transfer is not performed 7 14 2 Co
4. ccccccccssccssccecssssecsssceecccecseceussssesscaeeseeeensaneessrseseasensanenss 257 Table 4 6 SNMP ini File Pap hers sai ecrsssacseeaus asetivancs Aad V ean K 3 V nesses 258 Table 4 77 SIP ini File PASS i dl o o o k o 260 Table 4 38 Voices Mail ini File Parametar S sus 645duadd5a ninian add Keke Anni an ved K5a dk SSV ka Kask 277 Tabled PSTN ini File Faramello 34 oi n dole ea aaa eed aa aaia 279 Table 4 10 Analog Telephony ini File Parameters ssssssssssrssssisssrsnnrsssnnaaisresessnssarnnnnnsttnnaaiidesenanaaa 280 Table 4 11 Number Manipulation and Routing ini File PATAMEtETS eee 289 Table 4 12 Channelini File PargmeolorS sua ods E ial a AAL AT aN 298 Table 4 13 Auxiliary Configuration ini File Parameters sesessssseessseesssrnesrressssernsarsnnnsssnnaarireeesennaa 304 Table ets Usor MNommaton eis z oto 0 oo a o E k a 313 Table 7 1 Supported X Deta t Event Types ernadua ni aaia Aanand aaa 332 Table 7 2 Supported RADIUS AUS croisean nL 6 do A o Ek o 336 Table 7 3 Supporied CDR PTE NO 6 nd o ooo a nas 339 Table 8 1 Trafile Network Types and PT z isiosv sustv ldd odd zad de dd anaana ote Table 8 2 Example of VLAN and Multiple IPs Configuration eee eeeeeeenn gots Table 9 1 Supplied Sofware Pakage is cesses sok icatsccehadseaiacsaeeiincenstench de ASA o o 379 Table 10 1 MP 11x Functional SDRAOAUGTS zu ssd Jda nudn n d da k k skl a a dn 381 Table 10 2 MP 124 Functional Specification su ci
5. SIP User s Manual 7 IP Telephony Capabilities gt To configure the two devices for call communication take these 4 steps 1 For the first device 10 2 37 10 in the Endpoint Phone Number Table page refer to Configuring the Endpoint Phone Numbers on page 181 assign the phone numbers 101 to 104 to the device s endpoints Figure 7 7 Assigning Phone Numbers to Device 10 2 37 10 Channel s Phone Number Hunt Group ID 1 1 4 m 2 For the second device 10 2 37 20 in the Endpoint Phone Number Table page assign the phone numbers 201 to 204 to the device s endpoints Figure 7 8 Assigning Phone Numbers to Device 10 2 37 20 Channel s Phone Number Hunt Group ID 1 4 201 3 Configure the following settings for both devices In the Tel to IP Routing page refer to Tel to IP Routing Table on page 160 add the following routing rules a In the first row enter 10 for the destination phone prefix and enter 10 2 37 10 for the destination IP address i e IP address of the first device b Inthe second row enter 20 for the destination phone prefix and 10 2 37 20 for the destination IP address i e IP address of the second device These settings enable the routing from both devices of outgoing Tel to IP calls that start with 10 to the first device and calls that start with 20 to the second device Figure 7 9 Routing Calls Between Devices Dest Phone Prefix
6. 0 28800 0 Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined Not Defined SNS 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 209 If no IPSec methods are defined Encryption Authentication the default settings shown in the following table are applied Table 3 23 Default IKE Second Phase Proposals Proposal Encryption Proposal 0 3DES Proposal 1 3DES Proposal 2 DES Proposal 3 DES Version 5 6 95 Authentication SHA1 MD5 SHA1 MD5 November 2008 ca AudioCodes MediaPack Series Table 3 24 IPSec SPD Table Configuration Parameters Parameter Name IPSec Mode 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 IPsecPolicyKeyExc
7. Default_Cnonce Per Endpoint Enable None Optional 2 Configure the Proxy and Registration parameters according to the following table 3 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 4 To save the changes to flash memory refer to Saving Configuration on page 209 SIP User s Manual 112 Document LTRT 65411 SIP User s Manual 3 Web Based Management Table 3 28 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 Version 5 6 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 120 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 160 Click the right pointing arrow Lub button to open the Proxy Sets
8. 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 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 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
9. For detailed information on multiple routers support refer to Multiple Routers Support on page 368 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 63 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 63 Media Network Settings Available only in Multiple IP and Dual IP modes IP Address LocalMedialPAddress Subnet Mask LocalMediaSubnetMask Default Gateway Address LocalMediaDefaultGW 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
10. 0 Enable default 1 Disable On a secured RTP session this parameter determines whether to enable Encryption on transmitted RTCP packets 0 Enable default 1 Disable Determines the size in bytes of the Master Key Identifier MKI in SRTP Tx packets The range is 0 to 4 The default value is 0 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 80 m Web amp Telnet Access List refer to Configuring the Web and Telnet Access List on page 82 m Firewall Settings refer to Configuring the Firewall Settings on page 84 Certificates refer to Configuring the Certificates on page 86 m General Security Settings refer to Configuring the General Security Settings on page 90 m IPSec Table refer to Configuring the IPSec Table on page 94 IKE Table refer to Configuring the IKE Table on page 97 Version 5 6 79 November 2008 tall AudioCodes MediaPack Series 3 4 3 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 pe
11. 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 209 Table 3 15 Media Settings RTP RTCP Parameters Parameter Description Dynamic Jitter Buffer Minimum Minimum delay in msec for the Dynamic Jitter Buffer Delay DJBufMinDelay 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 334 Dynamic Jitter Buffer Dynamic Jitter Buffer frame error delay optimization factor Optimization Factor The valid range is 0 to 13 The default factor is 10 DJBufOptFactor Notes Set to 13 for data fax and modem calls For more information on Jitter Buffer refer to Dynamic Jitter Buffer Operation on page 334 RTP Redundancy Depth Determines whether the device generates redundant packets RTPRedundancyDepth 0 0 Disable the generation of redundant packets default 1 1 Enable the generation of RFC 2198 redundancy packets SIP User s Manual 74 Document LTRT 65411 SIP User s Manual Parameter Packing Factor RTPPackingFactor Basic RTP Packet Interval BasicRTPPacketlnterval RTP Directional Control RTPDirectionControl RFC 2833 TX Payload Type RFC2833TxPayloadType RFC 2833 RX Payload Type RFC2833RxPayloadType RFC 2198 Payload Type
12. 2 NFS Version 2 3 NFS Version 3 default Authentication method used for accessing the remote file system 0 Auth NULL 1 Auth UNIX default User ID used in authentication when using Auth UNIX The valid range is 0 to 65537 The default is 0 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 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 370 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 55 Before sending an 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 52 Version 5 6 63 November 2008 7a K tal AudioCodes MediaPack Series gt To configure static IP routing take these 3 s
13. Figure 3 73 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 86 17 10 8 6 18 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 209 3 4 4 4 5 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 236 Version 5 6 167 November 2008 A e AudioCodes Medi
14. 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 Reset Displayed on Toolbar of Submit Bun Reset Device Actions wv A Home 8 Help P Log off Reset Notification SIP User s Manual 24 Document LTRT 65411 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 m menu first level highest level m submenu second level contained within a menu m page item last level lowest level in a menu contained within a menu or submenu Figure 3 4 Terminology for Navigation Tree Levels Management K Dlagnosice Scenarios Search O Basic Full dnetwork Settings Amedia Settings security Settings Protocol Configuration Protocol Definition SIP General Parameters Proxy amp Registration Prox
15. 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 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 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 166 SIP User s Manual 162 Document LTRT 65411 SIP User s Manual Parameter Port PREFIX_DestPort Transport Type PREFIX_TransportType Dest IP Group ID PREFIX_DestIPGroupID IP Profile ID PREFIX_Profileld Status Charge Code PREFIX_MeteringCode Version 5 6 3 Web Based Management Description 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 SIPTr
16. SIP User s Manual 326 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 7 4 1 1 One Stage Dialing One stage dialing is when the FXO device receives an IP to Tel call off hooks the PBX line connected to the telephone and then immediately dials the destination telephone number In other words the IP caller doesn t dial the PSTN number upon hearing a dial tone Figure 7 2 Call Flow for One Stage Dialing FXO Gateway SIP Client F1 INVITE FXO seizes line FXO waits for dial tone from PBX if defined by Is VaitForDialTone and WaitF orDialTone F4 200 OK immediatley or after detecting polarity reversal or voice One stage dialing incorporates the following FXO functionality m Waiting for Dial Tone Enables the device to dial the digits to the Tel side only after detecting a dial tone from the PBX line The ini file parameter IsWaitForDialTone is used to configure this operation m Time to Wait Before Dialing Defines the time in msec between seizing the FXO line and starting to dial the digits The inifile parameter WaitForDialTime is used to configure this operation Note The ini file parameter IsWaitForDialTone must be disabled for this mode m Answer Supervision The Answer Supervision feature enables the FXO device to determine when a call is connected by using one of the following methods e Polarity Reversal device sends a 200 OK in response to an INVITE only when it det
17. lw Enable Proxy Keep Alive Disable Proxy Keep Alive Time 60 Proxy Load Balancing Method Disable Is Proxy Hot Swap No SIP User s Manual 120 Document LTRT 65411 SIP User s Manual 3 Web Based Management From the Proxy Set ID drop down list select an ID for the desired group Configure the Proxy parameters according to the following table Click the Submit button to save your changes 0 e U N To save the changes to flash memory refer to Saving Configuration on page 209 Table 3 29 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 Proxy 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 Hunt Group Settings page refer to Configuring the Hunt Group Settings on page 183 you can configure a Serving IP Group to where you want to route specific HuntGroup s endpoints while all other device endpoints 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 160 to configure the default Proxy Set if the parameter PreferRouteTable is setto 1 To summarize if the default Prox
18. 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 endpoint 1 Per Gateway Single Registration and Authentication for the entire device default 3 Per FXS Registration and Authentication for FXS endpoints Typically Authentication per endpoint is used for FXS interfaces where each endpoint registers and authenticates separately with its own user name and password Single Registration and Authentication Authentication Mode 1 is usually defined for FXO ports Enables setting an endpoint 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 Hunt Group Settings on page 183 and a specific endpoint 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 subseguent registration reguest 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 The out of service method is set according to the parameter
19. 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 1 133 November 2008 ca AudioCodes Parameter Silence Detection Period sec FarEndDisconnectSile ncePeriod Silence Detection Method FarEndDisconnectSile nceMethod Enable Fax Re Routing EnableFaxReRouting CDR and Debug CDR Server IP Address CDRSyslogServerlP CDR Report Level CDRReportLevel Debug Level GwDebugLevel SIP User s Manual MediaPack Series Description 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 Silence detection method 0 None Silence detection option is disabled 1 Packets Count According to packet count 2 Voice Energy Detectors According to energy and voice detectors default 3 All According to packet count and energy and voice detectors Enables or disables re routing of Tel to IP calls that are identified as fax calls
20. ETSI before ring DT AS 2 ETSI before ring RP AS 3 ETSI before ring LR DT AS 4 ETSI not ring related DT AS 5 ETSI not ring related RP AS 6 ETSI not ring related LR DT AS ETSIVMWITypeOneStandard Selects the ETSI Visual Message Waiting Indication VMWI Type 1 sub standard 0 ETSI VMWI between rings default 1 ETSI VMWI before ring DT AS 2 ETSI VMWI before ring RP AS 3 ETSI VMWI before ring LR DT AS 4 ETSI VMWI not ring related DT AS 5 ETSI VMWI not ring related RP_AS 6 ETSI VMWI not ring related LR DT AS BellcoreVMWITypeOneStand Selects the Bellcore VMWI sub standard ard 0 Between rings default 1 Not ring related SIP User s Manual 288 Document LTRT 65411 SIP User s Manual 4 ini File Configuration 4 4 11 Number Manipulation and Routing Parameters The number manipulation and routing related ini file configuration parameters are described in the table below Table 4 11 Number Manipulation and Routing ini File Parameters Parameter TrunkGroup DefaultNumber ChannelSelectMode TrunkGroupSettings Version 5 6 Description This ini file table parameter defines the device s endpoints and assigns them to Hunt Groups The format of this parameter is shown below TrunkGroup FORMAT TrunkGroup Index TrunkGroup TrunkGroupNum TrunkGroup FirstTrunkld TrunkGroup_LastTrunkld TrunkGroup FirstBChannel Trunk
21. Enable v Port 7 FXO Enabl Port 8 FXO Enable v 2 In the Destination Phone Number field corresponding to a port enter the telephone number that you want automatically dialed 3 From the Auto Dial Status drop down list select one of the following e Enable 1 The number in the Destination Phone Number field is automatically dialed if the phone is off hooked for FXS interfaces or a ring signal from PBX PSTN switch is generated to a port FXO interfaces e Disable 0 The automatic dialing feature for the specific port is disabled i e the number in the Destination Phone Number field is ignored SIP User s Manual 176 Document LTRT 65411 SIP User s Manual 3 Web Based Management e Hotline 2 When a phone is off hooked and no digit is dialed for a user defined interval Hotline Dial Tone Duration refer to DTMF 8 Dialing Parameters on page 125 the number in the Destination Phone Number field is automatically dialed applies to FXS and FXO interfaces 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 209 3 4 4 6 3 Caller Display Information The Caller Display Information page allows you to enable the device to send Caller ID information to IP when a call is made The called party can use this information for caller identification The information configured in this page is se
22. Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 298 Document LTRT 65411 SIP User s Manual Parameter FaxModemBypassM FaxModemNTEMode FaxBypassPayloadType CallerlIDTransportType ModemBypassPayloadType FaxModemRelayVolume FaxBypassOutputGain ModemBypassOutputGain T38MaxDatagram T38FaxMaxBufferSize DetFaxOnAnswerTone NTEMaxDuration EchoCancellerAggressiveN LP Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 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 73 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 Modem Bypass dynamic payload type The range is 0 127 The default value is 103 Determines the fax gain control The range 18 to 3 corresponds to 18 dBm to 3 dBm in 1 dB steps The default i
23. Parameter Description IP to Tel Routing Mode Determines whether to route IP calls to the Hunt Group before or RouteModelP2Tel after manipulation of destination number configured in Configuring the Number Manipulation Tables on page 151 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 Dest Host Prefix The request URI host name prefix of the incoming SIP INVITE PstnPrefix_DestHostPrefix 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 155 However the asterisk wildcard cannot be used to depict any source host prefix Source Host Prefix The From URI host name prefix of the incoming SIP INVITE PstnPrefix_SrcHostPrefix 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 155 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 Version 5 6 165 November 2008 ca AudioCodes Parameter
24. 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 Cisco compatible fax and modem bypass mode 0 NSE disabled default 1 NSE enabled Notes This feature can be used only if VxxModemTransportType 2 Bypass If 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 VxxModemTransportType 2 for all modems Configure the gateway parameter NSEPayloadType
25. 4923 30 h323 disconnect cause 22 0x16 4923 27 h323 call type VOIP 4923 26 h323 call origin Originate 4923 24 h323 conf id 02102944 600a1899 3fd61009 Oe2f3cc5 7 11 Call Detail Record The Call Detail Record CDR contains vital statistic information on calls made by the device 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 CDRSyslogServerlP For CDR in RADIUS format refer to Supported RADIUS Attributes on page 336 The following table lists the CDR fields that are supported Table 7 3 Supported CDR Fields Field Name Description ReportType Report for either Call Started Call Connected or Call Released Cid Port Number Callld SIP Call Identifier Trunk Physical Trunk Number always set to 1 as not applicable BChan Selected B Channel always set to 0 as not applicable Conld SIP Conference ID TG Trunk Group Number EPTyp Endpoint Type Orig Call Originator IP Tel Sourcelp Source IP Address Destlp Destination IP Address TON Source Phone Number Type NPI Source Phone Number Plan SrcPhoneNum Source Phone Number SrcNumBeforeMap Source Number Before Manipulation TON Destination Phone Number Type NPI Destination Phone Number Plan DstPhoneNum Destination Phone Number DstNumBeforeMap Destination Number Before Manipula
26. 7 3 2 3 7 3 2 4 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 the 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 To configure NSE mode perform the following configurations IsFaxUsed 0 FaxTransportMode 2 NSEMode 1 NSEPayloadType 100 V21ModemTransportType 2 V22ModemTransportType 2 V23ModemTransportType 2 V32ModemTransportType 2 V34ModemTransportType 2 BellModemTransportType 2 Fax Modem Transparent
27. 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 SIP User s Manual 366 Document LTRT 65411 SIP User s Manual 8 Networking Capabilities 8 2 2 8 23 To enable STUN perform the following m Enable the STUN feature using either the Web interface refer to Configuring the Application Settings on page 58 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 58 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 inifile parameter NATBindingDefaultTimeout to define the default NAT binding lifetime in seconds STUN is used to refresh the binding information after this time expires STUN onl
28. DNSPriServerIP DNS Secondary Server IP DNSSecServerlP STUN Settings Enable STUN EnableSTUN SIP User s Manual MediaPack Series Description Defines the time interval in seconds that the NTP client reguests 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 80 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 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 Qua
29. Figure 3 60 Coders Page Packetization Time Payload Type Silence Suppression 6723 1 30 v 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 SIP User s Manual 124 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 From the Rate drop down list select the bit rate in kbps for the selected coder 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 209 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 t
30. Gs ila CELLS Ol elas table Item Index Item Namel Item Name2 Item Name3 is the Format line valuel value2 value3 Item 1 valuel value3 These are the Data lines NTable Title This is the end of the table mark Refer to the following notes E 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 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 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 ina 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 large
31. Terminology for Navigation Tree Levels Navigation Tree in Basic and Full View Showing and Hiding Navigation Pane Toggling between Basic and Advanced Page View Expanding and Collapsing Parameter Groups Editing Symbol after Modifying Parameter Valu Value Reverts to Previous Valid Value Adding an Index Entry to a Table 33 Compacting a Web Interface Table 34 Searched Result Screen si z AD Scenario Creation Confirm Message B Box z GO Creating a Scenario 7 A si OF Scenario Loading Message Box ie OO Scenario Example me z OO Scenario File Page a40 Scenario Loading Message Box ais 42 Message Box for Confirming Scenario Deletion Confirmation Message Box for Exiting Scenario Mode Customizing Web Logo and Product Name Image Download Screen hes Toe for Current aa E E E E E E E E E ENE E T MP 124 Home Page Shortcut Menu when Clicking Port e g MP 11x Text Box for Typing Port Name e g MP 11x A E E RU Shortcut Menu when Clicking Port Port Settings e g MP 11x Basic Channel Information Page E ETER Shortcut Menu when Clicking Port Reset Channel eg MP 11x Log Off Contirmation BOK iisicacctadcdictoinausaiaianantiannaannne Web Session Logged Off IP Settings Page Confirmation Message for Accessing the Multiple Interface Table ee ere Interface Table P O KUN eV JE SV esas SE CR susan 56 Application
32. Via SIP 2 0 UDP 10 8 201 108 branch z9hG4bKacRKCVBud From lt sip 8000 10 8 201 108 gt tag 1c5354 ROM So OOO OOP 00 eg es Call ID 534366556655skKw 8000 1000 10 8 201 108 User Agent Audiocodes Sip Gateway MediaPack v 5 40 010 006 CSeq 18154 BYE Supported 100rel 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 MediaPack v 5 40 010 006 CSeq 18154 BYE Supported 100rel em Content Length 0 7 13 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 I
33. 3 second 8 6 second 4 30 second 5 60 second Note This parameter only takes effect from the next reset of the device ExtBootPReqEnable 0 Disable default 1 Enable extended information to be sent in BootP request SIP User s Manual 248 Document LTRT 65411 SIP User s Manual Parameter Serial Parameters DisableRS232 SerialBaudRate SerialData SerialParity SerialStop SerialFlowControl Version 5 6 4 ini File Configuration Description 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 recommend
34. Diverted party new destination of the forwarded call FXS or FXO device The served party FXS interface can be configured through the Web interface refer to Call Forward on page 178 or ini file to activate one of the call forward modes These modes are configurable per device s endpoints SIP User s Manual 360 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 7 14 5 7 14 6 When call forward is initiated the device sends a SIP 302 response with a contact that contains the phone number from the forward table and its corresponding IP address from the routing table or when a proxy is used the proxy s IP address For receiving call forward the device handles SIP 3xx responses for redirecting calls with a new contact Call Waiting The Call Waiting feature enables FXS devices to accept an additional second call on busy endpoints If an incoming IP call is designated to a busy port the called party hears call waiting tone several configurable short beeps and for Bellcore and ETSI Caller IDs can view the Caller ID string of the incoming call The calling party hears a Call Waiting Ringback Tone The called party can accept the new call using hook flash and can toggle between the two calls To enable Call Waiting m Set EnableCallWaiting 1 or using the Web interface refer to Supplementary Services on page 138 E Set EnableHold 1 Define the Call Waiting indication and Call Waiti
35. Don t Filter device doesn t 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 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 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 port phone line after the call is answered line offhooked FXS or seized FXO for IP to Tel calls 0 Disable Di
36. EFIX SrcHostPrefix PREFIX_TransportType PR EFIX 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 PREFIX_TransportType PREFIX_SrcTrunkGroupID PREFIX 0 guest 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 160 The parameters PREFIX_SrclPGroupID PREFIX_DestHostPrefix and PREFIX_SrcHostPrefix are currently not applicable and must be left empty or 1 They are used only for IP to IP routing supported in the next applicable release 290 Document LTRT 65411 SIP User s Manual Parameter PSTNPrefix Version 5 6 4 ini File Configuration Description 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 dec
37. FXSOOSBehavior 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 proxies and not only with the active one 119 November 2008 7a K tal AudioCodes MediaPack Series Parameter Description Mutual Authentication Mode Determines the device s mode of operation when Authentication and MutualAuthenticationMod Key Agreement AKA Digest Authentication is used e 0 Optional Incoming requests that don t include AKA authentication information are accepted default 1 Mandatory Incoming requests that do
38. For a detailed explanation on SAS and for configuring various SAS setups refer to Stand Alone Survivability SAS Feature on page 315 For additional SAS parameters configurable only using the ini file refer to SIP Configuration Parameters on page 260 Version 5 6 149 November 2008 7a c tal AudioCodes MediaPack Series gt To configure the Stand Alone Survivability parameters take these 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 67 SAS Configuration Page h Enable SAS Disable ew SAS Local SIP UDP Port 5080 SAS Default Gateway IP SAS Registration Time 20 E Short Number Length o m 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 209 Table 3 36 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
39. Jr Min Hook Flash Detection Period msec 300 Max Hook Flash Detection Period msec 700 2 Configure the hook flash 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 209 Table 3 17 Hook Flash Settings Parameters Parameter Description Min Flash Hook Detection Defines the minimum time in msec for detection of a hook flash event Period msec The valid range is 25 to 300 The default value is 300 MinFlashHookTime Detection is guaranteed for hook flash periods of at least 60 msec when setting the minimum time to 25 Hook flash signals that last a shorter period of time are ignored Notes This parameter is applicable only to FXS interfaces It s recommended to reduce the detection time by 50 msec from the desired value e g if you want to set the value to 200 msec then enter 150 msec i e 200 minus 50 Version 5 6 T November 2008 7a tal AudioCodes MediaPack Series Parameter Description Max Flash Hook Defines the hook flash period in msec for both analog and IP sides Detection Period msec For the IP side it defines the hook flash period that is reported to the IP FlashHookPeriod For the analog side it defines the following FXS interfaces Maximum hook flash detection period A longer signal is considered an off hook or on hook event FXS inte
40. Loading Auxiliary Files on page 210 The Coeff dat file consists of a set of parameters for the signal processor of the loop interface devices This parameter set provides control of the following AC and DC interface parameters DC battery feed characteristics AC impedance matching Transmit gain Receive gain Hybrid balance Frequency response in transmit and receive direction Hook thresholds Ringing generation and detection parameters SIP User s Manual 312 Document LTRT 65411 SIP User s Manual 6 Auxiliary Configuration Files This means for example that changing impedance matching or hybrid balance doesn t require hardware modifications so that a single device is able to meet requirements for different markets The digital design of the filters and gain stages also ensures high reliability no drifts over temperature or time and simple variations between different line types In future software releases it is to be expanded to consist of different sets of line parameters which can be selected in the ini file for each port Note To configure the FXO coefficients use the parameter CountryCoefficients described in Analog Telephony Parameters on page 279 6 5 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
41. MWI for remote extensions and voice mail applications Instead of subscribing to an MWI server to receive notifications of pending messages the FXO device receives subscriptions from the remote FXS device and notifies the appropriate extension when messages and the number of messages are pending The FXO device detects an MWI message from the Tel PBX side using any one of the following methods m 100 VDC sent by the PBX to activate the phone s lamp Stutter dial tone from the PBX m MWI display signal according to the parameter CallerIDType Version 5 6 349 November 2008 A c AudioCodes MediaPack Series Upon detection of an MWI message the FXO device sends a SIP NOTIFY message to the IP side When receiving this NOTIFY message the remote FXS device generates an MWI signal toward its Tel side Figure 7 11 MWI for Remote Extensions FXO Device FXS Device Remote PBX Extensions 7 13 4 4 Call Waiting for Remote Extensions When the FXO device detects a Call Waiting indication FSK data of the Caller Id CalleriIDType2 from the PBX it sends a proprietary INFO message which includes the caller identification to the FXS device Once the FXS device receives this INFO message it plays a call waiting tone and sends the caller ID to the relevant port for display The remote extension connected to the FXS device can toggle between calls using the Hook Flash button Figure 7 12 Call Waiting for Remote Ext
42. 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 following 6 steps 1 Access the Management Settings page as described in Configuring the Management Settings on page 199 Version 5 6 205 November 2008 7a E tal AudioCodes MediaPack Series 2 In the SNMP Trusted Managers field click the right pointing arrow u button the SNMP Trusted Managers page appears Figure 3 96 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 209 3 5 1 2 Co
43. Scenario For a description on the Navigation tree views refer to Navigation Tree on page 25 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 can create a Scenario Version 5 6 37 November 2008 ca AudioCodes 3 3 5 2 Accessing a Scenario MediaPack Series 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 A 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 Sennarios Search Available Parameter Basic Parameter List a hd O O Basic Full Scenario Max Digits In Phone Num 5 Scenario Name PBX Inter Digit Timeout for Overlap Dialing sec Interoperability 1 Define Coders LESE AKON 2 Define Max Digits bc 2nd Tx DTMF Option 3 Definie Voice Mail nmaa p 3rd Tx DTMF Option 4th Tx DTMF Option Sth Tx DTMF Option RFC 2833 Payload Type Hook Flash Option Digit Mapping Rules Dial Tone Durati
44. Sth Tx DTMF Option Feed Selected ZE DTMF Dialing j RFC 2833 Payload Type PUBSIP Advanced Parameters Page i Mook Flash Option B marepulstion Tables Digi Mapping fules routing Tables u Profile Dehrsbons Endoven Setinas Scenario Name PBX Enadle Special Digits Interoperability i Define Coders Deal Tone Duration sec Hotline Dial Tone Duration sec Default Destination Number Special Digit Representaton Added Scenario Step Scenario Mame PEX Intesoperab ty Defining Scenario Name Nex Button gt Step Name SIPPODIMF Defining Step Name pica Save 8 Firish Cancel Scenarios Oeti 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
45. Type V23ModemTransportTy pe V 32 Modem Transport Type V32ModemTransportTy pe V 34 Modem Transport Type V34ModemTransportTy pe Fax Relay Redundancy Depth FaxRelayRedundancyD epth Fax Relay Enhanced Redundancy Depth FaxRelayEnhancedRed undancyDepth Fax Relay ECM Enable FaxRelayECMEnable Fax Relay Max Rate bps FaxRelayMaxRate SIP User s Manual MediaPack Series Description 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 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 2 Determines whether the Error Corr
46. and the ringing signal stops for Ring Detection Timeout the FXO releases the IP call Ring Detection Timeout supports full ring cycle of ring on and ring off from ring start to ring start 7 4 2 4 FXO Supplementary Services The FXO supplementary services include the following Version 5 6 Hold Transfer toward the Tel side The ini file parameter LineTransferMode must be set to 0 default If the FXO receives a hook flash from the IP side using out of band or RFC 2833 the device sends the hook flash to the Tel side by performing one of the following e Performing a hook flash i e on hook and off hook e Sending a hook flash code defined by the ini file parameter HookFlashCode The PBX may generate a dial tone that is sent to the IP and the IP side may dial digits of a new destination Blind Transfer to the Tel side A blind transfer is one in which the transferring phone connects the caller to a destination line before ringback begins The ini file parameter LineTransferMode must be set to 1 331 November 2008 e AudioCodes MediaPack Series The blind transfer call process is as follows e FXO receives a REFER request from the IP side e FXO sends a hook flash to the PBX dials the digits that are received in the Refer To header and then drops the line on hook Note that the time between flash to dial is according to the WaitForDialTime parameter e PBX performs the transfer internally Hold
47. basic and advanced parameters refer to Displaying Basic and Advanced Parameters on page 29 m Display of parameter groups refer to Showing Hiding Parameter Groups on page 30 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 80 If a page is read only Read Only Mode is displayed at the bottom of the page SIP User s Manual 28 Document LTRT 65411 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 m 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 Advanced Parametr Un Declare RFC 2833 in SOP No ist Tx OTMF Option RFC 2833 and Tx OTMF Opbon Jed Tx DTMF Option 4th T
48. 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 73 November 2008 7a tal AudioCodes MediaPack Series 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 Toconfigure 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 44 RTP RTCP Settings Page General Settings Dynamic Jitter Buffer Minimum Delay Dynamic Jitter Buffer Optimization Factor RTP Redundancy Depth Packing Factor Basic RTP Packet Interval Defaut RTP Directional Control RTPTxAx RFC 2833 TX Payload Type RFC 2833 RX Payload Type RFC 2198 Payload Type Fox Bypass Payload Type Enable RFC 3389 CN Payload Type Analog Signal Transport Type RTP Base UDP Port Comfort Noise Generation Negotiation Remote RTP Base UDP Port RTP Multiplexing Local UDP Port RTP Multiplexing Remote UDP Port
49. 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 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 reguired 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 36 Document LTRT 65411 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 Gontguraton 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 Setbogs Declare AFC 2033 in SDP t Wsecunty Setungs ist Tx OTMF Option i Protocol Configuration 2nd Tx DTMF Option wd Protocol Oeafirebon Brd Tx DTMF Option SIP General Parameters 4th Tx DTMF Option Proxy Registrabon E9 m
50. 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 209 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 32 You can also configure the NFS table using the ini file table parameter NFSServers refer to Networking Parameters on page 236 SIP User s Manual 62 Document LTRT 65411 SIP User s Manual 3 Web Based Management Table 3 10 Network Settings NFS Settings Parameters Parameter Index Host Or IP Root Path NFS Version Authentication Type UID GID VLAN Type Description The row index of the remote file system The valid range is 0 to 4 The domain name or IP address of the NFS server If a domain name is provided a DNS server must be configured 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
51. wd AudioCodes MediaPack Series Table 3 55 SNMP VS Users ParameteT S n ccceseccieacsisssnecdeccnesdoosenponsessnsaabbuseniaavirtcorenesenseabortaoaaborenanaen 205 Table 3 56 Auxiliary Files DESGPIOIS u cisstesalecantasieasdadadinncdtpseiedantaaiiessbeaeueanpaainiedinsebatanbaainiesinaansceibedll 210 Table 3 57 Ethernet Port Information Parameters ccccccceceececceceeeeeeeeeecneeeeeeeeeseeeenanaeeeeeeteneeees 220 Table 3 56 Call Counters DESC rit aie cssivsinsesercxsandossbnennecsnsranirvbpsebessusaabavreniasinisldoipireniaabinttoaenvrennate 224 Table 3 59 Call Routing Status Parameters ccccccccccccssccssececsseeecsseceseeeccsseeeesseeeeeecsseesesseeeeseeess 226 Table 3 60 SAS Registered Users Parameters ccccccccccsccssssecsseeccsseceseeeecseeeesseceeeecseesesseeeeaeeess 228 Table 3 61 IP Connectivity PIN acest kodu ladniehiakabiasiba wns dn dened tddaedueebad wie kakoiuokubk n esUbk nisl 229 Table 4 1 Networking ini File Parameters ccccceecceesceeeeeeeeceeeceeeseeeeeaeesaeeceesaeseaeseaeseeeeereetees 236 Table 4 2 System ini File Parameters inais iannau ianea iaaa niaaa iaaii aa aE aani ada 244 Table 4 3 Web and Telnet ini File Parameters cccccccsccssscecsseeecsseceseeecseescsseeseeeeseesesseeseaeeens 250 Table 4 4 Security ini File PAraMetETS ee 2eee0 000000000 eee nn denn dan h ARA A RK Ann A ena tn nn 252 Table 4 5 RADIUS ini File Parameters
52. 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 42 Voice Settings Page Voice Volume 32 to 31 dB 0 Input Gain 32 to 31 dB 0 Silence Suppression Disable DTMF Transport Type MuteDTMF MF Transport Type RFC2833R alay MF DTMF Volume 31 to 0 dB 11 NTE Max Duration 1 Enable Answer Detector Disable Answer Detector Activity Delay 0 Answer Detector Silence Time 10 Answer Detector Redirection 0 Answer Detector Sensitivity DTMF Generation Twist 0 Echo Canceler Enable Version 5 6 67 November 2008 ca AudioCodes MediaPack Series 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 209 Table 3 13 Media Settings Voice Settings Parameters Parameter Voice Volume VoiceVolume Input Gain InputGain Silence Suppression EnableSilenceCompression Echo Canceler EnableEchoCanceller DTMF Transport Type DTMFTransportType MF Transport Type MFTransportType SIP User s Manual Description Voice ga
53. 100 258 Document LTRT 65411 SIP User s Manual Parameter SNMP Trap Parameters SNMPManagerTablelP x SNMPManagerTrapPort x SNMPManagerTrapUser x SNMPManagerlsUsed x SNMPManagerTrapSending Enable x SNMPTrapManagerHostNa me 4 ini File Configuration Description For a description of this parameter refer to Configuring the SNMP Managers Table on page 201 For a description of this parameter refer to Configuring the SNMP Managers Table on page 201 This parameter can be set to the 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 201 For a description of this parameter refer to Configuring the SNMP Managers Table on page 201 For a description of this parameter refer to Configuring the Management Settings on page 199 SNMP Community String Parameters SNMPReadOnlyCommunity String_x SNMPReadWriteCommunity String_x SNMPTrapCommunityStrin g SNMP v3 Users Parameters SNMPUsers Version 5 6 For a description of this parameter refer to Configuring the SNMP Community Strings on page 203 For a description of this parameter refer to Configuring the SNMP Community Strings on page 203 For a description of this parame
54. 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 FaxModemBypassBasicRtpPacketlInterval 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 100 300 Document LTRT 65411 SIP User s Manual Parameter V21ModemTransportType V22ModemTransportType V23ModemTransportType V32ModemTransportType V34ModemTransportType V34FaxTransportType UserDefinedToneDetectorE nable BellModemTransportType InputGain VoiceVolume RTPRedundancyDepth RFC2198PayloadType EnableSilenceCompression IsCiscoSCEMode Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer t
55. 152 o O O0 IN IPV4 lt IPAdressA gt 0 di 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 oOdmt nod tow ou ou al Ge fl 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 VBD 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 4 FXO Operating Modes This section provides a description of the FXO operating modes and device configurations for Tel to IP and IP to Tel calls 7 4 1 IP to Telephone Calls The FXO device provides the following operating modes for IP to Tel calls m One stage dialing refer to One Stage Dialing on page 327 e Waiting for dial tone Two Stage Dialing on page 328 e Time to wait before dialing e Answer supervision m Two stage dialing refer to Two Stage Dialing on page 328 Dialing time e Disconnect supervision refer to Call Termination Disconnect Supervision on FXO Devices on page 328 e DID wink refer to DID Wink on page 329
56. 3 6 1 2 Viewing the Ethernet Port Information a vient 3 6 1 3 Viewing Active IP Interfaces eee ieee k annsi 220 3 6 1 4 Viewing Device Information 3 6 1 5 Viewing Performance Statistics Ps W 36 1 6 Viewing Active AlATMS lt z iiukekotukusicorskeee cca mansi est z eee 220 3 6 2 Gateway Statistics E a kt Gael db 3 6 2 1 Call Counters Pe 3 6 2 2 Call Routing Status 3 6 2 3 Registration Status ja debated EEA AET A EN E E 3 6 2 4 SAS SBC Registered Users E E AE E sk navod v neddkao aj 3 6 2 5 IP Connectivity sce ob ato sis ho cae sad jo 2 4 iM File CORONA asus Ka P M A KA A KBA A NA AB NN SK O BAN 231 4 1 Secured Encoded ini File EE E PENEAN E E EE E du l S 42 The mi File Sie acini cain eee a dor BOM RIES aan A S AEAEE E 232 4 2 2 Structure of Individual ini File Parameters cccccccscesssecsseeeesteeecsseeesseeeeaee 232 4 2 3 Structure of ini File Table Parameters ccccccccccsscecssececeecsseeeesseeeesseeeseeeesaes 233 SIP User s Manual 4 Document LTRT 65411 SIP User s Manual Contents 5 2 Restoring Factory ARON OVA OKO O PAS KOA 305 6 aa anakaa P ossaa 6 2 Conf igurinc 73 Fax je M 7 34 74 Version 5 6 s November 2008 Gg wi AudioCodes MediaPack Series 7 4 1 3 Call Termination iene Supervision on FXO Devices 328 7 4 1 4 DID Wink Seperate ee ee en hud a ba 7 4 2 Telephone to IP Calls boson aa
57. AATA T AN ome 3 4 3 7 Configuring the IKE Table E O E N N OF 3 4 4 Protocol Configuration AEE EARE ARER E 3 4 4 1 Configuring the Protocol Definition Parameters dainatiaoeeinceiactunagetsecterzas NOW 3 4 4 2 Configuring the SIP Advanced ParameterS ccccccccccsceseeseesseeees 129 3 4 4 3 Configuring the Number Manipulation Tables E EIEEE A EATER EE SAAA Configuring the Routing Tables sisiiissievariisnissiiseniiasssriranasi sensas LOR 3 4 4 5 Configuring the Profile Definitions eee 169 3 4 4 6 Configuring the Endpoint Settings EE AE E A LEA 3 4 4 7 Configuring the Endpoint Phone Numbers E E A ON 3 4 4 8 Configuring the Hunt and IP Groups anarien iA 3 4 5 Advanced Applications PEN 3 4 5 1 Configuring the Voice Mail VM Parameters EOE 3 4 5 2 Configuring RADIUS Accounting Parameters 34 5 3 Configuring the FXO Parameters sinisini adia dd d nesss 3 5 Management Tab A PPR ore 3 5 1 Management Ganfeurstiaii 3 5 1 1 Configuring the Management Settings ET 3 5 1 2 Configuring the P d oe Sudslava dale dd Aha a OO 3 5 1 3 Maintenance Actions ee 3 5 2 Software Update 3 5 2 1 Loading Auxiliary Files 3 5 2 2 Software Upgrade Wizard 3 5 2 3 Backing Up and Restoring Configuration 3 6 Status amp Diagnostics Tab z ed od Ab esac to o 3 6 1 Status amp Diagnostics SE E 3 6 1 1 Viewing the Device S Syslog Messages
58. Compression Silence Suppression Packet Loss Concealment Echo Canceler Gain Control DTMF Transport in band DTMF Detection and Generation Call Progress Tone Detection and Generation Answer Detector Output Gain Control Input Gain Control Fax Modem Relay Fax Relay Modem Transparency Protocols VoIP Signaling Protocol Communication Protocols Line Signaling Protocols Processor Control Processor Control Processor Memory Signal Processors Version 5 6 10 Selected Technical Specifications Specification G 711 PCM at 64 kbps u law A law 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 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 G 711 appendix 1 G 723 1 G 729 a b G 165 and G 168 2000 64 msec Configurable Mute transfer in RTP payload or relay in compliance with RFC 2833 Dynamic range 0 to 25 dBm compliant with TIA 464B and Bellcore TR NWT 000506 32 tones single tone dual tones or AM tones configurable 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 Speech detection 32 dB to 31 dB in steps of 1 dB 32 dB to 31 dB in steps of 1 dB Group 3 fax relay up to 14 4 kbps with automatic fallback T 38 compliant real time fax relay Tolerant networ
59. Concealment Echo Canceler Gain Control DTMF Transport In Band DTMF Detection and Generation Answer Detector Call Progress Tone Detection and Generation Output Gain Control Input Gain Control Fax Modem Relay Fax Relay Modem Transparency Protocols VoIP Signaling Protocol Communication Protocols Line Signaling Protocols SIP User s Manual MediaPack Series Specification Note For country specific coefficients use the parameter CountryCoefficients Caller ID detection Bellcore GR 30 CORE Type 1 using Bell 202 FSK modulation ETSI Type 1 NTT Denmark India Brazil and DTMF ETSI CID ETS 300 659 1 Immediate or smooth to prevent erroneous ringing 12 16 KHz sinusoidal bursts applicable only to FXS interfaces By frequency 15 100 Hz and cadence patterns DC voltage generation TIA EIA 464 B V23 FSK data stutter dial tone and DTMF based G 711 PCM at 64 kbps u law A law G 723 1 MP MLO at 5 3 or 6 3 kbps G 726 at 32 kbps ADPCM G 729 CS ACELP 8 Kbps Annex A B 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 G 711 appendix 1 G 723 1 G 729 a b G 165 and G 168 2000 64 msec Configurable Mute transfer in RTP payload or relay in compliance with RFC 2833 Dynamic range 0 to 25 dBm compliant with TIA 464B and Bellcore TR NWT 000506 Speech detection 32 tones single tone du
60. Defined Second Proposal Authentication Type Not Defined Second Proposal DH 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 Fourth Proposal DH Group Not Defined 2 From the Policy Index drop down list select the peer you want to edit up to 20 peers can be configured 3 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 209 To delete a peer from the IKE table select it from the Policy Index drop down list click the button Delete 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 25 Default IKE First Phase Proposals Proposal Encryption Authentication DH Group Proposal 0 3DES SHA1 1024 Proposal 1 3DES MD5 1024 Proposal 2 3DES SHA1 786 Proposal 3 3DES MD5 786 SIP User s Manual 98 Document LTRT 65411 SIP User s Manual 3 Web Based M
61. Description interface refer to Reasons for Alternative Routing on page 168 Foran explanation on usng ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a description of this parameter refer to Advanced Parameters on page 129 Alternative Routing Parameters RedundantRoutingMode AltRoutingTel2IPEnable AltRoutingTel2IPMode AltRoutingTel2IPConnM ethod AltRoutingTel2IPKeepAli veTime AltRoutingToneDuration IPConnQoSMaxAllowed PL IPConnQoSMaxAllowed Delay Phone Context Parameters AddPhoneContextAsPre fix PhoneContext Version 5 6 For a description of this parameter refer to Proxy amp Registration Parameters on page 112 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to Mapping NPI TON to Phone Context on page 155 This ini file table
62. Dial Tone Duration HotLineToneDuration SIP User s Manual Description 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 FXO interfaces support the receipt of RFC 2833 Hook Flash signals The FXS interfaces send Hook Flash signals only if EnableHold is set to 0 Digit map pattern 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 maximum 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
63. Echo Canceller Non Linear Processor Mode off m Jitter buffering optimizations To configure fax modem transparent with events mode perform the following configurations m IsFaxUsed 0 E FaxTransportMode 3 m V21ModemTransportType 3 m V22ModemTransportType 3 m V23ModemTransportType 3 m V32ModemTransportType 3 m V34ModemTransportType 3 E BellModemTransportType 3 7 3 2 6 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 Version 5 6 Echo Canceller off Silence Compression off Echo Canceller Non Linear Processor Mode off Dynamic Jitter Buffer Minimum Delay 40 Dynamic Jitter Buffer Optimization Factor 13 323 November 2008 7a c tall AudioCodes MediaPack Series 7 3 2 7 7 3 3 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 m 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 c
64. FQDN DNS resolution is performed according to DNSQueryType For available notations that represent multiple numbers refer to Dialing Plan Notation on page 155 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to IP to Hunt Group Routing on page 163 For a description of this parameter refer to Tel to IP Routing Table on page 160 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 If enabled the device swaps the calling and called numbers received from the Tel side The INVITE message contains the 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 101 This ini file table parameter manipulates the destination number of Tel to IP calls The format of this parameter is as follows NumberMapTel2Ip FORMAT NumberMapTel2Ip Index NumberMapTel2Ip DestinationPrefix NumberMapTel2Ip SourcePrefi
65. FXO Enable w Port 7 FXO Enable v Port 8 FXO Enable 3 In the Tel to IP Routing page enter 10 in the Destination Phone Prefix field and the IP address of the FXS device 10 1 10 3 in the field IP Address Figure 7 18 FXO Tel to IP Routing Configuration Dest Phone Prefix Source Phone Prefix R Dest IP Address 1 10 101103 4 Inthe FXO Settings page refer to Configuring the FXO Parameters on page 195 set the parameter Dialing Mode to Two Stages IsTwoStageDial 1 7 13 5 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 and two Internet Telephony Service Providers ITSP using AudioCodes device Scenario In this example an Enterprise has depployed the 8 FXS port MediaPack The first four phones connected to MediaPack s FXS ports are to operate with ITSP 1 using UDP while the next four phones channels 5 8 are to operate with ITSP 2 using TCP ITSP 1 requires single registration i e one registration for all four phones while ITSP 2 requires registration per phone Each ITSP implements two servers for redundancy and load balancing The fi
66. 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 Instead 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 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 refer to Saving a Scenario to a PC on page 40 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 249 The Scenario dat file mu
67. Firewall Settings on page 84 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 336 Table 4 5 RADIUS ini File Parameters Parameter EnableRADIUS AAAlndications BehaviorUponRadiusTimeout MaxRADIUSSessions SharedSecret RADIUSRetransmission RadiusTO RADIUSAuthServerlP RADIUSAuthPort RADIUSAccServerlP RADIUSAccPort RadiusAccountingType DefaultAccessLevel RadiusLocalCacheMode Version 5 6 Description For a description of this parameter refer to Configuring the General Security Settings on page 90 For a description of this parameter refer to Configuring RADIUS Accounting Parameters on page 194 For a description of this parameter refer to Configuring the General Security Settings on page 90 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 90 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 pa
68. 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 62 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 209 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 reverts to Disable Throughout the DHCP procedure the BootP TFTP application 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 e
69. 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 RegistrarIP is configured The IP addresses list is 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 Determines whether Keep Alive with the Proxy is enabled or disabled Thi
70. 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 3 4 3 4 3 Self Signed Certificates The device is shipped with an operational self signed server 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 Version 5 6 To change the subject name and regenerate the self signed certificate take these 4 steps 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
71. Interfaces and VLANs AudioCodes Media Gateway Shared ___ omen Network Internet Separated Networks Scheme Version 5 6 371 November 2008 ca AudioCodes MediaPack Series For security the VLAN mechanism is activated only when the device is loaded from the flash memory Therefore when using BootP Load an ini file with VlanMode 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 52 The device must be connected to a VLAN aware switch and the switch s PVID must be egual to the device s native VLAN ID The mapping of an application to its CoS and traffic type is shown in the table below Application Debugging interface Telnet DHCP Web server HTTP SNMP GET SET Web server HTTPS IPSec IKE RTP traffic RTCP traffic T 38 traffic SIP SIP over TLS SIPS Syslog ICMP ARP listener SNMP Traps DNS client NTP NFS SIP User s Manual Traffic Network Types Management Management Management Management Management Management Determined by the service Media Media Media Control Control Management Management Determined by the initiator of the request Management DNS EnableDNSasOAM NTP EnableNTPasOAM NFSServers VlanType in the NFSServers table 372 Table 8 1 Traf
72. 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 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 235 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 45 November 2008 ca AudioCodes MediaPack Series 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 descripti
73. Media 8 Control Multiple IP networks 1 Dual IP OAM 8 Control Multiple IP networks 1 Dual IP OAM 8 Medial Multiple IP networks Note This parameter is not relevant when using Multiple Interface tables activated by clicking the Multiple Interface Table button L described below refer to Configuring the Multiple Interface Table on page 55 For detailed information on Multiple IPs refer to Multiple IPs on page 370 Version 5 6 53 November 2008 K tal AudioCodes MediaPack Series Parameter Single IP Settings IP Address Subnet Mask Default Gateway Address Description 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 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
74. 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 169 to apply certain adaptations to the channel used for fax modem e g to use the coder G 711 to set the jitter buffer optimization 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 V22ModemTransportType 0 V23ModemTransportType 0 V32ModemTransportType 0 SIP User s Manual 322 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 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 320 or Transparent with Events refer to Fax Modem Transparent with Events Mode on page 323 for modem 7 3 2 5 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 m Echo Canceller on or off for modems m
75. MyConference MLPP Multilevel Precedence and Preemption Call Priority Mode CallPriorityMode MLPP DiffServ MLPPDiffserv Precedence Ringing Type PrecedenceRingingTyp e 3 4 4 2 3 Metering Tones Enables Priority Calls handling 0 Disable Disable default 1 MLPP Priority Calls handling is enabled Defines the DiffServ value differentiated services code point 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 Defines the index of the Precedence Ringing tone in the Call Progress Tones CPT file This tone is used when the parameter CallPriorityMode is set to 1 and a Precedence call is received from the IP side The valid range is 1 to 16 The default value is 1 i e plays standard Ringing tone The FXS interfaces can generate 12 16 KHz metering pulses towards the Tel side e g for connection to a payphone or private meter Tariff pulse rate is determined according to an internal table This capability enables users to define different tariffs according to the source destination numbers and the time of day The tariff rate includes the time interval between the generated pulses and the number of pulses generated on answer Note SIP User s Manual The Metering Tones page is available only for FXS interfaces 144 Document LTRT 65411 SIP User s Manual 3 Web Based Management gt To configure
76. 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 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 41 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 Seeman File L 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 40 Document LTRT 65411 SIP User s Manual 3 Web Based Management Click the Get Scenario
77. Private TON Select the Number Type assigned to this entry If 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 SIP User s Manual 156 Document LTRT 65411 SIP User s Manual Parameter Phone Context 3 4 4 4 3 Web Based Management Description Level 0 Regional Local 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 The Phone Context SIP URI parameter Configuring the Routing Tables The Routing Tables submenu allows you to configure the device s call routing This submenu includes the following page items 3 4 4 4 1 Routing General Parameters refer to Routing General Parameters on page 157 Tel to IP Routing refer to Tel to IP Routing Table on page 160 IP to Hunt Group Routing refer to IP to Trunk Group Routing on page 163 Internal DNS Table refer to Internal DNS Table on page 166 Internal SRV Table refer to Internal SRV Table on page 167 Reasons for Alternative Routing refer to Reasons for Alternative Routing on page 168 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 To configure the general routing parameters ta
78. Proxy Sets Table Page vw Proxy Set ID 1 v Proxy Address Transport Type 10 33 37 77 UDP w 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 5wap No 3 In the IP Group Table page refer to Configuring the IP Groups on page 186 configure the two IP Groups 1 and 2 Assign Proxy Sets 1 and 2 to IP Groups 1 and 2 respectively Figure 7 21 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 Disable v Disable v ITSP 2 Disable v Disable v 4 In the Endpoint Phone Number Table page refer to Configuring the Endpoint Phone Numbers on page 181 configure Hunt Group ID 1 for channels 1 4 and Hunt Group ID 2 for channels 5 8 Figure 7 22 Configuring Hunt Groups Channel s Phone Number Hunt Group ID Tel Profile ID 1 4 5 6 5 In the Hunt Group Settings page refer to Configuring the Hunt Group Settings on page 183 configure Per Account registration for Hunt Group ID 1 and associate it with IP Group 1 Configure Per Endpoint reg
79. RTP multiplexing is disabled 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 3 4 2 4 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 SIP User s Manual 76 Document LTRT 65411 SIP User s Manual 3 Web Based Management Figure 3 45 General Media Settings Page v General Settings Max Echo Canceller Length Default 4 Enable Continuity Tones Disable 2 Configure the general 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 209 Table 3 16 Media Settings Parameters Parameter Description Max Echo Canceller Length N A Enable Continuity Tones N A 3 4 2 5 Configuring the Hook Flash Settings The Hook Flash Settings page allows you to configure hook flash parameters gt To configure the Hook Flash parameters take these 4 steps 1 Open the Hook Flash Settings page Configuration tab gt Media Settings menu gt Hook Flash Settings page item Figure 3 46 Hook Flash Settings Page vw
80. 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 N A 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 parameters 1 IP Address 2 Subnet Mask 3 Default Gateway IP Address 4 ActivityListToLog 201 November 2008 7a K tall AudioCodes MediaPack Series 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 199 2 In the SNMP Trap Destinations field click the right pointing arrow button the SNMP Trap Destinations page appears Figure 3 93 SNMP Trap Destinations Page IP Address Trap Port Trap Enable SNMP Manager 10 6 2 28 162 Enable W SNMP Manager 0 0 0 0 Enable W SNMP Manager 0 0 0 0 Enable SNMP Manager 0 0 0 0 i Enable W SNMP Manager
81. SRV2IP_Port1 SRV2IP_Dns2 SRV2IP_Priority2 SRV2IP_Weight2 SRV2IP_Port2 SRV2IP_Dns3 SRV2IP_Priority3 SRV2IP_Weight3 SRV2IP_Port3 SRV2IP For example SRV2IP SRVZ2IP 0 SrvDomain O0 Dnsname1 1 1 500 Dnsname2 2 2 501 0 0 0 SRV2IP Notes This parameter can include up to 10 indices lf the 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 167 237 November 2008 A L e AudioCodes MediaPack Series Parameter EnableSTUN STUNServerPrimarylP STUNServerSecondarylP 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 233 For a description of this parameter refer to Configuring the Application Settings on page 58 For a description of this parameter refer to Configuring the Application Settings on page 58 For a description of this parameter refer to Configuring the Application Settings on page 58 Defines the domain name for the Simple Traversal of User Datagram Protocol STUN s
82. Submit button the toolbar displays the word Reset refer to Toolbar on page 23 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 To lock the device take these 5 steps Open the Maintenance Actions page refer to Maintenance Actions on page 207 Under the LOCK UNLOCK group from the Graceful Option drop down list select one of the following options 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 SIP User s Manual 208 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 Cl
83. 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 required after the burst tone ends and the tone detection is reported For continuous tones this parameter is ignored 308 Document LTRT 65411 SIP User s Manual 6 Auxiliary Configuration Files e 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 e 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 e Third Signal On Time 10 msec Signal On period in 10 msec units for the third caden
84. To save the changes to the flash memory refer to Saving Configuration on page 209 SIP User s Manual 146 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 4 4 2 5 Keypad Features The Keypad Features page applicable only to FXS interfaces enables you to activate and deactivate the following features directly from the connected telephone s keypad m Call Forward refer to Call Forward on page 178 m Caller ID Restriction refer to Caller ID on page 177 m Hotline refer to Automatic Dialing on page 175 The Keypad Features page is available only for FXS interfaces The method used by the device to collect dialed numbers is identical to the method used during a regular call i e max digits interdigit timeout digit map etc The activation of each feature remains in effect until it is deactivated i e not deactivated after a call gt To configure the keypad features take these 4 steps 1 Open the Keypad Features page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Keypad Features page item Figure 3 66 Keypad Features Page v Forward Unconditional No Answer On Busy On Busy or No Answer Do Not Disturb Deactivate vy Caller ID Restriction Activate Deactivate Hotline Activate Deactivate v Transfer Blind w Call Waiting Activate Deactivate wv Reje
85. Tones generated by any PBX or telephone network Relevant parameters DisconnectOnBusyTone and DisconnectOnDialTone m Detection of silence The call is disconnected after silence is detected on both call directions for a specific configurable amount of time The call isn t disconnected immediately therefore this method should only be used as a backup option Relevant parameters EnableSilenceDisconnect and FarEndDisconnectSilencePeriod m Special DTMF code A digit pattern that when received from the Tel side indicates to the device to disconnect the call Relevant ini file parameter TelDisconnectCode m Interruption of RTP stream Relevant parameters BrokenConnectionEventTimeout and DisconnectOnBrokenConnection Note This method operates correctly only if silence suppression is not used Protocol based termination of the call from the IP side Note The implemented disconnect method must be supported by the CO or PBX 7 4 1 4 DID Wink The device s FXO ports support Direct Inward Dialing DID DID is a service offered by telephone companies that enables callers to dial directly to an extension on a PBX without the assistance of an operator or automated call attendant This service makes use of DID trunks which forward only the last three to five digits of a phone number to the PBX If for example a company has a PBX with extensions 555 1000 to 555 1999 and a caller dials 555 1234 the local central office CO
86. Transfer toward the IP side The FXO device doesn t initiate hold transfer as a response to input from the Tel side If the FXO receives a REFER request with or without replaces it generates a new INVITE according to the Refer To header 7 5 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 1 Supported X Detect Event Types Events Type Subtype Required Configuration CPT SIT SITDetectorEnable 1 UserDefinedToneDetectorEnable 1 FAX CED IsFaxUsed 0 or IsFaxUsed 0 and FaxTransportMode 0 modem VxxModemTransportType 3 PTT voice start EnableDSPIPMDetectors 1 voice end The X Detect event notification process is as follows 1 For 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 re
87. Type ProxyDNSQueryType Version 5 6 3 Web Based Management Description 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 user 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 ty
88. User s Manual Parameter Version 5 6 4 ini File Configuration Description 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 170 The first group of coders indices O 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 Packetlnterval 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 case 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 co
89. User s Manual 3 Web Based Management 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 Z http 10 33 4 161 EndOfProcess Microsoft Internet Explorer O CMP Version ID 5 404 000 Call Progress Tone File Name 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 21 and click OK a message box appears informing you of the new CMP file Figure 3 107 Message Box Informing of Upgraded CMP File Microsoft Internet Explorer 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 3 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 p
90. a call to that destination gt To view the IP connectivity information take these 2 steps 1 Inthe 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 3 118 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 L 2 3 4 5 6 7 8 9 SIP User s Manual 228 Document LTRT 65411 SIP User s Manual Column Name IP Address Host Name Connectivity Method Connectivity Status Ouality Status Quality Info DNS Status Version 5 6 3 Web Based Management Table 3 61 IP Connectivity Parameters Description The IP address can be one of the following IP address defined as the destination IP address in the Tel to IP Routing table IP address resolved from the host name defined as the destination IP address in the Tel to IP Routing table Host name or IP address as defined in the Tel to IP Routing table The method according to which the destination IP address is gueried periodically ICMP ping or SIP OPTIONS request The status of the IP address connectivit
91. 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 reguests 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 5080 SAS Default Gateway IP The default gateway used in SAS Emergency Mode When an SASDefaultGatewaylP incoming SIP INVITE is received and the destination Address Of Record is not included in the SAS database the reguest 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 150 Document LTRT 65411 SIP User s Manual Parameter SAS Registration Time SASRegistrationTime 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
92. 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 m 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 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 speech 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 v
93. authenticating digest a Hunt 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_ServedIPGroup Account ServinglPGroup Account Username Account Password Account HostName Account Register Account ContactUser Account For example Account FORMAT Account Index Account ServedTrunkGroup Account ServedlPGroup Account ServinglPGroup 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 The table item Account ServedlPGroup is currently not applicable and must be left empty or assigned the value 1 It is used only for IP to IP routing applications supported in the next applicable release You can define multiple table indices having the same ServedTrunkGroup with different ServinglPGroups username password HostName and ContactUser This provides the capability for registering the same Hunt Group to several ITSP s i e Serving IP Groups For configuring 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 188 Foran explanation on using ini file table parameters refer to Structure of ini File Table Par
94. be able to communicate with an external device network on its OAMP and Control networks IP routing rules must be used 8 8 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 21 2 Use the Software Upgrade Wizard refer to Software Upgrade Wizard on page 212 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 52 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 v VLAN Mode Enable w VLAN ID Settings Native VLAN ID OAM VLAN ID Control VLAN ID Media VLAN ID Version 5 6 373 November 2008 7a e AudioCodes MediaPack Series c Click the Submit button to save your changes 4 Configure the multiple IP parameters by completing the following steps a Inthe IP Settings page modify the IP parameters to correspond to the values shown in the figure below Note that th
95. both Tel and IP sides and protocols each of which reguires different system behavior Version 5 6 169 November 2008 A e AudioCodes MediaPack Series 3 4 4 5 1 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 page 160 m IP to Hunt Group Routing page refer to IP to Trunk Group Routing on page 163 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 endpoints channels For example specific ports 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
96. by a user or for the endpoint telephone number Note These symbols can always be used as the first digit of a dialed number even if you disable this parameter Defines the default destination phone number used if the received message doesn t contain a called party number and no phone number is configured in the Endpoint Phone Number table refer to Configuring the Endpoint Phone Numbers on page 181 The parameter is used as a starting number for the list of channels comprising all hunt 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 ff default 1 Numeric Uses the numerical values 10 and 11 3 4 4 2 Configuring the SIP Advanced Parameters The SIP Advanced Parameters submenu allows you to configure advanced SIP control protocol parameters This submenu contains the following page items Version 5 6 Advanced Parameters refer to General Parameters on page 129 Supplementary Services refer to Supplementary Services on page 138 Metering Tones refer to Metering Tones on page 144 Charge Codes refer to Charge Codes Table on page 146 Keypad Features refer to Keypad Features on page 147 Stand Alone Survivability refer to Stand Alone Survivability on page 149 129 November 2008 7a tall AudioCod
97. can t be configured on the fly 4 Current Disconnect The device disconnects the current of the FXS endpoint This option can t be configured on the fly 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 Itis assumed that all Ringback Tones are defined in sequence in the CPT file In case 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 1 137 November 2008 A c tal AudioCodes MediaPack Series Parameter Emergency Calls Emergency Numbers EmergencyNumbers Emergency Calls Regret Timeout EmergencyRegretTim eout Description Defines a list of numbers which are defined as emergency numbers When one of these numbers is dialed the outgoing INVITE message includes the Priority and Resource Priority headers If the user sets the phone on hook the call is not disconnected but instead a Hold Re INVITE reguest is sent to the remote p
98. clicking the p 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 52 Document LTRT 65411 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 35 IP Settings Page IP Settings Jr IP Networking Mode Single IP Network Single IP Settings IP Address 10 13 4 13 Subnet Mask 255 255 0 0 lS Default Gateway Address 1 0 13 01 w Multiple Interface Settings Multiple Interface Table U p VLAN Mode w VLAN ID Settings Native VLAN ID OAM VLAN ID Control VLAN ID Media VLAN ID w NAT Settings IS 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 refer to Saving Configuration on page 209 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
99. default 1 Client certificates are required 255 November 2008 A K tal AudioCodes MediaPack Series Parameter HTTPSRootFileName HTTPSPkeyFileName HTTPSCertFileName VoiceMenuPassword Internal Firewall Parameters AccessList SIP User s Manual 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 Enhanced 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 For a description of this parameter refer to Configuring the General Security Settings on page 90 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 AccessLis
100. 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 370 For a description of the Web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 32 SIP User s Manual 56 Document LTRT 65411 SIP User s Manual Parameter Table parameters Index Application Type IP Address Prefix Length Gateway Version 5 6 3 Web Based Management Table 3 8 Multiple Interface Table Parameters Description Description Index of each interface The range is 0 3 Note Each interface index must be unigue 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 allowed on the interface 3 OAM 8 Media Only OAMP and Media RTP applications are allowed on the interface 4 OAM 8 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 interf
101. emergency number e g 911 in North America which it then redirects the call directly to the PSTN through its FXO interface The emergency number is configured using the jini file parameter SASEmergencyNumbers for a detailed description refer to SIP Configuration Parameters on page 260 Figure 7 1 Device s SAS Agent Redirecting Emergency Calls to PSTN IP Centrex Network Emergency Calls e g 911 Routed to PSTN Phones SIP User s Manual 316 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 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 EnableSAS 1 SASLocalSIPUDPPort default 5080 IsProxyUsed 1 ProxylP 0 lt external proxy IP address device gt ProxylP 1 lt external proxy IP address SAS gt IsRegisterNeeded 1 for the device IsFallbackUsed 0 SASRegistrationTime lt expiration time that SAS returns in the 200 OK to REGISTER in Emergency mode gt default 20 SASDefaultGatewaylP lt SAS gateway IP address gt SASProxySet 1 7 2 Configuring the DTMF Transport Types You can control the way DTMF digits are transported over the IP network to the remote
102. enable Call Waiting per port refer to Call Waiting on page 180 For information on the Call Waiting feature refer to Call Waiting on page 361 For information on the Call Progress Tones file refer to Configuring the Call Progress Tones File Number of Call Waiting indications that are played to the called telephone that is connected to the device FXS only for Call Waiting The valid range is 1 to 100 indications The default value is 2 Time in seconds between consecutive call waiting indications FXS only for call waiting The valid range is 1 to 100 The default value is 10 Defines the interval in seconds before a call waiting indication is played to the port that is currently in a call FXS only The valid range is 0 to 100 The default time is 0 seconds Duration in msec of call waiting indications that are played to the port that is receiving the call FXS only The valid range is 100 to 65535 The default value is 300 Determines whether Caller ID is enabled 0 Disable Disable the Caller ID service default 1 Enable Enable the Caller ID service 141 November 2008 A c tal AudioCodes MediaPack Series Parameter Caller ID Type CallerIDType Hook Flash Code HookFlashCode Description If the Caller ID service is enabled then for FXS interfaces calling number and Display text from IP are sent to the device s port For FXO interfaces the Caller ID signal is detected
103. 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 8 Dialing Parameters on page 125 e TxDTMFOption 1 ini file 1 to 5 Tx DTMF Option field INFO Nortel Web interface refer to DTMF 8 Dialing Parameters on page 125 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 8 Dialing Parameters on page 125 e TxDTMFOption 3 ini file 1 to 5 Tx DTMF Option field INFO Cisco Web interface refer to DTMF 8 Dialing Parameters on page 125 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 enabl
104. ini file take these 4 steps 1 Save the ini file from the device to your PC using the Web interface refer to Backing Up and Restoring Configuration on page 217 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 217 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 6 235 November 2008 c tal AudioCodes MediaPack Series 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 DNSPriServerlP DNSSecServerlP SIP User s Manual Table 4 1 Networking ini File Parameters Description Defines the Ethernet connection mode type 0 10Base T half duplex Not applicable 1
105. 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 Password for the voice menu used for configuration and status To activate the menu connect a POTS telephone and dial three stars followed by the password The default value is 12345 For detailed information on the voice menu refer to the device s Installation Manual Determines whether the RADIUS application is enabled 0 Disable RADIUS application is disabled default 1 Enable RADIUS application is enabled 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 transmitte
106. isn t configured its Caller ID generation detection are determined according to the global parameter EnableCallerlD described in Supplementary Services on page 138 This parameter can appear up to 8 times for 8 port devices and up to 24 times for MP 124 devices To configure Call ID Permissions using the Web interface refer to Caller ID Permissions on page 179 Foran explanation on ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced 283 November 2008 V m C A AudioCodes Parameter FXSOOSBehavior NumberOfWaitingIndications TimeBetweenWaitingIndicati ons TimeBeforeWaitinglndication WaitingBeepDuration EnableCalleriD CallerIDType SubscriptionMode EnableMWI MWIAnalogLamp MWIDisplay EnableMWISubscription MWIServerlP SubscribeRetryTime MWIServerTransportType MWIExpirationTime StutterToneDuration PayPhoneMeteringMode MeteringType KeyCFUnCond KeyCFNoAnswer SIP User s Manual MediaPack Series Description Parameters on page 129 For a description of this parameter refer to Advan
107. modes for fax per modem type V 22 V 23 Bell V 32 V 34 m 1 38 fax relay refer to Fax Relay Mode on page 319 m Fax and modem bypass a proprietary method that uses a high bit rate coder refer to Fax Modem Bypass Mode on page 320 m NSE Cisco s Pass through bypass mode for fax and modem refer to Fax Modem NSE Mode on page 322 m Transparent passing the fax modem signal in the current voice coder refer to Fax Modem Transparent Mode on page 322 m 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 323 m G 711 Transport switching to G 711 when fax modem is detected refer to G 711 Fax Modem Transport Mode on page 323 m Fax fallback to G 711 if T 38 is not supported refer to Fax Fallback on page 324 Adaptations refer to automatic reconfiguration of certain DSP features for handling fax modem streams differently than voice T 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 320 m Automatically switching to T 38 mode without using SIP Re INVITE messages
108. need to use scroll bars The arrow button located just below the Navigation bar is used to hide and show the Navigation pane m To hide the Navigation pane click the left pointing arrow lt 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 displayed and the button is replaced by the left pointing arrow button Figure 3 6 Showing and Hiding Navigation Pane A Home map Show Hide Button Displayed Navigation QoS Settings Delene Selected Ermes _Pane _ BR a nom tabio entry me advance Appicanons Destination IP Address Oevtination Mask Gateway IP Address Hep Count Are Oot Sina c S Delete Destination 1P Row A d ress Delete Selected Entes Destmaton iP Address Destination Mask Gateway iP Address tte 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 27 November 2008 A c tal AudioCodes MediaPack Series 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 requi
109. 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 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 Determines whether the device when acting as client for TLS 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 PeerHostNameVerificationMode must be used as well Defines the Subject Name that is compared with the name 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
110. on page 138 You can also configure the Call Forward table using the ini file table parameter Fwdinfo refer to Analog Telephony Parameters on page 279 gt To configure Call Forward per port take these 4 steps 1 Open the Call Forward Table page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Call Forward page item Figure 3 82 Call Forward Table Page Gateway Port Port 1 FXS On busy 30 Forward Type Forward to Phone Time for No Reply Number Forward Port 2 FXS On busy 201 Port 3 FXS No Answer 1203 Port4 FXS Unconditional 202810211 Port 5 FXO Deactivate SIP User s Manual 178 Document LTRT 65411 SIP User s Manual 3 Web Based Management 2 Configure the Call Forward parameters for each port 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 209 Table 3 44 Call Forward Table Parameter Description Forward Type Determines the scenario for forwarding a call 0 Deactivate Don t forward incoming calls default 1 On Busy Forward incoming calls when the port is busy 2 Unconditional Always forward incoming calls 3 No Answer Forward incoming calls that are not answered within the time specified in the Time for No Reply Forward field 4 On Busy or No Answer Forward incoming
111. 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 m 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 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 125 e TxDTMFOption 0 ini file 1 to 5 Tx DTMF Option field Disable Web interface refer to DTMF amp Dialing Parameters on page 125 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 125 e TxDTMFOption 3 ini file 1 to 5 Tx DTMF Option field INFO Korea Web interface refer to DTMF amp Dialing Parameters on page 125 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 INF
112. 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 ooo sen w Priority Settings Value Reverted g See S kk Media Premium Priority bo E Network Priority E iE _ to Previou ia Ea da _ Valid Value _ Control Premium Priority Gold Priority Bronze Priority a Differential Services 3 3 3 4 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 3 3 3 5 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 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 32 Document LTRT 65411 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
113. performed per port or for the entire device If authentication is performed for the entire device the configuration in the Authentication page is ignored If either the user name or password field is omitted the port s phone number defined in Configuring the Endpoint Phone Numbers on page 181 and global password refer to the parameter Password described in Proxy amp Registration Parameters on page 112 are used instead You can also configure Authentication using the ini file table parameter Authentication refer to SIP Configuration Parameters on page 260 gt To configure the Authentication Table take these 5 steps 1 Set the Authentication Mode parameter to Per Endpoint refer to Proxy 8 Registration Parameters on page 112 2 Open the Authentication page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Authentication page item Figure 3 79 Authentication Page Gateway Port User Name Password Port 1 FXS Port 2 FXS Port 3 FXS Port4 FXS Port5 FKO Port 6 FRO Port 7 FXO Port 8 FXO 3 In the User Name and Password fields corresponding to a port enter the user name and password respectively 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 209 Versi
114. refer to Configuring the Hook Flash Settings on page 77 303 November 2008 ca AudioCodes MediaPack Series 4 4 13 Auxiliary Configuration Files Parameters The configuration files i e auxiliary files can be loaded to the device using the Web interface or a TFTP session refer to Auxiliary Files on page 210 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 13 Auxiliary Configuration ini File Parameters Parameter CallProgressTonesFilename FXSLoopCharacteristicsFileName PrerecordedTonesFileName UserInfoFileName SetDefaultOnlniFileProcess SaveConfiguration SIP User s Manual 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 The name and path of the file providing the FXS line characteristic parameters The name and path of the file containing the Prerecorded Tones The name and path of the file containing the User Information data 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 curren
115. refer to Automatically Switching to T 38 Mode without SIP Re INVITE on page 320 Version 5 6 319 November 2008 7a e AudioCodes MediaPack Series 7 3 2 1 1 7 3 2 1 2 7 3 2 2 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 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 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 To configure T 38 mode using SIP Re INVITE messages set IsFaxUsed to 1 Additional configuration parameters include the following E FaxRelayEnhancedRedundancyDepth m FaxRelayRedundancyDepth
116. 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 in 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 Determines the time period in milliseconds for which the device plays a tone to the endpoint on each Alternative Routing attempt When the device finishes playing the tone a new SIP INVITE message is sent toward the new destination The tone played is the Call Forward Tone i e Tone Type 25 in the CPT file The valid range is 0 to 20 000 The default is 0 i e no tone is played 159 November 2008 K tal AudioCodes MediaPack Series Parameter Description Max Allowed Packet Loss for Packet loss percentage at which the IP connection is considered a Alt Routing failure and Alternative Routing mechanism is activated IPConnQoSMaxAllowedPL The range is 1 to 20 The default value is 20 Max Allowed Delay for Alt Transmission delay in msec at which the IP connection is Routing msec considered a failure and Alternative Routing mechanism is activated IPConnQoSMaxAllowedDel The range is 100 to 1000 The default value is 250 ay 3 4 4 4 2 Tel to IP Routing Table The Tel to IP Routing page provides a table for configuring up to up to 50 ro
117. 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 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 90 For a description of this parameter refer to Configuring the General Security Settings on page 90 This ini file table parameter configures the IPSec SPD table The 253 November 2008 A K e AudioCodes MediaPack Series Parameter IK
118. 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 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 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 123 or one of the coder groups you defined in the Coder Group Settings page refer to Coder Group Settings on page 170 Repeat steps 2 through 6 for the next IP Profiles optional Click the Submit button to save your changes To save the changes to flash memory refer to Saving Configuration on page 209 Table 3 43 Description of Parameter Unique to IP Profile Parameter Description Number of Calls Limit Maximum numbe
119. selects the lowest channel number in the hunt group and then starts ascending again 2 Ascending Selects the lowest available channel It always starts at the lowest channel number in the hunt 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 hunt group When the device reaches the lowest channel number in the hunt group it selects the highest channel number in the hunt group and then starts descending again 4 Descending Selects the highest available channel Always starts at the highest channel number in the hunt 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 busy the call is released 6 By Source Phone Number Selects the device s channel according to the calling number Note For defining the channel select mode per Hunt Group refer to Configuring the Hunt Group Settings on page 183 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 a
120. 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 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 u 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 gomd attribute is added to the SDP in the following format For A law a gpmd 8 vbd yes ecan on For u law a gpmd 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 319 Detect Fax on Answer Determines when the device initiates a T 38 session for fax Tone transmission DetFax
121. subnet mask is 0 0 0 0 The device s default Gateway IP address in the Media network The default value is 0 0 0 0 54 Document LTRT 65411 SIP User s Manual 3 Web Based Management Parameter Description Multiple Interface Settings Multiple Interface Table Click the right pointing arrow L 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 55 VLAN For detailed information on the device s VLAN implementation refer to VLANS and Multiple IPs on page 370 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 identifier VLANControlVlanlD 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 I
122. t match the NAT mechanism is activated Consequently the remote IP address of the outgoing stream is replaced by the source IP address 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 333 238 Document LTRT 65411 SIP User s Manual Parameter EnableUDPPortTranslatio n NoOpEnable NoOplnterval 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 tim
123. 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 57 November 2008 A K e AudioCodes MediaPack Series Parameter VLAN ID Interface Name General Parameters VLAN Mode VIANMode Native VLAN ID VLANNativeVlanID Description interface address For configuring additional routing rules for other interfaces refer to Configuring the IP Routing Table on page 63 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 Settings on page 52 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 inte
124. their average is constant The valid range is 10 to 10 dB The default value is 0 dB 69 November 2008 7a K tall AudioCodes MediaPack Series 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 43 Fax Modem CID Settings Page Fax Transport Mode RelayEnable Caller ID Transport Type Mute Caller ID Type W 21 Modem Transport Type Disable 22 Modem Transport Type Enable Bypass 23 Modem Transport Type Enable Bypass 32 Modem Transport Type Enable Bypass 34 Modem Transport Type Enable Bypass Fax Relay Redundancy Depth 0 Fax Relay Enhanced Redundancy Depth 4 Fax Relay ECM Enable Enable Fax Relay Max Rate bps 14400bps Fax Modem Bypass Coder Type G ilAlaw 64 Fax Modem Bypass Packing Factor 1 Fax Bypass Output Gain 0 Modem Bypass Output Gain 0 Fax CNG Mode CMG Detector Mode Disable Standard Bellcore lt ses s INS KICK Disable 2 Configure the fax Modem and CID parameters according to the table below 3 Click the Submit button to save your changes 4 To save
125. this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 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 DigitPatternForwardOnBu sy DigitPatternForwardOnNo Answer DigitPatternForwardOnDN D DigitPatternForwardNoRe ason DigitPatternForwardOnBu syExt DigitPatternForwardOnNo AnswerExt DigitPatternForwardOnDN DExt DigitPatternForwardNoRe asonExt DigitPatternInternalCall DigitPatternExternalCall TelDisconnectCode DigitPatternDigitTolgnore SIP User s Manual For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this par
126. to Analog Telephony Parameters on page 279 or the embedded Web server s Automatic Dialing screen refer to Automatic Dialing on page 175 The SIP call flow diagram below illustrates Automatic Dialing Figure 7 4 Call Flow for Automatic Dialing SIP Client F1 INVITE Sent immediately if Caller ID detected otherwise sent after 2 rings or after 1 ring if RingsBeforeCallerl 0 FXO Gateway FXO detects rings on line FXO seizes line off hook only after receiving 200 OK even after receiving 183 to enable routing to voice mail on the PBX side 7 4 2 2 Collecting Digits Mode When automatic dialing is not defined the device collects the digits The SIP call flow diagram below illustrates the Collecting Digits Mode SIP User s Manual 330 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities Figure 7 5 Collecting Digits Mode SIP Client a FXO Gateway FXO detects Caller ID according to RingsBeforeCallerlD F1 INVITE Sent after collecting MaxDigits or after TimeBetweenDigits has expired or once digit strings DigitMapping match digit map 7 4 2 3 Ring Detection Timeout The operation of Ring Detection Timeout depends on the following No automatic dialing and Caller ID is enabled if the second ring signal doesn t arrive for Ring Detection Timeout the device doesn t initiate a call to the IP Automatic dialing is enabled if the remote party doesn t answer the call
127. 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 Defines the DiffServ value for the Gold CoS content The valid range is 0 to 63 The default value is 26 Defines the DiffServ value for the Bronze CoS content The valid range is 0 to 63 The default value is 10 66 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 67 m Fax Modem CID Settings refer to Configuring the Fax Modem CID Settings on page 69 RTP RTCP Settings refer to Configuring the RTP RTCP Settings on page 73 General Media Settings refer to Configuring the General Media Settings on page 76 Hook Flash Settings refer to Configuring the Hook Flash Settings on page 77 Media Security refer to Configuring Media Security on page 78 Channel parameters can be modified on the fly Changes take effect from the next call Some channel parameters can be configured per endpoint or call routing using profiles refer to Configuring the Profile Definitions on page 169 3 4 2
128. to the device 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 choose 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 E cmp mandatory compressed firmware file m jni configuration file m Auxiliary files CPT Call Progress Tone PRT Prerecorded Tones FXS Coefficient and USRINF User Info Warnings e Before upgrading the device to a new major software version e g from version 5 2 to 5 4 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 217 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 traf
129. would forward for example only 234 to the PBX The PBX would then ring extension 234 DID wink enables the originating end to seize the line by going off hook It waits for acknowledgement from the other end before sending digits This serves as an integrity check that identifies a malfunctioning trunk and allows the network to send a re order tone to the calling party The start dial signal is a wink from the PBX to the FXO device The FXO then sends the last four to five DTMF digits of the called number The PBX uses these digits to complete the routing directly to an internal station telephone or equivalent m DID Wink can be used for connection to EIA TIA 464B DID Loop Start lines m Both FXO detection and FXS generation are supported Version 5 6 329 November 2008 7a K tal AudioCodes MediaPack Series 742 7 4 21 Telephone to IP Calls The FXO device provides the following FXO operating modes for Tel to IP calls m Automatic Dialing refer to Automatic Dialing on page 330 m Collecting Digits Mode refer to Collecting Digits Mode on page 330 m Ring Detection Timeout refer to Ring Detection Timeout on page 331 m FXO Supplementary Services refer to FXO Supplementary Services on page 331 e _Hold Transfer Toward the Tel side e __Hold Transfer Toward the IP side Blind Transfer to the Tel side Automatic Dialing Automatic dialing is defined using the ini file parameter table TargetOfChannel refer
130. 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 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 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 0r3 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 call 3 Connect amp End Call CDR report is sent to Syslog at connection and at the end of each call 4 Start 8 Connect
131. 0 0 0 0 Enable W 3 Configure the SNMP trap managers 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 209 Note Only table row entries whose corresponding check boxes are selected are applied when clicking Submit otherwise settings revert to their defaults Table 3 53 SNMP Trap Destinations Parameters Description Parameter Description SNMP Manager Determines the validity of the parameters IP address and port SNMPManagerlsUsed x number of the corresponding SNMP Manager used to receive SNMP traps 0 Check box cleared Disabled default 1 Check box selected Enabled IP Address IP address of the remote host used as an SNMP Manager The SNMPManagerTablelP x device sends SNMP traps to these IP addresses Enter the IP address in dotted decimal notation e g 108 10 1 255 Trap Port Defines the port number of the remote SNMP Manager The device SNMPManagerTrapPort x sends SNMP traps to these ports The valid SNMP trap port range is 100 to 4000 The default port is 162 SIP User s Manual 202 Document LTRT 65411 SIP User s Manual 3 Web Based Management Parameter Description Trap Enable Activates or de activates the sending of traps to the corresponding SNMPManagerTrapSendin SNMP Manager gEnable x 0 Disable Sending is disabled 1 Enable Sending i
132. 000 10 8 201 108 user phone gt User Agent Audiocodes Sip Gateway MediaPack 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 s Phone Call C I Nel PAM LOR M2 Odes Ole t 0 0 m audio 4000 RTP AVP 8 96 a rtpmap 8 pcma 8000 a rtpmap 96 telephone event 8000 a fmtp 96 0 15 a ptime 20 SIP User s Manual 342 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities m F2 10 8 201 10 gt gt 10 8 201 108 TRYING 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 Mac lt Sijos 1OOO 10 S92 OKR OE Call ID 534366556655skKw 8000 1000 10 8 201 108 Server Audiocodes Sip Gateway MediaPack v 5 40 010 006 CSeg 18153 INVITE Content Length 0 m 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 MediaPack v 5 40 010 006 CSeg 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 g
133. 0000 50000 BLOCK 0 o 0940 ee es CO IC 4 Yes 104 00 255 255 0 0 4000 9000 Any 0 0 0 BLOCK 0 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 209 gt To edit a rule take these 4 steps 1 Inthe Edit Rule column select the rule that you want to edit Modify the fields as desired Click the Apply button to save the changes amp To save the changes to flash memory refer to Saving Configuration on page 209 SIP User s Manual 84 Document LTRT 65411 SIP User s Manual 3 Web Based Management gt To activate a de activated rule take these 2 steps 1 In the 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 Inthe 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 ru
134. 10Base T full duplex 2 100Base TX half duplex 3 100Base TX full duplex 4 Auto negotiate default For detailed information on Ethernet interface configuration refer to Ethernet Interface Configuration on page 365 For a description of this parameter refer to Configuring the IP Settings on page 52 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 129 For a description of this parameter refer to Configuring the IP Settings on page 52 For a description of this parameter refer to Configuring the IP Settings on page 52 236 Document LTRT 65411 SIP User s Manual 4 ini File Configuration Parameter DNS2IP SRV2IP Version 5 6 Descripti
135. 114 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 2 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 oo Nlolol lolololo Counter Number of Attempted Calls Number of Established Calls SIP User s Manual Table 3 58 Call Counters Description Description Indicates the number of attempted calls It is composed of established and 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 Indicates the number of established calls It is incremented as a result of one of 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 internal reasons RELEASE BECAUSE UNKNOWN REASON RELEASE BECAUSE REMOTE CANCEL CALL RELEASE
136. 160 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 default 1 Transmit amp Receive Send and receive RTP 2 Transmit Only Send RTP only 3 Receive Only Receive RTP only Enables Direct Inward Dialing DID using Wink Start signaling 0 Disable Disables DID Wink default 1 Enable Enables DID Wink If enabled the device can be used for connection to EIA TIA 464B DID Loop Start lines Both FXO detection and FXS generation are supported An FXO interface dials DTMF digits after a Wink signal is detected instead of a Dial tone An FXS interface generates the Wink signal after the detection of offhook instead of playing a Dial tone Defines the time interval in seconds between detection of offhook and generation of a DID Wink Applicable only to FXS interfaces The valid range is 0 to 1 000 The default value is 0 The time interval after the user hangs up the phone and before the call is disconnected FXS This allows the user to hang up and then pick up the phone before this timeout to continue the call conversation Thus it s also referred to as regret time The valid range is 0 to 255 in seconds The default value is 0 Disconnect and Answer Supervision Send Digit Pattern on Connect TelConnectCode Enable Polarity Reversal EnableReversalPolarit y SIP User s Manual Defines
137. 164 Local RS 232 terminal Web Management via HTTP or HTTPS Telnet UL60950 1 FCC part 15 Class B CE Mark EN 60950 1 EN 55022 EN 55024 EN 61000 3 2 EN 61000 3 3 EN 55024 386 Document LTRT 65411 SIP User s Manual 11 Term ADPCM A law AOR bps BootP CoS CMP CPT dB DHCP DID DiffServ DNS DR DSP DTMF ETSI FQDN FXS FXO GRUU ICMP IETF IKE IP IPSec ISO ITU ITU T Jitter kbps Mbps Version 5 6 11 Glossary Glossary Table 11 1 Glossary of Terms Meaning Adaptive Differential PCM voice compression Standard companding algorithm used in European digital communications systems to optimize the dynamic range of an analog signal for digitizing Address of Record Bits per second AudioCodes Proprietary Bootstrap Loader Utility Class of Service Compressed File device Firmware Call Progress Tones Decibels Dynamic Host Control Protocol Direct Inward Dial Differentiated Services Domain Name System or Server Debug Recording Digital Signal Processor or Processing Dual Tone Multiple Frequency Touch Tone European Telecommunications Standards Institute Fully Qualified Domain Name Foreign Exchange Station Foreign Exchange Office Globally Routable User Agent URIs Internet Control Message Protocol Internet Engineering Task Force Internet Key Exchange for IPSec Internet Protocol IP Security International Standards Organization International Te
138. 1x amp MP 124 SIP Release Notes MP 11x amp MP 124 SIP Installation Manual MP 11x SIP Fast Track Guide MP 124 SIP Fast Track Guide CPE Configuration Guide for IP Voice Mail Warning The device is supplied as a sealed unit and must only be serviced by qualified service personnel Notes The following naming conventions are used throughout this manual unless otherwise specified e The term device refers to the MediaPack series gateways The term MediaPack refers to MP 124 MP 118 MP 114 and MP 112 e The term MP 11x refers to the MP 118 MP 114 and MP 112 devices Note gt PRPP SIP User s Manual 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 IP 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 PBX i e telephone connected directly or indirectly to the device Tel to P refers to calls received from the PSTN PBX and destined for the IP network FXO Foreign Exchange Office is the interface replacing the analog telephone and connects to a Public Switched Telephone Network PSTN line from the Central Office CO or to a Private Branch Exchange PBX The FXO is designed to receive line voltage and ringing current supplied from the CO or the PBX just like an analog telephon
139. 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 Endpoint Phone Number 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 171 Fora description of using ini file table parameters refer to Structure of ini File Table Parameters on page 233 4 4 8 Voice Mail Parameters 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 8 Voice Mail ini File Parameters Parameter Description VoiceMaillnterface For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 Version 5 6 277 November 2008 A c tal AudioCodes MediaPack Series Parameter SMDI SMDITimeOut LineTransferMode WaitForDialTime MWIOnCode MWIotffCode MWISuffixCode MWISourceNumber Description For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of
140. 200 msec of the parameter s value plus 100 For example if 287 November 2008 7a K tal AudioCodes MediaPack Series Parameter Description CurrentDisconnectDuration is 200 msec then the detection range is 100 to 500 msec CurrentDisconnectDefaultThr Determines the line voltage threshold which when reached is eshold considered a current disconnect detection The valid range is 0 to 20 Volts The default value is 4 Volts Note Applicable only to FXO interfaces TimeToSampleAnalogLineVo Determines the frequency at which the analog line voltage is Itage sampled after offhook for detection of the current disconnect threshold The valid range is 100 to 2500 msec The default value is 1000 msec Note Applicable only to FXO interfaces AnalogCallerIDTimingMode Determines when Caller ID is generated 0 Caller ID is generated between the first two rings default 1 The device attempts to find an optimized timing to generate the Caller ID according to the selected Caller ID type Notes Applicable only to FXS interfaces When used with distinctive ringing the Caller ID signal doesn t change the distinctive ringing timing BellcoreCallerIDTypeOneSub Selects the Bellcore Caller ID sub standard Standard 0 Between rings default 1 Not ring related ETS ICallerlDTypeOneSubSta Selects the ETSI FSK Caller ID Type 1 sub standard FXS only ndard 0 ETSI between rings default 1
141. 2008 ca AudioCodes Parameter FirstCallRBTld EnableReasonHeader 3xxBehavior EnablePChargingVector EnableVMURI EmergencyRegretTimeout EmergencyNumbers MaxActiveCalls MaxCallDuration EnableBusyOut EnableDigitDelivery2IP EnableDigitDelivery Authentication SIP User s Manual MediaPack Series Description digits in band transparent of RFC 2833 in addition to out of band DTMF messages Note Usually this mode is not recommended For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 12
142. 3 37 78 branch z9hG4bKac862428454 From lt sip 101eGatewayName gt tag 1c862422082 To lt sip 101eGatewayName gt GaTe GIOVOTVOS25I1 200023 29A5 10 335 37 78 CSeg 3 REGISTER Contact lt sip 101 10 33 37 78 gt expires 3600 Expires 3600 User Agent Audiocodes Sip Gateway MP 118 FXS FXO v 5 40A 008 002 Content Length 0 3 4 4 8 2 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 and Serving IP Group ID in the Hunt Group Settings refer to Configuring the Hunt Group Settings on page 183 and Account refer to Configuring the Account Table on page 188 tables These call routing tables are used for identifying 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 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 120 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 th
143. 74 Endpoint Phone Number refer to Configuring the Endpoint Phone Numbers on page 181 Trunk IP Group refer to Configuring the Hunt and IP Groups on page 182 3 4 4 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 101 Proxy amp Registration refer to Proxy amp Registration Parameters on page 112 Proxy Sets Table refer to Proxy Sets Table on page 120 Coders refer to Coders on page 123 DTMF amp Dialing refer to DTMF amp Dialing Parameters on page 125 SIP User s Manual 100 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 4 4 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 submenu gt SIP General Parameters page item Figure 3 57 SIP General Parameters Page
144. 7a u wi AudioCodes CPE amp Access Analog Gateways SIP MediaPack MP 124 amp MP 11x User s Manual Version 5 6 Document LTRT 65411 November 2008 SIP User s Manual Contents Table of Contents 1 Overview T TT TT TT TEkEEEEE 15 11 DODO iea NR 1 2 MediaPack Features MP 11x M 1 2 2 MP 124 Hardware Features oak ZOO o 13 SIP OMG eco cose dk o o o ooo e oececoms tece ete 2 Configuration CONCOPIS c u uorierersnasarnizasaoss nas saonaoa n ni ana no as su innsin 1D 3 Web Based Management e s eemeereeseeeeseeeeeeeeeeoeoeeeeeeeeo ananas eee eoeennecne 21 3 4 16 figuring the QoS Se Version 5 6 3 November 2008 Gg wt AudioCodes MediaPack Series ae Media Sein Boni Grint Gente ier O 67 3 4 2 1 Configuring the Voice Settings gt 3 4 2 2 Configuring the Fax Modem CID Settings 3 4 2 3 Configuring the RTP RTCP Settings 3 4 2 4 Configuring the General Media Settings 3 4 2 5 Configuring the Hook Flash pt t mes eee 3 4 2 6 Configuring Media Security ach Ree banna SAS OBOU SOS aai S n 3 4 3 1 Configuring the Web User Accounts TE 3 4 3 2 Configuring the Web and Telnet Access List A E A 3 4 3 3 Configuring the Firewall Settings E EE S E EE E E 3 4 3 4 Configuring the Certificates RE A O 3 4 3 5 Configuring the General eea Settings ee A 3 4 3 6 Configuring the IPSec Table PA
145. 8 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 134 Document LTRT 65411 SIP User s Manual Parameter Misc Parameters Progress Indicator to IP ProgressIndicator2IP Enable Busy Out EnableBusyOut Version 5 6 3 Web Based Management Description 1 1 Flow debugging is enabled 2 2 Flow and device interface debugging are enabled 3 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 For Analog FXS FXO interfaces 0 No PI For Tel to IP calls the device sends 180 Ringing SIP response to IP after placing a call to a phone FXS or PBX FXO 1 Pl 1 8 Pl 8 For Tel to IP calls if EnableEarlyMedia 1 the device sends 183 Session Progress message with SDP immediately after a call is placed to a phone PBX This is used to cut through the voice path before the remote party answers the call enabling the originating party to listen to network Call Progress Ton
146. 80 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 parameters are included in the Remote Party ID RPID header 0 No 1 Yes default If RPID header is enabled EnableRPIHeader 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 106 Document LTRT 65411 SIP User s Manual Parameter Enable History Info Header EnableHistorylnfo Use Source Number as Display Name UseSourceNumberAsDi splayName 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 r
147. 9 This ini file table parameter defines a username and password combination for authenticating each device port The format of this parameter is as follows Authentication FORMAT Authentication_Index Authentication_Userld Authentication UserPassword Authentication Port Authentication Module Authentication Where Userld User name UserPassword Password Port Port number Module Module number 0 5 N A For example Authentication Authentication 1 david 14325 1 Authentication 2 Alex 18552 1 Authentication 3 user1 1234 1 Authentication 272 Document LTRT 65411 SIP User s Manual Parameter SITDetectorEnable SourcelPAddressInput 4 ini File Configuration Description Notes You can omit either the username or password using the sign If omitted the port s phone number is used for authentication The indexing of this ini file table parameter starts at 1 To configure the authentication username and password using the Web interface refer to Authentication on page 174 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 Enables or disables Special Information Tone SIT detection according to the ITU T recommendation E 180 Q 35 0 Disable default 1 Enable For a description of this parameter refer to Routing General Parameters on page 157 Stand A
148. 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 ApplicatonTypes IPv InterfaceMode IPAddress PrefixLength Gateway VianID InterfaceName 1 l 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 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 se
149. Answer Supervision EnableVoiceDetection Version 5 6 3 Web Based Management Description Notes The correct dial tone parameters should be configured in the Call Progress Tones file The device may take 1 to 3 seconds to detect a dial tone according to the dial tone configuration in the Call Progress Tones file Determines the delay before the device starts dialing on the FXO line in the following scenarios The delay between the time the line is seized and dialing begins during the establishment of an IP to Tel call Note Applicable only for one stage dialing when the parameter IsWaitForDialTone is disabled The delay between detection of a Wink and the start of dialing during the establishment of an IP to Tel call for DID lines EnableDIDWink is set to 1 For call transfer the delay after hook flash is generated and dialing begins The valid range in milliseconds is 0 to 20 000 i e 20 seconds The default value is 1 000 i e 1 second Defines the timeout in seconds for detecting the second ring after the first detected ring If automatic dialing is not used and Caller ID is enabled the device seizes the line after detection of the second ring signal allowing detection of caller ID sent between the first and the second rings If the second ring signal is not received within this timeout the device doesn t initiate a call to IP If automatic dialing is used the device initiates a call t
150. AudioCodes MediaPack Series Parameter AddTrunkGroupAsPrefix AddPortAsPrefix UseSourceNumberAsDis playName UseDisplayNameAsSour ceNumber AlwaysUseRouteTable Prefix SIP User s Manual Description TrunkGroupSettings 0 1 0 5 audiocodes user 1 TrunkGroupSettings 1 2 1 0 localname user1 2 TrunkGroupSettings Notes This parameter can include up to 24 indices For configuring HuntGroup Settings using the Web interface refer to Configuring Hunt Group Settings on page 183 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to Routing General Parameters on page 157 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to Proxy amp Registration Parameters on page 112 This ini file table parameter configures the Tel to IP Routing table for routing Tel to IP calls The format of this parameter is as follows PREFIX FORMAT PREFIX_Index P EFIX DestinationPrefix PREFIX_DestAddress PREFIX_SourcePrefix PREFIX_Profileld PREFIX_MeteringCode PREFIX_DestPort PREFIX_SrclPGroupID PREFIX_DestHostPrefix PREFIX_DestIPGroupID P
151. 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 224 Document LTRT 65411 SIP User s Manual Counter Percentage of Successful Calls ASR 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 Version 5 6 3 Web Based Management Description to No Answer counter The rest of the release reasons increment the Number of Failed Calls due to Other Failures counter The percentage of established calls from attempted calls 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 the call duration is zero Indicates the number of calls that were t
152. Codes Parameter UseSIPTgrp EnableGRUU UserAgentDisplaylnfo SIPSDPSessionOwner RetryAfterTime EnablePAssociatedURIHeader EnableContactRestriction RemoveToTagInFailureRespo nse ReRegisterOnConnectionFail ure SourceNumberPreference EnableRTCPAttribute OPTIONSUserPart SIP User s Manual MediaPack Series Description ProxySet IsProxyHotSwap ProxySet 0 0 60 0 0 ProxySet 1 1 60 1 0 ProxySet Notes This table parameter can include up 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 120 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on pa
153. Control Message Protocol ICMP redirections and caches them as routing rules with expiration time SIP User s Manual 368 Document LTRT 65411 SIP User s Manual 8 Networking Capabilities 8 6 8 7 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 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 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
154. D VLANMediaVLANID VLANNetworkServiceClas sPriority VLANPremiumServiceCla ssMediaPriority VLANPremiumServiceCla ssControlPriority VlanGoldServiceClassPrio 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 52 For a description of this parameter refer to Configuring the IP Settings on page 52 For a description of this parameter refer to Configuring the IP Settings on page 52 For a description of this parameter refer to Configuring the IP Settings on page 52 For a description of this parameter refer to Configuring the QoS Settings on page 65 For a description of this parameter refer to Configuring the QoS Settings on page 65 For a description of this parameter refer to Configuring the QoS Settings on page 65 For a description of this parameter refer to Configuring the QoS Settings on page 65 For a description of this parameter refer to Configuring the QoS Settings on page 65 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 C
155. Dest Phone Prefix PstnPrefix DestPrefix Source Phone Prefix PstnPrefix SourcePrefix Source IP Address PstnPrefix_SourceAddress Hunt Group ID PstnPrefix_TrunkGroupld Profile ID PstnPrefix_Profileld Source IP Group ID PstnPrefix_SrclPGroupID 3 4 4 4 4 Internal DNS Table MediaPack Series Description 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 155 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 Notation on page 155 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 157 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 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 The Hunt Group to which incoming SIP calls are assigned that match all or any combination inclu
156. E Parameters IPSec IKEDB Table SIP User s Manual Description format of this parameter is as follows IPSEC SPD TABLE Format SPD INDEX IPSecMode IPSecPolicyRemotelPAddress IPSecPolicySrcPort IPSecPolicyDStPort IPSecPolicyProtocol IPSecPolicyLifelnSec IPSecPolicyLifelnKB IPSecPolicyProposalEncryption X IPSecPolicyProposalAuthentication X IPSecPolicyKeyExchangeMethodindex IPSecPolicyLocallPAddressType IPSecPolicyRemoteTunnellPAddress IPsecPolicyRemoteSubnetMask MPSEC SPD TABLE For example IPSEC SPD TABLEJ Format SPD INDEX IPSecMode IPSecPolicyRemotelPAddress IpsecPolicySrcPort IPSecPolicyDStPort IPSecPolicyProtocol IPSecPolicyLifelnSec IPSecPolicyProposalEncryption 0 IPSecPolicyProposalAuthentication 0 IPSecPolicyProposalEncryption 1 IPSecPolicyProposalAuthentication 1 IPSecPolicyKeyExchangeMethodlndex 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 f
157. E ae 7 13 4 3 Message Waiting Indication for Remote Extensions E P T Ne 7 13 4 4 Call Waiting for Remote Extensions assbuenigasiimsigas 7 13 4 5 FXS Gateway Configuration 7 13 4 6 FXO Gateway Configuration saii 7 13 5 SIP Trunking between Enterprise and ITSPs 7 14 Working with Supplementary Services cccccccececeeeeeeeeeeeneeeeeneeeseeeeeeneeetaes 356 7 14 1 Call Hold and ROU IVE u 7 14 2 Consultation Alternate eee hi LAS Call TAn o recne a ee cer rere eer reer io dk CIV POR cnn ore deg 7 14 5 Call Waiting Pe eer re errr E AEE 7 14 6 Message Waiting NR a leat cacauscsaktud aceteaaascdeicaszatastica HAS EKV 7 14 7 Caller ID 7 14 7 1 Caller ID Detection Generation on n the Tel Side ii 7 14 7 2 Debugging a Caller ID Detection on FXO E koda dkho alk o n 7 14 7 3 Caller ID on the IP Side ER PEE T E 8 Nobvorkina Gap a M kok oa aa katka n lez 365 8 1 Ethernet Interface Configuration iii minis ee 8 2 NAT ni Address Translation duot 8 2 2 First Incoming Packet Mechanism EII ose wa E E ae ae AE do 367 8 23 No Op PK P PAVO R OUP AOR M putes edna read SOLO PAR REO O oor 367 Ga IF eI seneesa E ee 8 4 Robust Reception of RTP SireamS za ka o kua n kooakka aaa i D 8 5 Multiple Routers SUPPOMt ccc cece cess eee eeeeeeeeeeeeeeeeeeeeees PPS SEEK E a 368 86 Simple Network Time Protocol SUPPOT s uiadsi iu aaa eds is assiave SIP User s
158. E ane ees 7 4 2 1 Automatic Dialing screen pittance eeaenix 7 4 2 2 Collecting Digits Mode 7 4 2 3 Ring Detection Timeout sii 7 4 2 4 FXO Supplementary Services Salus a bh 7 5 Event Notification using X Detect Header 4 eeeeeeeeeeeeeeeeeeee een 332 76 RTP Multiplexing ThroughPacket scisciicschiiesitraintsssissisiasicsaieilsstoiaeieuennreee 7 7 Dynamic Jitter Buffer Operation ceeeen re m PREE EA 334 7 8 Configuring Alternative Routing Based on Connectivity and QoS PETET EE 335 7 8 1 Alternative Routing Mechanism s sie s hdd x s k d akinanani 335 7 8 2 Determining the Availability of Destination IP AddrESS6S 2 0 335 209 Relevan Pea S orninn AE EEES A ONEEN Eaa ee 7 9 Mapping PSTN Release Cause to SIP Response ee 336 7 10 Supported RADIUS MIB lt sikdiu ke einim u ebna aaae 336 7 11 Call Detail Record ET IT EP Ae Ce TEE n 339 7 12 Proxy or Registrar Registration Example E E E EEN KOHO A KOK KST AA TE l 7 13 Configuration Examples sos csi ixcreietecnciettecdanidainncans EEEE TEETER 342 7 13 1 SIP Call FlOW sicscccstradscsizcacees 7 13 2 SIP Authentication Example E E SERE 7 13 3 Establishing a Call between Two Devices Sh s ne re gta z di de 7 13 4 Remote PBX Extension Between FXO and FXS Devices 348 7 13 4 1 Dialing from Remote Extension Phone at FXS E e 7 13 4 2 Dialing from PBX Line or PSTN E E
159. ETSI Caller ID and MWI 2 NTT 4 Britain 16 DTMF ETSI 17 Denmark Caller ID and MWI 18 India 19 Brazil Notes Typically the Caller ID signals are generated detected between the first and second rings However sometimes the Caller ID is detected before the first ring signal In such a scenario configure RingsBeforeCallerlD to 0 Caller ID detection for Britain 4 is not supported on the device s FXO ports Only FXS ports can generate the Caller ID for Britain 4 To select the Bellcore Caller ID sub standard use the parameter BellcoreCallerlIDTypeOneSubStandard To select the ETSI Caller ID sub standard use the parameter ETSICallerlDTypeOneSubStandard To select the Bellcore MWI sub standard use the parameter Bellcore VMWITypeOneStandard To select the ETSI MWI sub standard use the parameter ETSIVMWITypeOneStandard If you define Caller ID Type as NTT 2 you need to define the NTT DID signaling form FSK or DTMF using NTTDIDSignallingForm 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 71 November 2008 ca AudioCodes Parameter V 23 Modem Transport
160. Figure 7 23 MediaPack Series z O Routing SWS PRE oz shoot babku kabuki deda A ka dobou olla bast ubo bud 226 6 Registration Status PACE oxidant nk einan nanasan au koks dada dud oh uou Ea dan 227 T SAS Registered Users Page assisi iiiad a iaaa aas aeaiaioii 227 SIP Comen vE PAE aia a ai i 228 Example of a User Information File 2eeee2eee00 000000 eee eee een nnii 314 Device s SAS Agent Redirecting Emergency Calls to PSTN eee 316 Call Flow for One Stage Dialing x o5abuuda udd b ckkad skala bikkkud tuku esk anaE ia Eia n 327 Call Flow f r Two Stage Dialing osten did bs nod ekk ruka s dass k 328 Call Flow for LONG DI eiia iinoa p db kauce 330 Golecha Digits ModE isar RO O O O O haan Ghee dan dee pasinisedaaas 331 SPIP ae a E a dapt a n kdo k shani E EA ce sklu e b sn 342 Assigning Phone Numbers to Device 10 2 37 10 een 347 Assigning Phone Numbers to Device 10 2 37 20 eee eeeennn n 347 Routing Calls Between DES s ddd ad co ok dd ko dd 347 FXO FXS Remote PBX Extension Example ee eee eee eeeeee eee O48 PR OKP SME PSOH o etre tr dd oo Eo TT ere Er rere 350 Call Waiting for Remote Ado code dou A at kose 350 Assigning Phone Nt S zon a dl o B ky ek Ba zlu ka Ch Automatic Dialing OV OP UN n ad o a oh no 351 Tene FE Found ad o ae 351 Assigning Phone Numbers to FXO sich ME AG T kk 351 Automatic Dialing Pld ei dod ak t ooo dod ok 352 FXO Tel to IP Routing Config
161. Group LastBChannel TrunkGroup FirstPhoneNumber TrunkGroup Profileld TrunkGroup Module TrunkGroup For example TrunkGroup TrunkGroup 4 3 0 0 TrunkGroup 4 3 0 0 TrunkGroup Notes 101 0 1 4 channels 201 0 0 4 channels This parameter can appear up to 8 times for 8 port devices and up to 24 times for MP 124 devices The parameters TrunkGroup_FirstTrunkld TrunkGroup LastTrunkld and TrunkGroup Module are not applicable For configuring this table in the Web interface refer to Configuring the Endpoint Phone Numbers on page 181 Fora description of ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a description of this parameter refer to DTMF amp Dialing Parameters on page 125 For a description of this parameter refer to SIP General Parameters on page 101 This ini file table parameter defines rules for port allocation per Hunt 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 ContactUse r TrunkGroupSettings ServinglPGroup TrunkGroupSettings For example TrunkGroupSettings 289 November 2008 K tal
162. 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 129 Each line in the file represents a mapping rule of a single PBX extension Up to 100 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 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 i eae Caller ID 30 A string that represents the user name for SIP Vaomapa registration 40 Password A string that represents the password for SIP 20 registration Version 5 6 313 November 2008 7a e AudioCodes MediaPack Series An example of a User Information file is shown in the figure below Figure 6 1 Example of a User Information File nix File Edt Format Help 401 6380001 DN401 UN401 401 402 6380002 DN402 UN402 401 403 6380003 DN403 UN403 401 404 6380004 DN404 UN404 401 405 6380005 DN405 UN405 401 406 6380006 DN406 UN406 401 407 6380007 DN407 UN407 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
163. IP implemented in the gateway complies with the Internet Engineering Task Force IETF RFC 3261 refer to http www ietf org Version 5 6 17 November 2008 A tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 18 Document LTRT 65411 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 m AnHTTP based Embedded Web Server Web interface using any standard Web browser described in Web based Management on page 21 m A configuration file referred to as the ini file refer to ini File Configuration on page 231 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 19 November 2008 A tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 20 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 3 1 3 2 Web Based Management The device s Embedded Web Server
164. Information file can be loaded to the device using the ini file UserlnfoFileName parameter described in Auxiliary Configuration Files Parameters on page 303 the Web interface refer to Loading Auxiliary Files on page 210 or by using the automatic update mechanism UserlnfoFileURL refer to the Product Reference Manual The maximum permissible size of the file is 10 800 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 gradually 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 SIP User s Manual 314 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 7 7 1 IP Telephony Capabilities This se
165. Manual 6 Document LTRT 65411 SIP User s Manual Contents 8 7 P Qos via Po Services DiffServ 8 ogratng ing the ini File 9 Supplied SIP Software Package ssssessee 10 10 1 MP 11x CAS a dh o aa 10 2 MP 124 SOC Caton sina a a kk a u a bd kakao ee Version 5 6 7 November 2008 CA AudioCodes tora ue Figure 1 1 Figure 3 1 Figure 3 2 Figure 3 3 Figure 3 4 Figure 3 5 Figure 3 6 Figure 3 7 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 2 Figure 3 25 Figure 3 26 Figure 3 27 Figure 3 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 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 User Defined Web Welcome Message after Login MediaPack Series List of Figures Typical MediaPack VoIP Applicator siisii iniinis naana naaa aaa aiia aiaia 16 Enter Network Password Screen i ee Main Areas of the Web Interface GUI 23 Reset Displayed on Toolbar 24
166. MediaPack Series 3 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 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 99 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 reguire 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
167. 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 154 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 4 4 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 160 and IP to Hunt Group Routing table refer to IP to Trunk Group Routing on page 163 The dialing notation applies to digits entered for the destination and source prefixes to represent multiple numbers Table 3 38 Dialing Plan Notations Notation Description Example n m Represents a range of 5551200 5551300 represents all numbers from numbers 5551200 to 5551300 Note Range of letters is 123 100 200 represents all numbers from not supported 123100 to 123200 n m Represents multiple 2 3 4 5 6 represents a one digit number that numbers Up to three digits starts with 2 3 4 5 or 6 can be used to denote 11 22 33 xxx represents a four digit number each number that starts 11 22 or 33 111 222 xxx represents a four digit number that starts 111 or 222 x Represents any single 54324 represents any number that starts with 54324 digit Pound sign Represents the end of a 54324xx represents a 7 digit number that starts with number 54324 at the end of a number A single Repre
168. NVITE 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 SIP User s Manual 344 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 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 To lt sip 122 10 1 1 200 gt Call 1De O342931S4 1 1200 User Agent Audiocodes Sip Gateway MediaPack v 5 40 010 006 CSeq 1 REGISTER Contact SI pr T2 CilO 2 010k 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 From lt Sip 122 10 2 2 222 gt tag 1c17940 TOn Sip 220I R222 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 equat
169. O 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 Dialing Parameters on page 125 m MGCPDTMFDetectionPoint DTMFVolume DTMFTransportType DTMFDigitLength and DTMFInterDigitInterval refer to Channel Parameters on page 298 SIP User s Manual 318 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 7 3 7 3 1 7 3 2 7 3 2 1 Fax and Modem Capabilities Fax Modem Operating Modes The device supports two modes of operations m Fax modem negotiation that isn t performed during the establishment of the call m VBD mode for V 152 implementation refer to Supporting V 152 Implementation on page 325 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 Fax Modem Transport Modes The device supports the following transport
170. OnAnswerTone 0 Initiate T 38 on Preamble The device to which the called fax is connected initiates a T 38 session on receiving HDLC Preamble signal from the fax default 1 Initiate T 38 on CED The device to which the called fax is connected initiates 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 SIP User s Manual 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 3 104 Document LTRT 65411 SIP User s Manual Parameter SIP Transport Type SIPTransportType SIP UDP Local Port LocalSIPPort SIP TCP Local Port TCPLocalSIPPort SIP TLS Local Port TLSLocalSIPPort Enable SIPS EnableSIPS Enable TCP Connection Reuse EnableTCPConnectionR euse 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 an
171. P 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 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 m 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 other source IP address and UDP port 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
172. P Server IP Scorarios Search address v NTP Settings Basic Full NTP Server IP Address 0000 za Mo e Ua Network Settings NTP UTC Offset ph u Enables or disables the IP Settings embedded Telnet server Application Settings POZE ne ta i Hours 24 Routing Table ITP Updated interval QoS Settings tUudMeda Settings v Telnet Settings n a Secunty Setong Embedded Telnet Server pay ewgeratien Telnet Server TCP Port dvencs An 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 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 47 November 2008 A wl AudioCodes MediaPack Series 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
173. P User s Manual 50 Document LTRT 65411 SIP User s Manual 3 Web Based Management Figure 3 32 Shortcut Menu when Clicking Port Reset Channel e g MP 11x wae Reset channel Reset channel Uplink Ready Power Port Settings Update Port Info 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 F button the Log Off confirmation message box appears Figure 3 33 Log Off Confirmation Box Microsoft Internet Explorer 2 Log Off 2 Click OK the Web session is logged off and the Log In button appears Figure 3 34 Web Session Logged Off Z http 10 13 4 13 HiddenPresslog0ff Microsoft Internet E E JEJ gt File Edit View Favorites Tools Help Ox O AAS Address http 10 13 4 13 HiddenPressL ogOFf Web session is logged off Internet 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 Web Interface on page 21 Version 5 6 51 November 2008 7a e AudioCodes MediaPack Series 3 4 3 4 1 3 4 1 1 Configuration Tab The Configuration tab on the Navigation bar displays all menus related to device configurati
174. Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to Proxy amp Registration Parameters on page 112 For a description of this parameter refer to Proxy amp Registration Parameters on page 112 For a description of this parameter refer to Proxy amp Registration Parameters on page 112 For a description of this parameter refer to Proxy amp Registration Parameters on page 112 For a description of this parameter refer to Proxy amp Registration Parameters on page 112 260 Document LTRT 65411 SIP User s Manual Parameter SIPReroutingMode EnableProxyKeepAlive ProxyKeepAliveTime DNSQueryType ProxyDNSQueryType ProxylP ProxySet Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy amp Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 This ini file table parameter configures the Proxy
175. RFC2198PayloadType Fax Bypass Payload Type FaxBypassPayloadType Enable RFC 3389 CN Payload Type EnableStandardSIDPayload Type Comfort Noise Generation Negotiation ComfortNoiseNegotiation Analog Signal Transport Type AnalogSignalTransportType RTP Base UDP Port BaseUDPPort Version 5 6 3 Web Based Management Description 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 Determines 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
176. RI 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 No 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 reguest 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 thi
177. RouteCauseTel2IP AltRouteCauseTel2IP 0 486 Busy Here AltRouteCauseTel2IP 1 480 Temporarily Unavailable AltRouteCauseTel2IP 2 408 No Response AltRouteCauseTel2 IP Notes The 408 reason can be used to specify no response from the remote party to the INVITE request This parameter can include up to 5 indices For defining the Reasons for Alternative Routing table using the Web interface refer to Reasons for Alternative Routing on page 168 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 This ini file table parameter configures call failure reason values received from the TelPSTN side If a call is released as a result of one of these reasons the device attempts to locate an alternative Hunt Group for the call in the IP to Hunt Group Routing table The format of this parameter is as follows AltRouteCauselP2Tel FORMAT AltRouteCauselP2Tel Index 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 For defining the Reasons for Alternative Routing table using the Web 296 Document LTRT 65411 SIP User s Manual Parameter FilterCalls2IP 4 ini File Configuration
178. 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 ProxylP FORMAT Proxylp Index Proxylp IpAddress Proxylp_TransportType Proxylp ProxySetld ProxylP For example ProxylP FORMAT Proxylp Index Proxylp IpAddress 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 ProxylP Notes 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 120 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 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 For example ProxySet FORMAT ProxySet Index ProxySet EnableProxyKeepAlive ProxySet ProxyKeepAliveTime ProxySet ProxyLoadBalancingMethod 261 November 2008 ca Audio
179. Settings PAGE eee GG 9 So ie A PaE MER O O T O OTO O O O IP Roning able PNE cc saaissscionchancensnan Aaaa EE ate Sad KA VOC odd kasta 64 E IMBP AC Eo ob S Dob So Vod T POURS ors TUNE c e s R O O ola Fax Modem CID Settings PPR AM E ROS SOR ESPA 0 RIPI REP ea A M O O O E General Media Settings Page Hook Flash Settings Page eee Media Security Page i m Web User Accounts Page for Users with Security A Administrator Privileges inoue TOIRINN 81 Web amp Telnet Access List Page Add New ee eases i Pe Web 8 Telnet Access List Table Firewall Settings Page Certificates Signing Request Page sete 86 IKE Table Listing Loaded Certificate Files 88 General Security S Page 90 IPSec Table i SEU KA ASAP AKG ka Sas SL S S eae KE Tahle Pa G6 ga od a ol bd ao ad Ie Document LTRT 65411 SIP User s Manual Contents Figure Figure Figure 3 Figure Figure 3 69 Figure 3 71 Figure 3 7 Figure Figure 3 73 Ban 3 T4 l Fi Ji Ils Count Page Figure 3 114 Ca Version 5 6 9 November 2008 C A AudioCodes Figure 3 11 Figure 3 11 Figure 3 11 Figure 3 11 Figure 6 1 Figure 7 1 Figure 7 2 Figure 7 3 Figure 7 4 Figure 7 5 Figure 7 6 Figure 7 7 Figure 7 8 Figure 7 9 Figure 7 10 Figure 7 11 Figure 7 12 Figure 7 13 Figure 7 14 Figure 7 15 Figure 7 16 Figure 7 17 Figure 7 18 Figure 7 19 Figure 7 20 Figure 7 21 Figure 7 22
180. 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 7 A s Destination Prefix Source Prefix Digits pa pi sak r Presentation Number 03 201 971 Allowed 1001 5 23 Restricted 123451001 0 8 Not Configured 30 40 px Not Configured 2001 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 e When the destination number is 035000 and source number is 20155 the source number is changed to 97120155 SIP User s Manual 152 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 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 Configure the Number Manipulation table according to the table below Click the Submit button to save your changes To save the changes to fl
181. Source Phone Prefix Dest IP Address gt 10 10 2 37 10 20 10 2 37 20 4 Make a call Pick up the phone connected to port 1 of the first device and dial 102 to the phone connected to port 2 of the same device Listen for progress tones at the calling phone and for the ringing tone at the called phone Answer the called phone speak into the calling phone and check the voice quality Dial 201 from the phone connected to port 1 of the first device the phone connected to port 1 of the second device rings Answer the call and check the voice quality Version 5 6 347 November 2008 A c tal AudioCodes MediaPack Series 7 13 4 Remote PBX Extension Between FXO and FXS Devices Remote PBX extension offers a company the capability of extending the power of its local PBX by allowing remote phones remote offices to connect to the company s PBX over the IP network instead of via PSTN This is as if the remote office is located in the head office where the PBX is installed PBX extensions are connected through FXO ports to the IP network instead of being connected to individual telephone stations At the remote office FXS units connect analog phones to the same IP network To produce full transparency each FXO port is mapped to an FXS port i e one to one mapping This allows individual extensions to be extended to remote locations To call a remote office worker a PBX user or a PSTN caller simply dial
182. TMF 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 TxDTMFOption 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 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 127 November 2008 A K tal AudioCodes MediaPack Series Parameter Hook Flash Option HookFlashOption Digit Mapping Rules DigitMapping Dial Tone Duration sec TimeForDialTone Hotline
183. 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 120 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 Reguest 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 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 for call routing when Proxy servers are unavailable 0 Disable Fallback is not used default 1 Enable Tel to IP Routing table is used when Prox
184. Typically this feature is used only when early media EnableEarlyMedia is used to establish the voice path before the call is answered Note This feature is applicable only for one stage dialing Determines the number of rings before the device starts detecting Caller ID 0 0 Before first ring 1 1 After first ring default 2 2 After second ring N A Determines whether the device disconnects the call upon detecting a busy tone 0 Enable Do not disconnect call on detection of busy tone 1 Disable Call is released if busy or reorder fast busy tones are detected on the device s FXO port default The device can disconnect a call after a dial tone is detected from the PBX 0 Disable Call isn t released 1 Enable Call is released if dial tone is detected on the device s FXO port default Note This option is in addition to the mechanism that disconnects a call when either busy or reorder tones are detected Defines the time interval in seconds after a call has ended and a new call can be accepted for IP to Tel FXO calls The valid range is 0 to 10 The default value is 1 Note Occasionally after a call ends and on hook is applied a delay is required before placing a new call and performing off hook This is necessary to prevent incorrect hook flash detection or other glare phenomena Management Tab The Management tab on the Navigation bar displays all menus relat
185. 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 traffic 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 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 Microsoft Internet Explore
186. When FXS ports receive Private or Anonymous strings in the From header they don t send the calling name or number to the Caller ID display If Caller ID name is detected on an FXO line EnableCallerlD 1 it is used instead of the Caller ID name defined on this page When the Presentation field is set to Restricted the Caller ID is sent to the remote side using only the P Asserted Identity and P Preferred Identity headers AssertedldMode To maintain backward compatibility when the strings Private or Anonymous are entered in the Caller ID Name field the Caller ID is restricted and the value in the Presentation field is ignored The value of the Presentation field can be overridden by configuring the Presentation field in the Source Number Manipulation table refer to Configuring the Number Manipulation Tables on page 151 You can also configure the Caller Display Information table using the ini file table parameter CallerDisplayInfo refer to Analog Telephony Parameters on page 279 3 4 4 6 4 Call Forward The Call Forwarding Table page allows you to forward redirect IP to Tel calls using SIP 302 response originally destined to specific device ports to other device ports or to an IP destination Ensure that the Call Forward feature is enabled default for the settings on this page to take effect To enable Call Forward use the parameter EnableForward Supplementary Services
187. While hearing Ringback transfer from alert e While speaking to C transfer from active Blind Transfer REFER 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 Transfer is initiated by sending REFER with REPLACES The device can receive and act upon receiving REFER with or without REPLACES The device can receive and act upon receiving INVITE with REPLACES in which case the old call is replaced by the new one The INVITE with REPLACES can be used to implement Directed Call Pickup Call Forward The following forms of call forward are supported Immediate incoming call is forwarded immediately and unconditionally Busy incoming call is forwarded if the endpoint is busy No Reply incoming call is forwarded if it isn t answered for a specified time On Busy or No Reply incoming call is forwarded if the port is busy or when calls are not answered after a specified time Do Not Disturb immediately reject incoming calls Upon receiving a call to Do Not Disturb call the 603 Decline SIP response code is sent Three forms of forwarding parties are available Served party party configured to forward the call FXS device Originating party party that initiated the first call FXS or FXO device
188. a K tal AudioCodes MediaPack Series Figure 3 103 Load a CMP file Page p hitp 10 13 4 13 Software Update Wizard Microsoft Interne CMP file Load a CMP file from your computer to the device INI fite a CPT file PRT file FXO file FXS file USRINF file FINISH Note At this stage you can quit the Software Update Wizard by clicking Cancel X 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 SIP User s Manual 214 Document LTRT 65411 SIP User s Manual 3 Web Based Management Figure 3 104 CMP File Loaded Successfully Message zh http 10 13 4 13 Software Update Wizard Microsoft Interne R CMP file INI file File MP118_SIP_F5 30A 012 005 cmp was CPT file successfully loaded into the device PRT file FXO file FXS file USRINF filo FINISH 5 Click one of the following buttons 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 6 Inthe Load an
189. a digit pattern to send to the Tel side after 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 Enables the polarity reversal feature 0 Disable Disable the polarity reversal service default 1 Enable Enable the polarity reversal service 132 Document LTRT 65411 SIP User s Manual Parameter Enable Current Disconnect EnableCurrentDisconn ect Disconnect on Broken Connection DisconnectOnBroken Connection Broken Connection Timeout BrokenConnectionEve ntTimeout Disconnect Call on Silence Detection EnableSilenceDisconn ect Version 5 6 3 Web Based Management Description If the polarity reversal service is enabled the FXS interface changes the line polarity on call answer and then changes it back on call release The FXO interface sends a 200 OK response when polarity reversal signal is detected applicable only to one stage dialing and releases a call when a second polarity reversal signal is detected Enables call release upon detection of a current disconnect signal 0 Disable Disable the current disconnect service default 1 Enable Enable the current disconnect service If the current disconnect service is enabled the FXO releases a call when a current disconnect signal is detected
190. a ssccediaseaiaccisensghansaaicesisawiaecsaneigivartideesacieeasmens 384 Table TI Clossary Of TEMS icccecresiccicigiacceivaccecies ae bd Vaude hdd vadu Khdn b vd aE oor SIP User s Manual 12 Document LTRT 6541 1 SIP User s Manual Notices Notice This document describes the AudioCodes MediaPack series Voice over IP VolP gateways 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 hitp www audiocodes com support Copyright 2008 AudioCodes Ltd All rights reserved This document is subject to change without notice Date Published November 17 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 acc
191. aPack Series gt To configure the Internal SRV table take these 9 steps 1 Open the Internal SRV Table page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Internal SRV Table page item Figure 3 74 Internal SRV Table Screen 3 4 4 4 6 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 reguired 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 209 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 t
192. ab gt Protocol Configuration menu gt Endpoint Settings submenu gt Caller ID Permissions page item Figure 3 83 Caller ID Permissions Page Gateway Caller Port ID Port1 FXS Enable Port 2 FXS Enable Port 3 FXS 2 From the Caller ID drop down list select one of the following e Enable Enables Caller ID generation FXS or detection FXO for the specific port e Disable Caller ID generation FXS or detection FXO for the specific port is disabled e Not defined Caller ID generation FXS or detection FXO for the specific port is determined according to the parameter Enable Caller ID described in Supplementary Services on page 138 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 209 3 4 4 6 6 Call Waiting The Call Waiting page allows you to enable or disable call waiting per device FXS port This page is applicable only to FXS interfaces Instead of using this page you can enable or disable call waiting for all the device s ports using the global call waiting parameter Enable Call Waiting refer to Supplementary Services on page 138 You can also configure the Call Waiting table using the ini file table parameter CallWaitingPerPort refer to SIP Configuration Parameters on page 260 SIP User s Manual 180 Document LTRT 65411 SIP User s Manual 3 Web B
193. able parameters refer to Structure of ini File Table Parameters on page 233 4 4 12 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 Parameter DJBufMinDelay DJBufOptFactor AnalogSignalTransportTyp e FaxTransportMode FaxRelayEnhancedRedund ancyDepth FaxRelayRedundancyDepth FaxRelayMaxRate FaxRelayECMEnable FaxModemBypassCoderTy pe CNGDetectorMode FaxCNGMode SIP User s Manual Table 4 12 Channel ini File Parameters Description For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax
194. abled Note Use of SRTP reduces the number of available channels a MP 124 18 available channels SIP User s Manual 78 Document LTRT 65411 SIP User s Manual Parameter Media Security Behavior MediaSecurityBehaviour Disable Authentication On Transmitted RTP Packets RTPAuthenticationDisableTx Disable Encryption On Transmitted RTP Packets RTPEncryptionDisableTx Disable Encryption On Transmitted RTCP Packets RTCPEncryptionDisableTx SRTP Settings Master Key Identifier MKI Size SRTPTxPacketMKISize 3 4 3 Security Settings 3 Web Based Management Description MP 118 6 available channels MP 114 3 available channels MP 112 No reduction Determines the device s mode of operation when SRTP is used 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 On a secured RTP session this parameter determines whether to enable Authentication on transmitted RTP packets 0 Enable default 1 Disable On a secured RTP session this parameter determines whether to enable Encryption on transmitted RTP packets
195. abrd t 2666 igr_pabrdit 2667 lgr pabrd f 2668 igr_psbrdif 2669 igr_pabrdit 2670 lgr psbrd f 2671 lgr psbrdif 2672 lgr psbrd f 2673 lgr psbrdif 2674 lgr psbrdex 2675 lgr flov 2676 lgr flov 2677 lgr psbrd f 2678 lgr pabrd f 2679 lgr psbrd f 2680 recy lt ON HOOK Ch 1 W1 0W HOOK EV I 1 0N HOOK EV i cpDigitMapHndir Stop Stoped 0 M1 CloseChannel ChannelNum 1 Open channel IsVoice n 1 IsT380n 1 IsVbdOn 0 I W1 OpenChannel on Trunk 1 BChannel 1 CID 1 with Voici W1 OpenChannel VoiceVolume 0 DTMFVolume 11 Inpu OpenChannel CoderType 15 Interval 4 N 1 W1 FAXTransportType 1 W1 ConfigFaxModemChannelParams NSEMode 0 CNGDetModev Detectors Amd 0 Ans 0 En 0 IBScmd Oxal 1 PS50SBoardinterface StopPlayTone Called recy lt OFF HOOK Ch 1 1 0FF_HOOK_EV W1 OFF HOOK EV UpdateChannelParams Channel 1 W1 ConfigFaxModemChannelParom NSENode 0 CWGDetNode i ActivateDigitMap for channel 1 MaxDialStringLength 219 November 2008 7a c tall AudioCodes MediaPack Series 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 t
196. ac gt lt sip 2400 10 6 210 5 gt expires 160 lt sip 2401 Proxies ac gt lt sip 2401 10 8 210 5 gt expires 180 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 6 210 5 gt expires 160 lt sip 2403 Proxies ac gt lt sip 2403 10 8 210 5 gt expires 180 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 6 210 5 gt expires 180 Version 5 6 227 November 2008 7a e AudioCodes MediaPack Series Table 3 60 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 available Contact SIP URI that can be used to contact that specific instance of the User Agent for subsequent requests 3 6 2 5 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 160 This information is available only if the parameter Enable Alt Routing Tel to IP refer to Routing General Parameters on page 157 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
197. ace 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 Note Each interface must be assigned a unique IP address 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 notation 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
198. ack Series Description For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 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 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 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 Fo
199. age 325 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 SIP User s Manual 324 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 7 3 3 1 7 3 3 2 7 3 4 Using Bypass Mechanism for V 34 Fax Transmission In this proprietary scenario the device uses bypass or NSE 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 FaxTransportMode 2 Bypass V34ModemTransportType 2 Modem bypass V32ModemTransportType 2 V23ModemTransportType 2 V22ModemTransportType 2 Configure the following parameters to use bypass mode for V 34 faxes and T 38 for T 30 faxes m FaxTransportMode 1 Relay V34ModemTransportType 2 Modem bypass V32ModemTransportType 2 V23ModemTransportType 2 V22ModemTransportType 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 m FaxTransportMode 1 Relay V34ModemTransportType 0 Transparent V32ModemTransportType 0 V23M
200. ail Settings Page v General Voice Mail Interface Line Transfer Mode 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 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 209 Version 5 6 191 November 2008 A L tal AudioCodes MediaPack Series Table 3 49 Voice Mail Parameters Parameter Description General Voice Mail Interface Enables the voice mail application on the device and determines VoiceMaillnterface the communication method used between the PBX and the device 0 None default 1 DTMF 2 SMDI Line Transfer Mod
201. aintenance 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 98 Maintenance Actions Page v Reset Configuration Reset Board Burn To FLASH Graceful Option w LOCK UNLOCK Lock LOCK Graceful Option No Current Admin State UNLOCKED w Save Configuration Burn To FLASH 3 5 1 3 1 Resetting the Device The Maintenance Actions page allows you to remotely reset the device In addition before resetting the device you can choose the following options Version 5 6 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 207 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 207 November 2008 A c tal AudioCodes
202. al Tone Duration sec 16 Hotline Dial Tone Duration sec 16 Enable Special Digits Disable 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 209 Table 3 31 DTMF and Dialing Parameters 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 Description Defines the maximum number of collected destination number digits that can be received i e dialed from the Tel side 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 5 Notes Digit Mapping Rules can be used instead Dialing ends when the maximum number of digits is dialed the Interdigit Timeout expires the key is dialed or a digit map pattern is matched Defines the time in seconds that the device waits between digits that are dialed by the calling party for Tel to IP calls 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 Defi
203. al tones or AM tones configurable 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 Group 3 fax relay up to 14 4 kbps with automatic fallback T 38 compliant real time fax relay Tolerant network delay up to 9 seconds round trip Auto switch to PCM or ADPCM on V 34 or V 90 modem detection SIP RFC 3261 RTP RTCP packetization IP stack UDP TCP RTP Remote software upload TFTP HTTP and HTTPS Loop start signaling 382 Document LTRT 65411 SIP User s Manual Function Processor Control Processor Control Processor Memory Signal Processors Interfaces FXS Telephony Interface FXO Telephony Interface Combined FXS FXO Network Interface RS 232 Interface Indicators Lifeline 10 Selected Technical Specifications Specification Motorola PowerQUICC 870 SDRAM 32 MB AudioCodes AC482 VoIP DSP 2 4 or 8 Analog FXS phone or fax ports loop start RJ 11 4 or 8 Analog FXO PSTN PBX loop start ports MP 118 4 FXS 8 4 FXO ports MP 114 2 FXS 8 2 FXO ports 10 100Base TX RS 232 Terminal Interface requires a DB 9 to PS 2 adaptor Channel status and activity LEDs The Lifeline provides a wired analog POTS phone connection to any PSTN or PBX FXS port when there is no power or the network fails Combined FXS FXO devices provide a Lifeline connection
204. alil 1De 11923010 33 253 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 anonymouseanonymous invalid gt tag 1c25298 To lt Sip 101 10 33 2 53 user phone gt Calis LOAL 3342 53 CSeq 1 INVITE Contact lt sip 100 10 33 2 53 gt X Detect Response CPT FAX Content Type Application X Detect Content Length xxx Type CPT Subtype SIT 7 6 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 to Configuring the RTP RTCP Settings on page 73 or to specific IP destinations using the IP Profile feature refer to IP Profile Settings on page 173 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 o
205. all and cuts through the voice channel if there is no other active call on the port even if the port is in off hook state When the call is terminated by the remote party the device plays a reorder tone for a user defined time configured by the parameter TimeForReorderTone and is then ready to answer the next incoming call without on hooking the phone The waiting call is automatically answered by the device when the current call is terminated configured by setting the parameter EnableCallWaiting to 1 Note This option is applicable only to FXS interfaces 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 210 0 Disable Disabled default 1 Enable Enabled Determines the behavior of undefined FXS endpoints as well as all FXS endpoints when a Busy Out condition exists 0 None Normal operation No response is provided to undefined endpoints A dial tone is played to FXS endpoints when a Busy Out condition exists 1 Reorder Tone The device plays a reorder tone to the connected phone PBX default 2 Polarity Reversal The device reverses the polarity of the endpoint marking it unusable relevant for example to PBX DID lines This option can t be configured on the fly 3 Reorder Tone Polarity Reversal Same as 2 and 3 combined This option
206. all 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 Message Waiting Indication MWI MWI Off Digit Pattern MWIOffCode MWI On Digit Pattern MWIOnCode Version 5 6 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 193 November 2008 ca AudioCodes Parameter MWI Suffix Pattern MWISuffixCode MWI Source Number MWISourceNumber SMDI Enable SMDI SMDI SMDI Timeout SMDITimeOut MediaPack Series Description Determines the digit code used by
207. alled transparent with events To configure fax modem transparent mode set IsFaxUsed to 2 Fax Fallback 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 the following adaptations m 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 m For G 711A law a gpmd 0 vbd yes ecan on m 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 325 m Using relay mode i e fallback to T 38 refer to Using Relay mode for both T 30 and V 34 faxes on p
208. allerlD EnableCallerlDTypeTwo disables enables the generation of Caller ID type 2 when the phone is off hooked used for call waiting RingsBeforeCallerlD sets the number of rings before the device starts detection of caller ID FXO only By default the device detects the caller ID signal between the first and second rings AnalogCallerlDTimimgMode determines the time period when a caller ID signal is generated FXS only By default the caller ID is generated between the first two rings PolarityReversalType some Caller ID signals use reversal polarity and or wink signals In these scenarios it is recommended to set PolarityReversalType to 1 Hard FXS only The Caller ID interworking can be changed using the parameters UseSourceNumberAsDisplayName and UseDisplayNameAsSourceNumber 7 14 7 2 Debugging a Caller ID Detection on FXO gt Version 5 6 To debug a Caller ID detection on an FXO interface take these 7 steps Verify that the parameter EnableCallerlD is set to 1 Verify that the caller ID standard and substandard of the device matches the standard of the PBX CallerlDType BellcoreCallerlIDTypeOneSubStandard and ETSICallerlIDTypeOneSubStandard Define the number of rings before the device starts detection of caller ID RingsBeforeCallerlD Verify that the coefficient file loaded to the device is correct if the caller ID signal is distorted the device won t recognize it Connect a phone to the ana
209. ameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 For a description of this parameter refer to Configuring the Voice Mail VM Parameters on page 190 278 Document LTRT 65411 SIP User s Manual 4 4 9 4 ini File Configuration PSTN Parameters The PSTN related ini file configuration parameters are described in the table below Parameter CallPriorityMode MLPPDiffserv PlayRBTone2Tel PlayRBTone2IP ProgressIndicator2IP TimeForReorderTone DisconnectOnBus
210. ameters on page 233 264 Document LTRT 65411 SIP User s Manual 4 ini File Configuration Parameter IPGroup NumberOfActiveDialogs PrackMode AssertedidMode PAssertedUserName UseAORInReferToHeader Version 5 6 Description 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 IPGroup ProxySetld IPGroup SIPGroupName IPGroup ContactUser IPGroup EnableSurvivability IPGroup ServinglPGroup IPGroup SIPReRoutingMode IPGroup AlwaysUseRouteTable IPGroup For example IPGroup FORMAT IPGroup Index IPGroup Type IPGroup Description IPGroup ProxySetld PGroup_SIPGroupName IPGroup ContactUser IPGroup EnableSurvivability IPGroup ServinglPGroup IPGroup SIPReRoutingMode IPGroup AlwaysUseRouteTable IPGroup 1 0 acme gateway 1 firstIPgroup 0 1 IPGroup 2 0 abc server 2 second Pgroup 0 1 IPGroup 3 0 IP phones 1 thirdiPGroup 0 1 0 IPGroup Notes 0 0 0 0 This table parameter can include up to 9 indices 1 9 The parameters IPGroup Type IPGroup EnableSurvivability and IPGroup_ServinglPGroup are currently not applicable and must be left empty or 1 These parameters are used only for IP to IP call routing applications supported in the next applicable release For configuring the IP Group table using the Web interface a
211. anagement 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 26 IKE Table Configuration Parameters Parameter Name Authentication Method IkePolicyAuthenticationMe thod Shared Key IKEPolicySharedKey IKE SA LifeTime sec IKEPolicyLifelnSec IKE SA LifeTime KB IKEPolicyLifelnKB Description Determines the authentication method for IKE 0 Pre shared Key default 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 86 Determines the pre shared key in textual format Both peers must register the same pre shared key for the 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 rec
212. and sent to IP in the SIP INVITE message as Display element For information on the Caller ID table refer to Caller ID on page 177 To disable enable caller ID generation per port refer to Call Forward on page 178 Defines one of the following standards for detection FXO and generation FXS of Caller ID and detection FXO generation FXS of MWI when specified signals 0 Standard Bellcore Caller ID and MWI default 1 Standard ETSI Caller ID and MWI 2 Standard NTT 4 Standard BT Britain 16 Standard DTMF Based ETSI 17 Standard Denmark Caller ID and MWI 18 Standard India 19 Standard Brazil Notes Typically the Caller ID signals are generated detected between the first and second rings However sometimes the Caller ID is detected before the first ring signal in such a scenario configure RingsBeforeCallerlD to 0 Caller ID detection for Britain 4 is not supported on the device s FXO ports Only FXS ports can generate the Britain 4 Caller ID To select the Bellcore Caller ID sub standard use the parameter BellcoreCallerlIDTypeOneSubStandard To select the ETSI Caller ID substandard use the parameter ETSICallerlDTypeOneSubStandard To select the Bellcore MWI sub standard use the parameter BellcoreVMWITypeOneStandard To select the ETSI MWI sub standard use the parameter ETSIVMWITypeOneStandard Determines a digit pattern that when received from the T
213. ansportType refer to SIP General Parameters on page 101 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 186 the reguest URI host name in the 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 120 The IP Profile ID configured in Configuring the Profile Definitions on page 169 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 An optional Charge Code 1 to 25 can be appl
214. arameter TrunkGroupSettings refer to Number Manipulation and Routing Parameters on page 289 gt To configure the Hunt Group Settings table take these 5 steps 1 Open the Hunt Group Settings page Configuration tab gt Protocol Configuration menu gt Hunt IP Group submenu gt Hunt Group Settings page item Figure 3 86 Hunt Group Settings Page Routing Index Hunt Group ID Serving IP Channel Select Mode Registration Mode Group ID gt Gatewsy Name Contact User 1 1 Cyclic Ascending Vv Per Gateway v v 2 v 3 v 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 Hunt 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 209 Table 3 46 Hunt Group Settings Parameters Description Parameter Description Hunt Group ID The Hunt Group ID that you want to configure TrunkGroupSettings TrunkGro The valid range is 1 99 upld Channels are assigned to Hunt Groups in the Endpoint Phone Numbers page refer to Configuring the Endpoint Phone Numbers on page 181 Version 5 6 183 November 2008 ca AudioCodes Parameter Channel Select Mode TrunkGroupSettings ChannelS electMode Registration Mode TrunkGroupSettings Registrati onMode SIP User s Manual MediaPack Series Descrip
215. arameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a desc
216. arameters 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 Version 5 6 217 November 2008 7a K tal AudioCodes MediaPack Series 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 Configuration File Figure 3 108 Configuration File Page Configuration File o S Fil 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 rese
217. arge Codes Table refer to Charge Codes Table on page 146 Metering Tone Type Defines the metering tone 12 or 16 kHz that is generated by FXS MeteringType interfaces 0 12 kHz 12 kHz metering tone default 1 16 kHz 16 kHz metering tone Note A suitable 12 or 16 KHz FXS Coefficient file must be used for FXS interfaces Charge Codes Table If you configured the Generate Metering Tones parameter to Internal Table access the Charge Codes Table page by clicking W For detailed information on configuring the Charge Codes table refer to Charge Codes Table on page 146 Version 5 6 145 November 2008 7a K tall AudioCodes MediaPack Series 3 4 4 2 4 Charge Codes Table The Charge Codes Table page is used to configure the metering tones and their time interval that the FXS interfaces generate to the Tel side To associate a charge code to an outgoing Tel to IP call use the Tel to IP Routing table The Charge Codes Table page is available only for FXS interfaces You can also configure the Charge Codes table using the ini file table parameter ChargeCode refer to Analog Telephony Parameters on page 279 gt Toconfigure the Charge Codes table take these 4 steps 1 Access the Charge Codes Table page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Charge Codes page item Alternatively you can also access this page from the Metering T
218. arted 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 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 Version 5 6 365 November 2008 7a K tal AudioCodes MediaPack Series 8 2 1 The following figure illustrates the device s supported NAT architecture Figure 8 1 Nat Functioning MediaPack 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 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 iss
219. arty Only if the remote party disconnects the call i e a BYE is received or a timer expires set by the parameter EmergencyRegretTimeout is the call terminated The list can include up to four different numbers where each number can be up to four digits long Example EmergencyNumbers 100 911 112 Note This parameter is applicable only to FXS interfaces Determines the time in minutes that the device waits before tearing down an emergency call defined by the parameter EmergencyNumbers Until this time expires an emergency call can only be disconnected by the remote party typically by a Public Safety Answering Point PSAP The valid range is 1 to 30 The default value is 10 Note This parameter is applicable only to FXS interfaces 3 4 4 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 356 gt To configure the supplementary services parameters take these 4 steps 1 Open the Supplementary Services page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Supplementary Services page item SIP User s Manual 138 Document LTRT 65411 SIP User s Manual 3 Web Based Management Figure 3 63 Supplementary Services Page Enable Hold Hold Format He
220. ased Management gt To configure Call Waiting take these 4 steps 1 Open the Caller Waiting page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Call Waiting page item Figure 3 84 Call Waiting Page Gateway Port Call waiting Configuration FXS FXS FRS FRS FXO 2 From the Call Waiting Configuration drop down list corresponding to the port you want to configure for call waiting select one of the following options e Enable Enables call waiting for the specific port When the device receives a call on a busy endpoint port it responds with a 182 response and not with a 486 busy The device plays a call waiting indication signal When hook flash is detected by the device the device switches to the waiting call The device that initiated the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received e Disable No call waiting for the specific port e Empty Call waiting is determined according to the global parameter Enable Call Waiting described in Supplementary Services on page 138 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 209 3 4 4 7 Configuring the Endpoint Phone Numbers The Endpoint Phone Number Table page allows you to activate the device s ports endpoints by defining telephone numbers for the endpoints an
221. ash memory refer to Saving Configuration on page 209 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 Trunk 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 155 Table 3 37 Number Manipulation Parameters Description Parameter Description Source Trunk Group The source Trunk Group 1 99 for Tel to IP calls To denote any _SreTrunkGroupID 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 Source IP Group The IP Group from where the IP to IP call originated Typically this IP _SrclPGroupID Group of an
222. ason in the NOTIFY body such as 486 if busy tone detected and generates an additional hook flash towards the FXO line to restore connection to the original call Digit Patterns 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 Determines the digit pattern used by the PBX to indicate call Internal forward on busy when the original call is received from an internal DigitPatternForwardOnBusy extension The valid range is a 120 character string Forward on No Answer Digit Determines the digit pattern used by the PBX to indicate call Pattern Internal forward on no answer when the original call is received from an DigitPatternForwardOnNoAn internal extension swer The valid range is a 120 character string SIP User s Manual 192 Document LTRT 65411 SIP User s Manual Parameter Forward on Do Not Disturb Digit Pattern Internal DigitPatternForwardOnDND Forward on No Reason Digit Pattern Internal DigitPatternForwardNoReas on Forward on Busy Digit Pattern External DigitPatternForwardOnBusy Ext Forward on No Answer Digit Pattern External DigitPatternForwardOnNoAn swerExt Forward on Do Not Disturb Digit Pattern External DigitPatternForwardOnDND Ext Forward on No Reason Digit Pattern External Di
223. at several policies can be associated with a single IKE entry The valid range is 0 to 19 The default value is 0 96 Document LTRT 65411 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 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 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 encrypti
224. atabase 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 SASDefaultGatewaylP i e the device itself 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 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 274 Document LTRT 65411 SIP
225. ateway MediaPack v 5 40 010 006 When configured the string UserAgentDisplaylnfo 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 109 November 2008 ca AudioCodes Parameter SDP Session Owner SIPSDPSessionOwner Subject SIPSubject Multiple Packetization Time Format MultiPtimeFormat Enable Semi Attended Transfer EnableSemiAttendedTra nsfer 3xx Behavior 3xxBehavior Enable P Charging Vector EnablePCharging Vector Enable VoiceMail URI EnableVMURI Retry After Time RetryAfterTime Enable P Associated URI Header EnablePAssociatedURI Header SIP User s Manual MediaPack Series Description 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 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 Packetizat
226. ation message box appears Figure 3 36 Confirmation Message for Accessing the Multiple Interface Table 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 ener 3 Click OK to confirm the Multiple Interface Table page appears Figure 3 37 Interface Table Page Index Application Type IP Address Prefix Length Gateway vian ID Interface Name 1 JM v 1013 413 16 101301 lo Al interfaces ly 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 To save the changes to flash memory refer to Saving Configuration on page 209 When adding more than one interface to the table ensure 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
227. ation numbers for Tel to IP calls using the Web interface refer to Configuring the Number Manipulation Tables on page 151 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 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 RemoveFromLetft NumberMaplp2Tel RemoveFromRight NumberMaplp2Tel LeaveFromRight NumberMaplp2Tel Prefix2Add NumberMaplp2Tel_ Suffix2Add NumberMaplp2Tel IsPresentationRestricted NumberMaplp2Tel For example NumberMaplp2Tel NumberMaplp2Tel 0 03 22 2 667 NumberMaplp2Tel Notes This table parameter can include up to 100 indices The parameter NumberMaplp2Tel IsPresentationRestricted is not applicable Set its value to The parameters NumberMapTel2lp_ SrclPGroupID NumberMaplp2Tel NumberType and 293 November 2008 A tal AudioCodes MediaPack Series Parameter SourceNumberMapTel2I P SIP User s Manual Description NumberMaplp2Tel_ NumberPlan are not applicable Set these to RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add and LeaveFromRight are applied if the called an
228. available on all FXS ports Note The Lifeline splitter for FXS devices is a special order option Connectors amp Switches Rear Panel Analog Lines AC Power Supply Socket Ethernet RS 232 Reset Button Physical Dimensions HxWxD Weight Environmental Mounting Power Management Configuration Management and Version 5 6 MP 118 8 analog lines 8 x RJ 11 connectors MP 114 4 analog lines 4 x RJ 11 connectors MP 112 2 analog lines 2 x RJ 11 connectors 100 240 0 3A max 10 100Base TX RJ 45 Console PS 2 port Resets the MP 11x 42 mm 1 65 in x 172 mm 6 8 in x 220 mm 8 7 in 0 5 kg Approx Operational 5 to 40 C 41 to 104 F Storage 25 to 70 C 77 to 158 F Humidity 10 to 90 non condensing Desktop 19 inch rack and wall mounting Note The rack mount is a special order option 100 240 VAC Nominal 50 60 Hz HTTP based Embedded Web Server Web browser or ini file SNMP v2c SNMP v3 383 November 2008 ca AudioCodes Function Maintenance Type Approvals Safety and EMC MediaPack Series Specification Syslog according to RFC 3164 Local RS 232 terminal Web Management via HTTP or HTTPS Telnet UL 60950 1 FCC part 15 Class B CE Mark EN 60950 1 EN 55022 EN 55024 EN61000 3 2 EN61000 3 3 EN55024 10 2 MP 124 Specifications The table below lists the main technical specifications of the MP 124 Table 10 2 MP 124 Functional Specificatio
229. aylnfo Index CallerDisplayInfo DisplayString CallerDisplaylnfo IsCidRestricted CallerDisplayInfo Port CallerDisplaylnfo Module CallerDisplayInfo Where DisplayString Caller ID string IsCidRestricted Restriction 0 not restricted default 1 restricted Port Port number Module Module number 0 5 N A For example CallerDisplayInfo CallerDisplaylnfo 1 Mark M 0 5 Caller ID on channel 5 CallerDisplayInfo Notes The indexing of this ini file table parameter starts at 1 The numbering of channels starts with 0 To configure Caller Display Information using the Web interface refer to Caller ID on page 177 For a description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 This ini file table parameter forwards IP to Tel calls using SIP 302 response based on the device s port to which the call is routed The format of this parameter is as follows FwdInfo FORMAT Fwdlnfo Index Fwdlnfo Type FwdInfo Destination Fwdinfo_NoReplyTime Fwdlnfo Port FwdInfo Module Fwadlnfo Where Type Forward Type for a list of options refer to Call Forward on page 178 Destination Telephone number or URI number IP address to which the call is forwarded NoReplyTime Timeout in seconds for No Reply If you have set the Forward Type for this port to No Answer 3 enter the number of seconds the device wa
230. ayloadFormat MinFlashHookTime FlashHookOption FlashHookPeriod Version 5 6 4 ini File Configuration Description Defines the deviation in Hz allowed for the detection of each signal freguency 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 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 Determines whether the device adds the Blind Transfer code KeyBlindTransfer to the dialed destination number 0 Disable default 1 Enable Note This parameter is applicable to FXO and FXS interfaces 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 refer to Configuring the Hook Flash Settings on page 77 For a description of this parameter refer to DTMF amp Dialing Parameters on page 125 For a description of this parameter
231. ays 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 server refer to the Product Reference Manual gt To activate the Message Log take these 3 steps 1 Inthe Advanced Parameters page refer Advanced Parameter on page 129 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 2 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 id 14h 43m 9s Lid 14h 43m 9s 11d 14h 43m 95s iid 14h 43m 93s 11d 14h 43m 9s did 14h 43 99 1id 14h 43m 9s 1id 14h 43m 92 1id 14h 43m 9s 1id 1dh 43m 99 1id 14h 43m 98s 1id 14h 43m 9s 1id 14h 43m 98 1id 14h 43m 9s Lid 14h 43 98 lid 14h 43m 95s 1id 14h 43m 93 1id 14h 43m 9s Version 5 6 lgr psbrdex 2662 lgr flov 2663 lgr flov 2664 igr_pabrdit 2665 lgr p
232. 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 Version 5 6 367 November 2008 8 3 8 4 8 5 A c tal AudioCodes MediaPack Series 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 NoOplnterval For a description of the RTP No Op ini file parameters refer to Networking Parameters on page 236 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 236 AudioCodes default payload type is 120 m 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 duplication of the sequence number Note Receipt of No Op packets is always supported IP Multicasting The device supports I
233. ber 2008 ca AudioCodes Parameter Registration Time RegistrationTime Re registration Timing RegistrationTimeDivider Registration Retry Time RegistrationRetryTime Registration Time Threshold RegistrationTimeThresho Id Re register On INVITE Failure RegisterOnlnviteFailure ReRegister On Connection Failure ReRegisterOnConnection Failure Miscellaneous parameters Gateway Name SIPGatewayName SIP User s Manual MediaPack Series Description Defines the time interval 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 RegistrationTimeThres
234. ber further using the Number Manipulation tables refer to Number Manipulation and Routing Parameters on page 289 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 FXO Blind Transfer modes LineTransferMode 1 2or3 The valid range is a string of up to 9 characters The default is an empty string For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 This ini file table parameter defines call waiting per port The format of this parameter is as follows CallWaitingPerPort FORMAT CallWaitingPerPort_Index CallWaitingPerPort IsEnabled CallWaitingPerPort_Port CallWaitingPerPort Module CallWaitingPerPort Where IsEnabled Enables 1 or disables 0 call waiting Port Port number 268 Document LTRT 65411 SIP User s Manual Parameter CHRRTimeout EnableCallWaiting 3WayConferenceMode Version 5 6 4 ini File Configuration Description Module Module number For example CallWaitingPerPor
235. ble TelnetServerPort TelnetServerldleDisconnect SSHServerEnable SSHServerPort 4 ini File Configuration Description For a description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 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 only 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 41 For a description of this parameter refer to Configuring the Application Settings on page 58 For a description of this parameter refer to Configuring the Application Settings o
236. ble Page v Add Phone Context s Prefix Phone Context Index NPI Phone Context Unknown Unknown unknown com Private Level 2 Regional host com na e1b4 host com E 164 Public National 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 209 Several rows with the same NPI TON or Phone Context 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 289 Table 3 39 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 Called and Calling numbers 0 Disable Disable default 1 Enable Enable NPI Select the Number Plan assigned to this entry 0 Unknown Unknown default 1 E 164 Public E 164 Public 9 Private
237. calls when the port is busy or when calls are not answered within the time specified in the Time for No Reply Forward field 5 Do Not Disturb Immediately reject incoming calls Forward to Phone The telephone number or URI lt number gt lt IP address gt to where the Number call is forwarded Note If this field only contains a telephone number and a Proxy isn t used the forward to phone number must be specified in the Tel to IP Routing table refer to Tel to IP Routing Table on page 160 Time for No Reply If you have set the Forward Type for this port to No Answer enter the Forward number of seconds the device waits before forwarding the call to the phone number specified 3 4 4 6 5 Caller ID Permissions The Caller ID Permissions page allows you to enable or disable per port the Caller ID generation for FXS interfaces and detection for FXO interfaces If a port isn t configured its Caller ID generation detection are determined according to the global parameter EnableCallerlID described in Supplementary Services on page 138 Note You can also configure the Caller ID Permissions table using the ini file table parameter EnableCallerlD refer to Analog Telephony Parameters on page 279 Version 5 6 179 November 2008 7a K tal AudioCodes MediaPack Series gt To configure Caller ID Permissions per port take these 4 steps 1 Open the Caller ID Permissions page Configuration t
238. 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 PeerHostNameVerificationMode is set to 1 or 2 3 4 3 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 252 SIP User s Manual 94 Document LTRT 65411 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 55 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 Releated Key Exchange Method Index SA Lifetime sec SA Lifetime KB First Proposal Encryption Type First Proposal Authentication Type Second Proposal Encryption Type Second Proposal Authentication Type Third Proposal Encryption Type Third Proposal Authentication Type Fourth Proposal Encryption Type Fourth Proposal Authentication Type Transport Control 0 0 0
239. ccording 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 FADN 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 maximum 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 246 Document LTRT 65411 SIP User s Manual Parameter PrtFileURL CptFileURL FXSCoeffFileURL TLSRootFileUrl TLSCertFileUrl UserInfoFileURL AutoUpdateCmpFile AutoUpdateFreguency AutoUpdatePredefinedTim e ResetNow Version 5 6 4 ini File Configuration Description 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
240. ce 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 m Title of the table The name of the table in square brackets e g MY_TABLE_NAME m 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 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 m 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 IMY TABLE NAMEJ Version 5 6 233 November 2008 7a L e AudioCodes MediaPack Series The following displays an example of the structure of an ini file table parameter Title
241. ce ON OFF cycle Can be omitted if there isn t a third cadence 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 Recommended values from 10 to 25 Notes e 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 e 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 Lo
242. ce delay ms Average voice jitter ms Total RTP packets TX Total RTP packets RX Total call atternpts Reset Statistics gt To reset the performance statistics to zero take the following step m Click the Reset Statistics button SIP User s Manual 222 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 m Source unit from which the alarm was raised m Description brief explanation of the alarm m 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 48 gt To view the list of alarms take this step m Open the Active Alarms page Status amp Diagnostics tab gt Status amp Diagnostics menu gt Active Alarms page item Figure 3 113 Active Alarms 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 including 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 Cal
243. ced Parameters on page 129 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Proxy amp Registration Parameters on page 112 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer t
244. celPAddressInput SIP User s Manual Description Determines whether the device s Hunt Group ID is added as a prefix to the destination phone number for Tel to IP calls 0 No Dont add Hunt Group ID as prefix default 1 Yes Add Hunt Group ID as prefix to called number Notes This option can be used to define various routing rules To use this feature you must configure the Hunt Group IDs refer to Configuring the Endpoint Phone Numbers on page 181 Determines whether the port number is added as a prefix to the called number for Tel to IP calls 0 No Don t add port number as prefix default 1 Yes Enable add port number as prefix If enabled the port number single digit in the range 1 to 8 for 8 port devices two digits in the range 01 to 24 for MP 124 is added as a prefix to the called destination phone number This option can be used to define various routing rules 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 163 from a telephone number for an IP 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 Hunt Group Routing table is scanned for a matching prefix If such a prefix is fo
245. come 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 KK KKKK KKK KK A KKK KK K KK K KKK K K K M WelcomeMessage 2 This is a Welcome message WelcomeMessage 3 KK KK KK K KKK K KKK K KK KKK K KK KKK K M WelcomeMessage Note Each index represents a line of text in the Welcome message box Up to 20 indices can be defined 46 Document LTRT 65411 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 button the Help topic pertaining to the opened page appears as shown below Figure 3 25 Help Topic for Current Page faAudioC J MP 118 FXS Z FXO v Sub met Bum 6 Help os Lo oft TTI Stabs Centiguaten Maragemert 3 Diagnostics Defines the NT
246. cording to the parameter HoldFormat Receiving Hold Retrieve m When an active call receives a Re INVITE message with either the IP address 0 0 0 0 or the inactive string in SDP the device stops sending RTP and plays a local Held tone m When an active call receives a Re INVITE message with the sendonly string in SDP the device stops sending RTP and listens to the remote party In this mode it is expected that on hold music or any other hold tone is played over IP by the remote 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 138 Version 5 6 357 November 2008 7a E tal AudioCodes MediaPack Series The device also supports double call hold for FXS interfaces where the called party which has been placed on hold by the calling party can then place the calling party on hold as well and make a call to another destination The flowchart below provides an example of this type of call hold Figure 7 28 Double Hold SIP Call Flow Endpoint C Endpoint A Endpoint B Endpoint D l INVITE sendrecv l l 200 OK sendrecv I I I I I I Conversation gt l I I I I I I I I I I I I I I I 1 INVITE Hold inactive l 200 OK inactive I l 200 OK sendrecv U 1 INVITE Hold inactive l 200 OK inactive l eae Dial Conversation gt
247. ct Anonymous Call Activate Deactivate 2 Configure the Keypad Features 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 209 Version 5 6 147 November 2008 A L tal AudioCodes MediaPack Series Table 3 35 Keypad Features Parameters Description Parameter Description Forward Note The forward type and number can be viewed in the Call Forward table refer to Call Forward on page 178 Unconditional Keypad sequence that activates the immediate call forward option KeyCFUnCond No Answer KeyCFNoAnswer Keypad sequence that activates the forward on no answer option On Busy KeyCFBusy Keypad seguence that activates the forward on busy option On Busy or No Answer Keypad sequence that activates the forward on busy or no answer KeyCFBusyOrNoAnswer option Do Not Disturb Keypad seguence that activates the Do Not Disturb option KeyCFDoNotDisturb immediately reject incoming calls To activate the reguired forward method from the telephone 1 Dial the preconfigured seguence number on the keypad a dial tone is heard 2 Dial the telephone number to which the call is forwarded terminate the number with a confirmation tone is heard Deactivate Keypad seguence that deactivates any of the call forward options KeyCFDeact After the seguence is pressed a confirmati
248. ct Name field enter the DNS name and then click Generate CSR A textual certificate signing reguest 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 reguest 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 MIIDkzCCAnugAwIBAgIEAgAAADANBgkghkiG9w0BAOOFADA MOswCOYDVOOGEWJIGUj ETMBEGA1UEChMKO2VyadGlwb3N0ZTEDMBkGA1UBAxMSO2VydGlwb3N0ZSBTZXJ2ZXVy MB4XDTk4MDYyNDA4MDAwMFoXDTE4MDYyNDA4MDAwMFowPzELMAkGA1UEBhMCR1IxEz ARBgNVBAOTCkN1cnRpcG9zdGUxGzAZBgNVBAMTEkN1cnRpcG9zdGUgU2VydmV1cjcCC ASEwDOYJKoZIhvcNAOEBBOADggEOADCCAOkCggEAPgd4MziR4spWldGRx8borhZkon WnNm Yhb7 4067ecf1janH7GcN SXsfx7jJpreWULf7v7Cvpr4R7gIJcmdHIntmf7 JPM5n6cDBv17uSW63er7NkVnMFHwK10aGFLMybFkzaeGrvFm4k31RefiXxDmu0Oe FhJ gHYezYHf44LvPRPwhSrzi9 Aq308pWDguJuZDIUP1F1jMa LPwvREXf FcUW w END CERTIFICATE 6 Set the parameter Secured Web Connection HTTPS to HTTPS Only 0 refer to Configuring the General Security Settings on page 90 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 7 In the Certifica
249. ct Reference Manual The raw data files must be recorded with the following characteristics E Coders G 711 A law or G 711 u law m Rate 8 kHz m 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 210 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 Coefficient File The Coeff_FXS dat file is used to provide best termination and transmission quality adaptation for different line types for FXS interfaces This adaptation is performed by modifying the telephony interface characteristics such as DC and AC impedance feeding current and ringing voltage The coeff dat auxiliary file is produced specifically for each market after comprehensive performance analysis and testing and can be modified on request The current file supports US line type of 600 ohm AC impedance and 40 V RMS ringing voltage for REN 2 To load the coeff dat file to the device use the Web interface or specify the FXS coeff dat file name in the device s ini file refer to
250. ction describes the device s IP telephony capabilities 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 typically 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 parall
251. d Accounting AAA AAAlndications indications 0 None No indications default 3 Accounting Only Only accounting indications are used 3 4 5 3 Configuring the FXO Parameters The FXO Settings page allows you to configure the device s specific FXO parameters Note The FXO Settings page is available only for FXO interfaces Version 5 6 195 November 2008 7a K tal AudioCodes MediaPack Series gt To configure the FXO parameters take these 4 steps 1 Open the FXO Settings page Configuration tab gt Advanced Applications menu gt FXO Settings page item Figure 3 91 FXO Settings Page Dialing Mode Waiting for Dial Tone Answer Supervision Ring Detection Timeout sec 8 Rings before Detecting Caller ID 1 Guard Time Between Calls 1 Two Stages No Time to Wait before Dialing msec 1000 Reorder Tone Duration sec 255 No Send Metering Message to IP No Disconnect Call on Detection of Busy Tone Enable Disconnect On Dial Tone Disable 2 Configure the FXO 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 209 Parameter Dialing Mode IsTwoStageDial Waiting For Dial Tone IsWaitForDialTone SIP User s Manual Table 3 51 FXO Parameters Description Description Determines the dialing m
252. d assigning them to Hunt Groups and profiles Each endpoint i e channel must be assigned a unique phone number In other words no two endpoints can have the same phone number You can also configure the endpoint phone numbers using the ini file table parameter TrunkGroup refer to Number Manipulation and Routing Parameters on page 289 gt To configure the Endpoint Phone Number table take these 4 steps 1 Open the Endpoint Phone Number Table page Configuration tab gt Protocol Configuration menu gt Endpoint Number submenu gt Endpoint Phone Number page item Version 5 6 181 November 2008 7a i K tall AudioCodes MediaPack Series Figure 3 85 Endpoint Phone Number Table Page Channel s Phone Number Hunt Group ID Profile ID 2 Configure the endpoint phone numbers according to the table below You must enter a number in the Phone Number fields for each port that you want to use 3 Click the Submit button to save your changes or click the Register or Un Register buttons to save your changes and to register unregister to a Proxy Registrar 4 To save the changes to the flash memory refer to Saving Configuration on page 209 Table 3 45 Endpoint Phone Number Table Description Parameter Description Channel s The port numbers channels endpoints as labeled on the device s rear panel To enable a device channel you must enter the port channel number You can enter a ran
253. d 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 151 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 This ini file table parameter manipulates the source phone number for Tel to IP calls The format of this parameter is as follows SourceNumberMapTel2Ip FORMAT SourceNumberMapTel2Ip Index SourceNumberMapTel2lp_DestinationPrefix SourceNumberMapTel2lp_SourcePrefix SourceNumberMapTel2Ip SourceAddress SourceNumberMapTel2Ip NumberType SourceNumberMapTel2lp_NumberPlan SourceNumberMapTel2lp_RemoveFromLeft SourceNumberMapTel2lp_RemoveFromRight SourceNumberMapTel2lp_LeaveFromRight SourceNumberMapTel2lp_Prefix2Add So
254. d 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 f using 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 80 91 November 2008 ca AudioCodes Parameter RADIUS Authentication Server IP Address RADIUSAuthServerlP RADIUS Authentication Server Port RADIUSAuthPort 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 RadiusVSAVendorlD RADIUS VSA Access Level Attribute RadiusVSAAccessAttribute EtherDiscover Setting EtherDiscover Operation Mode IPSec Setting Enable IP Security EnablelPSec SIP User s Manual MediaPack Series Description IP address of the RADIUS authentication server Port number of the RADIUS authentication server The default value is 1645 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 respon
255. d 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 messages 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
256. ded 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 Fora list of supported coders refer to Coders on page 123 To configure the Coders table in the Web interface refer to Coders on page 123 Fora description of using ini file table parameters refer to Structure of ini File Table Parameters on page 233 275 November 2008 ca AudioCodes Parameter IPProfile TelProfile SIP User s Manual MediaPack Series Description 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 IpPreference 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 IPProfile EnableEarlyMedia IPProfile P
257. 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 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 Confirmation 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
258. des MediaPack Series Table 3 54 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 default 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 199 2 Inthe SNMP V3 Table field click the right pointing arrow u button the SNMP V3 Settings page appears Figure 3 95 SNMP V3 Setting Page Add Compact Delete P Apply User Name Authentication Protocol Privacy Protocol Authentication Key Privacy Key Read Write None v None 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 refe
259. ding only a single parameter of the parameters described above The IP Profile configured in IP Profile Settings on page 173 that is assigned to the routing rule The source IP Group 1 9 associated with the incoming IP to Tel 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 188 for obtaining authentication user name password for this call 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 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 236 SIP User s Manual 166 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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
260. dministrator gt Toreplace 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 52 Certificates Signing Reguest Page Certificate Signing Reguest Subject Name Generate CSR Copy the certificate signing request and send it to your Certification Authority for signing erate self signed to create a self signed cert ate using the subject name provided above e the device will be out of service onfiguration and rese Pres ton Generat t eate elf signed certific Important this is a lengthy operation during this tim After the t ave r 1 operation is complete 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 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 SIP User s Manual 86 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 In the Subje
261. dress 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 Version 5 6 83 November 2008 7a E tal AudioCodes MediaPack Series 3 4 3 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 firewall 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 252 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 51 Firewall Settings Page Action Edit Is Rule Local Port Packet Burst Match Rule Active Source IP Subnet Mask Range Protocol Size Byte rate Bytes u Oo No mgmt customer com 255 255 255 255 0 80 TCP 0 0 0 ALLOW 0 2 O No 192 0 0 0 255 0 0 0 0 65535 Any 0 4
262. ds 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 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 BootPSelectiveEnable 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 BootPDelay The interval between the device s startup and the first BootP DHCP request that is issued by the device 1 1 second default 2
263. e SIP User s Manual 220 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 3 111 Active IP Interfaces Page Index Application Type Address Type Interface Mode IP Address TER Gateway WLAN 10 Interface Name IPv4 Manual 10 13 4 13 16 10 13 0 1 d 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 General Settings MAC Address oo908fO84fo9 Serial Number 544665 Board Type 56 Device Up Time Od 0h 23m 3s 48th Device Administrative State Unlocked Device Operational State Enabled Flash Size bytes 8388608 RAM Size bytes 33554432 CPU Speed MHz 40
264. e TargetOfChannel Where Destination Destination phone number Type 1 Destination phone number is automatically dialed if phone is off hooked for FXS interface or ring signal is applied to port FXO interface 0 automatic dialing is disabled 2 enables Hotline when a phone is off hooked and no digit is pressed for HotLineToneDuration the destination phone number is automatically dialed Port Port number Module Module number 0 5 N A For example TargetOfChannel TargetOfChannel 2 108 1 7 Automatic dialing on port 7 TargetOfChannel Notes The indexing of this ini file table parameter starts at 1 The numbering of channels starts at 0 Define this parameter for each device port that implements Automatic Dialing This parameter can appear up to 8 times for 8 port devices and up to 24 times for MP 124 devices To configure the Automatic Dialing Table using the Web interface refer to Automatic Dialing on page 175 For an explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 281 November 2008 ca AudioCodes Parameter CallerDisplaylInfo Fwdinfo SIP User s Manual MediaPack Series Description This ini file table parameter enables the device to send Caller ID information to IP when a call is made The format of this parameter is as follows CallerDisplayInfo FORMAT CallerDispl
265. e An FXO VoIP device interfaces between the CO PBX line and the Internet FXS Foreign Exchange Station is the interface replacing the Exchange i e the CO or the PBX and connects to analog telephones dial up modems and fax machines The FXS is designed to supply line voltage and ringing current to these telephone devices An FXS VoIP device interfaces between the analog telephone devices and the Internet 14 Document LTRT 65411 SIP User s Manual 1 Overview 1 Overview This manual provides you with information for configuring and operating the VolP analog MediaPack series devices listed in the table below Table 1 1 Supported MediaPack Series Configurations Product Name MP 124 MP 118 MP 114 MP 112 FXS SSS v Combined FXS Number of FXO Channels x x 24 v 4 4 8 v 2 2 4 x x 2 1 1 The MP 112 differs from the MP 114 and MP 118 in that its configuration excludes the RS 232 connector Lifeline option and outdoor protection Gateway Description The MediaPack series analog Voice over IP VolP Session Initiation Protocol SIP media gateways hereafter referred to as device are cost effective cutting edge technology products These stand alone analog VolP devices provide superior voice technology for connecting legacy telephones fax machines and Private Branch Exchange PBX systems to IP based telephony networks as well as for integration with new IP based PBX architectures These device
266. e proposal number 0 to 3 1 DES CBC 99 November 2008 A e AudioCodes MediaPack Series Parameter Name Description 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 IKEPolicyProposalAuthenti depicts the proposal number 0 to 3 cation_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 up to Group four proposals For the ini file parameter X depicts the proposal IKEPolicyProposalDHGrou number 0 to 3 X 0 DH 786 Bit 1 DH 1024 Bit Not Defined default 3 4 4 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 Definition Parameters on page 100 SIP Advanced Parameters refer to Configuring the SIP Advanced Parameters on page 129 Manipulation Tables refer to Configuring the Number Manipulation Tables on page 151 Routing Tables refer to Configuring the Routing Tables on page 157 Profile Definitions refer to Configuring the Profile Definitions on page 169 Endpoint Settings refer to Configuring the Endpoint Settings on page 1
267. e Determines the call transfer method used by the device LineTransferMode 0 None IP default 1 Blind PBX blind transfer After receiving a REFER message from the IP side the FXO sends a hook flash to the PBX dials the digits that are received in the Refer To header and then immediately drops the line on hook The PBX performs the transfer internally 2 Semi Supervised PBX Semi Supervised transfer After receiving a REFER message from the IP side the FXO sends a hook flash to the PBX and then dials the digits that are received in the Refer To header If no Busy or Reorder tones are detected within approximately 2 seconds the device completes the call transfer by releasing the line otherwise the transfer is cancelled the device sends a SIP NOTIFY message with a failure reason in the NOTIFY body such as 486 if busy tone detected and generates an additional hook flash towards the FXO line to restore connection to the original call 8 Supervised PBX Supervised transfer After receiving a REFER message from the IP side the FXO sends a hook flash to the PBX and then dials the digits that are received in the Refer To header The FXO waits for connection of the transfer call and if speech is detected e g hello within approximately 2 seconds the device completes the call transfer by releasing the line otherwise the transfer is cancelled the device sends a SIP NOTIFY message with a failure re
268. e 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 289 To configure the Tel to IP Routing table take these 5 steps Open the Tel to IP Routing page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Tel to IP Routing page item Figure 3 71 Tel to IP Routing Page Routing Index 10 W Tel To IP Routing Mode Route cals betore maraton Dest Phone Prefix Source Phone Prefix gt Dest IP Address 10 100 1033456 346 From the Routing Index drop down list select the range of entries that you want to add 161 November 2008 A c tal AudioCodes MediaPack Series 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 209 Table 3 41 Tel to IP Routing Table Parameters Description Parameter Description Tel to IP Routing Mode Determines whether to route Tel calls to IP before or after RouteModeTel2IP mani
269. e 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 10 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 SIP User s Manual 370 Document LTRT 65411 SIP User s Manual 8 Networking Capabilities 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 m Outgoing packets from the device to the switch All outgoing packets are tagged each according to its interface Control Media or OAMP If the device s native VLAN ID is ident
270. e 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 J Subnet Mask 255 255 0 0 HS Default Gateway Address Control Network Settings IS IP Address 10 32 174 50 J Subnet Mask 255 255 0 0 HS Default Gateway Address Media Network Settings I IP Address 10 33 174 50 J 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 p button alongside the label Multiple Interface Table refer to Configuring the Multiple Interface Table on page 55 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 VlanID InterfaceName lo 10 33 17450 hs 103301 Media b Click the Submit button to save your changes Note Configure the OAM parame
271. e device if you have loaded a Call Progress Tones file refer to Resetting the Device on page 207 211 November 2008 A K tal AudioCodes MediaPack Series 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 208 You can schedule automatic loading of updated auxiliary files using HTTP HTTPS FTP or NFS refer to the Product Reference Manual 3 5 2 2 You can also load the Auxiliary files using 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 303 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
272. e 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 Configuring the device to use and operate with the SAS capabilities refer to Configuring SAS on page 316 m Configuring SAS emergency call routing refer to Configuring Emergency Calls on page 316 Version 5 6 315 November 2008 A e AudioCodes MediaPack Series 7 1 1 Configuring SAS For configuring the device to operate with SAS perform the following configurations IsProxyUsed 1 ProxylP 0 lt SAS agent s IP address i e the device gt ProxylP 1 lt external Proxy server IP address gt IsRegisterNeeded 1 for the device RegistrarlP SIPDestinationPort 5080 IsUserPhonelnFrom 0 don t use user phone in From Header IsFallbackUsed 0 EnableProxyKeepAlive 1 enables keep alive with Proxy using OPTIONS EnableSAS 1 SASLocalSIPUDPPort default 5080 SASRegistrationTime lt expiration time that SAS returns in the 200 OK to REGISTER in Emergency mode gt default 20 B m F m IsUserPhone 0 don t use user phone in SIP URL E a B m a SASDefaultGatewayIP lt SAS gateway IP address gt m SASProxySet 1 7 1 2 Configuring Emergency Calls The device s SAS agent can be configured to detect a user defined
273. e 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 Notes Applicable only if single device registration is used i e Authentication Mode is set to Authentication Per gateway The Authentication table can be used instead refer to Authentication on page 174 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 Passw Note The Authentication table can be used instead refer to Authentication on page 174 118 Document LTRT 65411 SIP User s Manual Parameter Cnonce Cnonce Authentication Mode AuthenticationMode Set Out Of Service On Registration Failure OOSOnRegistrationFail Challenge Caching Mode SIPChallengeCachingMo de Version 5 6 3 Web Based Management Description
274. e 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 intervals 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 247 November 2008 A L tal AudioCodes MediaPack Series Parameter Description 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 BootP and TFTP Parameters 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 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 sen
275. e 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 is 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 52 For a description of this parameter refer to Configuring the Management Settings on page 199 For a description of this parameter refer to Configuring the Management Settings on
276. e this mode define the following e RxDTMFOption 0 ini file Declare RFC 2833 in SDP field No Web interface refer to DTMF 8 Dialing Parameters on page 125 317 November 2008 A L e AudioCodes MediaPack Series e TxDTMFOption 2 ini file 1 to 5 Tx DTMF Option field NOTIFY Web interface refer to DTMF 8 Dialing Parameters on page 125 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 m 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 8 Dialing Parameters on page 125 e TxDTMFOption 4 ini file 1 to 5 Tx DTMF Option field RFC 2833 Web interface refer to DTMF amp Dialing Parameters on page 125 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
277. e 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 260 gt To configure IP Groups take these 4 steps 1 Open the IP Group Table page Configuration tab gt Protocol Configuration menu gt Hunt IP Group submenu gt IP Group Table page item Figure 3 87 IP Group Table Page Always se Route Table SIP Re Routing U Description SIP Group Name Contact User Mode Standard v Disable Standard v Disable v Standard v Disable v Standard v l Disable v 2 Configure the IP group parameters according to the table below SIP User s Manual 186 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 209 Parameter Description Proxy Set ID SIP Group Name Contact User SIP Re Routing Mode Version 5 6 Table 3 47 IP Group Parameters Description Description 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 120 to associate wi
278. eRouting SIPGatewayName IsProxyUsed ProxyName AlwaysSendToProxy PreferRouteTable 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 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 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General
279. ection 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 72 Document LTRT 65411 SIP User s Manual Parameter Fax Modem Bypass Coder Type FaxModemBypassCode rType Fax Modem Bypass Packing Factor FaxModemBypassM Fax Bypass Output Gain FaxBypassOutputGain Modem Bypass Output Gain ModemBypassOutputG ain Fax CNG Mode FaxCNGMode CNG Detector Mode CNGDetectorMode T 38 Max Datagram Size T38MaxDatagram Version 5 6 3 Web Based Management Description 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 is 31 to 31 dB in 1 dB steps The default is 0 i e no gain
280. ects a polarity reversal e Voice Detection device sends a 200 OK in response to an INVITE only when it detects the start of speech or ringback tone from the Tel side Note that the IPM detectors must be enabled Version 5 6 327 November 2008 7a 5 c tall AudioCodes MediaPack Series 7 4 1 2 7 4 1 3 Two Stage Dialing Two stage dialing is when the IP caller is required to dial twice The caller initially dials to the FXO device and only after receiving a dial tone from the PBX via the FXO device dials the destination telephone number Figure 7 3 Call Flow for Two Stage Dialing FXO G amp Client F1 INVITE FXO seizes line Two stage dialing implements the Dialing Time feature Dialing Time allows you to define the time that each digit can be separately dialed By default the overall dialing time per digit is 200 msec The longer the telephone number the greater the dialing time The relevant parameters for configuring Dialing Time include the following m DTMFDigitLength 100 msec time for generating DTMF tones to the PSTN PBX side m DTMFinterDigitInterval 100 msec time between generated DTMF digits to PSTN PBX side Call Termination Disconnect Supervision on FXO Devices The FXO Disconnect Supervision enables the device s FXO ports to monitor call progress tones from a PBX or from the PSTN This allows the FXO to determine when the call has terminated on the PBX side and thereby prevents anal
281. ed 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 Proxy Sets Table on page 120 associated with these Serving IP Groups A Hunt 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 Note You can also configure the Account table using the ini file table parameter Account refer to SIP Configuration Parameters on page 260 gt To configure Accounts take these 5 steps 1 Open the Account Table page Configuration tab gt Protocol Configuration menu gt Hunt IP Group submenu gt Account Table page item Figure 3 88 Account Table Page Compact oup ServinglPGroup Username 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 209 Note For a descri
282. ed to device management These menus appear in the Navigation tree and include the following m Management Configuration refer to Management Configuration on page 198 m Software Update refer to Software Update on page 210 SIP User s Manual 198 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 5 1 3 5 1 1 Management Configuration The Management 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 199 m Regional settings refer to Configuring the Regional Settings on page 206 m Maintenance Actions refer to Maintenance Actions on page 207 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 92 Management Settings Page wv Syslog Settings Syslog Server IP Address Syslog Server Port Enable Syslog Analog Ports Filter SNMP Settings SNMP Trap Destinations SNMP Community String SNMP V3 Table SNMP Trusted Managers Disable SNMP Trap Manager Host Name w Activity Types to Report via Activity L
283. ed 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 control 0 None default 1 Hardware 249 November 2008 A tal AudioCodes MediaPack Series 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 82 For a description of this parameter
284. efer to IP to Hunt Group Routing on page 163 configure that INVITEs with ITSP1 as the hostname in the From URI are routed to Hunt Group 1 and INVITEs with ITSP2 as the hostname in the From URI are routed to Hunt Group 2 In addition configure calls received from ITSP1 as associated with IP Group 1 Figure 7 26 Configuring IP to Hunt Group Routing pd P Profile Source Dest Host Prefix Source Host Prefix Dest Phone Profix Source Phone Prefix Source IP Address 7 Group o IPGroup ID ITSP1 1 1 ITSP2 9 In the Tel to IP Routing page refer to Tel to IP Routing Table on page 160 configure Tel to IP routing rules for calls from Hunt Group 1 to IP Group 1 and from Hunt Group 2 to IP Group 2 Version 5 6 355 November 2008 7a K tal AudioCodes MediaPack Series Figure 7 27 Configuring Tel to IP Routing Sre Trunk Group 0 Dest Phone Prefix Source Phone Prefix Dest IP Address on IP Profile ID 7 14 Working with Supplementary Services The device supports the following supplementary services Call Hold and Retrieve refer to Call Hold and Retrieve on page 356 Consultation Alternate refer to Consultation Alternate on page 359 Call Transfer refer to Call Transfer on page 359 Call Forward 3xx Redirect Responses refer to Call Forward on page 360 Call Waiting 182 Queued Response refer to Call Waiting on page 361 Message Waiting Indication MWI re
285. el 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 separate analog or digital gateway For PSTN fallback the local VoIP gateway should be equipped with analog FXO lines for PSTN connectivity In this way the Enterprise preserves its capability for internal and outgoing calls 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 th
286. el 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 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 SIP User s Manual 172 Document LTRT 65411 SIP User s Manual 3 Web Based Management 6 From the Coder Group drop down list select the Coder Group refer to Coder Group Settings on page 170 or the device s default coder refer to Coders on page 123 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 209 3 4 4 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 160 and IP to Hunt Group Routing page refer to IP to Trunk Group Routing on page 163 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 fi
287. el side indicates a Hook Flash event The valid range is a 25 character string The default is a null string Message Waiting Indication MWI Parameters Enable MWI EnableMWI SIP User s Manual Enables Message Waiting Indication MWI 0 Disable Disabled default 1 Enable MWI service is enabled Notes This parameter is applicable only to FXS interfaces The device supports only the receipt of SIP MWI NOTIFY messages the device doesn t generate these messages For detailed information on MWI refer to Message Waiting Indication on page 361 142 Document LTRT 65411 SIP User s Manual Parameter MWI Analog Lamp MWIAnalogLamp MWI Display MWIDisplay Subscribe to MWI EnableMWISubscription MWI Server IP Address MWIServerlP MWI Server Transport Type MWIServerTransportTy pe MWI Subscribe Expiration Time MWIExpirationTime Stutter Tone Duration StutterToneDuration Version 5 6 3 Web Based Management Description Enables visual display of MWI 0 Disable Disable default 1 Enable Enables visual Message Waiting Indication by supplying line voltage of approximately 100 VDC to activate the phone s lamp Note This parameter is applicable only for FXS interfaces Determines whether MWI information is sent to the phone display 0 Disable MWI information isn t sent to display default 1 Enable The device generates an MWI mes
288. elephony related ini file configuration parameters are described in the table below Table 4 10 Analog Telephony ini File Parameters Parameter Description Prefix2ExtLine Defines a string prefix e g 9 dialed for an external line that when identified causes the device s FXS port to play a secondary dial tone and then restart digit collection The valid range is a 1 character string The default is an empty string Note This parameter is applicable only to FXS interfaces PrecedenceRingingType For a description of this parameter refer to Supplementary Services on page 138 FXONumberOfRings Defines the number of rings before the device s FXO interface answers a Call When set to 0 the FXO seizes the line after one ring When set to 1 the FXO seizes the line after two rings The valid range is 0 to 255 The default is 0 seconds Note If caller ID is enabled and if the number of rings defined by the parameter RingsBeforeCallerlD is greater than the number of rings defined by this parameter the greater value is used CountryCoefficients Determines the FXO line characteristics AC and DC according to USA or TBR21 standard 66 TBR21 70 United States default ChargeCode This ini file table parameter configures metering tones and their time intervals that the device s FXS interface generates to the Tel side The format of this parameter is as follows ChargeCode FORMAT ChargeCode Index ChargeCode EndTime1 C
289. ember 2008 ca AudioCodes 3 4 4 1 2 Proxy amp Registration Parameters MediaPack Series 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 on page 226 gt To configure the Proxy amp Registration parameters take these 4 steps 1 Open the Proxy amp Registration page Configuration tab gt Protocol Configuration menu gt Protocol Definition submenu gt Proxy amp Registration page item Figure 3 58 Proxy amp Registration Page Use Default Proxy Proxy Name Redundancy Mode Proxy IP List Refresh 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 Subscription Mode Number of RTX Before Hot Swap Use Gateway Name for OPTIONS User Name Password Cnonce Authentication Mode Set Out Of Service On Registration Failure Challenge Caching Mode Mutual Authentication Mode No Parking 60 Disable No Disable Routing T able Standard Mode Disable A Record A Record Per Endpoint 3 No Default_Passwd
290. ensions FXO Device FXS Device Remote PBX Extensions Plays Call Waiting Tone and Sends Caller ID 7 13 4 5 FXS Gateway Configuration The procedure below describes how to configure the FXS device at the remote PBX extension gt Toconfigure the FXS interface take these 3 steps 1 In the Endpoint Phone Numbers page refer to Configuring the Endpoint Phone Numbers on page 181 assign the phone numbers 100 to 107 to the device s endpoints Figure 7 13 Assigning Phone Numbers Channel s Phone Number Hunt Group ID 1 8 100f SIP User s Manual 350 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 2 In the Automatic Dialing page refer to Automatic Dialing on page 175 enter the phone numbers of the FXO device in the Destination Phone Number fields When a phone connected to Port 1 off hooks the FXS device automatically dials the number 200 Figure 7 14 Automatic Dialing Configuration Gateway Destination Phone Auto Dial Port Number Status Port 1 FXS Enable W Port 2 FXS Enable Port 3 FXS 202 Enable Port4 FXS Enable PortS FXS i Enable Port 6 FXS Enable Port 7 FRS Enable Pot 8 FXS 3 In the Tel to IP Routing page refer to Tel to IP Routing Table on page 160 enter 20
291. ent 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 97 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 Secure Hypertext Transport Protocol HTTPS Parameters HTTPSOnly HTTPSPort HTTPSCipherString WebAuthMode HTTPSReguireClientCertific ate Version 5 6 For a description of this parameter refer to Configuring the General Security Settings on page 90 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 90 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
292. 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 335 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 168 Alternative routing using this table is commonly 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 th
293. er in this table is set to 1 In addition for a SIP call that is identified by both the Served Trunk Group and Serving 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 Hunt Group Settings table or Tel to IP Routing table refer to Tel to IP Routing Table on page 160 For IP to Tel calls the Serving IP Group is the Source IP Group ID defined in the IP to Hunt Group Routing table refer to IP to Hunt Group Routing on page 163 Note If no match is found in this table for incoming or outgoing calls the username and password defined in the Authentication table refer to Authentication on page 174 or the global parameters UserName and Password defined on the Proxy amp Registration page refer to Proxy Registration Parameters on page 112 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 Y
294. er s Manual MediaPack Series Description Defines the duration in milliseconds for which the device waits for a disconnection from the Tel side after the Blind Transfer Code KeyBlindTransfer has been identified When this timer expires a SIP REFER message is sent toward the IP side If this parameter is set to 0 the REFER message is immediately sent The valid range is 0 to 1 000 000 The default is 0 This ini file table parameter determines whether the device rejects incoming anonymous calls on FXS interfaces The format of this parameter is as follows RejectAnonymousCallPerPort FORMAT RejectAnonymousCallPerPort_Index RejectAnonymousCallPerPort_Enable RejectAnonymousCallPerPort Where Enable accept 0 default or reject 1 incoming anonymous calls For example RejectAnonymousCallPerPort RejectAnonymousCallPerPort 0 0 RejectAnonymousCallPerPort 1 1 RejectAnonymousCallPerPort If enabled when a device s FXS interface receives an anonymous call it responds with a 433 Anonymity Disallowed SIP response Notes This parameter is applicable only to FXS interfaces This parameter is per device This parameter can appear up to 8 times for 8 port MP 11x devices and up to 24 times for MP 124 devices The double dollar symbol represents the default value Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a desc
295. erence feature is used 0 Conference initiating INVITE sent by the device uses the ConferencelD concatenated with a unique identifier as the Request UR default 1 Conference initiating INVITE sent by the device uses only the ConferencelD as the Reques URI If 3wayConferenceMode is set to 0 the Conference initiating INVITE sent by the device uses the ConferencelD concatenated with a unique identifier as the Request URI This same Request URI is set as the Refer To header value in the REFER messages that are sent to the two remote parties If 3wayConferenceMode is set to 1 the Conference initiating INVITE sent by the device only uses the ConferencelD as the Reques URI The media server sets the Contact header of the 200 OK response to the actual unique identifier Conference URI to be used by the participants This Conference URI is included 269 November 2008 ca AudioCodes Parameter Enable3WayConference ConferenceCode ConferencelD Send180ForCallWaiting HookFlashCode UseSIPURIForDiversionHeade r FXOAutoDialPlayBusyTone EnableComfortTone WarningToneDuration FirstCallWaitingTonelD SIP User s Manual MediaPack Series Description by the device in the Refer To header value in the REFER messages sent by the device to the remote parties The remote parties join the conference by sending INVITE messages to the media server using this conference URI For a description of this parameter
296. erminated 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 225 November 2008 7a K tal AudioCodes MediaPack Series 3 6 2 2 Call Routing Statu
297. erver s address used for retrieving all STUN servers with an SRV query The STUN client can perform the required 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 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
298. es 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 Per Account in the Hunt Group Settings table refer to Configuring the Hunt Group Settings on page 183 for the specific Hunt 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 189 November 2008 A K tal AudioCodes MediaPack Series Parameter Description REGISTER sip audiocodes SIP 2 0 Via SIP 2 0 UDP 10 33 37 78 branch z9hG4bKac13 97582418 From lt sip ContactUsereHostName 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 MP 118 FXS_FXO v 5 40A 008 002 Content Length 0 Notes The Trunk Group account registration is not effected by the parameter IsRegisterNeeded f registration to an IP Group s fails for all the accounts defined in this table for a specific Hunt Group and if this Group includes all the channels in the the Hunt Group is set to Out Of Service if the parameter OOSOnRegistrationFail is set to 1 refer to Proxy amp Registration Parameters on page 112 Contact User Defines the AOR use
299. es such as ringback tone or other network announcements 1 Not Configured Default values are used The default for FXO interfaces is 1 The default for FXS interfaces is 0 Determines whether the Busy Out feature is enabled 0 Disable Busy out feature is not used default 1 Enable Busy ou feature is enabled When Busy Out is enabled and certain scenarios exist the device performs the following A reorder tone determined by FXSOOSBehavior is played when the phone is off hooked These behaviors are 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 Notes The FXSOOSBehavior parameter controls the behavior of the FXS endpoints when a Busy Out or Graceful Lock occurs FXO endpoints during Busy Out and Lock are inactive Refer to the LifeLineType parameter for complementary optional behavior 135 November 2008 ca AudioCodes Parameter Default Release Cause DefaultReleaseCause Dela
300. es MediaPack Series 3 4 4 2 1 Advanced Parameters 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 Configuration menu gt SIP Advanced Parameters submenu gt Advanced Parameters page item Figure 3 62 Advanced Parameters Page v General IP Security Disable Filter Calls to IP Dorit Fiter W Enable Digit Delivery to Tel Disable Enable Digit Delivery to IP Disable RTP Only Mode Disable Enable DID Wink Disable Delay Before DIO Wink 0 Reanswer Time 0 PSTN Alert Timeout 180 Disconnect and Answer Supervision Send Digit Pattern on Connect Enable Polarity Reversal Disable Enable Current Disconnect Disable Disconnect on Broken Connection Yes Broken Connection Timeout 100 msec 100 Disconnect Call on Silence Detection No Silence Detection Period sec 120 Silence Detection Method Voice Energy Detectors Enable Fax Re Routing Disable COR and Debug CDR Server IP Address COR Report Level Debug Level Misc Parameters Progress Indicator to IP Enable Busy Out Default Release Cause Max Number of Active Calls Max Call Duration min Enable LAN Watchdog Enable Calls Cut Through Enable User Information Usage Out Of Service Behavior Delay After Reset sec Emergency Calls Emergency Numbers Emergency Ca
301. es 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 249 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 a RADIUS server refer to the Product Reference Manual For secured HTTP connection HTTPS refer to the Product Reference Manual 3 4 3 2 Configuring the Web and Telnet Access List The Web 8 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 WebAccessLi
302. es 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 276 Document LTRT 65411 SIP User s Manual 4 ini File Configuration Parameter Description TelProfile InputGain TelProfile VoiceVolume TelProfile EnableReversePolarity TelProfile EnableCurrentDisconnect TelProfile EnableDigitDelivery TelProfile EnableEC TelProfile MWIlAnalog TelProfile MWIDisplay TelProfile FlashHookPeriod TelProfile EnableEarlyMedia TelProfile ProgressIndicator2IP TelProfile TimeForReorderTone TelProfile EnableDIDWink TelProfile IsTwoStageDial TelProfile DisconnectOnBusyTone TelProfile Indicates common 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 0 0 0 TelProfile
303. esktop wall or in a 19 inch rack The device provides a variety of management and provisioning tools including an HTTP based embedded Web server Telnet and Simple Network Management Protocol SNMP The user friendly Web interface provides remote configuration using any standard Web browser such as Microsoft Internet Explorer Version 5 6 15 November 2008 A e AudioCodes MediaPack Series The figure below illustrates a typical MediaPack VoIP application Figure 1 1 Typical MediaPack VoIP Application Phone S PSTN M s FAX MediaPack MediaPack FXO FXS Router MediaPack M FXS 4 1 2 MediaPack Features This section provides a high level overview of some of the many device supported features For more updated information on the device s supported features refer to the latest MP 11x amp MP 124 SIP Release Notes 1 2 1 MP 11x Hardware Features The MP 11x series hardware features include the following m Combined FXS FXO devices four FXS and four FXO ports on the MP 118 two FXS and two FXO ports on the MP 114 MP 11x compact rugged enclosure only one half of a 19 inch rack unit 1 U high m Lifeline provides a wired phone connection to the PSTN line that becomes active upon a power or network failure combined FXS FXO devices provide a Lifeline connection that s available on all FXS ports m LEDs on the front panel that provide information on the device s operating status a
304. esponse 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 SIP Reason Code ISDN Redirecting Reason 302 Moved Temporarily Call Forward Universal 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 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 IP Display Name remains empty default 1 Yes 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
305. essed 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 VolP 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 terms are used throughout this manual Hexadecimal notation is indicated by Ox preceding the number Version 5 6 13 November 2008 7a wi AudioCodes Related Documentation Document LTRT 523xx where xx is the document version MediaPack Series Manual Name Product Reference Manual LTRT 656xx LTRT 598xx LTRT 529xx LTRT 532xx LTRT 665xx MP 1
306. esses 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 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 Determines whether Caller ID is permitted Not Configured privacy is determined according to the Caller ID table refer to Caller ID on page 177 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 Tel to IP source number manipulation If Presentation is set to Restricted and Asserted Identity
307. eters 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 8 295 November 2008 K tal AudioCodes MediaPack Series Parameter SecureCallsFromliP AltRouteCauseTel2IP AltRouteCauselP2Tel SIP User s Manual Description 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 151 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a description of this parameter refer to Advanced Parameters on page 129 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 WitRouteCauseTel2IP For example Alt
308. evice are listed in the table below Coder Name G 711 A law g711Alaw64k G 711 U law g711Ulaw64k G 729 9729 G 723 1 97231 G 726 9726 Transparent Transparent G 711A law VBD g711AlawVbd G 711U law VBD g711UlawVbd T 38 t38fax gt To configure the device s coders take these 9 steps Table 3 30 Supported Coders Packetization Time 10 20 default 30 40 50 60 80 100 120 10 20 default 30 40 50 60 80 100 120 10 20 default 30 40 50 60 80 100 30 default 60 90 10 20 default 30 40 50 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 Always 64 Always 64 Always 8 5 3 0 6 3 1 default 16 0 24 1 32 2 default 40 3 Always 64 Always 64 Always 64 N A Payload Type Always 8 Always 0 Always 18 Always 4 Dynamic 0 120 Dynamic 0 120 Dynamic 0 120 Dynamic 0 120 N A MediaPack Series Silence Suppression Disable 0 Enable 1 Disable 0 Enable 1 Disable 0 Enable 1 Enable w o Adaptations 2 Disable 0 Enable 1 Disable 0 Enable 1 Disable 0 Enable 1 N A N A N A 1 Open the Coders page Configuration tab gt Protocol Configuration menu gt Protocol Definition submenu gt Coders page item Coder Name
309. ew 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 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 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 187 November 2008 K tal AudioCodes MediaPack Series Parameter Description This parameter is disregarded if the parameter AlwaysSendToProxy is set to 1 Always Use Route Determines the Request URI host name in outgoing INVITE messages Table 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 160 as the Reguest URI host name in outgoing INVITE messages instead of the value entered in the SIP Group Name field 3 4 4 8 3 Configuring the Account Table The Account Table page allows you to define accounts per Hunt Groups referr
310. ew of the Web related parameters Files CallProgressTonesFileName cpusa dat SIP User s Manual 232 Document LTRT 65411 SIP User s Manual 4 ini File Configuration 4 23 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 configured 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 devi
311. eypad Features on refer to Keypad Features on refer to Keypad Features on refer to Keypad Features on refer to Keypad Features on refer to Keypad Features on refer to Keypad Features on refer to Keypad Features on refer to Keypad Features on refer to Keypad Features on 0 Flash hook default only the phone s Flash button is used according to the following scenarios 1 During an existing call if the user presses Flash the call is put on hold a dial tone is heard and the user is able to initiate a second call Once the second call is established on hooking transfers the first held call to the second call 2 During an existing call if a call comes in call waiting pressing Flash places the active call on hold and answers the waiting call pressing Flash again toggles between these two calls 1 Flash hook digit a sequence of Flash 1 holds a call or toggles between two existing calls Flash 2 makes a call transfer Flash keys sequence timeout the time the device waits for digits after the user presses the Flash Hook button Flash Hook Digit mode when the parameter FlashKeysSequenceStyle is set to 1 285 November 2008 ca AudioCodes Parameter BlindTransferDisconnectTim eout RejectAnonymousCallPerPor t IsTwoStageDial IsWaitForDialTone FXOBetweenRingTime RingsBeforeCallerlD DisconnectOnDialTone GuardTimeBetweenCalls NTTDIDSignallingForm SIP Us
312. f 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 Version 5 6 333 November 2008 7 7 A e AudioCodes MediaPack Series RTP Multiplexing must be enabled on both devices When VLANS are imlemented the RTP Multiplexing mechanism is not supported 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 guality 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 processor in evenly spaced intervals The device uses a dynamic jitter buffer that can be configured using the following two parameters m Minimum delay DJBufMinDelay 0 msec to 150 msec Defines the starting jitter capacity of the buffer For example
313. fault 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 ld headers value 0 Disable default 110 Document LTRT 65411 SIP User s Manual 3 Web Based Management Parameter Description 1 Enable Note P Associated URIs in registration responses is handled only if the device is registered per endpoint Source Number Determines the SIP header used to determine the Source Number in Preference incoming INVITE messages SourceNumberPreferen empty string Use device s internal logic for header preference ce default FROM Use the Source Number received in the From header The valid range is a string of up to 10 characters The default is an empty string Forking Handling Mode Determines how the device reacts to forking of outgoing INVITE ForkingHandlingMode 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 ForkingHandlingM
314. fer to Message Waiting Indication on page 361 m Caller ID refer to Caller ID on page 362 To activate these supplementary services Hold Transfer Forward Waiting and MWI on the device enable each service s corresponding parameter either from the Web interface or via the ini file All call participants must support the specific supplementary service that is used When working with certain application servers such as BroadSof 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 7 14 1 Call Hold and Retrieve Initiating Call Hold and Retrieve m Active calls can be put on hold by pressing the phone s hook flash button m The party that initiates the hold is called the holding party the other party is called the held party m After a successful Hold the holding party hears a Dial tone HELD TONE defined in the device s Call Progress Tones file m Call retrieve can be performed only by the holding party while the call is held and active m The holding party performs the retrieve by pressing the telephone s hook flash button After a successful retrieve the voice is connected again m Hold is performed by sending a Re INVITE message with IP address 0 0 0 0 or SIP User s Manual 356 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities a sendonly in the SDP ac
315. fer to Configuring the IP Settings on page 52 LocalOAMDefaultGW For a description of this parameter refer to Configuring the IP Settings on page 52 Multiple Interface Table 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 Interface Table IPv6InterfaceMode InterfaceTable IPAddress Interface Table PrefixLength InterfaceTable Gateway Interface Table VlanlD InterfaceTable InterfaceName InterfaceTable 0 6 0 192 168 85 14 16 192 168 0 1 1 myAll Mnterface Table For example InterfaceTable FORMAT InterfaceTable Index InterfaceTable ApplicationTypes Interface Table IPv6InterfaceMode InterfaceTable IPAddress Interface Table PrefixLength InterfaceTable Gateway Interface Table VlanlD InterfaceTable InterfaceName Interface Table 0 0 0 192 168 85 14 16 0 0 0 0 1 ManagementlF 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 Mnterface Table The above example configures three network interfaces OAMP Control and Media applications InterfaceTable Notes To configure the Multiple Interface table using the Web interface refer to Configuring the Multiple Interface Table on page 55 Fora description of confi
316. fic Network Types and Priority Class of Service Priority Bronze Bronze Network Bronze Bronze Bronze Determined by the service Premium media Premium media Premium media Premium control Premium control Bronze Determined by the initiator of the request Network Bronze Network Depends on traffic type Control Premium control Management Bronze Gold Document LTRT 65411 SIP User s Manual 8 Networking Capabilities 8 8 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 373 and ini file refer to Integrating Using the ini File on page 375 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 P Address jjask IP Address VD Rule OAMP 10 31 174 50 255 255 0 0 0 0 0 0 4 83 4 87 X Control 10 32 174 50 255 255 0 0 0 0 0 0 5 130 33 4 6 Media 10 33 174 50 255 255 0 0 10 33 01 6 The values provided in this section are only used as an example Since a default Gateway is available only for the Media network for the device to
317. fic on the device before initiating the wizard by performing a graceful lock refer to Locking and Unlocking the Device on page 208 SIP User s Manual 212 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 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 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 Software Upgrade Wizard Start Software Upgrade Click the button to start the software upgrade process 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 Version 5 6 213 November 2008 7
318. figuration 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 207 2 Under the Save Configuration group click the BURN button a confirmation message appears when the configuration successfully saves Version 5 6 209 November 2008 A c tal AudioCodes MediaPack Series Saving configuration to the non volatile memory may disrupt traffic on 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 208 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 207 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 m Load Auxiliary Files refer to Loading Auxiliary Files on page 210 m Software Upgrade Wizard refer to Software Upgrade Wizard on page 212 m Configuration File refer to Backing Up and Restoring Configurati
319. figuration tab gt Advanced Applications menu gt RADIUS Parameters page item Figure 3 90 RADIUS Parameters Page Enable RADIUS Access Control Disable v Accounting Server IP Address 0 0 0 0 Accounting Port 1646 RADIUS Accounting Type At Call Release Indications None 2 Configure the RADIUS accounting 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 209 SIP User s Manual 194 Document LTRT 65411 SIP User s Manual 3 Web Based Management Table 3 50 RADIUS Parameters Description Parameter Description Enable RADIUS Access Enables or disables the RADIUS application Control o o EnableRADIUS 0 Disable disables RADIUS application default 1 Enable enables RADIUS application Accounting Server IP IP address of the RADIUS accounting server Address RADIUSAccServerlP Accounting Port Port of the RADIUS accounting server RADIUSAccPort The default value is 1646 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 an
320. file refer to Configuring the Call Progress Tones File in the Product Reference Manual If the MWI display is configured the number of waiting messages is also displayed If the MWI lamp is configured the phone s lamp on a phone that is equipped with an MWI lamp is lit The device can subscribe to the MWI server per port usually used on FXS or per device used on FXO Version 5 6 361 November 2008 7a c tal AudioCodes MediaPack Series To configure MWI set the following parameters EnableMWI or using the Web interface refer to Supplementary Services on page 138 MWIServerlP or using the Web interface refer to Supplementary Services on page 138 MW1AnalogLamp or using the Web interface refer to Supplementary Services on page 138 MW Display or using the Web interface refer to Supplementary Services on page 138 StutterToneDuration or using the Web interface refer to Supplementary Services on page 138 EnableMW Subscription or using the Web interface refer to Supplementary Services on page 138 MWIExpirationTime or using the Web interface refer to Supplementary Services on page 138 SubscribeRetryTime or using the Web interface refer to Supplementary Services on page 138 SubscriptionMode or using the Web interface refer to Proxy amp Registration Parameters on page 112 CallerlDType determines the standard for detection of MWI signals or using the Web interface refer t
321. file ID Profile Name w Profile Parameters Profile Preference Fax 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 Enable Polarity Reversal Disable Enable Current Disconnect Disable MWI Analog Lamp Disable MWI Display Disable Echo Canceler Enable Flash Hook Period 700 Enable Early Media Disable Progress Indicator to IP Not Configured Time For Reorder Tone sec 255 Enable DID Wink Disable Dialing Mode Two Stages Disconnect Call on Detection of Busy Tone Enable wv Coder Group Coder Group Default Coder Group 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 T
322. 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 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 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 FXS coefficients file and the location of the server IP address or FQDN from where it is loaded For example http server_name file https server_nameffile 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_nam
323. for the destination phone prefix and 10 1 10 2 for the IP address of the FXO device Figure 7 15 Tel to IP Routing Configuration Dest Phone Prefix Source Phone Prefix Dest IP Address 1 20 Note For the transfer to function in remote PBX extensions Hold must be disabled at the FXS device i e Enable Hold 0 and hook flash must be transferred from the FXS to the FXO HookFlashOption 4 7 13 4 6 FXO Gateway Configuration The procedure below describes how to configure the FXO device to which the PBX is directly connected gt To configure the FXO device take these 4 steps 1 In the Endpoint Phone Numbers page assign the phone numbers 200 to 207 to the device s FXO endpoints Figure 7 16 Assigning Phone Numbers to FXO Ports Channelis Phone Number Hunt Group ID 1 8 200 Version 5 6 351 November 2008 7a c tal AudioCodes MediaPack Series 2 In the Automatic Dialing page enter the phone numbers of the FXS device in the Destination Phone Number fields When a ringing signal is detected at Port 1 the FXO device automatically dials the number 100 Figure 7 17 Automatic Dialing Configuration Gateway Destination Phone Auto Dial Port Number Status Port 1 FXO Enable Port 2 FXO Enable Port 3 FXO Enable Port 4 FXO Enable Port 5 FXO Enable Port 6
324. g the number of busy endpoints by the total number of enabled endpoints 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 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 FarEndDisconnectSilence Period FarEndDisconnectSilence Method FarEndDisconnectSilence Threshold For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 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 a
325. g 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 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 41 QoS Settings Page w Priority Settings Network Priority Media Premium Priority Control Premium Priority Gold Priority t Bronze Priority wv Differential Services lS Network QoS J Media Premium QoS Control Premium QoS 1 Gold QoS lS 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 209 Version 5 6 65 November 2008 ca AudioCodes MediaPack Series Table 3 12 QoS Settings Parameters Parameter Priority Settings Network Priority VLANNetworkServiceClassPriority Media Premium Priority VLANPremiumServiceClassMediaPriority 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 f
326. ge 101 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 8 Registration Parameters on page 112 For a description of this parameter refer to SIP General Parameters on page 101 Enables or disables the use of the rtcp attribute in the outgoing SDP 0 Disable 1 Enable default Defines the User Part value of the Request URI for outgoing SIP OPTIONS requests If no value is configured the endpoint number is used 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 262 Document LTRT 65411 SIP User s Manual Parameter UseGatewayNameForOptions IsProxyHotSwap HotSwapRtx ProxyRedundancyMode ProxyLoadBalancingMethod ProxylPListRefreshTime IsFallbackUsed UserName Password Cnonce SIPChallengeCachingMode MutualAuthenticationMode IsRegisterNeeded RegistrariP RegistrarTransportType RegistrarName GWRegistrationName AuthenticationMode OOSOnRegistrationFail RegistrationTime RegistrationTimeDivider RegistrationRetryTime Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Proxy 8 Registration P
327. ge 305 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 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 6 231 November 2008 A L e AudioCodes MediaPack Series 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 232 m Table parameters which include multiple individual parameters refer to Structure of ini File Table Parameters on page 233 Structure Rules The ini file must adhere to the following format rules E The ini file name
328. ge of ports using the format n m where n represents the lower port number and m the higher port number For example 1 4 specifies ports 1 through 4 Phone Number The telephone number that is assigned to the channel For a range of channels only enter the first telephone number Subsequent channels are assigned the next consecutive telephone number For example if you enter 400 for channels 1 to 4 then channel 1 is assigned phone number 400 channel 2 is assigned phone number 401 and so on These numbers are also used for port allocation for IP to Tel calls if the Hunt Group s Channel Select Mode is set to By Phone Number Note If the 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 user1 Hunt Group ID The Hunt Group ID 1 99 optionally assigned to the channel s The same Hunt Group ID can be assigned to multiple channels The Hunt Group ID defines a group of common channel behavior for routing IP to Tel calls If an IP to Tel call is assigned to a Hunt Group the call is routed to the channel s that are assigned to the same Hunt Group ID You can also configure the Hunt Group Settings table refer to Configuring the Trunk Group Settings on page 183 to determine the method in which new calls are assigned to channels within the Hunt Groups Note If you enter a Hunt Group ID you must confi
329. gitPatternForwardNoReas onExt Internal Call Digit Pattern DigitPatternInternalCall External Call Digit Pattern DigitPatternExternalCall Disconnect Call Digit Pattern TelDisconnectCode Digit To Ignore Digit Pattern DigitPatternDigitTolgnore 3 Web Based Management 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 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 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 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 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 digit pattern used by the PBX to indicate call forward with no reason when the original c
330. 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 Determines 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 Notes This parameter is applicable only to FXS interfaces To enable the device to send a 183 180 SDP responses set EnableEarlyMedia to 1 f 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 0 Don t 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 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 Hunt Group to which the call belongs is used according to RFC 4904 For example INVITE sip 163055501 00 tgr
331. gular SAS Normal Emergency logic same as option 0 but when in Normal mode incoming REGISTER requests are ignored 273 November 2008 ca AudioCodes Parameter SASBindingMode SASEnableENUM SASRegistrationManipulation SASEmergencyNumbers Profile Parameters CoderName SIP User s Manual MediaPack Series Description 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 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 d
332. gure below illustrates the example setup SIP User s Manual 352 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities Figure 7 19 SIP Trunking Example Architecture Proxy Set 1 P 10 33 37 77 IP 10 33 37 79 ITSP 1 IP Group 1 POTS Phones i Hunt Group ID 1 AudioCodes MediaPack Hunt LA Group ID 2 POTS Phones ITSP 2 IP Group 2 Proxy Set 2 IP 10 8 8 40 IP 10 8 8 10 X PSTN Network gt To configure call routing between Enterprise and two ITSPs using the device take these 9 steps 1 Enable the device to register to a Proxy Registrar server using the parameter IsRegisterNeeded in the Proxy 8 Registration page refer to Proxy amp Registration Parameters on page 112 2 Inthe Proxy Sets Table page refer to Proxy Sets Table on page 120 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 Version 5 6 353 November 2008 7a K tal AudioCodes MediaPack Series 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 similar configuration for Proxy Set ID 2 but using different IP addresses Figure 7 20 Configuring Proxy Set ID 1 in the
333. gure the IP to Hunt Group Routing Table page refer to IP to Hunt Group Routing which assigns incoming IP calls to the appropriate Hunt Group If you do not configure the IP to Hunt Group Routing Table calls are not established Profile ID The Tel Profile ID refer to Tel Profile Settings on page 171 assigned to the endpoint s SIP User s Manual 182 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 4 4 8 Configuring the Hunt and IP Groups The Hunt IP Group menu allows you to configure groups of channels This submenu includes the following page items m Hunt Group Settings refer to Configuring the Hunt Group Settings on page 183 m IP Group Table refer to Configuring the IP Groups on page 186 m Account Table refer to Configuring the Account Table on page 188 3 4 4 8 1 Configuring the Hunt Group Settings The Hunt Group Settings page is mainly used to select the method for which IP to Tel calls are assigned to channels within each Hunt Group If no method is selected for a specific Hunt Group the setting of the global parameter ChannelSelectMode in the SIP General Parameters page refer to SIP General Parameters on page 101 applies In addition this page also defines the method for registering Hunt Groups to selected Serving IP Group IDs if defined You can add up to entries in this table Note You can also configure the Hunt Group Settings table using the ini file table p
334. guring ini file table parameters refer to Structure of ini File Table Parameters on page 233 Differential Services For detailed information on IP QoS via Differentiated Services refer to IP QoS via Differentiated Services DiffServ on page 369 NetworkServiceClassDiffS For a description of this parameter refer to Configuring the QoS erv Settings on page 65 SIP User s Manual 242 Document LTRT 65411 SIP User s Manual Parameter PremiumServiceClassMed iaDiffServ PremiumServiceClassCon trolDiffServ GoldServiceClassDiffServ BronzeServiceClassDiffSe rv NFS Table Parameter NFSServers Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Configuring the QoS Settings on page 65 For a description of this parameter refer to Configuring the QoS Settings on page 65 For a description of this parameter refer to Configuring the QoS Settings on page 65 For a description of this parameter refer to Configuring the QoS Settings on page 65 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 NFSServe
335. hangeMethodIndex 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 IPSec is When an FQDN is used a DNS applied to server must be configured outgoing DNSPriServerIP packets Determines the local interface to which ae the encryption is applied applicable to f defined for multiple IPs and VLANs these 0 OAM OAMP interface default parameters 1 Control Control interface Defines the source port to which the IPSec mechanism is applied 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 O Any protocol default 17 UDP 6 TOP Any other protocol type defined by IANA Internet Assigned Numbers Authority Determines the index for the corresponding IKE entry Note th
336. hannel but not declared in the SDP Note The IsCiscoSCEMode parameter is only relevant when the 301 November 2008 ca AudioCodes Parameter EnableEchoCanceller ECNLPMode EchoCancellerAggressiveN LP EnableStandardSIDPayload Type ComfortNoiseNegotiation RTPSIDCoeffNum DTMFVolume DTMFGenerationTwist DTMFInterDigitInterval DTMFDigitLength RxDTMFHangOverTime TxDTMFHangOverTime DTMFTransportType AnswerDetectorSensitivity RFC2833PayloadType UserDefinedToneDetectorE nable SIP User s Manual MediaPack Series Description selected coder is G 729 For a description of this parameter refer to Configuring the Voice Settings on page 67 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 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 Determines the number of spectral coefficients added to an SID packet being sent according to RFC 3389 Valid only if EnableStandardSIDPayloadType 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 o
337. hargeCode Pulselnterval1 ChargeCode PulsesOnAnswer1 ChargeCode EndTime2 ChargeCode Pulselnterval2 ChargeCode PulsesOnAnswer2 ChargeCode EndTime3 ChargeCode Pulselnterval3 ChargeCode PulsesOnAnswer3 ChargeCode EndTime4 ChargeCode Pulselnterval4 ChargeCode_PulsesOnAnswer4 ChargeCode Where EndTime Period 1 4 end time Pulselnterval Period 1 4 pulse interval PulsesOnAnswer Period 1 4 pulses on answer For example ChargeCode ChargeCode 1 7 30 1 14 20 2 20 15 1 0 60 1 ChargeCode 2 5 60 1 14 20 1 0 60 1 ChargeCode 3 0 60 1 ChargeCode 0 6 3 1 12 2 1 18 5 2 0 2 1 ChargeCode SIP User s Manual 280 Document LTRT 65411 SIP User s Manual Parameter TargetOfChannel Version 5 6 4 ini File Configuration Description Notes The parameter can appear up to 25 times i e up to 25 different metering rules can be defined To configure the Charge Codes table using the Web interface refer to Charge Codes Table For an explanation on configuration using ini file table parameters refer to Structure of ini File Table Parameters on page 233 This ini file table parameter defines telephone numbers that are automatically dialed when a specific port is used The format of this parameter is as follows TargetOfChannel FORMAT TargetOfChannel_Index TargetOfChannel_Destination TargetOfChannel_Type TargetOfChannel Port TargetOfChannel_ Modul
338. he TCP or UDP protocols and the appropriate port numbers as defined on the device 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 85 November 2008 7a K tal AudioCodes MediaPack Series Parameter Description Action Upon Match AccessList Allow Type Action upon match i e Allow or Block Match Count A read only field providing the number of packets accepted rejected AccessList MatchCount by the specific rule 3 4 3 4 Configuring the Certificates The Certificates page is used for the following m Replacing the server certificate refer to Server Certificate Replacement on page 86 m Replacing the client certificates refer to Client Certificates on page 88 m Regenerating Self Signed Certificates refer to Self Signed Certificates on page 89 Updating the private key using HTTPSPkeyFileName as described in the Product Reference Manual 3 4 3 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 a
339. he Web interface is accessed displaying the Home page for a detailed description of the Home page refer to Using the Home Page on page 48 Note If access to the device s Web interface is denied Unauthorized due to Microsoft Internet Explorer security settings perform the following 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 22 Document LTRT 65411 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 O tak 424 8 hetp 20 13 4 13 Title Bar 3 A lod vice v Submit Bun Device Actions 4 Home Hen a pe A Status Cortguaton Management 7 Disgnostice S nanos Seach Bore PoranoterUst a v Syslog Setti
340. he device or endpoints are registered to a SIP Registrar Proxy server gt To view Registration status take this step m Open the Registration Status page Status amp Diagnostics tab gt Gateway Statistics menu gt Registration Status page item Figure 3 116 Registration Status Page Registered Per Gateway NO w Status Gateway Port Status Port1 FXS NOT REGISTERED Port 2 FXS NOT REGISTERED Port 3 FXS NOT REGISTERED Port4 FXS NOT REGISTERED Port5 FXO NOT REGISTERED Port6 FKO NOT REGISTERED Port 7 FXO NOT REGISTERED Ports FXO NOT REGISTERED If a channel is registered then REGISTERED is displayed in the Status column corresponding to the channel otherwise NOT REGISTERED is displayed If registration is per device then YES is displayed alongside Registered Per Gateway otherwise NO is displayed 3 6 2 4 SAS SBC Registered Users The SAS Registered Users page displays a list of up to 25 Stand Alone Survivability SAS registered users The SAS feature is configured in the SAS Configuration page refer to Stand Alone Survivability on page 149 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 3 117 SAS Registered Users Page Address Of Record Contact lt sip 2400 Proxies
341. he 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 IsCiscoSCEMode 3 4 4 1 5 DTMF amp Dialing Parameters The DTMF 8 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 8 Dialing page Configuration tab gt Protocol Configuration menu gt Protocol Definition submenu gt DTMF amp Dialing page item Version 5 6 125 November 2008 ca AudioCodes Figure 3 61 DTMF amp Dialing Page MediaPack Series Max Digits In Phone Num 30 Inter Digit Timeout for Overlap Dialing sec 4 Declare RFC 2833 in SDP No ist Tx DTMF Option RFC 2833 2nd Tx DTMF Option 3rd Tx DTMF Option 4th Tx DTMF Option 5th Tx DTMF Option RFC 2833 Payload Type 96 Hook Flash Option Not Supported Digit Mapping Rules Di
342. his 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 duplex mode and speed You can also access this page from the Home page refer to Using the Home Page on page 48 For detailed information on the Ethernet redundancy scheme refer to Ethernet Interface Redundancy For detailed information on the Ethernet interface configuration refer to Ethernet Interface Configuration on page 365 gt To view Ethernet port information take the following step m 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 Port 1 Duplex Mode Port 1 Speed Table 3 57 Ethernet Port Information Parameters Parameter Description 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 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 55 and that are currently activ
343. hold 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 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 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 116 Document LTRT 65411 SIP User s Manual Parameter Gateway Registration Name GWRegistrationName DNS Query Type DNSQueryType Proxy DNS Query
344. ical 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 m 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 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 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
345. ick the LOCK button a confirmation message box appears requesting you to confirm device Lock Figure 3 100 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 EE 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 current 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 207 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 con
346. ied to each routing rule to associate it with an entry in the Charge Code table refer to Charge Codes Table on page 146 163 November 2008 tall AudioCodes MediaPack Series 3 4 4 4 3 IP to Trunk Group Routing Table The IP to Hunt Group Routing Table page provides a table for routing incoming IP calls to groups of channels FXS FXO endpoints called Hunt Groups Hunt Group ID s are assigned to the device s channels in the Endpoint Phone Number page refer to Configuring the Endpoint Phone Numbers on page 181 You can add up to 24 IP to Hunt Group routing rules in the table The IP to Hunt Group Routing Table page appears only if the parameter EnableSBC is set to 0 default in SBC Configuration If this parameter is enabled the Inbound IP Routing Table page appears instead refer to Inbound IP Routing Table for a description of this page The IP to Tel calls are routed to Hunt 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 Hunt Group the call is sent to the device s channels pertaining to that Hunt Group The specific channel within the Hunt Group to which the call is sent is determined according to the Hunt Group s channel selection mode This channel selection mode can be defined per Hunt Group refer to Configuring the Trun
347. ils the device sends information on the test results of each hardware component to the Syslog server 0 Rapid and Enhanced self test mode 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 0 Disable device s watch dog 1 Enable device s watch dog default Defines the Lifeline phone type The Lifeline phone is available on port 1 of MP 11x FXS devices and on ports 1 4 of the MP 118 FXS FXO devices The Lifeline is activated upon one of the following options 0 Power down default 1 Power down or when link is down physical disconnect 2 Power down or when link is down or on network failure logical link disconnect Notes To enable Lifeline switching on network failure LAN watch dog must be activated EnableLANWatchDog 1 This parameter is only applicable to FXS interface For a description of this parameter refer to Advanced Parameters on page 129 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 A
348. imal notation or FQDN If an FQDN is used DNS resolution is performed according to DNSQueryType If the 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 155 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 This ini file table parameter configures the routing of IP to Tel calls to Hunt Groups The format of this parameter is as follows PSTNPrefix FORMAT PstnPrefix Index PstnPrefix DestPrefix PstnPrefix TrunkGroupld PstnPrefix SourcePrefix PstnPrefix SourceAddress PstnPrefix Profileld PstnPrefix SrcIPGroupID PstnPrefix DestHostPrefix PstnPrefix SrcHostPrefix PSTNPrefix For example PSTNPrefix FORMAT PstnPrefix Index PstnPrefix_DestPrefix PstnPrefix TrunkGroupld PstnPrefix SourcePrefix PstnPrefix SourceAddress PstnPrefix Profileld PstnPrefix_SrclPGroupID PstnPrefix De
349. in control in decibels This parameter sets the level for the transmitted IP to Tel signal The valid range is 32 to 31 dB The default value is 0 dB Pulse code modulation PCM input gain control in decibels This parameter sets the level for the received Tel to IP signal The valid range is 32 to 31 dB The default value is 0 dB Silence Suppression is a method for conserving bandwidth on VoIP 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 If EnableSilenceCompression is 0 annexb no If EnableSilenceCompression is 1 annexb yes If EnableSilenceCompression is 2 and IsCiscoSCEMode is 0 annexb yes 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 Tran
350. incoming INVITE is determined classified using the Version 5 6 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 for Tel gt IP Calls page If this Source IP Group has a Serving IP Group then all calls 153 November 2008 ca AudioCodes Parameter Destination Prefix DestinationPrefix Source Prefix SourcePrefix Source IP SourceAddress Stripped Digits From Left L RemoveFromLetft Stripped Digits From Right RemoveFromRight Prefix to Add Prefix2Add Suffix to Add Suffix2Add Number of Digits to Leave LeaveFromRight Presentation IsPresentationRestricted SIP User s Manual MediaPack Series Description 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 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 addr
351. ing 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 server 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 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 115 Novem
352. 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 by not selecting an ini file and by clearing the Use existing configuration check box Version 5 6 215 November 2008 7a tal AudioCodes MediaPack Series Figure 3 105 Load an ini File Page p http 10 13 4 13 Software Update Wizard Microsoft Interne CMP file Load an ini fle from your computer to the device bas Send Fie CPT file Use existing configuration The Device will revert to default configuration if no ERTMS configuration is chosen FXO file FXS 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 SIP User s Manual 216 Document LTRT 65411 SIP
353. ion 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 8 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 m QoS The QoS of an IP connection is determined according to RTCP 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 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 8 3 Relevant Parameters The following parameters described in Routing General Parameters on page 157 are used to configure the Alternative Routing mechanism m AltRoutingTel2IPEnable m AltRoutingTe
354. ion 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 ptime 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 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 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 de
355. ion 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 method type is REGISTER e Using SIP protocol sip e Proxy IP from inifile 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 Version 5 6 345 November 2008 e AudioCodes MediaPack Series 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 NCIS joe 1022222 SIP 2 0 Via SIP 2 0 UDP 10 1 1 200 From lt sip 122 10 1 1 200 gt tag 1c23940 mos sEog 1220110112005 Call ID 654982194 10 1 1 200 Server Audiocodes Sip Gateway MediaPack v 5 40 010 006 CSeq 1 REGISTER Conltde Hi sHi prea 2 O0 2008 Expires 3600 Authorizatio
356. 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 Display Name is not empty 107 November 2008 ca AudioCodes Parameter Use Display Name as Source Number UseDisplayNameAsSou rceNumber Enable Contact Restriction EnableContactRestricti on Play Ringback Tone to IP PlayRBTone2IP Play Ringback Tone to Tel PlayRBTone2Tel Use Tgrp Information UseSIPTgrp SIP User s Manual MediaPack Series 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
357. is sent in a 200 OK response to an incoming REGISTER message when in SAS Emergency Mode The valid range is O 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 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 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
358. is shown below 11xS OOT 1 7 xxx 8xxXxxxxx XXXXXXX XX 9 1 XXXXXXXXXX 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 Duration in seconds that the dial tone is played FXS interface plays the dial tone after the phone is picked up off hook while FXO interface plays the dial tone after port is seized in response to ringing from PBX PSTN The default time is 16 Notes During play of dial tone the device waits for DTMF digits This parameter is not applicable when Automatic Dialing is enabled Duration in seconds of the Hotline dial tone If no digits are received during this duration the device initiates a call to a user defined number refer to Automatic Dialing on page 175 The valid range is 0 to 60 The default is 16 Note This parameter is applicable for both FXS and FXO interfaces 128 Document LTRT 65411 SIP User s Manual 3 Web Based Management Parameter Enable Special Digits IsSpecialDigits Default Destination Number DefaultNumber Special Digit Representation UseDigitForSpecialDTMF Description Determines whether the asterisk and pound digits can be used 0 Disable Use or to terminate number collection refer to the parameter UseDigitForSpecialDTMF Default 1 Enable Allows and for telephone numbers dialed
359. istration for Hunt Group ID 2 and associated it with IP Group 2 SIP User s Manual 354 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities Figure 7 23 Configuring Hunt Groups Settings Serving Hunt Registration IP Channel Select Mode Gateway Name Contact User Group ID Mode Group ID i cyotic Ascending Pe Account x 1 cycic Ascending V Per Endpoint v 12 v 6 In the Authentication page refer to Authentication on page 174 for channels 5 8 i e Hunt Group ID 2 define for each channel the registration authentication user name and password Figure 7 24 Configuring Username and Password for Channels 5 8 in Authentication Page Gateway Port User Name Password Port 1 FxS Port 2 FXS Port 3 FXS Port 4 FXS Port5 FXS Port 6 FXS Port FXS Port 8 FX5 7 In the Account Table page refer to Configuring the Account Table on page 188 configurea single Account for Hunt Group ID 1 including an authentication user name and password and enable registration for this Account to ITSP 1 i e Serving IP Group is 1 Figure 7 25 Configuring Accounts Index ServedTrunkGroup ServinglPGroup Username Password HostName Register ah 1 TSPluser 1234 rse1 h 8 In the IP to Hunt Group Routing page r
360. ith 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 Each 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 41 Version 5 6 35 November 2008 7a L tall AudioCodes MediaPack Series 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
361. its before forwarding the call to the phone number specified Port Port number Module Module number 0 5 For example FwdInfo Fwdinfo 1 1 1001 2 FwdInfo 2 1 2003 10 5 1 1 2 282 Document LTRT 65411 SIP User s Manual Parameter EnableCallerlD EnableDIDWink DelayBeforeDIDWink EnableReversalPolarity EnableCurrentDisconnect CutThrough Version 5 6 4 ini File Configuration Description Fwdlnfo 3 3 2005 30 2 Fwdlnfo Notes The indexing of this parameter starts at 1 The device ports starts at 0 This parameter can appear up to 24 times for MP 124 To configure the Call Forward table using the Web interface refer to Call Forward on page 178 For an explanation on ini file table parameters refer to Structure of ini File Table Parameters on page 233 This ini file table parameter configures Caller ID permissions The format of this parameter is as follows EnableCallerlD FORMAT EnableCallerl D Index EnableCallerlD IsEnabled EnableCallerlD Port EnableCallerlD Module EnableCallerlD Where IsEnabled Enables 1 or disables 0 default Caller ID Port Port number Module Module number 0 5 N A For example EnableCallerlD EnableCallerlD 1 1 3 EnableCalleriD 2 0 EnableCallerlD Notes The indexing of this ini file table parameter starts at 1 The numbering of ports starts at 0 fa port
362. ive 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 Based on Connectivity and QoS on page 335 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 335 To receive quality information displayed in the Quality Status and Quality Info fields in IP Connectivity on page 228 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
363. iveVlanId 4 Routing Table Configuration IP Routing table parameters RoutingTableDestinationsColumn 130 33 4 6 83 4 87 6 RoutingTableDestinationMasksColumn 255 255 255 255 255 9 235 455 0 RoutingTableGatewaysColumn 10 32 0 1 10 31 0 1 RoutingTableInterfacesColumn 2 0 RoutingTableHopsCountColumn 20 20 Class Of Service parameters VlanNetworkServiceClassPriority 7 VlanPremiumServiceClassMediaPriority VlanPremiumServiceClassControlPriority 6 VlanGoldServiceClassPriority 4 VlanBronzeServiceClassPriority 2 NetworkServiceClassDiffServ 48 PremiumServiceClassMediaDiffServ 46 PremiumServiceClassControlDiffServ 40 GoldServiceClassDiffServ 26 BronzeServiceClassDiffServ 10 Application Type for applications EnableDNSasOAM 1 EnableSCTPasControl 1 EnableNTPasOAM 1 a 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 SIP User s Manual 376 Document LTRT 65411 SIP User s Manual 8 Networking Capabilities Instead of using the ini file table parameter InterfaceTable you can configure multiple IPs and VLANs using the individual ini file parameters as shown below VLAN Configuration VlanMode 1 Vla
364. k Group Settings on page 183 or for all Hunt Groups using the global parameter ChannelSelectMode refer to SIP General Parameters on page 101 When a call release reason defined in Reasons for Alternative Routing on page 168 is received for a specific IP to Tel call an alternative Hunt 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 Hunt Group ID You can also configure the IP to Hunt Group Routing table using the ini file table parameter PSTNPrefix refer to Number Manipulation and Routing Parameters on page 289 gt To configure the IP to Hunt Group Routing table take these 5 steps 1 Open the IP to Hunt Group Routing page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt IP to Hunt Group Routing page item SIP User s Manual 164 Document LTRT 65411 SIP User s Manual 3 Web Based Management Figure 3 72 IP to Hunt Group Routing Page Matching Rules Tel Destination Rules 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 failure refer to Saving Configuration on page 209 Table 3 42 IP to Hunt Group Routing Table Description
365. k delay up to 9 seconds round trip Auto switch to PCM or ADPCM on V 34 or V 90 modem detection SIP RFC 3261 RTP RTCP packetization IP stack UDP TCP RTP Remote software upload TFTP HTTP and HTTPS Loop start signaling Motorola PowerQUICC 860 SDRAM 64 MB AudioCodes AC482 VolP DSP 385 November 2008 ca AudioCodes Function Interfaces FXS Telephony Interface Network Interface RS 232 Interface Indicators Connectors amp Switches Rear Panel 24 Analog Lines Ethernet RS 232 AC power supply socket Front Panel Reset Button Physical Enclosure Dimensions Weight Environmental Mounting Electrical Management Configuration Management and Maintenance Type Approvals Safety and EMC SIP User s Manual MediaPack Series Specification 24 Analog FXS phone or fax ports loop start RJ 11 10 100Base TX RS 232 Terminal Interface DB 9 Channel status and activity LEDs 50 pin Telco shielded connector 10 100Base TX RJ 45 shielded connector DB 9 console port 100 240 0 8A max Resets the MP 124 a 1U 19 inch rack Width x height x depth 445 mm 17 5 in x 44 5 mm 1 75 in x 269 mm 10 6 in 1 8 kg 4 Ib Operational 5 to 40 C 41 to 104 F Storage 25 to 70 C 77 to 158 F Humidity 10 to 90 non condensing Rack mount or desktop 100 240 VAC Nominal 50 60 Hz HTTP based Embedded Web Server Web browser or ini file SNMP v2c SNMP v3 Syslog RFC 3
366. ke these 4 steps Open the Routing General Parameters page Configuration tab gt Protocol Configuration menu gt Routing Tables submenu gt Routing General Parameters page item Figure 3 70 Routing General Parameters Page General Parameters Add Hunt Group ID as Prefix dd Trunk ID as Prefix Replace Empty Destination with B channel Phone Number Add NPI and TON to Called Number dd NPI and TON to Calling Number No IP to Tel Remove Routing Table Prefix No Source IP Address Input Enable Alt Routing Tel to IP Alt Routing Tel to IP Mode Alt Routing Tel to IP Connectivity Method Alt Routing Tel to IP Keep Alive Time 60 Alternative Routing Tone Duration ms 0 Max Allowed Packet Loss for Alt Routing 20 JE IEA SIP Contact Header Disable Both ICMP Fing L Max Allowed Delay for Alt Routing msec 250 Version 5 6 157 November 2008 A K tal AudioCodes MediaPack Series 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 209 Table 3 40 Routing General Parameters Description Parameter Add Hunt Group ID as Prefix AddTrunkGroupAsPrefix Add Trunk ID as Prefix AddPortAsPrefix IP to Tel Remove Routing Table Prefix RemovePrefix Source IP Address Input Sour
367. ken 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 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 SIP General Parameters on page 101 Defines the port with relation to RTP port for sending and receiving T 38 packets 0 Use the RTP port 2 to send 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 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this para
368. king 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 IsFaxUsed 0 FaxTransportMode 2 V21ModemTransportType 2 V22ModemTransportType 2 V23ModemTransportType 2 V32ModemTransportType 2 V34ModemTransportType 2 BellModemTransportType 2 Additional configuration parameters e FaxModemBypassCoderType e FaxBypassPayloadType e ModemBypassPayloadType e FaxModemBypassBasicRTPPacketlnterval 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 Version 5 6 321 November 2008 7a K tal AudioCodes MediaPack Series
369. l 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 Table 7 2 Supported RADIUS Attributes Attribute Attribute VSA Value i Number Name No Psae Format anpe iis Request Attributes String Start 1 sseN m Account number or calling up to 15 5421385747 Acc party number or blank digits Stop long Acc Start 4 NSE Peddie Ohne Numeric 192 168 14 43 ASC Address requesting device Stop Acc Start Service Nb Acc 6 Type Type of service requested Numeric 1 login Stop Acc H323 Up to rhe 26 Incoming 1 SIP call identifier 32 Sto Conf Id octets p Acc H323 2 6 Ran oie 23 IP address of the remote Numeric Stop SIP User s Manual 336 Document LTRT 65411 SIP User s Manual Attribute Number 26 26 26 26 26 26 26 26 26 26 30 31 40 Version 5 6 Attribute Name Address H323 Conf ID H323 Setup Time H323 Call Origin H323 Call Type H323 Connect Time H323 Disconnect Time H323 Disconnect Cause H323 Gw ID SIP Call ID Call Terminator Called Station ID Calling Station ID Acct Status VSA No 24 25 26 27 28 29 30 33 34 35 Purpose gateway H 323 SIP call identifier Setup time in NTP format 1 The call s originator Answering IP or Originator PSTN Pr
370. l2IPMode Version 5 6 335 November 2008 A c tal AudioCodes MediaPack Series IPConnQoSMaxAllowedPL IPConnQoSMaxAllowedDelay 7 9 Mapping PSTN Release Cause to SIP Response The device s FXO interface interoperates between the SIP network and the PSTN PBX This interoperability includes the mapping of PSTN PBX Call Progress Tones to SIP 4xx or 5xx responses for IP to Tel calls The converse is also true for Tel to IP calls the SIP 4xx or 5xx responses are mapped to tones played to the PSTN PBX When establishing an IP to Tel call the following rules are applied m If the remote party PSTN PBX is busy and the FXO device detects a Busy tone it sends 486 Busy to IP If it detects a Reorder tone it sends 404 Not Found no route to destination to IP In both cases the call is released Note that if DisconnectOnBusyTone is set to 0 the FXO device ignores the detection of Busy Reorder tones and doesn t release the call m For all other FXS FXO release types caused when there are no free channels in the specific Hunt Group or when an appropriate rule for routing the call to a Hunt Group doesn t exist or if the phone number isn t found the device sends a SIP response to IP according to the parameter DefaultReleaseCause This parameter defines Q 931 release causes Its default value is 3 which is mapped to the SIP 404 response By changing its value to 34 the SIP 503 response is sent Other causes can be used as wel
371. larmSeverity 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 199 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 can 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 100 The default value is
372. ld Timeout Enable Transfer Transfer Prefix Enable Call Forward Enable Call Waiting Waiting Beep Duration Enable Caller ID Hook Flash Code Caller ID Type Call Hold Reminder Ring Timeout Number of Call Waiting Indications Time Between Call Waiting Indications 10 Time Before Waiting Indications 0 0 0 0 l Enable Enable v Enable v 2 pd 300 Disable Standard Bellcore v v Message Waiting Indication MWI Parameters Enable MWI MWI Analog Lamp MwT Display Subscribe to MWI MWI Server IP Address MWI Server Transport Type MWI Subscribe Expiration Time Stutter Tone Duration MWI Subscribe Retry Time Disable v Disable v Disable v No v Not Configured v 7200 2000 v Conference 4 Enable 3 Way Conference Establish Conference Code Conference ID w MLPP Call Priority Mode MLPP Diffserv Precedence Ringing Type 50 E 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 209 Version 5 6 139 November 2008 ca A
373. le 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 209 Parameter Is Rule Active Source IP AccessList Source IP Subnet Mask AccessList Net Mask Local Port Range AccessList Start Port AccessList End Port Protocol AccessList Protocol Packet Size AccessList Packet Size Byte Rate AccessList Byte Rate Burst Bytes AccessList Byte Burst Version 5 6 Table 3 21 Internal Firewall Parameters Description 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 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 The destination UDP TCP ports on this device to which packets are sent The valid range is 0 to 65535 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 protocol number in the range of 0 Any to 255 Note This field also accepts the abbreviated strings SIP and HTTP Specifying these strings implies selection of t
374. le table parameter IPProfile refer to SIP Configuration Parameters on page 260 gt To configure the IP Profile settings take these 9 steps 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 78 IP Profile Settings Page h Profile ID Profile Name wv Profile Parameters Profile Preference Fax Signaling Method Dynamic Jitter Buffer Minimum Delay msec Dynamic Jitter Buffer Optimization Factor RTP IP DiffServ Signaling DiffServ RTP Redundancy Depth Remote RTP Base UDP Port CNG Detector Mode Disable Modems Transport Type Enable Bypass NSE Mode Disable Play Ringback Tone to IP Don t Play Enable Early Media Disable Progress Indicator to IP Not Configured Echo Canceler Enable Media Security Behavior Preferable KCAL Number of Calls Limit 1 Copy Destination Number to Redirect Number Disable Disconnect on Broken Connection Yes wv Coder Group Coder Group Default Coder Group Version 5 6 173 November 2008 A c tal AudioCodes MediaPack Series From the Profile ID drop down list select an identification number for the IP Profile In the Profile Name field enter an arbitrary name that allows you to easily identify the IP Profile From the Profile Preference drop down list
375. leProxyKeepAlive 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 first Proxy Registrar server If there is no response from the first Proxy Registrar server after a specific number of retransmissions configured by the parameter HotSwapRtx 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 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 325 You can also configure the Coders table using the ini file table parameter CoderName refer to SIP Configuration Parameters on page 260 Version 5 6 123 November 2008 ca AudioCodes The coders supported by the d
376. lecommunications Union Telecommunications section of the ITU Variation of interpacket timing interval Kilobit per second 1 000 bits per second Megabit per second Million bits per second 387 November 2008 A K tal AudioCodes MediaPack Series Term MIB MLPP ms or msec MWI NAPTR NAT NPI NTP OAMP OSI PBX PCM PKI POTS PRT PSTN PVID QoS RFC RTCP RTP SA SAS SDP SIP SMDI SME SNMP SRTP SRV SSH SSL STUN TCP TCP IP SIP User s Manual Meaning Management Information Base Multilevel Precedence and Preemption Millisecond a thousandth part of a second Message Waiting Indicator Naming Authority Pointer Network Address Translation Numbering Plan Indicator Network Time Protocol Operations Administration Maintenance and Provisioning Open Systems Interconnection Industry Standard Private Branch Exchange Pulse Code Modulation Public Key Infrastructures Plain Old Telephone System or Service Prerecorded Tones File Public Switched Telephone Network Port VLAN ID VLAN ID assignment to Ethernet packet by switch Quality of Service Request for Comment issued by IETF Real Time Transport RTP Control Protocol Real Time Transport Protocol Security Associations contains encryption keys and profile 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 Enterp
377. lete 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 refer to Saving Configuration on page 209 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 44 Document LTRT 65411 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 235 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
378. liary Files on page 210 m 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 210 6 1 Configuring the Call Progress Tones File The Call Progress Tones CPT and Distinctive Ringing auxiliary file used by the device is a binary file with file extension dat This file is comprised of two sections The first section contains the definitions of the Call Progress Tones levels and frequencies that are detected generated by the device The second section contains the characteristics of the distinctive ringing signals that are generated by the device refer to Configuring the Distinctive Ringing Section of the ini File on page 310 You can either use one of the supplied device auxiliary dat files or create your own file 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 210 Note Only the dat file can be loaded to the device You can create up t
379. lified Domain Names FQDN in the Tel to IP Routing table 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 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 not require any special behavior from them For detailed information on STUN refer to STUN on page 366 60 Document LTRT 65411 SIP User s Manual Parameter STUN Server Primary IP STUNServerPrimarylP STUN Server Secondary IP STUNServerSecondaryIP NFS Settings NFS Table DHCP Settings Enable DHCP DHCPEnable Version 5 6 3 Web Based Management Description Notes For defining the STUN server domain name use the ini file parameter STUNServerDomainName refer to Networking Parameters on page 236 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
380. lls Reanswer Timeout 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 209 SIP User s Manual 130 Document LTRT 65411 SIP User s Manual Parameter General IP Security SecureCallsFromIP Filter Calls to IP FilterCalls2IP Enable Digit Delivery to IP EnableDigitDelivery2I P Enable Digit Delivery to Tel EnableDigitDelivery Version 5 6 3 Web Based Management Table 3 32 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 160 This is useful in preventing unwanted SIP calls or messages and or VolP 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 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 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 112 0
381. log line of the PBX instead of to the device s FXO interface and verify that it displays the caller ID Configure the following parameters e FXOSeizeLine 0 e RTPOnlyMode 1 or 2 the RTP is sent without SIP signaling e Coder G 711 e Inthe Tel to IP Routing table route all calls to the PC used for capturing e EnableCalleriD 0 e RingsBeforeCallerlD 0 e Set the automatic dialing to hotline e g TargetOfChannel7 9005 2 e HotLineToneDuration 0 e CallerlDTransportType 0 The above settings allow the FXO to send RTP by immediately seizing the line after receiving the first ring at your PC 363 November 2008 A tal AudioCodes MediaPack Series 7 Capture the RTP using Wireshark you can also use DSP trace and send the file to AudioCodes 7 14 7 3 Caller ID on the IP Side Caller ID is provided by the From header containing the caller s name and number for example From David lt SIP 101 10 33 2 2 gt tag 35dfsgasd45dg If Caller ID is restricted received from Tel or configured in the device the From header is set to From anonymous lt anonymouseanonymous invalid gt tag 35dfsgasd45dg The P asserted or P preferred headers are used to present the originating party s caller ID even when the caller ID is restricted These headers are used together with the Privacy header m If Caller ID is restricted e The From header is set to anonymo
382. lone Survivability SAS Parameters EnableSAS SASLocalSIPUDPPort SASDefaultGatewaylP SASRegistrationTime SASLocalSIPTCPPort SASLocalSIPTLSPort SASProxySet RedundantSASProxySet SASSurvivabilityMode Version 5 6 For a description of this parameter refer to Stand Alone Survivability on page 149 For a description of this parameter refer to Stand Alone Survivability on page 149 For a description of this parameter refer to Stand Alone Survivability on page 149 For a description of this parameter refer to Stand Alone Survivability on page 149 For a description of this parameter refer to Stand Alone Survivability on page 149 For a description of this parameter refer to Stand Alone Survivability on page 149 For a description of this parameter refer to Stand Alone Survivability on page 149 For a description of this parameter refer to Stand Alone Survivability on page 149 Determines the Survivability mode used by the SAS application 0 Standard All incoming 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 SASProxySet is available 2 Ignore REGISTER Use re
383. ls Count refer to Call Counters on page 224 Call Routing Status refer to Call Routing Status on page 226 Registration Status refer to Registration Status on page 226 SAS SBC Registered Users refer to SAS SBC Registered Users on page 227 IP Connectivity refer to IP Connectivity on page 228 Note The Gateway Statistics pages don t refresh automatically To view updated information re access the required page Version 5 6 223 November 2008 7a K tal AudioCodes MediaPack Series 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 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 or Tel to IP Calls Count page item the figure below shows the IP to Tel Calls Count page vw Figure 3
384. m 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 configurations m IsFaxUsed 0 E FaxTransportMode 1 m Additional configuration parameters FaxRelayEnhancedRedundancyDepth FaxRelayRedundancyDepth FaxRelayECMEnable 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 a Performs certain jitter buffering optimizations SIP User s Manual 320 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 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 pac
385. meter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 267 November 2008 V A AudioCodes Parameter EnableEarlyMedia EnableTransfer XferPrefix EnableMicrosofExt XferPrefixIP2Tel EnableHold HoldFormat HeldTimeout EnableForward CallWaitingPerPort SIP User s Manual MediaPack Series Description For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 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 e622125519100104 Once modified the device can then manipulate the num
386. mous Call Note You can reject anonymous calls per device using RejectAnonymousCallPerPort refer to Analog Telephony Parameters on page 279 Activate Keypad seguence that activates the reject anonymous call option KeyRejectAnonymousCal whereby the device rejects incoming anonymous calls After the I seguence is pressed a confirmation tone is heard Deactivate KeyRejectAnonymousCal Keypad sequence that de activates the reject anonymous call option IDeact After the seguence is pressed a confirmation tone is heard 3 4 4 2 6 Stand Alone Survivability The SAS Configuration 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 VoIP 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
387. 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 may 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 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 Name REMARK Parameter Value Parameter Value An example of an ini file containing individual ini file parameters is shown below SYSTEM Params syslogsenverrP 101372769 EnableSyslog 1 These are a few of the system related parameters WEB Params LogoWidth 339 WebLogoText My Device UseWeblogo 1 These are a f
388. n Digest username 122 realm audiocodes com nonce 11432d6bce58ddf02e3b5e1c77c010d2 uri 10 2 2 222 response b9c45d0234ababf5ddf5c704029b38cf 7 Upon receiving this request and if accepted by the Proxy the proxy returns a 200 OK response closing the REGISTER transaction SIP 2 0 200 OK Via SIP 2 0 UDP 10 1 1 200 From lt sip 122 10 1 1 200 gt tag 1c23940 mos egijos 220101 12005 Call ID 654982194e10 1 1 200 Cseg 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 13 3 Establishing a Call between Two Devices This section provides an example on configuring two AudioCodes devices with FXS interfaces for establishing call communication After configuration you can make calls between telephones connected to the same device or between the two devices In the example the IP address of the first device is 10 2 37 10 and its endpoint numbers are 101 to 104 The IP address of the second device is 10 2 37 20 and its endpoint numbers are 201 to 204 In this example a SIP Proxy is not used Internal call routing is performed using the device s Tel to IP Routing table SIP User s Manual 346 Document LTRT 65411
389. n page 58 For a description of this parameter refer to Configuring the Application Settings on page 58 For a description of this parameter refer to Configuring the Application Settings on page 58 For a description of this parameter refer to Configuring the Application Settings on page 58 Customizing the Web Appearance Parameters For detailed information on customizing the Web interface interface refer to Customizing the Web Interface on page 43 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 MediaPack The string can be up to 29 characters 251 November 2008 ca AudioCodes Parameter UseWebLogo WebLogoText LogoWidth LogoFileName 4 4 4 MediaPack Series 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
390. n page 67 For a description of this parameter refer to Configuring the Voice Settings on page 67 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 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 67 For a description of this parameter refer to Configuring the Voice Settings on page 67 For a description of this parameter refer to DTMF amp Dialing Parameters on page 125 Enables or disables detection of User Defined Tones signaling 0 Disable 1 Enable 302 Document LTRT 65411 SIP User s Manual Parameter UDTDetectorFreguencyDevi ation CPTDetectorFreguencyDevi ation MGCPDTMFDetectionPoint KeyBlindTransferAddPrefix VoiceP
391. n page 78 For a description of this parameter refer to General Parameters on page 129 For a description of this parameter refer to General Parameters on page 129 For a description of this parameter refer to Configuring the General Security Settings on page 90 For a description of this parameter refer to Configuring the General Security Settings on page 90 252 Document LTRT 65411 SIP User s Manual Parameter SIPSReguireClientCertificat e PeerHostNameVerification Mode VerifyServerCertificate TLSRemoteSubjectName OCSPEnable OCSPServerlP OCSPServerPort OCSPDefaultResponse EnableSecureStartup SSHAdminKey SSHReguirePublicKey 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 90 For a description of this parameter refer to Configuring the General Security Settings on page 90 For a description of this parameter refer to Configuring the General Security Settings on page 90 For a description of this parameter refer to Configuring the General Security Settings on page 90 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
392. n t include AKA authentication information are rejected 3 4 4 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 gualified domain name FADN You can define up to six Proxy Sets each having a unigue 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 186 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 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 260 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 59 Proxy Sets Table Page Proxy Set ID Proxy Address
393. n the IP Group table refer to Configuring the IP Groups on page 186 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 160 Note If the parameter PreferRouteTable is set to 1 refer to Proxy 8 Registration Parameters on page 112 the routing rules in the Tel to IP Routing Table prevail over the selected Serving IP Group ID The host name used in the From header in INVITE messages and as a host name in From To headers in REGISTER reguests If not configured the global parameter SIPGatewayName is used instead This is used as the user part in the Contact URI in INVITE messages and as a user part in From To and Contact headers in REGISTER reguests This is applicable only if the field Registration Mode is set to Per Account and the Registration through the Account table is successful 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 185 November 2008 A c tal AudioCodes MediaPack Series 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 3
394. n 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 SIP User s Manual 64 Document LTRT 65411 SIP User s Manual Parameter Metric RoutingTableHopsCountColumn Interface RoutingTablelnterfacesColumn 3 Web Based Management Description Multiple Interface Table on page 55 The maximum number of allowed routers hops 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 eguals 0 are local routes set automatically by the device Specifies the interface network type to which the 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 55 3 4 1 6 Configuring the QoS Settings The QoS Settings page is used for configurin
395. nOamVlanId 4 VlanNativeVlanId 4 VlanControlVlanId 5 VlanMediaVlanID 6 Multiple IPs Configuration EnableMultipleIPs 1 LocalMediaIPAddress 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 ASS 255 255 0 RoutingTableGatewaysColumn 10 32 0 1 10 31 0 1 RoutingTableInterfacesColumn 1 0 RoutingTableHopsCountColumn 20 20 Version 5 6 377 November 2008 A tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 378 Document LTRT 65411 SIP User s Manual 9 Supplied SIP Software Package 9 Supplied SIP Software Package The table below lists the standard SIP software package supplied with the SIP device File Name Ram cmp file MP124 SIP xxx cmp MP118 SIP xxx cmp ini files SlPgw MP124 ini SIPgw_fxs_MP118 ini SIPgw_fxs_MP114 ini SIPgw_fxs_MP112 ini MP1xx_Coeff_FXS dat Usa tones xx dat Usa tones xx ini Utilities DConvert ACSyslog BootP CPTWizard MIB Files Version 5 6 Table 9 1 Supplied Software Package Description Image file containing the software for the MP 124 FXS device Common Image file Image file containing
396. nce period of the cadence The distinctive ringing section of the ini file format contains the following strings NUMBER OF DISTINCTIVE RINGING PATTERNS Contains the following key e Number of Distinctive Ringing Patterns defining the number of Distinctive Ringing signals that are defined in the file Ringing Pattern X Contains the Xth ringing pattern definition starting from 0 and not exceeding the number of Distinctive Ringing patterns defined in the first section minus 1 using the following keys e Ring Type Must be equal to the Ringing Pattern number e Freq Hz Frequency in hertz of the ringing tone e First Burst Ring On Time 10 msec Ring On period in 10 msec units for the first cadence on off cycle e First Burst Ring Off Time 10 msec Ring Off period in 10 msec units for the first cadence on off cycle e Second Burst Ring On Time 10 msec Ring On period in 10 msec units for the second cadence on off cycle e Second Burst Ring Off Time 10 msec Ring Off period in 10 msec units for the second cadence on off cycle e Third Burst Ring On Time 10 msec Ring On period in 10 msec units for the third cadence on off cycle e Third Burst Ring Off Time 10 msec Ring Off period in 10 msec units for the third cadence on off cycle e Fourth Burst Ring On Time 10 msec Ring Off period in 10 msec units for the fourth cadence on off cycle e Fourth Burs
397. nd for a description of the items in this n file table refer to Configuring the IP Groups on page 186 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 Defines 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 5 The default value is 5 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 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 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 265 November 2008 V u 3 AudioCodes Parameter UseTelURIForAssertedID EnableRPlheader IsUserPhone IsUserPhonelnFrom IsUseToHeaderAsCalledNumb er EnableHistoryInfo SIPSubject MultiPtimeFormat EnableReasonHeader EnableSemiAttendedTransfer SIP183Behavior EnablePtime EnableUserlnfoUsage HandleReasonHeader EnableSilenceSuppInSDP SIP User s Manual MediaP
398. nd the network interface m Reset button on the rear panel for restarting the MP 11x and for restoring the MP 11x parameters to their factory default settings SIP User s Manual 16 Document LTRT 65411 SIP User s Manual 1 Overview 1 2 2 MP 124 Hardware Features The MP 124 hardware features include the following m MP 124 19 inch 1U rugged enclosure provides up to 24 analog FXS ports using a single 50 pin Telco connector m LEDs on the front panel that provide information on the device s operating status and the network interface m Reset button on the front panel for restarting the MP 124 and for restoring the MP 124 parameters to their factory default settings 1 3 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 S
399. nd frequencies that the device uses The default CPT file is U S A SIP User s Manual 210 Document LTRT 65411 SIP User s Manual 3 Web Based Management File Type Prerecorded Tones User Info Version 5 6 Description The dat PRT file enhances the device s capabilities of playing a wide range of telephone exchange tones that cannot be defined in the Call Progress Tones file 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 To load an auxiliary file to the device using the Web interface take these 6 steps Open the Load Auxiliary Files page Management tab gt Software Update menu gt Load Auxiliary Files page item Figure 3 101 Load Auxiliary Files Page INI file Ce ees FXS Coefficient file Bee ae Call Progress Tones file Prerecorded Tones file Browse Lead Fite User Info file 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 Click the Load File button corresponding to the file you want to load Repeat steps 2 through 3 for each file you want to load To save the loaded auxiliary files to flash memory refer to Saving Configuration on page 209 To reset th
400. ne to the PBX immediately after the phone is off hooked 2 Dial the destination number e g phone number 201 The DTMF digits are sent over IP directly to the PBX All the audible tones are generated from the PBX such as ringback busy or fast busy tones One to one mapping occurs between the FXS ports and PBX lines 3 The call disconnects when the phone connected to the FXS goes on hook Dialing from PBX Line or PSTN The procedure below describes how to dial from a PBX line i e from a telephone directly connected to the PBX or from the PSTN to the remote PBX extension i e telephone connected to the FXS device gt To dial from a telephone directly connected to the PBX or from the PSTN take this step m Dial the PBX subscriber number e g phone number 101 in the same way as if the user s phone was connected directly to the PBX As soon as the PBX rings the FXO device the ring signal is sent to the phone connected to the FXS device Once the phone connected to the FXS device is off hooked the FXO device seizes the PBX line and the voice path is established between the phone and PBX There is one to one mapping between PBX lines and FXS device ports Each PBX line is routed to the same phone connected to the FXS device The call disconnects when the phone connected to the FXS device is on hooked Message Waiting Indication for Remote Extensions The device supports the relaying of Message Waiting Indications
401. ned with an NTT Caller ID signal The format of this parameter is as follows EnableDID FORMAT EnableDID_Index EnableDID IsEnable EnableDID_Port EnableDID_Module EnableDID Where IsEnable Enables 1 or disables 0 default Japan NTT Modem DID support Port Port number Module Module number N A For example EnableDID EnableDID 0 1 2 EnableDID Notes This parameter is applicable only to FXS interfaces Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 Disables the generation of Caller ID type 2 when the phone is off hooked Caller ID type 2 also known as off hook Caller ID is sent to a currently busy telephone to display the caller ID of the waiting call 0 Caller ID type 2 isn t played 1 Caller ID type 2 is played default Defines the voltage change slope during polarity reversal or wink 0 Soft reverse polarity default 1 Hard reverse polarity Notes This parameter is applicable only to FXS interfaces Some Caller ID signals use reversal polarity and or Wink signals In these cases it is recommended to set PolarityReversalType to 1 Hard The duration in msec of the current disconnect pulse The range is 200 to 1500 The default is 900 Notes This parameter is applicable for both FXS and FXO interfaces The FXO interface detection range is approximately
402. negotiation methods are prioritized according to the 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 125 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 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 DTMF 271 November
403. nes the supported Receive DTMF negotiation method 0 No Don t declare RFC 2833 telephony event parameter in SDP 3 Yes Declare RFC 2833 telephony event parameter in SDP default The device is designed to always be receptive to RFC 2833 DTMF 126 Document LTRT 65411 SIP User s Manual Parameter 1 to 5 Tx DTMF Option TxDTMFOption RFC 2833 Payload Type RFC2833PayloadType Version 5 6 3 Web Based Management Description 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 Determines a single or several preferred transmit DTMF negotiation methods 0 Not Supported No negotiation DTMF digits are sent according to the parameters DTMFTransportType and RFC2833PayloadType default 1 INFO Nortel Sends DTMF digits according to IETF lt draft choudhuri sip info digit 00 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 D
404. 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 Registration mode per Hunt 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 112 or username if GWRegistrationName is not configured 0 Per Endpoint Each channel in the Hunt 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 registrations are sent by endpoints pertaining to the Hunt Group For example if the device is configured globally to register all its endpoints using the parameter ChannelSelectMode you can exclude some 184 Document LTRT 65411 SIP User s Manual Parameter Serving IP Group ID TrunkGroupSettings ServinglP Group Gateway Name TrunkGroupSettings GatewayN ame Contact User TrunkGroupSettings ContactU
405. nfiguring 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 97 Regional Settings Page Minutes Seconds 16 23 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 updated If the device is configured to obtain the date and time from an SNTP server refer to Configuring the Application Settings on page 58 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 369 After performing a hardware reset the date and time are returned to their defaults and therefore should be updated SIP User s Manual 206 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 207 Lock and unlock the device refer to Locking and Unlocking the Device on page 208 Save the configuration to the device s flash memory refer to Saving Configuration on page 209 To access the M
406. ng Ringback tones in the Call Progress Tones file You can define up to four Call Waiting indication tones refer to the parameter FirstCallWaitingTonelD in SIP Configuration Parameters on page 260 m To configure the Call Waiting indication tone cadence modify the following parameters NumberOfWaitingIndications WaitingBeepDuration and TimeBetweenWaitingIndications or using the Web interface refer to Supplementary Services on page 138 m To configure a delay interval before a Call Waiting Indication is played to the currently busy port use the parameter TimeBeforeWaitingIndication or using the Web interface refer to Supplementary Services on page 138 This enables the caller to hang up before disturbing the called party with Call Waiting Indications Applicable only to FXS modules Both the calling and called sides are supported by FXS modules the FXO modules support only the calling side To indicate Call Waiting the device sends a 182 Call Queued response The device identifies a Waiting Call when a 182 Call Queued response is received Message Waiting Indication Support for Message Waiting Indication MWI according to IETF lt draft ietf sipping mwi 04 txt gt including SUBSCRIBE to MWI server The FXS device can accept an MWI NOTIFY message that indicates waiting messages or that the MWI is cleared Users are informed of these messages by a stutter dial tone The stutter and confirmation tones are defined in the CPT
407. ng SNMP Trusted Managers on SIP User s Manual 200 Document LTRT 65411 SIP User s Manual Parameter Enable SNMP DisableSNMP Trap Manager Host Name SNMPTrapManagerHostName 3 Web Based Management Description page 205 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 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 Version 5 6 Changes made on the fly to parameters Loading of auxiliary files e g via Certificate page
408. ngs Basic S Full Syslog Server IP Address Management Settings Syslog Server Port Management Configuration Enable Syslog Regione Sethogs Memtenance Acbons software Update A Activity Types to Report vie Activity Log Messages The Web GUI is composed of the following main 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 43 For customizing the product name refer to Customizing the Product Name on page 46 m Toolbar Provides frequently required command buttons for configuration refer to Toolbar on page 23 m Navigation Pane Consists of the following areas e Navigation bar Provides tabs for accessing the configuration menus refer to Navigation Tree on page 25 creating a Scenario refer to Scenarios on page 35 and searching ini file parameters that have corresponding Web interface parameters refer to Searching for Configuration Parameters on page 34 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 m Work pane Displays configuration pages where all configuration is performed refer to Working with Configuration Pages on page 27 Version 5 6 23 November 2008 7a K tal AudioCodes MediaPack Series 3 3 1 Toolbar The toolbar provides command buttons fo
409. ni File Configuration As an alternative to configuring the device using the Web interface as described in Web Based Management on page 21 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 217 The ini file configuration parameters are saved in the device s non 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 235 The ini file parameters that are configurable in the Web interface are described in Web Based Management on page 21 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 pa
410. ni file table parameters AltRouteCauseTel2IP and AltRouteCauselP2Tel refer to Number Manipulation and Routing Parameters on page 289 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 Alternative Routing page item Figure 3 75 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 3 In the Tel to IP Reasons group select up to four different call failure reasons that invoke an alternative Tel to IP routing 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 209 3 4 4 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 170 m Tel Profile Settings refer to Tel Profile Settings on page 171 m P Profile Settings refer to IP Profile Settings on page 173 Implementing the device s Profile features provides the device with high level adaptation when connected to a variety of eguipment at
411. ns Function Channel Capacity Available Ports FXS Functionality FXS Capabilities Additional Features Polarity Reversal Wink Metering Tones Distinctive Ringing Message Waiting Indication SIP User s Manual Specification 24 analog ports Short or Long Haul Automatic Detection REN3 up to 9 km 30 000 feet using a 24 AWG line Note The lines have been tested under the following conditions ring voltage greater than 32 Vrms offhook loop current greater than 20 mA all lines ring simultaneously Lightning and high voltage protection for outdoor operation Caller ID generation Bellcore GR 30 CORE Type 1 using Bell 202 FSK modulation ETSI Type 1 NTT Denmark India Brazil British and DTMF ETSI CID ETS 300 659 1 Programmable Line Characteristics Battery feed line current hook thresholds AC impedance matching hybrid balance Tx amp Rx frequency response Tx amp Rx Gains Note For a specific coefficient file please contact AudioCodes Configurable ringing signal up to 4 cadences and frequency from 15 to 200 Hz Loop backs for testing and maintenance Immediate or smooth to prevent erroneous ringing 12 16 KHz sinusoidal bursts Applicable only to FXS interfaces By freguency 15 100 Hz and cadence patterns DC voltage generation TIA EIA 464 B V23 FSK data Stutter dial tone DTMF based 384 Document LTRT 65411 SIP User s Manual Function Voice 4 Tone Characteristics Voice
412. ns TimeBetweenWaitinglnd ications Time Before Waiting Indications TimeBeforeWaitingIndic ation Waiting Beep Duration WaitingBeepDuration Enable Caller ID EnableCallerlD Version 5 6 3 Web Based Management Description Determines whether Call Forward is enabled 0 Disable Disable the Call Forward service 1 Enable Enable Call Forward service using REFER default For FXS interfaces the Call Forward table refer to Call Forward on page 178 must be defined to use the Call Forward service 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 an FXS interface receives a call on a busy endpoint it responds with a 182 response and not with a 486 busy The device plays a call waiting indication signal When hook flash is detected the device switches to the waiting call The device that initiated the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received Notes The device s Call Progress Tones file must include a Call Waiting Ringback tone caller side and a Call Waiting tone called side FXS only The EnableHold parameter must be enabled on both the calling and the called side You can use the ini file table parameter CallWaitingPerPort to
413. nsultation Alternate m The consultation feature is relevant only for the holding party applicable only to the FXS module m After holding a call by pressing hook flash the holding party hears a dial tone and can now initiate a new call which is called a consultation call m While hearing a dial tone or when dialing to the new destination before dialing is complete the user can retrieve the held call by pressing hook flash m The held call can t be retrieved while Ringback tone is heard After the consultation call is connected the user can switch between the held and active call by pressing hook flash 7 14 3 Call Transfer There are two types of call transfers m Consultation Transfer REFER and REPLACES 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 AdialsC Version 5 6 359 November 2008 K tal AudioCodes MediaPack Series 7 14 4 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
414. nswering of the call 0 Disable Early Media is disabled default 1 Enable Enables Early Media 102 Document LTRT 65411 SIP User s Manual Parameter 183 Message Behavior SIP183Behaviour Session Expires Time SIPSessionExpires Minimum Session Expires MinSE Session Expires Method SessionExpiresMethod Asserted Identity Mode AssertedidMode Version 5 6 3 Web Based Management Description Note that to send a 183 response you must also set the parameter ProgressIndicator2IP to 1 If it is equal to 0 180 Ringing response is sent Defines the response of the device upon receipt of a SIP 183 response 0 Progress A 183 response without SDP does not cause the device to play a ringback tone default 1 Alert 183 response is handled by the device as if a 180 Ringing response is received and the device plays a ringback tone 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 O 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
415. nt in an INVITE message in the From header For information on Caller ID restriction according to destination source prefixes refer to Configuring the Number Manipulation Tables on page 151 gt To configure the Caller Display Information take these 5 steps 1 Open the Caller Display Information page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Caller Display Information page item Figure 3 81 Caller Display Information Page Gateway Port FXS Private Restricted v Caller ID Name Presentation FXS Susan C Restricted v FXS Lee P Allowed V FXS Ronaldino E Restricted FXO Hung L Allowed ov IL FXO alowed M FXO Allowed FXO Allowed 2 In the Caller ID Name field corresponding to the desired port enter the Caller ID string up to 18 characters 3 From the Presentation drop down list select one of the following e Allowed 0 sends the string defined in the Caller ID Name field when a Tel to IP call is made using the corresponding device port e Restricted 1 the string defined in the Caller ID Name field is not sent 4 Click the Submit button to save your changes 5 To save the changes to flash memory refer to Saving Configuration on page 209 Version 5 6 177 November 2008 e AudioCodes MediaPack Series
416. nternet 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 65 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 220 You can also configure this table using the ini file table parameter Interface Table refer to Networking Parameters on page 236 Version 5 6 55 November 2008 A tal AudioCodes MediaPack Series 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 52 2 Under the Multiple Interface Settings group click the right arrow um button alongside Multiple Interface Table a confirm
417. nts 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 display all the menus By default the Basic option is selected Figure 3 5 Navigation Tree in Basic and Full View Status p Status Contiguraton Management X Diagnostics Contiguration Management 3 Diagnostics Scenarios Search Scenarios Search Basic O Full O Basic Full barge asic a Network Settings network Settings Full Navigation Navigation Tree media settings HB Media Settings Tree View _ View Option protocol Configuration P security Setting Option _ P advance Applications Protocol Configuration ti H Advance Applications All Menus Note When in Scenario mode refer to Scenarios on page 35 the Navigation tree is displayed in Full view i e all menus are displayed in the Navigation tree SIP User s Manual 26 Document LTRT 65411 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
418. o Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 Determines the V 34 fax transport method 0 Transparent 1 Relay default 2 Bypass 3 Transparent with Events 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 67 For a description of this parameter refer to Configuring the Voice Settings on page 67 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 For a description of this parameter refer to Configuring the Voice Settings on page 67 Determines whether 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 c
419. o Metering Tones on page 144 For a description of this parameter refer to Metering Tones on page 144 For a description of this parameter refer to Keypad Features on page 147 For a description of this parameter refer to Keypad Features on page 147 284 Document LTRT 65411 SIP User s Manual Parameter KeyCFBusy KeyCFBusyOrNoAnswer KeyCFDoNotDisturb KeyCFDeact KeyCLIR KeyCLIRDeact KeyHotLine KeyHotLineDeact KeyBlindTransfer KeyCallWaitingDeact KeyCallWaiting KeyRejectAnonymousCall KeyRejectAnonymousCallDe act FlashKeysSequenceStyle FlashKeysSequenceTimeout Version 5 6 4 ini File Configuration Description For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 For a description of this parameter page 147 Flash keys sequence style refer to Keypad Features on refer to Keypad Features on refer to Keypad Features on refer to K
420. o Supplementary Services on page 138 ETSIVMWITypeOneStandard for a description refer to Analog Telephony Parameters on page 279 BellcoreVMWITypeOneStandard for a description refer to Analog Telephony Parameters on page 279 7 14 7 Caller ID This section discusses the device s Caller ID support 7 14 7 1 Caller ID Detection Generation on the Tel Side By default generation and detection of Caller ID to the Tel side is disabled To enable Caller ID set the parameter EnableCallerID to 1 When the Caller ID service is enabled For FXS the Caller ID signal is sent to the device s port For FXO the Caller ID signal is detected The configuration for Caller ID is described below Use the parameter CallerlDType to define the Caller ID standard Note that the Caller ID standard that is used on the PBX or phone must match the standard defined in the device Select the Bellcore caller ID sub standard using the parameter BellcoreCallerlDTypeOneSubStandard Select the ETSI FSK caller ID sub standard using the parameter ETSICallerlIDTypeOneSubStandard SIP User s Manual 362 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities Enable or disable per port the caller ID generation for FXS and detection for FXO using the Generate Detect Caller ID to Tel table EnableCallerlD If a port isn t configured its caller ID generation detection are determined according to the global parameter EnableC
421. o 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 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 m 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 Version 5 6 307 November 2008 7a K tal AudioCodes MediaPack Series 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
422. o 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 recommended 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 rest
423. o IP calls an alternative IP address is provided for IP to Tel calls an alternative Hunt Group is provided Refer to Tel to IP Routing Table on page 160 for information on defining an alternative IP address refer to IP to Trunk Group Routing on page 163 for information on defining an alternative Hunt Group You can use the Reasons for Alternative Routing page for the following example scenarios m 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 129 The device also plays a tone to the endpoint whenever an alternative route is used This tone is played for a user defined time using the ini file parameter AltRoutingToneDuration refer to Routing General Parameters on page 157 SIP User s Manual 168 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 subseguently attempted twice to establish the call unsuccessfully You can also configure alternative routing using the i
424. o IP when the ringing signal is detected The FXO line is seized only if the remote IP party answers the call If the remote party doesn t answer the call and the second ring signal is not received within this timeout the device releases the IP call This parameter is typically set to between 5 and 8 The default is 8 Note This parameter is applicable only for Tel to IP calls Busy or Reorder tone duration in seconds that the device plays before releasing the line The valid range is 0 to 254 The default is 0 seconds Typically after playing a Reorder Busy tone for the specified duration the device starts playing an Offhook Warning tone Notes Selection of Busy or Reorder tone is performed according to the release cause received from IP Refer also to the parameter CutThrough described in Advanced Parameters on page 129 Enables sending of 200 OK upon detection of speech fax or modem 1 Yes device sends 200 OK to INVITE messages when speech fax modem is detected 0 No 200 OK is sent only once the device completes dialing default 197 November 2008 ca AudioCodes Parameter Rings before Detecting Caller ID RingsBeforeCallerlD Send Metering Message to IP SendMetering2IP Disconnect Call on Busy Tone Detection DisconnectOnBusyTone Disconnect on Dial Tone DisconnectOnDialTone Guard Time Between Calls GuardTimeBetweenCalls 3 5 MediaPack Series Description
425. ode for IP to Tel FXO calls 0 One Stage One stage dialing 1 Two Stages Two stage dialing default If two stage dialing is enabled the device seizes one of the PSTN PBX lines without performing any dialing connects the remote IP user to the PSTN PBX and all further signaling dialing and Call Progress Tones is performed directly with the PBX without the device s intervention If one stage dialing is enabled the device seizes one of the available lines according to the parameter ChannelSelectMode and dials the destination phone number received in the INVITE message Use the parameter IsWaitForDialTone to specify whether the dialing must start after detection of the dial tone or immediately after seizing the line Determines whether the device waits for a dial tone before dialing the phone number for IP to Tel FXO calls 0 No Don t wait for dial tone 1 Yes Wait for dial tone default When one stage dialing and this parameter are enabled the device dials the phone number to the PSTN PBX line only after it detects a dial tone If this parameter is disabled the device immediately dials the phone number after seizing the PSTN PBX line without listening for a dial tone 196 Document LTRT 65411 SIP User s Manual Parameter Time to Wait before Dialing msec WaitForDialTime Ring Detection Timeout sec FXOBetweenRingTime Reorder Tone Duration sec TimeForReorderTone
426. ode value once a SIP 200 OK response is received the device uses the RTP information and re opens the voice stream if necessary Enable Reason Header Enables disables the usage of the SIP Reason header EnableReasonHeader 0 Disable 1 Enable default Retransmission Parameters SIP T1 Retransmission The time interval in msec between the first transmission of a SIP Timer msec message and the first retransmission of the same message SipT1Rtx 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 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 SIP T2 Retransmission The maximum interval in msec between retransmissions of SIP Timer msec messages SipT2Rtx 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 SIP Maximum RTX Maximum number of UDP transmissions first transmission plus SIPMaxRtx retransmissions of SIP messages The range is 1 to 30 The default value is 7 Version 5 6 111 Nov
427. odemTransportType 0 V22ModemTransportType 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 the 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 123 Version 5 6 325 November 2008 A c tal AudioCodes MediaPack Series 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 Below is an example of media descriptions of an SDP indicating support for V
428. og Messages Parameters Value Change 4uxiliary Files Loading 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 3 Click the Submit button to save your changes 4 To save the changes to flash memory refer to Saving Configuration on page 209 Version 5 6 199 November 2008 A K tal AudioCodes MediaPack Series Table 3 52 Management Settings Parameters Parameter Description Syslog Seitings 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 numbe
429. og trunks i e lines to the PBX from getting stuck when the called phone hangs up The PBX doesn t disconnect the call but instead signals to the device that the call is disconnected using one of the following methods m Detection of polarity reversal current disconnect The call is immediately disconnected after polarity reversal or current disconnect is detected on the Tel side assuming the PBX CO produces this signal This is the recommended method Relevant parameters EnableReversalPolarity EnableCurrentDisconnect CurrentDisconnectDuration CurrentDisconnectDefaultThreshold and TimeToSampleAnalogLineVoltage SIP User s Manual 328 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities m Detection of Reorder Busy Dial and Special Information Tone SIT tones The call is immediately disconnected after a Reorder Busy Dial or SIT tone is detected on the Tel side assuming the PBX CO generates this tone This method requires the correct tone frequencies and cadence to be defined in the Call Progress Tones file If these frequencies are not known define them in the CPT file the tone produced by the PBX CO must be recorded and its frequencies analyzed refer to Adding a Reorder Tone to the CPT File in the Reference Manual This method is slightly less reliable than the previous one You can use the CPTWizard described in Call Progress Tones Wizard in the Reference Manual to analyze Call Progress
430. oice 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 O 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 packets 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 SIP User s Manual 334 Document LTRT 65411 SIP User s Manual 7 IP Telephon
431. ol 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 button which is located near the bottom of the page in which you are working modifications to parameters with on the fly capabilities 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 207 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 209 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 31 November 2008 7a K tal AudioCodes MediaPack Series 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
432. omized Title bar with a different image logo and product name Figure 3 22 Customizing Web Logo and Product Name A Home 6 a gt 6 Log oft Default Logo Customized Customized logo Product Name Version 5 6 43 November 2008 7a L tal AudioCodes MediaPack Series 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 21 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 ESSE 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 comp
433. ommended 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 231 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 Determines the time in seconds the SA negotiated in the first IKE session main mode is valid After the time expires the SA is re negotiated The default value is 28800 i e 8 hours Determines the lifetime in kilobytes that the SA negotiated in the 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 negotiations 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 IKEPolicyProposalEncrypti on X Version 5 6 Determines the encryption type used in the main mode negotiation for up to four proposals For the ini file parameter X depicts th
434. on This ini file table parameter configures the internal DNS table for resolving host names into IP addresses Up to four different IP addresses in dotted decimal notation can be assigned to a host name The format of this parameter is as follows Dns2lp FORMAT Dns2Ip Index Dns2Ip DomainName Dns2lp FirstipAddress Dns2Ip SecondlpAddress Dns2lp ThirdipAddress Dns2lp FourthlpAddress Dns2Ip For example Dns2lp 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 If 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 166 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 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
435. on These menus appear in the Navigation tree and include the following Network Settings refer to Network Settings on page 52 Media Settings refer to Media Settings on page 67 Sigtran Configuration refer to Sigtran Configuration Security Settings refer to Security Settings on page 79 Protocol Configuration refer to Protocol Configuration on page 100 Advanced Applications refer to Advanced Applications on page 190 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 52 m Application Settings refer to Configuring the Application Settings on page 58 m P Routing Table refer to Configuring the IP Routing Table on page 63 m QoS Settings refer to Configuring the QoS Settings on page 65 Configuring 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
436. on on page 217 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 307 For information on deleting these files from the device refer to Device Information on page 221 Table 3 56 Auxiliary Files Descriptions File Type Description ini 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 FXS Coefficient This file contains the analog telephony interface configuration information for the device This information includes telephony interface characteristics such as DC and AC impedance feeding current and ringing voltage This file is specific to the type of telephony interface that the device supports In most cases you are required to load this type of file Note Use the parameter CountryCoefficients described in Analog Telephony Parameters on page 279 to configure the FXO coefficients Call Progress This is a region specific telephone exchange dependent file that contains the Tones Call Progress Tones CPT levels a
437. on sec Hotline Dial Tone Duration sec Enable Special Digits Default Destination Number Special Digit Representation 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 or light blue background To navigate between Scenario Steps you can perform one of the following m inthe Navigation tree click the required Scenario Step SIP User s Manual 38 Document LTRT 65411 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 o 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 Na
438. on 5 6 175 November 2008 7a K tal AudioCodes MediaPack Series 3 4 4 6 2 Automatic Dialing The Automatic Dialing page allows you to define a telephone number that is automatically dialed when an FXS or FXO port is used e g off hooked After a ring signal is detected on an Enabled FXO port the device initiates a call to the destination number without seizing the line The line is seized only after the call is answered After a ring signal is detected on a Disabled or Hotline FXO port the device seizes the line You can also configure automatic dialing using the ini file table parameter TargetOfChannel refer to Analog Telephony Parameters on page 279 You can configure the device to play a Busy Reorder tone to the Tel side upon receiving a SIP 4xx 5xx or 6xx response from the IP side i e Tel to IP call failure using the ini file parameter FXOAutoDialPlayBusyT one refer to SIP Configuration Parameters on page 260 gt To configure Automatic Dialing take these 5 steps 1 Open the Automatic Dialing page Configuration tab gt Protocol Configuration menu gt Endpoint Settings submenu gt Automatic Dialing page item Figure 3 80 Automatic Dialing Page Gateway Destination Phone Auto Dial Port Number Status Port 1 FXS Enable v Port 2 FXS Hotline Port 3 FXS Enable Port4 FXS Enable Port 5 FXO Enable v Port 6 FXO
439. 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 58 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 58 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 indicate 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 bi
440. on its port while the FXS interface generates a Current Disconnect Pulse after a call is released from IP The current disconnect duration is determined by the parameter CurrentDisconnectDuration The current disconnect threshold FXO only is determined by the parameter CurrentDisconnectDefaultThreshold The frequency at which the analog line voltage is sampled is determined by the parameter TimeToSampleAnalogLineVoltage Determines whether the device releases the call if RTP packets are not received within a user defined timeout 0 No 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 address 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
441. on on using the ini file refer to Modifying an ini File on page 235 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 1 Enable product name change The text string that replaces the product name The default is MediaPack 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 Microsoft Internet Explorer SRR AR ARERR EER ok oko oko oko oo ok oo ooo o oko o EE o sok EE REE AE patat t i a t i i Welcome to the Embedded Web Server faata ok a a se ooo k ok o oko ok oko ok o kok kok ok o ok ok os ooo ooo ok kok ok k ok k o o o ko k o ko ob ko ko ko ko oko k o oko k o ko oko ok o oko oko oko ok o oko o ok Table 3 5 ini File Parameter for Welcome Login Message Parameter WelcomeMessage SIP User s Manual Description Defines the Wel
442. on tone is heard Caller ID Restriction Note The caller ID presentation can be viewed in the Caller Display Information table refer to Caller ID on page 177 Activate Keypad seguence that activates the restricted Caller ID option After KeyCLIR the seguence is pressed a confirmation tone is heard Deactivate Keypad seguence that deactivates the restricted Caller ID option After KeyCLIRDeact the sequence is pressed a confirmation tone is heard Hotline Note The destination phone number and the auto dial status can be viewed in the Automatic Dialing table refer to Automatic Dialing on page 175 Activate Keypad sequence that activates the delayed hotline option KeyHotLine To activate the delayed hotline option from the telephone perform the following Dial the preconfigured sequence number on the keypad a dial tone is heard Dial the telephone number to which the phone automatically dials after a configurable delay terminate the number with a confirmation tone is heard Deactivate Keypad sequence that deactivates the delayed hotline option After KeyHotLineDeact the sequence is pressed a confirmation tone is heard Transfer Blind Keypad sequence that activates blind transfer for Tel to IP calls There KeyBlindTransfer are two possible scenarios Option 1 After this sequence is dialed the current call is put on hold using Re INVITE a dial tone is played to the phone and then phone n
443. on 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 3 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 252 Version 5 6 97 November 2008 7a c tal AudioCodes MediaPack Series gt To configure the IKE table take these 5 steps 1 Open the IKE Table page Configuration tab gt Security Settings menu gt IKE Table page item Figure 3 56 IKE Table Page vw Policy Index 0 State Does not exist Internet Key Exchange table row does not exist o Authentication Method Pre shared Key Shared Key 00000 IKE SA LifeTime sec 28800 IKE SA LifeTime KB 0 First Proposal Encryption Type Not Defined First Proposal Authentication Type Not Defined First Proposal DH Group Not Defined Second Proposal Encryption Type Not
444. ones page refer to Metering Tones on page 144 Figure 3 65 Charge Codes Table Page v Table Index 0 4 v Time Period 1 Time Period 2 Time Period 3 Time Period 4 Index End Pulse Pulses end Pulse Pulses End pulse Pulses End Pulse Pulses Time Interval Time Interval n Time Interval Time Interval Answer Answer Answer Answer o 07 30 1 14 20 2 20 15 1 00 60 1 05 60 1 14 20 1 00 60 1 00 60 1 2 Define up to 25 different charge codes each charge code is defined per row Each charge code can include up to four different time periods in a day 24 hours Each time period is composed of the following e The end of the time period in a 24 rounded hour s format e The time interval between pulses in tenths of a second e The number of pulses sent on answer The first time period always starts at midnight 00 It is mandatory that the last time period of each rule ends at midnight 00 This prevents undefined time frames in a day The device selects the time period by comparing the device s current time to the end time of each time period of the selected Charge Code The device generates the Number of Pulses on Answer once the call is connected and from that point on it generates a pulse each Pulse Interval If a call starts at a certain time period and crosses to the next the information of the next time period is used 3 Click the Submit button to save your changes 4
445. ontrol This parameter applies to both Multiple IPs and VLAN mechanisms Multiple IPs Determines the network type for 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 52 Note This parameter is not applicable when configuring multiple interfaces using the ini file table parameter Interface Table For a description of this parameter refer to Configuring the IP Settings on page 52 For a description of this parameter refer to Configuring the IP Settings on page 52 For a description of this parameter refer to Configuring the IP Settings on page 52 241 November 2008 A tal AudioCodes MediaPack Series Parameter Description LocalControllPAddress For a description of this parameter refer to Configuring the IP Settings on page 52 LocalControlSubnetMask For a description of this parameter refer to Configuring the IP Settings on page 52 LocalControlDefaultGW For a description of this parameter refer to Configuring the IP Settings on page 52 LocalOAMIPAddress For a description of this parameter refer to Configuring the IP Settings on page 52 LocalOAMSubnetMask For a description of this parameter re
446. option to configure registration mode per Hunt 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 RegistrationTimeDivider is 50 If registration per 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 Version 5 6 341 November 2008 7a c tal AudioCodes MediaPack Series 7 13 7 13 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 6 SIP Call Flow 10 8 201 10 10 8 201 108 Phone 1000 INVITE F1 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 8000e10 8 201 108 gt tag 1c5354 To lt sip 1000 10 8 201 10 gt Call ID 534366556655skKw 8000 1000 10 8 201 108 CSeq 18153 INVITE Contact lt sip 8
447. or 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 94 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 This ini file table parameter configures the IKE table The format of this parameter is as follows IPSec_IKEDB_ Table Format IKE_DB_INDEX IKEPolicySharedKey IKEPolicyProposalEncryption X IKEPolicyProposalAuthentication X IKEPolicyProposalDHGroup X IKEPolicyLifelnSec IKEPolicyLifelnKB IkePolicyAuthenticationMethod MPSEC IKEDB TABLE For example 254 Document LTRT 65411 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 differ
448. or 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 369 Network QoS NetworkServiceClassDiffServ Media Premium QoS PremiumServiceClassMediaDiffServ Control Premium QoS PremiumServiceClassConirolDiffServ Gold QoS GoldServiceClassDiffServ Bronze QoS BronzeServiceClassDiffServ 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 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 determined by the following according to priority PDiffServ value in the selected IP Profile PremiumServiceClassMediaDiffServ Defines the DiffServ value for Premium Control CoS content only if ControllPDiffserv is not set in the selected IP Profile The valid range is 0
449. ora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 This ini file table parameter manipulates the source number for IP to Tel calls The format of this parameter is as follows SourceNumberMaplp2Tel FORMAT SourceNumberMaplp2Tel Index SourceNumberMaplp2Tel DestinationPrefix SourceNumberMaplp2Tel SourcePrefix SourceNumberMaplp2Tel SourceAddress SourceNumberMaplp2Tel NumberType SourceNumberMaplp2Tel NumberPlan SourceNumberMaplp2Tel RemoveFromLetft 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 The parameters SourceNumberMaplp2Tel_ NumberType SourceNumberMaplp2Tel NumberPlan and SourceNumberMaplp2Tel IsPresentationRestricted are not applicable Set these to RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add and LeaveFromRight 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 Param
450. ot 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 m 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 P 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 Proxy 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 SIP User s Manual 160 Document LTRT 65411 SIP User s Manual 3 Web Based Management Version 5 6 Assign Profiles to destination addresses also when a Proxy is used 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
451. otocol type or family used on this leg of the call Connect time in NTP format Disconnect time in NTP format Q 931 disconnect cause code Name of the gateway SIP Call ID The call s terminator PSTN terminated call Yes IP terminated call No Destination phone number Calling Party Number ANI Account Request Type 337 7 IP Telephony Capabilities Value Format Example Up to 32 octets String Answer String Originate etc String VolP String String Numeric String SIPIDString String abcde ac com String Yes No String 8004567145 String 2427456425 String 5135672127 Numeric 1 start 2 stop AAA Acc Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Stop Acc Stop Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Stop Acc Start Acc Stop Acc Start Acc Stop Acc Start November 2008 ca AudioCodes Attribute Number 41 42 43 44 46 47 48 61 Attribute Response Attributes 26 44 VSA Value MediaPack Series 1 Name No Purpose Format Example AAA Type start or stop Acc Note start isn t Stop supported on the Calling Acc Card application z Start Acct Delay No of seconds tried in Ace i sending a particular Numeric 5 Time Stop record Acc Number of octets arrak received for that call Numeric ron duration Acct Output Numbe
452. ove user defined defaults and restore factory default values take this step m 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 default 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 6 305 November 2008 A tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 306 Document LTRT 65411 SIP User s Manual 6 Auxiliary Configuration Files 6 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 Auxi
453. p 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 Hunt Group number is added to the tgrp parameter value in the Contact header of outgoing SIP messages If a Hunt Group number is not associated with the call the tgrp 108 Document LTRT 65411 SIP User s Manual Parameter Enable GRUU EnableGRUU User Agent Information UserAgentDisplayInfo Version 5 6 3 Web Based Management Description 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 Contact 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 unigue ID that identifies the de
454. page If you modify a parameter in its primary configuration page orini file that also 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 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 123 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 are defined For G 729
455. 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 m On the toolbar click the Home G7 icon the Home page is displayed Figure 3 26 MP 11x Home Page Uplink Ready Power The number and type of channels displayed in the Home page depends on the device s model e g MP 118 or MP 114 The table below describes the areas of the Home page Table 3 6 Description of the Areas of the Home Page Item Label Description Alarms 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 To view a list of active alarms in the Active Alarms page refer to Viewing Active Alarms on page 222 click the Alarms area SIP User s Manual 48 Document LTRT 65411 SIP User s Manual 3 Web Based Management Item Label Description Channel Ports Displays the status of the ports channels red line not connected only applicable to FXO devices grey channel inactive blue handset is off hook green active RTP stream You can also view the channel s port settings refer to Viewing Port Information on page 50 reset the port refer to Releasing an Analog Channel on page 50 and as
456. page 199 For a description of this parameter refer to Configuring the Management Settings on page 199 239 November 2008 ca AudioCodes Parameter SyslogOutputMethod BaseUDPport RemoteBaseUDPPort L1L1ComplexTxUDPPort L1L1ComplexRxUDPPort NTPServerlP NTPServerUTCOffset NTPUpdatelnterval MediaPack Series 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 73 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 For a description of this parameter refer to Configuring the RTP RTCP Settings on page 73 For a description of this parameter refer to Configuring the Application Settings on page 58 For a description of this parameter refer to Configuring the Application Settings on page 58 For a description of this parameter refer to Configuring the Application Settings on page 58 IP Routing Table parameters The IP routing ini file pa
457. 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 Tel 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 297 November 2008 ca AudioCodes Parameter MediaPack Series Description 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 first 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 155 For a description on using ini file t
458. pe 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 NAPTR 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 ProxylP domain com 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 117 November 2008 ca AudioCodes Parameter Subscription Mode SubscriptionMode Number of RTX Before Hot Swap HotSwapRtx Use Gateway Name for OPTIONS UseGatewayNameForOp
459. port 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 8 Client Verify Subject Name when acting as a 93 November 2008 ca AudioCodes Parameter TLS Client Verify Server Certificate VerifyServerCertificate TLS Remote Subject Name TLSRemoteSubjectName MediaPack Series Description 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
460. 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 Notes If EnableProxyKeepAlive is set to 1 or 2 the device monitors the connection with the Proxies by using keep alive messages OPTIONS or Version 5 6 121 November 2008 A tal AudioCodes MediaPack Series Parameter Transport Type Proxy Load Balancing Method ProxyLoadBalancin gMethod Enable Proxy Keep Alive EnableProxyKeepA live SIP User s Manual Description 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 101 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
461. ption of the Web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 32 SIP User s Manual 188 Document LTRT 65411 SIP User s Manual Parameter Served Trunk Group Serving IP Group Username Password HostName Register Version 5 6 3 Web Based Management Table 3 48 Account Parameters Description Description The Hunt Group ID for which the device performs registration and or authentication to a destination IP Group i e Serving IP Group For Tel to IP calls the Served Trunk Group is the source Hunt Group from where the call initiated For IP to Tel calls the Served Trunk Group is the HuntGroup ID defined in the IP to Hunt Group Routing table refer to IP to Hunt Group Routing on page 163 For defining Hunt Groups refer to Configuring the Endpoint Phone Numbers on page 181 The destination IP Group ID defined in Configuring the IP Groups on page 186 to where the REGISTER requests if enabled are sent or Authentication is performed The actual destination to where the REGISTER reguests are sent is the IP address defined for the Proxy Set ID refer to Proxy Sets Table on page 120 associated with this IP Group This occurs only in the following conditions The parameter Registration Mode is set to Per Account in the Hunt Group Settings table refer to Configuring the Hunt Group Settings on page 183 The parameter Regist
462. pulation 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 Src Trunk Group ID The source Hunt Group for Tel to IP calls PREFIX_SrcTrunkGroupID The range is 1 99 Notes If this parameter is not required in the routing rule leave the field empty To denote any Hunt Group you can enter the asterisk symbol Dest Phone Prefix Represents a called telephone number prefix The prefix can be 1 to PREFIX_DestinationPrefix 19 digits long An asterisk represents all numbers Source Phone Prefix Represents a calling telephone number prefix The prefix can be 1 to PREFIX_SourcePrefix 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 155 Dest IP Address The destination IP address in dotted decimal notation to where PREFIX_DestAddress these calls must be sent Domain names e g domain com can be used instead of IP addresses
463. r 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 interface Accessing the Web Interface The Web interface can be opened using any standard Web browser refer to Computer Requirements on page 21 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 80 Version 5 6 21 November 2008 A e AudioCodes MediaPack Series 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 Realm UserName EE v Password s 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 t
464. r using SyslogServerlP 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 Advanced Parameters on page 129 For information on the Syslog refer to the Product Reference Manual Logs are also sent to the RS 232 serial port For information on establishing a serial communications link with the device refer to the device s nstallation Manual SNMP Settings For detailed information on the SNMP parameters that can be configured via the ini file refer to SNMP Parameters on page 258 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 Ll button to configure the SNMP trap destinations refer to Configuring the SNMP Trap Destinations Table on page 201 SNMP Community String Click the arrow L button to configure the SNMP community strings refer to Configuring the SNMP Community Strings on page 203 SNMP V3 Table Click the arrow gt button to configure the SNMP V3 users refer to Configuring SNMP V3 Table on page 204 SNMP Trusted Managers Click the arrow L button to configure the SNMP Trusted Managers refer to Configuri
465. r a description of this parameter refer to Advanced Parameters on page 129 Determines whether the device uses the value of the incoming SIP Reason header for Release Reason mapping 0 Disregard Reason header in incoming 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 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 266 Document LTRT 65411 SIP User s Manual Parameter EnableRport IsFaxUsed T38UseRTPPort DefaultReleaseCause IPAlertTimeout SIPPSessionExpires SessionExpiresMethod MINSE SIPMaxRtx SipT1Rtx SipT2Rtx Version 5 6 4 ini File Configuration Description Enables disables the usage of the rport parameter in the Via header 0 Enabled 1 Disabled default The device adds an rpor parameter to the Via header of each outgoing SIP message The first Proxy that receives this message sets the rpor 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 rpor tag the destination port of the response is ta
466. r 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 ALO WA o QO 255 PREFIX 1 AQ AW 1 OPP PREFIX 2 SO W dls o W 255 PREFIX 3 AQ 10 3 W 255 PREFIX 0 OF OF 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 234 Document LTRT 65411 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 E AERO 2 ROT 222 CoderName FORMAT CoderName Index CoderName Type CoderName PacketInterval CoderName rate CoderName PayloadType CoderName Sce CoderName 1 g7231 90 CoderName 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
467. r 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 5 Advanced Applications The Advanced Applications menu allows you to configure advanced SIP based applications This menu includes the following page items m Voice Mail Settings refer to Configuring the Voice Mail Parameters on page 190 m RADIUS Parameters refer to Configuring RADIUS Accounting Parameters on page 194 m FXO Settings refer to Configuring the FXO Parameters on page 195 3 4 5 1 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 FXO interfaces For detailed information on voice mail refer to the CPE Configuration Guide for Voice Mail User s Manual Note The Voice Mail page is only available for devices providing FXO interfaces 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 SIP User s Manual 190 Document LTRT 65411 SIP User s Manual 3 Web Based Management Figure 3 89 Voice M
468. r of concurrent calls If the profile is set to some limit the 3 4 4 6 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 Configuring the Endpoint Settings The Endpoint Settings submenu allows you to configure port specific parameters This submenu includes the following page items Authentication refer to Authentication on page 174 Automatic Dialing refer to Automatic Dialing on page 175 Caller Display Information refer to Caller Display Information on page 177 Call Forward refer to Call Forward on page 178 Caller ID Permissions refer to Caller ID Permissions on page 179 Call Waiting refer to Call Waiting on page 180 SIP User s Manual 174 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 4 4 6 1 Authentication The Authentication page defines a user name and password for authenticating each device port Authentication is typically used for FXS interfaces but can also be used for FXO interfaces The Authentication Mode parameter refer to Proxy amp Registration Parameters on page 112 determines whether authentication is
469. r of octets sent for N meric Stop Octets that call duration Acc Start Acct A unique accounting Ace i identifier match start amp String 34832 Session ID Stop stop ce Acct For how many seconds Sto Session the user received the Numeric A p i cc Time service Acct Input Number of packets Numeric Stop Packets received during the call Acc Acct Output Number of packets sent N meric Stop Packets during the call Acc i Start NAS Port PRT pon Ypa DF 0 Acc device on which the callis String Type f Asynchronous Stop active Ace H323 The reason for failing Return 103 authentication 0 ok Numeric z re Code other number failed p A unigue accounting zp identifier match start String Pie stop Below is an example of RADIUS Accounting where the non standard parameters are preceded with brackets Accounting Request 361 Use name 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 4841 8800 acct input packets 122 acct output packets 220 called station id 201 SIP User s Manual 338 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 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 0e2f3cc5
470. r 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 4 Submit Applies parameter settings to the device refer to Saving Configuration on page 209 Note This icon is grayed out when not applicable to the currently opened page Burn Saves parameter settings to flash memory refer to Saving Configuration on page 209 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 217 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 217 Reset opens the Maintenance Actions page for resetting the device refer to Resetting the Device on page 207 Software Upgrade Wizard opens the Software Upgrade Wizard page for upgrading the device s software refer to Software Upgrade Wizard on page 212 Home Opens the Home page refer to Using the Home Page on page 48 6 Help Opens the Online Help topic of the currently opened configuration page in the Work pane refer to Getting Help on page 47 Log off Logs off a session with the Web interface refer to Logging Off the Web Interface on page 51
471. r s Manual string is defined according to the following rules ame is equal to RegistrarName if configured The RegistrarName can 340 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities be any string m Otherwise the servername is equal to RegistrarlP either FQDN or numerical IP address if configured m Otherwise the 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 channel according to the parameter AuthenticationMode There is also an
472. r to Saving Configuration on page 209 For a description of the web interface s table command buttons e g Duplicate and Delete refer to Working with Tables on page 32 You can also configure SNMP v3 users using the ini file table parameter SNMPUsers refer to SNMP Parameters on page 258 SIP User s Manual 204 Document LTRT 65411 SIP User s Manual 3 Web Based Management Table 3 55 SNMP V3 Users Parameters Parameter Index SNMPUsers Index User Name SNMPUsers Username Authentication Protocol SNMPUsers AuthProtocol Privacy 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 3 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
473. rameter refer to Configuring the General Security Settings on page 90 For a description of this parameter refer to Configuring the General Security Settings on page 90 For a description of this parameter refer to Configuring RADIUS Accounting Parameters on page 194 For a description of this parameter refer to Configuring RADIUS Accounting Parameters on page 194 For a description of this parameter refer to Configuring RADIUS Accounting Parameters on page 194 For a description of this parameter refer to Configuring the General Security Settings on page 90 For a description of this parameter refer to Configuring the General Security Settings on page 90 257 November 2008 ca AudioCodes Parameter RadiusLocalCacheTimeout RadiusVSAVendorlD RadiusVSAAccessAttribute 4 4 6 MediaPack Series Description For a description of this parameter refer to Configuring the General Security Settings on page 90 For a description of this parameter refer to Configuring the General Security Settings on page 90 For a description of this parameter refer to Configuring the General Security Settings on page 90 SNMP Parameters The SNMP related ini file configuration parameters are described in the table below Parameter DisableSNMP SNMPPort SNMPTrustedMGR_x KeepAliveTrapPort SendKeepAliveTrap SNMPSysOid SNMPTrapEnterpriseOid acUserInputAlarmDescripti on acUserInputA
474. rameters are array parameters Each parameter configures a specific column in the IP routing table The first entry in each 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 63 RoutingTableDestination MasksColumn For a description of this parameter refer to Configuring the IP Routing Table on page 63 RoutingTableGatewaysCo lumn For a description of this parameter refer to Configuring the IP Routing Table on page 63 RoutingTableHopsCountC olumn For a description of this parameter refer to Configuring the IP Routing Table on page 63 RoutingTablelnterfacesCo For a description of this parameter refer to Configuring the IP lumn Routing Table on page 63 VLAN Parameters VLANMode For a description of this parameter refer to Configuring the IP Settings on page 52 SIP User s Manual 240 Document LTRT 65411 SIP User s Manual Parameter VLANNativeVLANID VLANOamVLANID VLANControlVLANI
475. ration followed by a Stutter tone Both tones are defined in the CPT file After this duration a dial tone is played The range is 1 000 to 60 000 The default is 2 000 i e 2 seconds Notes This parameter is applicable only to FXS interfaces The MWI tone takes precedence over the Call Forwarding Reminder tone For detailed information on MWI refer to Message Waiting Indication on page 361 143 November 2008 ca AudioCodes Parameter MWI Subscribe Retry Time SubscribeRetry Time Conference Parameters Enable 3 Way Conference Enable3WayConference Establish Conference Code ConferenceCode Conference ID ConferencelD MediaPack Series Description Subscription retry time in seconds after last subscription failure The default is 120 seconds The range is 10 to 7200 Enables or disables the 3 Way Conference feature 0 Disable Disable default 1 Enable Enables 3 way conferencing Defines the digit pattern which upon detection generates the Conference initiating INVITE when 3 way conferencing is enabled Enable3WayConference is set to 1 The valid range is a 25 character string The default is Hook Flash Defines the Conference Identification string up to 16 characters The default value is conf The device uses this identifier in the Conference initiating INVITE that is sent to the media server when Enable3WayConference is set to 1 For example ConferencelD
476. rect date and time client certificates cannot work gt 1 To enable two way client certificates take these 5 steps Set the parameter Secured Web Connection HTTPS to HTTPS Only 0 in Configuring the General Security Settings on page 90 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 Open the Certificates Signing Request page refer to Server Certificate Replacement on page 86 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 SIP User s Manual 88 Document LTRT 65411 SIP User s Manual 3 Web Based Management When the operation is complete set the ini file parameter HTTPSRequireClientCertificates to 1 Save the configuration refer to Saving Configuration on page 209 and then restart the device When a user connects to the secured Web server If the 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 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
477. red tab e Configuration refer to Configuration Tab on page 52 e Management refer to Management Tab on page 198 e Status 8 Diagnostics refer to Status amp Diagnostics Tab on page 218 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 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 23 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 26 To get Online Help for the currently opened page refer to Getting Help on page 47 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 80 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
478. refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 Determines the SIP response code for indicating call waiting 0 Use 182 Queued response to indicate call waiting default 1 Use 180 Ringing response to indicate call waiting For a description of this parameter refer to Supplementary Services on page 138 Sets the URI format in the SIP Diversion header 0 tel default 1 sip Determines whether the FXO device plays a Busy Reorder tone to the TDM side if a Tel to IP call is rejected by a SIP error response 4xx 5xx or 6xx The FXO device seizes the line off hook if a SIP error response is received and plays a Busy Reorder tone to the TDM side for the duration defined by the parameter TimeForReorderTone After playing the tone the line is released on hook 0 Disable default 1 Enable Determines whether the device plays a Comfort Tone Tone Type 18 to the FXS FXO endpoint after a SIP INVITE is sent and before a 18x response is received 0 Disable default 1 Enable Defines the duration in seconds for which Off Hook Warning Tone is played to the user The valid range is 1 to 2 147 483 647 The default is 600 Note A negative value indicates that the tone is played infinitely Dete
479. refer to Configuring the General Security Settings on page 90 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 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 nkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkkk ni 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 displayed 250 Document LTRT 65411 SIP User s Manual Parameter DisableWebConfig HTTPport ScenarioFileName Telnet Parameters TelnetServerEna
480. ress 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 370 in which it operates 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 373 For detailed information on configuring the multiple IP parameters refer to Networking Parameters on page 236 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 63 Th
481. rface 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 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 SIP User s Manual 58 Document LTRT 65411 SIP User s Manual 3 Web Based Management Figure 3 38 Application Settings Page w NTP Settings NTP Server IP Address 10 1 1 11 NTP UTC Offset Hours 0 Minutes NTP Updated Interval Hours 24 Minutes wv Telnet Settings Embedded Telnet Server Enable Unsecured Telnet Server TCP Port 23 Telnet Server Idle Timeout 0 SSH Server Enable Disable SSH Server Port 22 w DNS Settings l DNS Primar
482. rfaces Hook flash generation period upon detection of a SIP INFO message containing a hook flash signal FXO interfaces Hook flash generation period The valid range is 25 to 3 000 The default value is 700 Note For FXO interfaces a constant of 100 msec must be added to the required hook flash period For example to generate a 450 msec hook flash set this parameter to 550 3 4 2 6 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 47 Media Security Page General Media Security Settings Media Security Disable Media Security Behavior Preferable Disable Authentication On Transmitted RTP Packets 0 Disable Encryption On Transmitted RTP Packets 0 Disable Encryption On Transmitted RTCP Packets 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 209 Table 3 18 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 en
483. rformed 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 Table 3 19 Web User Accounts Access Levels and Privileges Numeric Representation Privileges Access Level Security 200 Administrator Read write privileges for all pages read write privileges for all pages except Administratar mm security related pages which are read only No access to security related and file loading pages read only access to the other pages Vaer Monitor an 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 20 Default Attributes for the Web User Accounts Account Attribute User Name Password Access Level Case Sensitive Case Sensiti
484. ription of this parameter refer to Configuring the FXO Parameters on page 195 For a description of this parameter refer to Configuring the FXO Parameters on page 195 For a description of this parameter refer to Configuring the FXO Parameters on page 195 For a description of this parameter refer to Configuring the FXO Parameters on page 195 For a description of this parameter refer to Configuring the FXO Parameters on page 195 For a description of this parameter refer to Configuring the FXO Parameters on page 195 Determines the type of Direct Inward Dialing DID signaling support for NTT Japan modem DTMF or Freguency Shift Keying FSK based signaling The devices can be connected to Japan s NTT PBX using Modem DID lines These DID lines are used to deliver a called number to the PBX 286 Document LTRT 65411 SIP User s Manual Parameter EnableDID EnableCallerlIDTypeTwo PolarityReversalType CurrentDisconnectDuration Version 5 6 4 ini File Configuration Description 0 FSK based signaling default 1 DTMF based signaling Note This parameter is applicable only to FXS interfaces This ini file table parameter enables support for Japan NTT Modem Direct Inward Dialing DID FXS interfaces can be connected to Japan s NTT PBX using Modem DID lines These DID lines are used to deliver a called number to the PBX The DID signal can be sent alone or combi
485. ription of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration 263 November 2008 V m C A AudioCodes Parameter RegisterOnlnviteFailure RegistrationTimeThreshold ZeroSDPHandling ForkingHandlingMode Account SIP User s Manual MediaPack Series Description Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 For a description of this parameter refer to Proxy 8 Registration Parameters on page 112 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 SDP 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 101 This ini file table parameter configures the Account table for registering and or
486. rise 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 Transmission Control Protocol Transmission Control Protocol Internet Protocol 388 Document LTRT 65411 SIP User s Manual Term TFTP TLS TON UA UDP URI SIP URIs VBD VLAN VoIP VoP VPN u Law Version 5 6 11 Glossary Meaning 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 Virtual Private Network A companding algorithm used in the digital telecommunication systems 389 November 2008 7a u wi AudioCodes CPE amp Access Analog Gateways SIP MediaPack MP 124 amp MP 11x User s Manual Version 5 6 7a m A wi AudioCodes vy www audiocodes com
487. rmines the index of the first Call Waiting Tone in the CPT file This feature enables the called party to distinguish between four different call origins e g external vs internal calls The device plays the tone received in the play tone CallWaitingTone parameter of an INFO message the value of this parameter 1 The valid range is 1 to 100 The default value is 1 not used Note It is assumed that all Call Waiting Tones are defined in sequence in the CPT file 270 Document LTRT 65411 SIP User s Manual Parameter RTPOnlyMode TimeoutBetween100And18x RxDTMFOption TxDTMFOption DisableAutoDTMFMute Version 5 6 4 ini File Configuration Description For a description of this parameter refer to Advanced Parameters on page 129 Defines the timeout in msec between receiving a 100 Trying response and a subseguent 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 For a description of this parameter refer to DTMF 8 Dialing Parameters on page 125 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
488. rmines 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 333 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 disabled 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
489. rogressIndicator2IP IPProfile EnableEchoCanceller 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 1 1 IPProfile_2 name2 55 5 55 5 55 55 55 5 40 IPProfile Notes This parameter can appear up to 9 times i e indices 1 9 Indicates 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 and 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 and IP to Hunt 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 173 Fora description of using ini file table parameters refer to Structure of ini File Table Parameters on page 233 This ini file table parameter configur
490. rs UID NFSServers GID NFSServers VlanType NFSServers For example NFSServers FORMAT NFSServers Index NFSServers HostOrlP 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 audio 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 62 Fora description of configuring ini file table parameters refer to Structure of ini File Table Parameters on page 233 243 November 2008 A c tal AudioCodes MediaPack Series 4 4 2 System Parameters The system related ini file configuration parameters are described in the table below Parameter EnableDiagnostics WatchDogStatus LifeLineType GWAppDelayTime 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 fa
491. s 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 gt To view the call routing status take this step m Open the Call Routing Status page Status amp Diagnostics tab gt Gateway Statistics menu gt Calls Routing Status page item Figure 3 115 Call Routing Status Page v Current Call Routing Method Current Proxy Not Used Current Proxy State Parameter Current Call Routing Method Current Proxy Current Proxy State SIP User s Manual Table 3 59 Call Routing Status Parameters Description Proxy Proxy server is used to route calls Routing Table preferred to Proxy The Tel to IP Routing table 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 112 Not Used Proxy server isn t defined IP address and FQDN if exists of the Proxy server with which the device currently operates 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 226 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 6 2 3 Registration Status The Registration Status page displays whether t
492. s 6 dBm fax gain control For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 For a description of this parameter refer to Configuring the Fax Modem CID Settings on page 69 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 101 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 removed only in the first half a second of the incoming IP signal 0 Disable default 1 Enable 299 November 2008 A c tal AudioCodes MediaPack Series Parameter FaxModemBypassBasicRT PPacketinterval FaxModemBypassDJBufMi nDelay EnableFaxModemInbandNe tworkDetection NSEMode NSEPayloadType SIP User s Manual Description Determines the basic frame size that is used during fax modem bypass sessions
493. s 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 as configured by the parameter ProxyKeepAliveTime If set to Using REGISTER the SIP REGISTER message is sent every user 122 Document LTRT 65411 SIP User s Manual Parameter Proxy Keep Alive Time ProxyKeepAliveTim e Is Proxy Hot Swap IsProxyHotSwap 3 4 4 1 4 Coders 3 Web Based Management Description 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 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 Enab
494. s 0 to 255 The default value is 35 N A Enables disables the Internet Protocol security IPSec on the device 92 Document LTRT 65411 SIP User s Manual Parameter Dead Peer Detection Mode IPSecDPDMode 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 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 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 DPD 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 sup
495. s are designed and tested to be fully interoperable with leading softswitches and SIP servers The device is best suited for small and medium sized enterprises SME branch offices or residential media gateway solutions The device enables users to make local or international telephone and or fax calls over the Internet between distributed company offices using their existing telephones and fax These calls are routed over the existing network ensuring that voice traffic uses minimum bandwidth The device also provides SIP trunking capabilities for Enterprises operating with multiple Internet Telephony Service Providers ITSP for VolP services The device supports the SIP protocol enabling the deployment of VolP solutions in environments where each enterprise or residential location is provided with a simple media gateway This provides the enterprise with a telephone connection i e RJ 11 connector and the capability to transmit voice and telephony signals over a packet network The device provides FXO and or FXS analog ports for direct connection to an enterprise s PBX FXO and or to phones fax machines and modems FXS Depending on model the device can support up to 24 simultaneous VoIP calls The device is also equipped with a 10 100Base TX Ethernet port for connection to the network The device provides LEDs for indicating operating status of the various interfaces The device is a compact unit that can be easily mounted on a d
496. s 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 To configure the SNMP community strings take these 5 steps 1 Access the Management Settings page as described in Configuring the Management Settings on page 199 2 In the SNMP Community String field click the right pointing arrow gt button the SNMP Community String page appears Figure 3 94 SNMP Community Strings Page Delete Community String Access Level Read Only Read Only Read Only m N n a Read Only Read Only Read Write Read Write Read Write Read Write i a e e 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 209 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 Version 5 6 203 November 2008 e AudioCo
497. s 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 114 Document LTRT 65411 SIP User s Manual Parameter 3 Web Based Management Description 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 reguest 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 RegistrarlP Registrar Transport Type RegistrarTransportType Version 5 6 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 Registration Time Note The device sends a REGISTER reguest for each channel or for the entire device accord
498. s the PBX extension that is mapped to the remote FXS port This section provides an example on how to implement a remote telephone extension through the IP network using 8 port FXO and 8 port FXS devices In this configuration the FXO device routes calls received from the PBX to the Remote PBX Extension connected to the FXS device The routing is transparent as if the telephone connected to the FXS device is directly connected to the PBX The following is required m One FXO device with ports connected directly to the PBX lines shown in the figure below One FXS device for the remote PBX extension Analog phones POTS PBX one or more PBX loop start lines LAN network Figure 7 10 FXO FXS Remote PBX Extension Example FXO Device FXS Device 10 1 10 2 y 10 1 10 3 PBX Line PBX Line Phone 100 Phone 101 vd none 8101 rom Phone 100 Phone 201 Remote PBX Extensions SIP User s Manual 348 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 7 13 4 1 7 13 4 2 7 13 4 3 Dialing from Remote Extension Phone at FXS The procedure below describes how to dial from the remote PBX extension i e phone connected to the FXS device gt To make a call from the FXS device take these 3 steps 1 Off hook the phone and wait for the dial tone from the PBX This is as if the phone is connected directly to the PBX The FXS and FXO devices establish a voice path connection from the pho
499. s 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 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 Lifeline is activated if enabled If the self test fails the device restarts to overcome internal fatal communication error Notes Enable LAN Watchdog is relevant only if the Ethernet connection is full duplex LAN Watchdog is not applicable to MP 118 136 Document LTRT 65411 SIP User s Manual Parameter Enable Calls Cut Through CutThrough Enable User Information Usage EnableUserlnfoUsage Out Of Service Behavior FXSOOSBehavior First Call Ringback Tone ID FirstCallRBTId Version 5 6 3 Web Based Management Description Enables users to receive incoming IP calls while the port is in off hook state 0 Disable Disabled default 1 Enable Enabled If enabled the FXS interface answers the c
500. sabled default 1 Enable Enable Digit Delivery feature for the FXO FXS device Notes The called number can include characters p 1 5 seconds pause and d detection of dial tone If character d is used it must be the first digit in the called number The character p can be used several 131 November 2008 A K tal AudioCodes MediaPack Series Parameter RTP Only Mode RTPOnlyMode Enable DID Wink EnableDIDWink Delay Before DID Wink DelayBeforeDIDWink Reanswer Time RegretTime Description times For example for FXS FXO interfaces the called number can be as follows d1005 dpp699 p9p300 To add the d and p digits use the usual number manipulation rules To use this feature with FXO interfaces configure the device to operate in one stage dialing mode If this parameter is enabled it is possible to configure the FXS FXO interface to wait for dial tone per destination phone number before or during dialing of destination phone number Therefore the parameter IsWaitForDialTone configurable for the entire device is ignored The FXS interface send SIP 200 OK responses only after the DTMF dialing is complete 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
501. sage determined by the parameter CallerlDType which is displayed on the MWI display Note This parameter is applicable only to FXS interfaces Enables subscription to an MWI server 0 No Disables MWI subscription default 1 Yes Enables subscription to MWI to MWIServerlP address Note Use the parameter SubscriptionMode described in Proxy amp Registration Parameters on page 112 to determine whether the device subscribes per endpoint or per the entire device MWI server s IP address If provided the device subscribes to this IP address The MWI server address can be configured as a numerical IP address or as a domain name If not configured the Proxy IP address is used instead Determines the transport layer used for outgoing SIP dialogs initiated by the device to the MWI 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 The MWI subscription expiration time in seconds The default is 7200 seconds The range is 10 to 72000 Duration in msec of the played stutter dial tone indicating enabled Call Forward or waiting message s The Stutter tone is played instead of a regular dial tone when Call Forward is enabled on the specific port or when Message Waiting Indication MWI is received The tone is composed of a Confirmation tone which is played for a user defined duration StutterToneDu
502. se 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 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 i
503. selected 6 In the Payload Type field if the payload type for 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 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 209 3 4 4 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 Endpoint Phone Number Table page thereby applying different behaviors to different channels i e ports Note You can also configure Tel Profiles using the ini file table parameter TelProfile refer to SIP Configuration Parameters on page 260 Version 5 6 171 November 2008 7a K tal AudioCodes MediaPack Series 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 77 Tel Profile Settings Screen v Pro
504. 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 129 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Advanced Parameters on page 129 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 Indication 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 245 November 2008 A K tal AudioCodes MediaPack Series Parameter RAlLowThreshold RAILoopTime Description 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 dividin
505. sents any number represents any number i e all numbers asterisk 3 4 4 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 including numbers that start with 551 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 Tel 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 Version 5 6 155 November 2008 7a c tal AudioCodes MediaPack Series gt Toconfigure 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 Ta
506. ser Version 5 6 3 Web Based Management Description endpoints from being registered by assigning them to a Hunt Group and configuring the Hunt 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 188 Notes To enable Hunt Group registrations configure the global parameter IsRegisterNeeded to 1 This is unnecessary for Per Account registration mode f no mode is selected the registration is performed according to the global registration parameter ChannelSelectMode refer to Proxy 8 Registration Parameters on page 112 If the device is configured globally ChannelSelectMode to register Per Endpoint and a Hunt Group comprising four FXO endpoints is configured to register Per Gateway the device registers all endpoints except the first four endpoints The Hunt Group of these four endpoints sends a single registration request The Serving IP Group ID to where INVITE messages initiated by this Hunt Group s endpoints are sent The actual destination to where these INVITE messages are sent is to the Proxy Set ID refer to Proxy Sets Table on page 120 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 i
507. should be executed during maintenance time Open the Certificates page refer to Server Certificate Replacement on page 86 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 Save configuration refer to Saving Configuration on page 209 and then restart the device for the new certificate to take effect 89 November 2008 A K tal AudioCodes MediaPack Series 3 4 3 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 steps 1 Open the General Security Settings page Configuration tab gt Security Settings menu gt General Security Settings page item Figure 3 54 General Security Settings Page v HTTP Authentication Mode Digest When Possible i Secured Web Connection HTTPS HTTP and HTTPS Voice Menu Password 12345 v General RADIUS Setting Enable RADIUS Access Control Disable Use RADIUS for Web Telnet Login Disable RADIUS Authentication Server IP Address 0 0 0 0 RADIUS Authentication Server Port 1645 RADIUS Shared Secret eeccccee w General RADIUS Authentication Default Access Level 200 Device Behavior Upon RADIUS Timeout Verify Access Locall
508. sign a name to the port refer to Assigning a Name toa Port on page 49 Uplink MP 11x If clicked the Ethernet Port Information page opens displaying Ethernet port LAN MP 124 configuration settings refer to Viewing Ethernet Port Information on page 220 Fail Currently not supported Ready Currently not supported Power Always lit green indicating power received by the device 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 28 Shortcut Menu when Clicking Port e g MP 11x Uplink 2 From the shortcut menu choose Update Port Info a text box appears Figure 3 29 Text Box for Typing Port Name e g MP 11x Port name or description ApplyPortlnfo 3 Type a brief description for the port and then click Apply Port Info Version 5 6 49 November 2008 A c tal AudioCodes MediaPack Series 3 3 8 2 Viewing Analog Port Information The Home page allows you to view detailed information on a specific FXS or FXO analog port such as RTP RTCP and voice settings gt To view detailed port information take these 3 steps 1 Click the port for which you want to view port settings the shortcu
509. son Header Enable v wv Retransmission Parameters SIP T1 Retransmission Timer msec 500 SIP T2 Retransmission Timer msec 4000 SIP Maximum RTX 7 Version 5 6 101 November 2008 A tal AudioCodes MediaPack Series 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 209 Table 3 27 SIP General Parameters Protocol Definition Parameter PRACK Mode PRACKMode Channel Select Mode ChannelSelectMode Enable Early Media EnableEarlyMedia 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 Port channel allocation algorithm for IP to Tel calls 0 By Dest Phone Number Selects the device s channel according to the called number defined in the Endpoint Phone Number table Configuring the Endpoint Phone Numbers on page 181 default 1 Cyclic Ascending Selects the next available channel in an ascending cyclic order Always selects the next higher channel number in the hunt group When the device reaches the highest channel number in the hunt group it
510. sparent 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 TxDTMF Option or RxDTMFOption Not Applicable 68 Document LTRT 65411 SIP User s Manual Parameter DTMF Volume 31 to 0 dB DTMFVolume Enable Answer Detector EnableAnswerDetector Answer Detector Activity Delay AnswerDetectorActivityDelay Answer Detector Silence Time AnswerDetectorSilenceTime Answer Detector Redirection AnswerDetectorRedirection Answer Detector Sensitivity AnswerDetectorSensitivity DTMF Generation Twist DTMF GenerationTwist Version 5 6 3 Web Based Management Description DTMF gain control value in decibels to the or analog side The valid range is 31 to 0 dB The default value is 11 dB 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 Defines the range in decibels between the high and low freguency components in the DTMF signal Positive decibel values cause the higher freguency component to be stronger than the lower one Negative values cause the opposite effect For any parameter value both components change so that
511. sponse 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 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 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 X DETECT e Type CPT FAX PTT e Subtype xxx according to the defined subtypes of each type SIP User s Manual 332 Document LTRT 65411 SIP User s Manual 7 IP Telephony Capabilities 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 anonymouseanonymous invalid gt tag 1c25298 To lt Sip 101 10 33 2 53 user phone gt Calil 1p3 UO 0 334258 CSeq 1 INVITE Contact lt sip 100 10 33 2 53 gt X Detect Request CPT FAX SEP 20200 OK Via SIP 2 0 UDP 10 33 2 53 branch z9hG4bKac5906 From anonymous lt sip anonymouseanonymous invalid gt tag 1c25298 To lt sip 101e10 33 2 53 user phone gt tag 1c19282 C
512. st be located in the same folder as the ini file For a detailed description on BootP refer to the Product Reference Manual 41 November 2008 A tall AudioCodes MediaPack Series 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 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 Microsoft Internet Explorer A 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 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 41 Loading an ini file with the ScenarioFileName parameter set to no value i e ScenarioFileName SIP User s Manual 42 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 3 5 7 3 3 6 3 3 6 1 Exiting Scenario Mode When you wan
513. st x refer to Web and Telnet Parameters on page 249 SIP User s Manual 82 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 49 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 8 Telnet Access List table Figure 3 50 Web amp Telnet Access List Table Delete Authorized IP Row Address Delete Selected Addresses Note Delete all rows to allow access from any IP address to WEB Telnet 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 209 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 ad
514. stHostPrefix PstnPrefix SrcHostPrefix 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 Hunt Group Routing Table on page 163 To support the In Call Alternative Routing feature you can use two entries that support the same call but assigned with a different HuntGroup 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 Hunt Groups for IP to Tel calls is according to 291 November 2008 K tal AudioCodes MediaPack Series Parameter RemovePrefix RouteModelP2Tel RouteModeTel2IP SourceManipulationMod e SwapTel2IPCalled amp Calli ngNumbers AddTON2RPI NumberMapTel2IP SIP User s Manual Description 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 8 255 If the source IP address SourceAddress includes an
515. 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 file name can be up to 47 characters The logo file name can be used to replace AudioCodes default Web logo with a user defined logo Security Parameters The security related ini file configuration parameters are described in the table below Parameter EnableMediaSecurity MediaSecurityBehaviour SRTPTxPacketMKISize RTPAuthenticationDisableT x RTPEncryptionDisableTx RTCPEncryptionDisableTx EnableSIPS TLSLocalSIPPort TLSVersion TLSReHandshakelnterval SIP User s Manual Table 4 4 Security ini File Parameters Description For a description of this parameter refer to Configuring Media Security on page 78 For a description of this parameter refer to Configuring Media Security on page 78 For a description of this parameter refer to Configuring Media Security on page 78 For a description of this parameter refer to Configuring Media Security on page 78 For a description of this parameter refer to Configuring Media Security on page 78 For a description of this parameter refer to Configuring Media Security o
516. t 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 1 To change the user name of an account perform the following a In the 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 Version 5 6 81 November 2008 A tal AudioCodes MediaPack Series 2 To change the password 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 attribut
517. t CallWaitingPerPort 0 0 1 1 CallWaitingPerPort 1 1 2 1 CallWaitingPerPort If enabled when an FXS interface receives a call on a busy endpoint it responds with a 182 response and not with a 486 busy The device plays a call waiting indication signal When hook flash is detected the device switches to the waiting call The device that initiates the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received Notes If this parameter is not configured default use the global parameter EnableCallWaiting refer to Supplementary Services on page 138 The numbering of channels starts at 0 This parameter can appear up to eight times for 8 port devices and up to 24 times for MP 124 The device s Call Progress Tones file must include a call waiting Ringback tone caller side and a call waiting tone called side FXS interfaces only The EnableHold parameter must be enabled on both the calling and the called sides To define call waiting using the Web interface refer to Call Waiting on page 361 Fora description on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 Defines the mode of operation when the 3 Way Conf
518. t 10 8 201 108 200 OK SIP 2 0 200 OK 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 CSeq 18153 INVITE Contact lt sip 1000 10 8 201 10 user phone gt Server Audiocodes Sip Gateway MediaPack 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 wal o AudiocodesGW 30221 87035 IN IP4 10 8 201 10 s Phone Call GSN We 2108210 10 ic 0 0 m audio 7210 RTP AVP 8 96 a rtpmap 8 pcma 8000 a ptime 20 a rtpmap 96 telephone event 8000 ESemMros96 W 15 m F5 10 8 201 108 gt gt 10 8 201 10 ACK Version 5 6 343 November 2008 tall AudioCodes MediaPack Series 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 MediaPack v 5 40 010 006 CSeq 18153 ACK 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 mM F6 10 8 201 108 gt gt 10 8 201 10 BYE BYE sip 1000 10 8 201 10 user phone SIP 2 0
519. t Ring Off Time 10 msec Ring Off period in 10 msec units for the fourth cadence on off cycle In SIP the distinctive ringing pattern is selected according to Alert Info header that is included in the INVITE message For example Alert Info lt Bellcore dr2 gt or Alert Info lt http Bellcore dr2 gt dr2 defines ringing pattern 2 If the Alert Info header is missing the default ringing tone 0 is played SIP User s Manual 310 Document LTRT 65411 SIP User s Manual 6 Auxiliary Configuration Files 6 2 1 Examples of Ringing Signals m Below is an example of a ringing burst Three ringing bursts followed by repeated ringing of 1 sec on and Bsc Corian NUMBER OF DISTINCTIVE RINGING PATTERNS Number of Ringing Patterns 1 Ringing Pattern 0 Ring Type 0 Freq Hz 25 First Burst Ring On Time 10msec 30 First Burst Ring Off Time 10msec 30 Second Burst Ring On Time 10msec 30 Second Burst Ring Off Time 10msec 30 Third Burst Ring On Time 10msec 30 Third Burst Ring Off Time 10msec 30 Fourth Ring On Time 10msec 100 Fourth Ring Off Time 10msec 300 m Below is an example of various ringing signals NUMBER OF DISTINCTIVE RINGING PATTERNS Number of Ringing Patterns 3 Regular North American Ringing Pattern Ringing Pattern 0 Ring Type 0 Freq Hz 20 First Ring On Time 10msec 200 First Ring Off Time 10msec 400 GR 506 CORE Ringing Pattern 1 Ringing Pa
520. t Defined Fourth Proposal Encryption Type Not Defined Fourth Proposal DH Group Not Defined 2 First Proposal Authentication Type HMAC SHA 1 35 First Proposal DH Group DH 1024 B1T Second Proposal Authentication Type Not Defined Second Proposel DH Group Not Defined Third Proposal Authentication Type Not Defined Fourth Proposal Authentication Type Not Defined Authentication Method Pre zhared Key Shared Key IKE SA LifeTime sec IKE SA LifeTime KB SIX ISSCC Gis Click the Apply button to load the certificates future IKE negotiations are now performed using the new certificates 3 4 3 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 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 369 to obtain the current date and time Without the cor
521. t 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 AccessList 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 If 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 84 256 Document LTRT 65411 SIP User s Manual Parameter AccessList MatchCount 4 4 5 4 ini File Configuration Description Fora description of configuring with ini file table parameters refer to Structure of ini File Table Parameters on page 233 For a description of this parameter refer to Configuring the
522. t menu appears Figure 3 30 Shortcut Menu when Clicking Port Port Settings e g MP 11x Uplink Q Ready Power 2 From the shortcut menu click Port Settings the Basic Channel Information page appears Figure 3 31 Basic Channel Information Page SIP Basic RTP RTCF Voice Settings v Channel Identifier 4 Status Inactive Call ID 0 Endpoint ID Call Duration sec 0 Call Type Voice Call Destination 10 13 4 13 Coder G71lAlaw 64 Last Current Disconnect Duration Line Current mA Line Voltage V Hook 0 Onhook 1 Off hook Ringf0 Off 1 On Line Connected 0 Disconnected 1 Connected Polarity state O Normal 1 Reversed 2 M A Line polarity O Positive 1 Negative Message Waiting Indication 0 Off 1 On ol olo olol l ol ol olo 3 To view RTP RTCP or voice settings click the relevant button 3 3 8 3 Resetting an Analog Channel The Home page allows you to inactivate reset an FXO or FXS analog channel This is sometimes useful in scenarios for example when the device FXO is connected to a PBX and the communication between the two can t be disconnected e g when using reverse polarity gt To reset a channel take this step m Click the required FXS or FXO port icon and then from the shortcut menu choose Reset Channel the channel is changed to inactive i e the port icon is displayed in grey SI
523. t settings 1 Enable default 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 304 Document LTRT 65411 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 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 rem
524. t to close the Scenario mode after using it for device configuration follow the procedure below Toclose 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 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 43 m Product s name displayed on the Title bar refer to Customizing the Product Name on page 46 m Login welcome message refer to Creating a Login Welcome Message on page 46 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 44 or text refer to Replacing the Corporate Logo with Text on page 45 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 cust
525. ted allowed per port using keypad features KeyCLIR and KeyCLIRDeact FXS only AssertedidMode defines the header that is used in the generated INVITE request to deliver the caller ID P Asserted Identity or P preferred Identity Use the parameter UseTelURIForAssertedID to determine the format of the URI in these headers sip or tel EnableRPlheader enables Remote Party ID RPI headers for calling and called numbers for Tel to IP calls SIP User s Manual 364 Document LTRT 65411 SIP User s Manual 8 Networking Capabilities 8 8 1 8 2 Networking Capabilities Ethernet Interface Configuration The device s Ethernet connection can be configured using the ini file parameter EthernetPhyConfiguration for one of the following modes m Manual mode e 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 t
526. teps 1 Open the IP Routing Table page Configuration tab gt Network Settings menu gt IP Routing Table page item Figure 3 40 IP Routing Table Page I Destination IP Address Destination Mask Gateway IP Address Metric Interface f Delete Row Delete Selected Entries lt Add a new tabl entry 00 Destination IP Address Destination Mask Gateway IP Address Metric 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 Routing TableDestinationsColumn network Destination Mask Specifies the subnet mask of the destination host RoutingTableDestinationMasksColumn network 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 i
527. ter refer to Configuring the SNMP Community Strings on page 203 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 Group 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 ini file table refer to Configuring SNMP V3 Users on page 204 Foran explanation on using ini file table parameters refer to Structure of ini File Table Parameters on page 233 259 November 2008 ca AudioCodes 4 4 7 MediaPack Series SIP Configuration Parameters The SIP related ini file configuration parameters are described in the table below Parameter ReliableConnectionPersistent Mode SIPTransportType TCPLocalSIPPort SIPDestinationPort EnableTCPConnectionReuse SIPTCPTimeout LocalSIPPort EnableFaxR
528. ters only if the OAM networking parameters are different from the networking parameters used in the Single IP Network mode SIP User s Manual 374 Document LTRT 65411 SIP User s Manual 8 Networking Capabilities 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 a Open the IP Routing Table page refer to Configuring the IP Routing Table on page 63 Figure 8 6 Static Routes for OAM Control in IP Routing Table Hop Count Delete Destination IP Gateway IP Row Address Destination Mask Address TTL Interface O 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 209 and reset the device refer to Resetting the Device on page 207 8 8 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 Getting Started with VLANS and Multiple IPs on page 373 gt To integrate the device into a multiple IPs network withVLANs using the ini file take
529. tes 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 209 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 uniguely identify the device The server certificate can also be loaded via ini file using the parameter HTTPSCertFileName Version 5 6 87 November 2008 7a c tal AudioCodes MediaPack Series gt To apply the loaded certificate for IPsec negotiations take these 2 steps Open the IKE Table page refer to Configuring the IKE Table on page 97 the Loaded Certificates Files group lists the newly uploaded certificates as shown below Figure 3 53 IKE Table Listing Loaded Certificate Files Loaded Certificate Files hd Server Certificate File Loaded Trusted Root File Loaded Lappy Policy Index O State Exasts First Proposal Encryption Type Trige DES CBC Second Proposal Encryption Type Not Defined Third Proposal Encryption Type Not Defined Third Proposal DH Group No
530. th the IP Group All INVITE messages configured to be sen 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 The reguest 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 8 Registration Parameters on page 112 is used instead The value range is a string of up to 49 characters The default is an empty field Defines the user part for the From To and Contact headers of SIP REGISTER messages and the user part for the Contact header of 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 188 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 n
531. 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 5 The default value is 1 i e no redundant Proxy Set 3 4 4 3 Configuring 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 m 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 For detailed information on Caller ID refer to Caller Display Information on page 177 Version 5 6 151 November 2008 7a e AudioCodes MediaPack Series The number manipulation is configured in the following tables m For Tel to IP calls e Destination Phone N
532. the Metering tones take these 4 steps 1 Open the Metering Tones page Configuration tab gt Protocol Configuration menu gt SIP Advanced Parameters submenu gt Metering Tones page item Figure 3 64 Metering Tones Page vw Generate Metering Tones Disable Metering Tone Type 16 KHz Charge Codes Table 2 Configure the Metering tones parameters 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 209 In the Tel to IP Routing table refer to Tel to IP Routing Table on page 160 assign a charge code rule to the routing rules you require When a new call is established the Tel to IP Routing table is searched for the destination IP addresses Once a route is found the Charge Code configured for that route is used to associate the route with an entry in the Charge Codes table Table 3 34 Metering Tones Parameters Parameter Description Generate Metering Tones Determines the method used to configure the metering tones that are PayPhoneMeteringMode generated to the Tel side 0 Disable Metering tones aren t generated default 1 Internal Table Metering tones are generated according to the internal table configured by the parameter ChargeCode Notes This parameter is applicable only to FXS interfaces If you select Internal Table you must configure the Ch
533. the changes to flash memory refer to Saving Configuration on page 209 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 101 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 70 Document LTRT 65411 SIP User s Manual Parameter Caller ID Transport Type CallerlIDTransportType Caller ID Type CallerlIDType V 21 Modem Transport Type V21ModemTransportTy pe V 22 Modem Transport Type V22ModemTransportTy pe Version 5 6 3 Web Based Management Description 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 from the voice stream default Defines one of the following standards for detection FXO and generation FXS of Caller ID and detection FXO and generation FXS of MWI when specified signals 0 Bellcore Caller ID and MWI default 1
534. 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 in the 0 931 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 8 NEC ICS Note When the RS 232 connection is used for SMDI messages Serial SMDI it cannot be used for other 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 5 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 Con
535. the software for MP 11x FXS devices Sample ini file for MP 124 FXS device Sample ini file for MP 118 FXS devices Sample ini file for MP 114 FXS devices Sample ini file for MP 112 FXS devices Telephony interface configuration file for MediaPack FXS devices Default loadable Call Progress Tones dat file Call Progress Tones ini file used to create dat file TrunkPack Downloadable Conversion Utility to create Call Progress Tones files Syslog server BootP TFTP configuration utility Call Progress Tones Wizard MIB library for SNMP browser 379 November 2008 A tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 380 Document LTRT 65411 SIP User s Manual 10 Selected Technical Specifications 10 Selected Technical Specifications The main technical specifications of the MP 11x and MP 124 devices are listed in the following subsections Note All specifications in this document are subject to change without prior notice 10 1 MP 11x Specifications The table below lists the main technical specifications of the MP 11x Table 10 1 MP 11x Functional Specifications Function Specification Channel Capacity Available Ports MP 112 2 ports MP 114 4 ports MP 118 8 ports The MP 112 differs from the MP 114 and MP 118 Its configuration excludes the RS 232 connector the Lifeline option and outdoor protection MP 11x FXS Functionality FXS Capabilities Short or Long Haul A
536. 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 VlanNativeVlanld 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 Version 5 6 375 November 2008 A tal AudioCodes MediaPack Series 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 0 O VW B3LUW 50 16 OsO 0 0 4 OANA InterfaceTable 0 i 10 33 174150 16 10 33 01 6 Wechies InterfaceTable 0 2 10 32 174 50 16 0 0 0 0 5 Comezolls InterfaceTable VLAN related parameters VlanMode 1 VlanNat
537. ti ons User Name UserName Password Password SIP User s Manual MediaPack Series Description 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 NAPTR query is not performed Note When enabled NAPTR SRV queries are used to discover Proxy servers even if the parameter DNSQueryType is disabled Determines the method the device uses to subscribe to an MWI server 0 Per Endpoint Each endpoint subscribes separately typically used for FXS interfaces default 1 Per Gateway Single subscription for the entire device typically used for FXO interfaces 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 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 th
538. 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 SIP 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 105 November 2008 ca AudioCodes Parameter Use user phone in From Header IsUserPhonelnFrom Use Tel URI for Asserted Identity UseTelURIForAssertedl D 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 MediaPack Series 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 1
539. 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 are 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 Identity 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 Tel or configured in the device the From header is set to lt anonymous anonymous invalid gt 103 November 2008 A tal AudioCodes MediaPack Series Parameter Description Fax Signaling Method Determines the SIP signaling method for establishing and transmitting a IsFaxUsed fax
540. tion The method in which IP to Tel calls are assigned to channels pertaining to a Hunt Group 0 By Dest Phone Number Selects the device s channel according to the called number defined in the Endpoint Phone Number refer to Configuring the Endpoint Phone Numbers on page 181 1 Cyclic Ascending default Selects the next available channel in an ascending cyclic order The next highest channel number in the Hunt Group is always selected When the highest channel number in the Hunt Group is reached the lowest channel number in the Hunt Group is selected and then it starts ascending again 2 Ascending Selects the lowest available channel The lowest channel number in the Hunt 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 Hunt Group is always first selected When the lowest channel number in the Hunt Group is reached it selects the highest channel number in the Hunt Group and then start descending again 4 Descending Selects the highest available channel The highest channel number in the Hunt 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
541. tion Durat Call Duration Coder Selected Coder Version 5 6 339 November 2008 ca AudioCodes Field Name Intrv Rtplp Port TrmSd TrmReason Fax InPackets OutPackets PackLoss RemotePackLoss Uniqueld SetupTime ConnectTime ReleaseTime RTPdelay RTPjitter RTPssrc RemoteRTPssrc RedirectReason TON MeteringPulses NPI RedirectPhonNum MediaPack Series Description Packet Interval RTP IP Address Remote RTP Port Initiator of Call Release 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 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 Proxy or Registrar Registration Example Below is an example of Proxy and Registrar Registration ER sip servername SIP 2 0 SIP 2 0 UDP 212 179 22 229 branch z9hG4bRaC7AU2 34 RegistrationName sipgatewayname gt tag 1c29347 lt sip GWRegistrationName sipgatewayname gt 3821217922229 ER 7 12 REGIST VIA From lt sip GW TOs Calh E TD TOAS Seq 1 REGIST Expires 3600 Contact SEs Or Contents Tenge GWRegistrationName 212 179 22 229 ag The servername m The servern SIP Use
542. ts 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 8 Diagnostics on page 219 m Gateway Statistics refer to Gateway Statistics on page 223 SIP User s Manual 218 Document LTRT 65411 SIP User s Manual 3 Web Based Management 3 6 1 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 Message Log refer to Viewing the Device s Syslog Messages on page 219 Ethernet Port Information refer to Viewing Ethernet Port Information on page 220 Active IP Interfaces refer to Viewing Active IP Interfaces on page 220 Device Information refer to Viewing Device Information on page 221 Performance Statistics refer to Viewing Performance Statistics on page 222 Active Alarms refer to Viewing Active Alarms on page 222 3 6 1 1 Viewing the Device s Syslog Messages The Message Log page displ
543. ts 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 Version 5 6 369 November 2008 7a e AudioCodes MediaPack Series 8 8 8 8 1 8 8 2 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 236 For the mapping of an application to its class of service refer to IEEE 802 1p Q VLANs and Priority on page 370 VLANS and Multiple IPs Multiple IPs Media Control and Management OAMP traffic in the device can be assigned one of the following IP addressing schemes m Single IP address for all traffic i e for Media Control and OAMP m Separate IP add
544. ttern 1 Ring Type 1 Freq Hz 20 First Ring On Time 10msec 200 First Ring Off Time 10msec 400 GR 506 CORE Ringing Pattern 2 Ringing Pattern 2 Ring Type 2 Freq Hz 20 First Ring On Time 10msec 80 First Ring Off Time 10msec 40 Second Ring On Time 10msec 80 Second Ring Off Time 10msec 400 6 3 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 Version 5 6 311 November 2008 7a e AudioCodes MediaPack Series Note The Prerecorded tones are used only for generation of tones Detection of tones is performed according to the CPT file 6 4 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 Produ
545. ttings are applied Version 5 6 33 November 2008 7a c tal AudioCodes MediaPack Series 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 index entry 6 is re assigned index number 2 Figure 3 12 Compacting a Web Interface Table PrefixLength Gateway VlanID InterfaceName Duplicate Compact Index ApphcationTypes IPv6InterfaceMode IPAddress PrefixLength Gateway VianIO InterfaceName 3 3 4 To delete an existing index table entry take these 3 steps In the Index column select the index corresponding to the table row that you want to delete Click Delete the table row is removed from the table 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 To search for ini file parameters configurable in the Web interface take these 4 steps On the Na
546. type for CN 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 Determines the analog signal transport type 0 Ignore Analog Signals Ignore default 1 RFC2833 Analog Signal Relay Transfer hookflash via RFC 2833 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 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 75 November 2008 ca AudioCodes Parameter Remote RTP Base UDP Port RemoteBaseUDPPort RTP Multiplexing Local UDP Port L1L1ComplexTxUDPPort RTP Multiplexing Remote UDP Port L1L1ComplexRxUDPPort MediaPack Series Description 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 Dete
547. udioCodes Table 3 33 Supplementary Services Parameters Parameter Enable Hold EnableHold Hold Format HoldFormat Held Timeout HeldTimeout Call Hold Reminder Ring Timeout CHRRTimeout Enable Transfer EnableTransfer Transfer Prefix xferPrefix SIP User s Manual MediaPack Series Description Allows users connected to the device to place a call on hold 0 Disable Disables the Hold service 1 Enable Enables the Hold service default If the Hold service is enabled a user can place the call on hold or remove from hold using the hook flash On receiving a Hold request the remote party is placed on hold and hears the hold tone Note To use this service the devices at both ends must support this option 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 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 seconds after which the call is released Defines the timeout in seconds for applying the Call Hold Reminder Ring If a user hangs up
548. ues the following mechanisms are available m STUN refer to STUN on page 366 m First Incoming Packet Mechanism refer to First Incoming Packet Mechanism on page 367 m RTP No Op packets according to the avt rtp noop draft refer to No Op Packets on page 367 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 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
549. umber 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 m 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 SourceNumberMapIP2Tel 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 Hunt 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 163 and Tel to IP Routing Mode parameter RouteModeTel2IP described in Tel to IP Routing Table on page 160 For configuring number manipulation using ini file table parameters NumberMapIP2Tel NumberMapTel2IP SourceNumberMapIP2Tel and SourceNumberMapTel2IP refer to Number Manipulation and Routing Parameters on page 289 gt Toconfigure 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
550. umber collection starts Option 2 A Hook Flash is pressed the current call is put on hold a dial tone is played to the phone and then digit collection starts After this sequence is identified the device continues the collection SIP User s Manual 148 Document LTRT 65411 SIP User s Manual 3 Web Based Management Parameter Description of the destination phone number For both options after the phone number is collected it s sent to the transferee in a SIP REFER reguest without a Replaces header The call is then terminated and a confirmation tone is played to the phone If the phone number collection fails due to a mismatch a reorder tone is played to the phone Notes This parameter is applicable to FXO and FXS interfaces but for FXO the Web interface does not display this parameter It is possible to configure whether the KeyBlindTransfer code is added as a prefix to the dialed destination number by using the parameter KeyBlindTransferAddPrefix refer to Channel Parameters on page 298 Call Waiting Note The call waiting can be viewed in the Call Waiting table refer to Call Waiting on page 361 Activate Keypad seguence that activates the Call Waiting option After the KeyCallWaiting seguence is pressed a confirmation tone is heard Deactivate Keypad seguence that deactivates the Call Waiting option After the KeyCallWaitingDeact seguence is pressed a confirmation tone is heard Reject Anony
551. und e g 21 then before the call is routed to the corresponding Hunt 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 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 158 Document LTRT 65411 SIP User s Manual Parameter Enable Alt Routing Tel to IP AltRoutingTel2IPEnable Alt Routing Tel to IP Mode AltRoutingTel2IPMode Alt Routing Tel to IP Connectivity Method AltRoutingTel2IPConnMeth od Alt Routing Tel to IP Keep Alive Time AltRoutingTel2IPKeepAlive Time Alternative Routing Tone Duration ms AltRoutingToneDuration Version 5 6 3 Web Based Management 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 Alternat
552. uration k o meee SIP Trunking Example Archie itt uza kazde ka deasa du t ea dd dada ud dne K ES ke K dna zku uakka Oe Configuring Proxy Set ID 1 in the Proxy Sets Table Page ded ok bike ad oka od ad dk k 354 Configuring IP Groups 1 and 2 in the IP eske Table E EPE E ET 354 Configuring Hunt Groups haa dean tadeeee 354 Configuring Hunt Groups Settings 355 Figure 7 24 Configuring Username and Password for Channels 58 i in n Authentication Page sicnatade 355 Figure 7 25 Configuring Accounts A ints B00 Figure 7 26 Configuring IP to Hunt Group Routing 00 Figure 7 27 Configuring Tello1P PROMO 346 ada sedi u d da a bud zla u dc dle dn diod u n 356 Figure 2223 Double Rold SIP Call FOM chs na sud dovedu da a d 358 Fouc e i Nat EO MOK z zd bla o oto Ako o de ae dn A 366 Figure 8 2 Multiple Network Interfaces and VLANs wet Figure 8 3 VLAN Configuration in the IP Settings Page ARE dak ud la 00 T Figure 8 4 OAM Control Media IP Configuration in the IP Settings PAGE umu gt 374 Figure 8 5 Multiple Interface Table Page ee errr errs 374 Figure 8 6 Static Routes for OAM Control in n IP Routing PRA aale tateutaassaasca MMS SIP User s Manual 10 Document LTRT 65411 SIP User s Manual Contents List of Tables ED O od o ho aa WO Table e Table lt Table 3 2 Table Table 3 Table lt Table Table Table Table Table Version 5 6 1 November 2008 Gg
553. urceNumberMapTel2lp_Suffix2Add SourceNumberMapTel2Ip IsPresentationRestricted NumberMapTel2Ip SreTrunkGroupID NumberMapTel2Ip SrcIPGroupID SourceNumberMapTel2Ip For example SourceNumberMapTel2Ip 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 The parameters SourceNumberMapTel2Ip NumberType SourceNumberMapTel2lp_NumberPlan are not applicable Set these to RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add LeaveFromRight NumberType NumberPlan and IsPresentationRestricted are applied if the called and calling numbers match the DestinationPrefix and SourcePrefix conditions 294 Document LTRT 65411 SIP User s Manual 4 ini File Configuration Parameter SourceNumberMapIP2Te I Version 5 6 Description 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 To configure manipulation of source numbers for Tel to IP calls using the Web interface refer to Configuring the Number Manipulation Tables on page 151 F
554. us lt anonymous anonymous invalid gt e The Privacy id header is included e The P Asserted Identity or P preferred Identity header shows the caller ID m f Caller ID is allowed e The From header shows the caller ID e The Privacy none header is included e The P Asserted Identity or P preferred Identity header shows the caller ID In addition the caller ID and presentation can be displayed in the Calling Remote Party ID header The Caller Display Information table CallerDisplayInfo is used for the following m FXS interfaces to define the caller ID per port that is sent to IP FXO interfaces to define the caller ID per port that is sent to IP if caller ID isn t detected on the Tel side or when EnableCallerlD 0 m FXS and FXO interfaces to determine the presentation of the caller ID allowed or restricted m To maintain backward compatibility when the strings Private or Anonymous are set in the Caller ID Name field the caller ID is restricted and the value in the Presentation field is ignored The value of the Presentation field that is defined in the Caller Display Information table can be overridden by configuring the Presentation parameter in the Tel to IP Source Number Manipulation table Therefore this table can be used to set the presentation for specific calls according to Source Destination prefixes The caller ID can be restric
555. uting 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 If this parameter is enabled the Outbound IP Routing Table page appears instead refer to Outbound IP Routing Table for a description of this page This routing table associates called and or calling telephone number prefixes originating from a specific Hunt 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 server 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 E 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 n
556. utomatic Detection Ringer Equivalency Number REN 3 per FXS port up to 9 km 30 000 feet using a 24 AWG line Note The lines have been tested under the following conditions ring voltage greater than 30 Vrms offhook loop current greater than 20 mA all lines ring simultaneously Lightning and high voltage protection for outdoor operation Caller ID generation Bellcore GR 30 CORE Type 1 using Bell 202 FSK modulation ETSI Type 1 NTT Denmark India Brazil British and DTMF ETSI CID ETS 300 659 1 Programmable Line Characteristics Battery feed line current hook thresholds AC impedance matching hybrid balance Tx amp Rx frequency response Tx amp Rx Gains Note For a specific coefficient file please contact AudioCodes Configurable ringing signal up to four cadences and frequency from 15 to 200 Hz Loopback for testing and maintenance MP 11x FXO Functionality FXO Capabilities Short or Long Haul P P only to MP Lightning and high voltage protection for outdoor operation ote Programmable Line Characteristics AC impedance matching hybrid balance Tx amp Rx frequency response Tx amp Rx Gains ring detection threshold DC characteristics Version 5 6 381 November 2008 ca AudioCodes Function Additional Features Polarity Reversal Wink Metering Tones Distinctive Ringing Message Waiting Indication Voice 4 Tone Characteristics Voice Compression Silence Suppression Packet Loss
557. uxiliary 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 244 Document LTRT 65411 SIP User s Manual Parameter ECHybridLoss GwDebugLevel CDRReportLevel CDRSyslogServerlP HeartBeatDestIP HeartBeatDestPort HeartBeatintervalmsec EnableRAIl RAIHighThreshold Version 5 6 4 ini File Configuration Description 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 name or password SPC Sensitive Parameters Value Change Changes made to sensitive parameters 1 IP Address 2 Subnet Mask 3 Default Gateway IP Address 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
558. ve Primary Account Admin Admin Security Administrator Note The Access Level cannot be changed for this account type Secondary Account User User User Monitor SIP User s Manual 80 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 48 Web User Accounts Page for Users with Security Administrator Privileges Current Logged User Admin w 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 w Account Data for User User 2 User Name User 2 Change User Name Access Level Administrator Change Access Level wv Fill in the following 3 fields to change the password _ Current Password New Password Confirm New Password Change Password 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 accoun
559. ven if the parameter doesn t appear in the ini file 61 November 2008 A c tal AudioCodes MediaPack Series 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 58 2 Under the NFS Settings group click the right arrow Lu button alongside NFS Table the NFS Settings page appears Figure 3 39 NFS Settings Page Add Apply Delete Host Or IP Root Path NFS Version Authentication Type User ID Vlan Type NFS Version 3 v 1 v Enable v 3 In the Ad 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
560. vice instance The global unigue 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 requests 2xx responses to SUBSCRIBE NOTIFY requests REFER requests and 2xx responses to REFER Note If the GRUU contains the opague URI parameter the device obtains the AOR for the user by stripping the parameter The resulting URI is the AOR 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 G
561. vigation bar click the Search tab the Search engine appears in the Navigation pane SIP User s Manual 34 Document LTRT 65411 SIP User s Manual 3 Web Based Management 2 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 the required string you can use the Search History drop down list to select the string saved from a previous search 3 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 4 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 Search History VLANURON ESERVICECLASSPRIOR A n ority for the Bronze servic t funi ality VLANNATIVEVLANID Sets the native VLAN jenifer TWORKGERVICECLASSPRIE ANSott as priority for Network service 1 gt Note If the searched parameter is not located a notification message is displayed 3 3 5 Working with Scenarios The Web interface allows you to create your own menu w
562. vigation 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 In the 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 39 November 2008 7a e AudioCodes MediaPack Series 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
563. vleg Server IP Address Syslog Server Port Enable Syslog Analog Ports Fiter v SNMP Semang SNMP Trap Destinabons SNMP Community String SNMP VI Table SNMP Trusted Managers 1 Omable SNMP Trap Manager Most Name Achwny Types to Report vie Activity Log Messages lt Parameters Vatse Charge Auplary Ties Loeding Device Reset flesh Memory Burning Device Software Update Access to Restricted Domanns Non Avorved Access Sensfsve Parameters Value Change Y Lv Syslog Settings Sytlog Server IP Address Syslog Server Port Enable Syslog Analog Ports Miter v SNMP Setengs SNMP Trap Destinatons SNMP Correnenty String SNMP V3 Table SNMP Trusted Managers Dsable SNMP Trap Manager Most Name a Achyty Types to Report ves Achvty Log Messages SIP User s Manual 30 Group Title Expanded Para meter Group Somi Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 Figure 3 9 Editing Symbol after Modifying Parameter Value aos Settings C wv Pnority Settings L Network Priority Media Premium Priority Edit Symb
564. w 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 Version 5 6 309 November 2008 6 2 7a K tal AudioCodes MediaPack Series Configuring the Distinctive Ringing Section of the ini File Distinctive Ringing is only applicable to FXS interface Using the distinctive ringing section of this auxiliary file you can create up to 16 distinctive ringing patterns Each ringing pattern configures the ringing tone frequency and up to four ringing cadences The same ringing frequency is used for all the ringing pattern cadences The ringing frequency can be configured in the range of 10 to 200 Hz with a 5 Hz resolution Each of the ringing pattern cadences is specified by the following parameters Burst Ring On Time Configures the cadence to be a burst cadence in the entire ringing pattern The burst relates to On time and the Off time of the same cadence It must appear between First Second Third Fourth string and the Ring On Off Time This cadence rings once during the ringing pattern Otherwise the cadence is interpreted as cyclic it repeats for every ringing cycle Ring On Time specifies the duration of the ringing signal Ring Off Time specifies the sile
565. while a call is still on hold then the FXS interface immediately rings the extension for the duration specified by this parameter If the user off hooks the phone the call becomes active The valid range is 0 to 600 The default value is 30 Note This parameter is applicable only to FXS interfaces Determines whether call transfer is enabled 0 Disable Disable the call transfer service 1 Enable Enable the call transfer service using REFER default If the transfer service is enabled the user can activate Transfer using hook flash signaling If this service is enabled the remote party performs the call transfer 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 forwarded number from the originally dialed number 140 Document LTRT 65411 SIP User s Manual Parameter Enable Call Forward EnableForward Enable Call Waiting EnableCallWaiting Number of Call Waiting Indications NumberOfWaitingIndica tions Time Between Call Waiting Indicatio
566. x NumberMapTel2Ip SourceAddress NumberMapTel2Ip NumberType NumberMapTel2Ip NumberPlan NumberMapTel2Ip RemoveFromLetft NumberMapTel2Ip RemoveFromRight NumberMapTel2Ip LeaveFromRight NumberMapTel2Ip Prefix2Add NumberMapTel2Ip Suffix2Add NumberMapTel2Ip IsPresentationRestricted NumberMapTel2Ip SreTrunkGrouplD NumberMapTel2Ip __ SrcIPGroupID NumberMapTel2Ip 292 Document LTRT 65411 SIP User s Manual Parameter NumberMapIP2Tel Version 5 6 4 ini File Configuration Description For example NumberMapTel2Ip NumberMapTel2lp 0 01 0 0 2 97 1 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 parameters NumberMapTel2lp_ SrclPGroupID NumberMapTel2lp_ NumberType and NumberMapTel2Ip NumberPlan 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 RemoveFromRight LeaveFromRight Prefix2Add and Suffix2Add Parameters can be skipped by using two dollar signs To configure manipulation of destin
567. x DTMF 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 Digit Timeout fer Overlap Drang z sec Declare RFC 2633 m SOP No ist Tx DTMF Option AFC 287 Zed Tx DTMF Option Parar Pa Sed Tx DTMF Option M 4th Tx DTMF Option in Darker Blue Sth Tx DTMF Option RFC 2833 Payload Type Hook flash Opten a Advanced Parameters mal Tone Durabon se gt r in Lighter Blue Hotine Dial Tone Durabon 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 25 configuration pages display all their parameters i e the Advanced Parameter List view is displayed Version 5 6 29 November 2008 ca AudioCodes 3 3 3 2 2 Showing Hiding Parameter Groups MediaPack Series 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 Syslog Settings Sy
568. y Local RADIUS Password Cache Mode Reset Timer Upon Access Local RADIUS Password Cache Timeout sec 300 RADIUS VSA Vendor ID 5003 RADIUS VSA Access Level Attribute 35 w EtherDiscover Setting lS EtherDiscover Operation Mode Unconfigured Device Only w IPSec Setting Enable IP Security Disable Dead Peer Detection Mode Disabled v TLS Settings TLS version SSL 2 0 3 0 and TLS 1 0 TLS Client Re Handshake Interval 0 Je TLS Mutual Authentication Disable Peer Host Name Verification Mode Disable TLS Client Verify Server Certificate Disable TLS Remote Subject Name 2 Configure 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 209 SIP User s Manual 90 Document LTRT 65411 SIP User s Manual 3 Web Based Management Table 3 22 General Security Parameters Parameter HTTP Authentication Mode WebAuthMode Secured Web Connection HTTPS HTTPSOnly Voice Menu Password VoiceMenuPassword General RADIUS Settings Enable RADIUS Access Control EnableRADIUS Use RADIUS for Web Telnet Login WebRADIUSLogin Version 5 6 Description Determines the authentication mode for the Web interface 0 Basic Mode Basic authentication clear text
569. y After Reset sec GWAppDelayTime Max Number of Active Calls MaxActiveCalls Max Call Duration min MaxCallDuration Enable LAN Watchdog EnableLanWatchDog SIP User s Manual MediaPack Series Description Default Release Cause 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 For an explanation on mapping PSTN release causes to SIP responses refer to Mapping PSTN Release Cause to SIP Response on page 336 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 maximum number of channels Define
570. y Capabilities 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 7 8 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 7 8 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 160 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 the 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 Destinat
571. y Server IP t DNS Secondary Server IP v STUN Settings Enable STUN Disable J STUN Server Primary IP 0 0 0 0 STUN Server Secondary IP 0 0 0 0 w NFS Settings NFS Table v DHCP Settings t 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 209 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 369 NTP Server IP Address IP address in dotted decimal notation of the NTP server NTPServerlP The default IP address is 0 0 0 0 i e internal NTP client is disabled NTP UTC Offset Defines the Universal Time Coordinate UTC offset in seconds NTPServerUTCOffset from the NTP server The default offset is 0 The offset range is 43200 to 43200 Version 5 6 59 November 2008 ca AudioCodes Parameter NTP Update Interval NTPUpdatelnterval Telnet Settings Embedded Telnet Server TelnetServerEnable Telnet Server TCP Port TelnetServerPort Telnet Server Idle Timeout TelnetServerldleDisconnect SSH Server Enable SSHServerEnable SSH Server Port SSHServerPort DNS Settings DNS Primary Server IP
572. y Set is used the INVITE message is sent according to the following preferences To the Hunt Group s Serving IP Group ID as defined in the Hunt Group Settings table According to the Tel to IP Routing table 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 Account Table on page 188 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 EnableProxyKeepAlive 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
573. y Sets Table ICoders DTMF amp Dialing sIP Advanced Parameters t manipulation Tables routing Tables Profile Definitions Wendpoint Settings t Endpoint Number BHunt 1P Group Bd advanced Applications gt To view menus in the Navigation tree take this step m On the Navigation bar select the required tab e Configuration refer to Configuration Tab on page 52 e Management refer to Management Tab on page 198 e Status amp Diagnostics refer to Status amp Diagnostics Tab on page 218 Version 5 6 25 November 2008 7a e AudioCodes MediaPack Series 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 signs to collapse the menus and submenus 2 Select the required page item the page opens in the Work pane 3 3 2 1 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 preve
574. y 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 157 Determines the QoS according 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 AoS 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 229 November 2008 A tal AudioCodes MediaPack Series Reader s Notes SIP User s Manual 230 Document LTRT 65411 SIP User s Manual 4 ini File Configuration 4 i
575. y 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 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 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
576. y servers are unavailable When the device falls back to its Tel to IP Routing table the device continues scanning for a Proxy When the device locates an active Proxy it switches from internal routing back to Proxy routing Note To enable the redundant Proxies mechanism set the parameter EnableProxyKeepAlive to 1 or 2 113 November 2008 ca AudioCodes 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 SIP User s Manual MediaPack Series Description 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 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 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 U
577. yTone EnableVoiceDetection DigitMapping TimeBetweenDigits MaxDigits TimeForDialTone RegretTime Version 5 6 Table 4 9 PSTN ini File Parameters Description For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to Supplementary Services on page 138 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to SIP General Parameters on page 101 For a description of this parameter refer to Advanced Parameters on page 129 For a description of this parameter refer to Configuring the FXO Parameters on page 195 For a description of this parameter refer to Configuring the FXO Parameters on page 195 For a description of this parameter refer to Configuring the FXO Parameters on page 195 For a description of this parameter refer to DTMF amp Dialing Parameters on page 125 For a description of this parameter refer to DTMF amp Dialing Parameters on page 125 For a description of this parameter refer to DTMF amp Dialing Parameters on page 125 For a description of this parameter refer to DTMF amp Dialing Parameters on page 125 For a description of this parameter refer to Advanced Parameters on page 129 279 November 2008 A tall AudioCodes MediaPack Series 4 4 10 Analog Telephony Parameters The analog t
578. 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 260 SIP User s Manual 170 Document LTRT 65411 SIP User s Manual 3 Web Based Management 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 76 Coder Group Settings Page Yv Coder Group ID Silence Suppression Coder Name Packetization Time Payload Type 30 v IIs v Disabled 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

Audiocodes MediaPack 114

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