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LTRT-65606 MediaPack & Mediant 1000 SIP Analog

1. Note 2 Each coder should appear only once Note 3 The ptime specifies the maximum packetization time the gateway receives Note 4 G 729B is supported if the coder G 729 is selected and EnableSilenceCompression equals 1 or 2 ini file note 1 This parameter CoderName_ID can appear up to 20 times five coders in four coder groups ini file note 2 The coder name is case sensitive ini file note 3 Enter in the format CoderName ptime For example the following three coders belong to coder group with ID 1 CoderName_1 g711Alaw64k 20 CoderName_1 g711Ulaw64k 40 CoderName_1 g7231 90 TrunkGroup_x TrunkGroup_ lt Hunt Group ID gt lt Starting channel gt lt Ending channel gt lt Phone Endpoint Phone Number Number gt lt Tel Profile ID gt Table For example TrunkGroup_1 1 4 100 TrunkGroup_2 5 8 200 1 Note 1 The numbering of channels starts with 1 Note 2 Hunt Group ID can be set to any number in the range 1 to 99 Note 3 When x Hunt Group ID is omitted the functionality of the TrunkGroup parameter is similar to the functionality of ChannelList and Channel2Phone parameters Note 4 This parameter can appear up to 8 times for MP 108 gateways and up to 24 times for MP 124 gateways Note 5 An optional Tel ProfilelD 1 to 5 can be applied to each group of channels DisableAutoDTMFMute Enables disables the automatic mute of DTMF digits when out of band DTMF transmission is used 0
2. c Parameters can be skipped by using the sign for example NumberMapTel2IP 01 2 972 0 0 NumberMaPTel2IP 03 2 667 0 0 22 Note Number Plan amp Type can optionally be used in Remote Party ID RPID header by using the EnableRPlHeader and AddTON2RPI parameters MediaPack and Mediant 1000 SIP Release Notes 54 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History Previous Version Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name SourceNumberMapTel2IP Source Phone Number Manipulation Table for Tel gt IP calls SourceNumberMapIP2Tel Source Phone Number Manipulation Table for IP gt Tel calls TargetOfChannelX Automatic Dialing Table Version 4 6 Description SourceNumberMapTel2IP a b c d e f g h a Source number prefix b Number of stripped digits from the left or if in brackets are used from right A combination of both options is allowed c String to add as prefix or if in brackets are used as suffix A combination of both options is allowed d Number of remaining digits from the right e Number Plan used in RPID header Number Type used in RPID header Destination number prefix f g h Calling number presentation 0 to allow presentation 1 to restrict presentation The b to f and h manipulation rules are applied if t
3. used to authenticate the gateway to the RADIUS server Should be a cryptographically strong password Determines the number of RADIUS retransmission retries for the same request The valid range is 1 to 10 The default value is 3 Determines the time interval measured in seconds the gateway waits for a response before a RADIUS retransmission is issued The valid range is 1 to 30 The default value is 10 Sets the VLAN functionality 0 Disable default 1 Enable 2 PassThrough N A Sets the native VLAN identifier PVID Port VLAN ID The valid range is 1 to 4094 The default value is 1 Sets the OAM Operation Administration and Management VLAN identifier The valid range is 1 to 4094 The default value is 1 Sets the control VLAN identifier The valid range is 1 to 4094 The default value is 2 Sets the media VLAN identifier The valid range is 1 to 4094 The default value is 3 Sets the priority for Network service class content The valid range is 0 to 7 The default value is 7 Sets the priority for the Premium service class content and media traffic The valid range is 0 to 7 The default value is 6 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 1 What s New Release in 4 6 Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name VlanPremiumServiceClassC ontro Priority Control Premium Priority VlanGol
4. 1 16 kHz metering tone Note Suitable 12 kHz or 16 KHz coeff file must be used for both FXS and FXO gateways IniFileURL Specifies the name of the ini file and the location of the TFTP server from which the gateway loads the ini and configuration files For example tftp 192 168 0 1 filename tftp 192 10 77 13 config lt MAC gt Note The optional string lt MAC gt is replaced with the gateway s MAC address Therefore the gateway requests an ini file name that contains its MAC address This option enables loading different configurations for specific gateways MediaPack and Mediant 1000 SIP Release Notes 48 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History Previous Version Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name CmpFileURL GWRegistrationName Gateway Registration Name RegistrarName Registrar Name RegistrationTimeDivider Re registration Timing IPAlertTimeout Tel to IP No Answer Timeout MINSE Minimum Session Expires MaxActiveCalls Max Number of Active Calls IsUserPhonelnFrom Use user phone in From header UseSourceNumberAsDispla yName Use Source Number as Display Name UseGatewayNameForOptio ns Use Gateway Name for OPTIONS Version 4 6 Description Specifies the name of the cmp file and the location of the TFTP server
5. 1xx Response Supported Comments 100 Trying Yes The SIP gateway generates this response upon receiving of Proceeding message from ISDN or immediately after placing a call for CAS signaling 180 Ringing Yes The SIP gateway generates this response for an incoming INVITE message On receiving this response the gateway waits for a 200 OK response 181 Call is being Yes The SIP gateway does not generate these responses However the gateway forwarded does receive them The gateway processes these responses the same way that it processes the 100 Trying response MediaPack and Mediant 1000 SIP Release Notes 28 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 2 SIP Compatibility Table 2 5 1xx SIP Responses 1xx Response Supported Comments 182 Queued Yes The SIP gateway generates this response in Call Waiting service When SIP gateway receives 182 response it plays a special waiting Ringback tone to Tel side 183 Session Yes The SIP gateway generates this response if Early Media feature is enabled Progress and if the gateway plays a Ringback tone to IP 2 3 5 2 2xx Response Successful Responses Table 2 6 2xx SIP Responses 2xx Response Supported Comments 200 OK Yes 202 Accepted Yes 2 3 5 3 3xx Response Redirection Responses Table 2 7 3xx SIP Responses 3xx Response Supported Comments 300 Multiple Choice Yes The gateway responds with an ACK and resends the request to first in the contact list new
6. IP Profile Settings lt Profile Name gt lt Preference gt lt Coder Group ID gt lt IsFaxUsed gt lt DJBufMinDelay gt lt DJBufOptFactor gt lt IpDiffServ gt lt ControllPDiffServ gt lt EnableSilenceCompression gt lt RTPRedundancyDepth gt lt RemoteBaseUDPPort gt Preference 1 10 The preference option is used to determine the priority of the Profile If both IP and Tel profiles apply to the same call the coders and other common parameters of the preferred Profile will be applied to that call If the Preference of the Tel and IP Profiles is identical the Tel Profile parameters will be applied For example IPProfile_1 name1 2 1 0 10 13 15 44 1 1 6000 IPProfile_2 name2 5 5 1 Not configured the default value of the parameter is used Common parameter used in both IP and Tel profiles Note 1 The IP ProfilelD can be used in the Tel2IP and IP2Tel routing tables Prefix and PSTNPrefix parameters Note 2 Profile Name assigned to a ProfilelD enabling User s to identify it intuitively and easily Note 3 This parameter can appear up to 4 times IsFaxUsed Determines the SIP signaling method used to establish and convey a fax session after a Fax Signaling Method fax is detected 0 No fax negotiation using SIP signaling default 1 Initiates T 38 fax relay 2 Initiates fax using the coder G 711 A law u law with adaptations refer to note 1 3 Initiates T 38 fa
7. RADIUSRetransmission RADIUSTo VLAN Parameters VilanMode VLAN Mode VianNativeVianID Native VLAN ID VianOamVlanID OAM VLAN ID VianControlVlanID Control VLAN ID VianMediaVlanID Media VLAN ID VianNetworkServiceClassPr iority Network Priority VianPremiumServiceClass MediaPriority Media Premium Priority MediaPack and Mediant 1000 SIP Release Notes 20 Enables disables the RADIUS application 0 RADIUS application is disabled default 1 RADIUS application is enabled Note In the current version RADIUS is used only for HTTP authentication CDR over RADIUS isn t supported Uses RADIUS queries for Web and Telnet interface authentication 0 Disabled default 1 Enabled When enabled logging to the gateway s Web and Telnet embedded servers is performed via a RADIUS server The gateway contacts a predefined server and verifies the given username and password pair against a remote database in a secure manner Note 1 The parameter EnableRADIUS must be set to 1 Note 2 RADIUS authentication requires HTTP basic authentication meaning the username and password are transmitted in clear text over the network Therefore users are recommended to set the parameter HttpsOnly 1 to force the use of HTTPS since the transport is encrypted IP address of the RADIUS authentication server Port number of the RADIUS authentication server The default value is 1645 Secret
8. disable the CLIR feature directly from their phone it can also be configured in the Embedded Web Server For example KeyCLIR 43 KeyCLIRDeact 44 To activate the CLRI option from the telephone Press the preconfigured KeyCLIR sequence number on the keypad a confirmation tone is heard To deactivate the CLRI option press the KeyCLIRDeact sequence after the sequence is pressed a confirmation tone is heard Note This option is applicable only to FXS gateways Enable MWI message waiting indication 0 Disabled default 1 Enabled This parameter is applicable only to FXS gateways Note The MP 1xx only supports reception of MWI 0 Disable MWI subscription default 1 Enable subscription to MWI to MWIServerlP address MWI subscription expiration time in seconds The default is 7200 seconds The range is 10 to 72000 Subscription retry time in seconds The default is 120 seconds The range is 10 to 7200 MWI server IP address If provided the gateway subscribes to this IP address Can be configured as a numerical IP address or as a domain name If not configured the Proxy IP address is used instead 0 Disable default 1 Enable visual Message Waiting Indication supplies line voltage of approximately 100 VDC to activate the phone s lamp This parameter is applicable only to FXS gateways 0 MWI information isn t sent to display default 1 MWI information is sent to display If enabled
9. h Not applicable set to i Source IP address obtained from the Contact header in the INVITE message The b to d manipulation rules are applied if the called and calling numbers match the a g and i conditions The manipulation rules are executed in the following order b d and c Parameters can be skipped by using the sign for example NumberMapIP2Tel 01 2 972 034 10 13 77 8 NumberMapIP2Tel 03 2 667 22 Note 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 NumberMapTel2IP Manipulates the destination number for Tel to IP calls Destination Phone Number NumberMapTel2IP a b c d e f g Manipulation Table for Tel gt IP calls a Destination number prefix b Number of stripped digits from the left or if brackets are used from the right A combination of both options is allowed c String to add as prefix or if brackets are used as suffix A combination of both options is allowed d Number of remaining digits from the right e Number Plan used in RPID header f Number Type used in RPID header g Source number prefix The b to f manipulation rules are applied if the called and calling numbers match the a and g conditions The manipulation rules are executed in the following order b d and
10. In addition a name field was added to the NTT Caller ID This field is available in NTT Caller ID type 1 onhook and type 2 Detection and bypass of Bell 103 modem signal is now supported and controlled Relevant parameter BellModemTransportType MP 11x only You can now use the Reset button located on the MP 11x rear panel to restore the networking parameters to their factory default values Fax CNG tone detection was improved by increasing the detection duration FXO gateways only A new DTMF pattern that when received from the Tel side indicates the gateway to disconnect the call Relevant ini file parameter TelDisconnectCode The default base UDP port was changed to 6000 Relevant parameter BaseUDPPort MediaPack and Mediant 1000 SIP Release Notes 8 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 1 What s New Release in 4 6 1 2 SIP New Features 17 18 19 20 21 22 23 24 25 Version 4 6 MP 1xx FXO and Mediant 1000 FXO only Voice Mail VM application The MP 1xx can now be used to mediate between an analog PBX and an IP VM application The supported architecture includes an MP 1xx connected to a PBX using voice mail lines and connected to a voice mail application via the IP network The MP 1xx communicates with the PBX by using either Simplified Message Desk Interface SMDI via the serial RS 232 connection MP 1xx FXO only or special in band DTMF digit
11. SMDI on the gateway 0 Normal serial default 1 Enable RS 232 SMDI interface 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 Determines the time in msec that the gateway waits for an SMDI Call Status message before or after a Setup message is received This parameter is used to synchronize the SMDI and analog interfaces If the timeout expires and only an SMDI message was received the SMDI message is dropped If the timeout expires and only a Setup message was received the call is established The valid range is 0 to 10000 10 seconds The default value is 2000 Determines the transfer method used by the gateway 0 IP default 1 PBX blind transfer 17 June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name Description WaitForDialTime This parameter is applicable only to FXO gateways Wait For Dial Time It determines the delay before the gateway starts dialing on the FXO line in the following scenarios Note Replaces the obsolete 1 The delay between the time the line is seized and dialing is begun during the parameter establishment of an IP gt Tel call FXOWaitForDialTime Note Applicable only to MP 10x FXO for single stage dialing when waiting for dia
12. oltage SubscriptionMode Version 4 6 Description Determines if the local Tel to IP routing table takes precedence over a Proxy for routing calls 0 Only Proxy is used to route calls default 1 The Proxy checks the Destination IP Address field in the Tel to IP Routing table for a match with the outgoing call Only if a match is not found a Proxy is used Note Applicable only if Proxy is not always used AlwaysSendToProxy 0 SendinviteToProxy 0 0 Remove the ptime header from SDP 1 Include the ptime header in SDP default Defines the amount of time in seconds the gateway s operation is delayed after a reset cycle The valid range is 0 to 600 The default value is 5 seconds Note This feature helps to overcome connection problems caused by some LAN routers or IP configuration parameters change by a DHCP Server Determines the line voltage threshold which when reached is considered a current disconnect detection Note Applicable only to MP 10x FXO gateways The valid range is 0 to 20 Volts The default value is 4 Volts Determines the frequency at which the analog line voltage is sampled after offhook for detection of the current disconnect threshold Note Applicable only to MP 10x FXO gateways The valid range is 100 to 2500 msec The default value is 1000 msec Determines the method the gateway uses to subscribe to an MWI server 0 Per endpoint Each endpoint subscrib
13. patterns Relevant Parameters VoiceMaillnterface DigitPatternForwardOnBusy DigitPatternForwardOnNoAnswer DigitPatternForwardOnDND DigitPatternForwardNoReason DigitPatternInternalCall DigitPatternExternalCall SMDI SMDITimeOut LineTransferMode SerialData SerialParity SerialStop SerialFlowControl WaitForDialTime MWIOffCode MWIOnCode TelDisconnectCode MP 11x and Mediant 1000 only Support was added for TLS and SIPS Secured SIP connections The gateway initiates a TLS connection if its selected transport type is TLS Ifa TLS connection is initiated to the gateway it responds using TLS even if TLS isn t enabled When SIPS is enabled TLS is used all the way to the destination over multiple hops TLS and SIPS use the Certificate Exchange process described in the MediaPack SIP User s Manual Relevant parameters TLSLocalSIPPort EnableSIPS SIPTransportType SIPSRequireClientCertificate Support was added for SIP over TCP A TCP session is established if the selected transport type is TCP or if a TCP connection is initiated by a remote gateway even if the local gateway isn t configured to use TCP Relevant parameters TCPLocalSIPPort SIPTransportType It is now possible to configure a specific destination port in addition to the IP address for the Proxy Registrar and Destination IP Address entries of the Tel to IP Routing table Relevant parameters ProxylP RegistrarlP Prefix Distinctive Call Waiting Tones
14. per port the automatic dialing configuration TargetOfChannel lt Port gt lt Phone gt lt Mode gt Port 0 to 7 for MP 108 0 to 23 for MP 124 Phone An auto dialed phone string mode 0 Normal collect digits mode 1 Auto Dial the gateway immediately dials after the phone is off hooked mode 2 Hotline the gateway dials if no digits were collected during a dial tone duration 55 June 2005 Ce AudioCodes MediaPack and Mediant 1000 SIP Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name TimeForDialTone Dial Tone Duration CallerDisplayInfox Caller ID Table KeyCFUncond KeyCFNoAnswer KeyCFBusy KeyCFDeact KeyCFBusyOrNoAnswer KeyCFDoNotDisturb Keypad Features KeyHotLine KeyHotLineDeact Keypad Features MediaPack and Mediant 1000 SIP Release Notes 56 Description Time in seconds that the dial tone is played The default time is 16 seconds FXS gateway ports play the dial tone after phone is picked up while FXO gateway ports play the dial tone after port is seized in response to ringing Note 1 During play of dial tone the gateway waits for DTMF digits Note 2 TimeForDialTone is not applicable when Automatic Dialing is enabled CallerDisplaylnfo lt channel gt lt Caller ID string gt lt Restriction gt Restriction 0 The CallerID is not restricted default Restriction 1 The CallerlD is res
15. 13 June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Description Parameter Name eseripiio RegistrariP IP address and optionally port number of Registrar server Registrar IP Address Enter the IP address in dotted format notation for example 201 10 8 1 lt 5080 gt Note 1 If not specified the REGISTER request is sent to the primary Proxy server refer to Proxy IP address parameter Note 2 When port number is specified DNS SRV queries aren t performed even if EnableSRVQuery is set to 1 Prefix Prefix lt Destination Phone Prefix gt lt Destination IP Address gt lt Source Phone Prefix gt lt Profile ID gt For example Prefix 20 10 2 10 2 202 1 Prefix 10 340 451 xxx 10 2 10 6 1 Prefix gateway domain com Note 1 lt destination source phone prefix gt can be single number or a range of numbers Note 2 This parameter can appear up to 50 times Note 3 Parameters can be skipped by using the sign for example Prefix 10 2 10 2 202 1 Note 4 The lt Destination IP Address gt field can be either in dotted format notation or a FQDN This field can also include a selected port to use in the format lt IP Address gt lt Port gt FirstCallWaitingTonelD Determines the index of the first Call Waiting Tone in the CPT file This feature enables t
16. 20 40 60 80 100 120 default 20 Determines the default transport layer used for outgoing SIP calls initiated by the gateway 0 UDP default 1 TCP 2 TLS SIPS applicable to MP 11x and Mediant 1000 only Note It is recommended to use TLS to communicate with a SIP Proxy and not for direct gateway gateway communication Local TCP port used to receive SIP messages The default value is 5060 Applicable to MP 11x and Mediant 1000 only Enables secured SIP SIPS connections over multiple hops 0 Disabled default 1 Enabled When SIPTransportType 2 TLS and EnableSIPS is disabled TLS is used for the next network hop only When SIPTransportType 2 TLS or 1 TCP and EnableSIPS is enabled TLS is used through the entire connection over multiple hops Note If SIPS is enabled and SIPTransportType UDP the connection fails Applicable to MP 11x and Mediant 1000 only Local TLS port used to receive SIP messages The default value is 5061 Note The value of TLSLocalSIPPort must be different to the value of TCPLocalSIPPort Applicable to MP 11x and Mediant 1000 only 0 The gateway doesn t require client certificate default 1 The gateway when acting as a server for the TLS connection requires reception of client certificate to establish the TLS connection Note The SIPS certificate files can be changed using the parameters HTTPSCertFileName and HTTPSRootFileName
17. 4 We MediaPack and Mediant 1000 SIP Release Notes General When viewing this manual on CD Web site or on any other electronic copy all cross references are hyperlinked Click on the page or section numbers shown in blue to reach the individual cross referenced item directly To return back to the point from where you accessed the cross reference press the ALT and lt keys Trademarks AC logo Ardito AudioCoded AudioCodes AudioCodes logo IPmedia Mediant MediaPack MP MLQ NetCoder Stretto TrunkPack VoicePacketizer and VolPerfect are trademarks or registered trademarks of AudioCodes Limited All other products or trademarks are property of their respective owners 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 and only Industry standard terms are used throughout this manual The symbol 0x indicates hexadecimal notation Related Documentation Document Manual Name LTRT 654xx e g LTRT 65401 MediaPack SIP User s Manual LTRT 614xx MP 1xx Fast Track Installation Guide LTRT 615xx MP 11x Fast Track Installation Guide LTRT 657xx Analog Mediant 1000 SIP User s Manual LTRT 659xx Analog M
18. 6 ETSI VMWI not ring related LR_DT_AS MediaPack and Mediant 1000 SIP Release Notes 16 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 1 What s New Release in 4 6 Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name BellcoreVMWITypeOneStan dard SNMPTrapManagerHostNa me Trap Manager Host Name WebAccessList_x Web and Telnet Access List Screen RTPS IDCoeffNum BellModemTransportType EnablelPAddrTranslation EnableParametersMonitorin g Description Selects the Bellcore VMWI sub standard 0 Between rings default 1 Not ring related Defines a 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 Defines up to ten IP addresses that are permitted to access the gateway s Web and Telnet interfaces Access from an undefined IP address is denied This security feature is inactive the gateway can be accessed from any IP address when the table is empty For example WebAccess_List_0 10 13 2 66 WebAccessList_1 10 13 77 7 The default value is 0 0 0 0 the gateway can be accessed from any IP address Determines the number
19. Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name Description IniFileURL Specifies the name of the ini file and the location of the server IP address or FQDN from which the gateway loads the ini file The ini file can be loaded using TFTP HTTP or HTTPS For example tftp 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 Note 1 When using HTTP or HTTPS the date and time of the ini file are validated Only more recently dated ini files are loaded Note 2 The optional string lt MAC gt is replaced with the gateway s MAC address Therefore the gateway requests an ini file name that contains its MAC address This option enables loading different configurations for specific gateways IniFileTemplateURL Specifies the name of a second ini file in addition to IniFile URL and the location of the server IP address or FQDN from which it is loaded http server_nameffile https server_name file PrtFileURL Specifies the name of the Prerecorded Tones file and the location of the server IP address or FQDN from which it is loaded http server_nameffile https server_name file CptFileURL Specifies the name of the CPT file and the location of the server IP address or FQDN from which it is loaded http server_name file https server_nameffile FXOCoeffFileURL Specifies the name of the
20. Proxy is Trusted default If Proxy is not Trusted the P asserted header is not used 0 TON PLAN parameters aren t included in the RPID header 1 TON PLAN parameters are included in the RPID header default If RPID header is enabled EnableRPlHeader 1 and AddTON2RPI 1 it is possible to configure the calling and called number type and number plan using the Number Manipulation tables for Tel gt IP calls Defines that the T 38 packets will be received using the same Rx port as RTP packets 0 Use the RTP port 2 to receive T 38 packets default 1 Use the same port as the RTP port to receive T 38 packets Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History Previous Version Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name IPProfile_ID IP Profile Settings TelProfile_ID Tel Profile Settings Version 4 6 Description IPProfile_ lt Profile ID gt lt Profile Name gt lt Preference gt lt Coder Group ID gt lt IsFaxUsed gt lt DJBufMinDelay gt lt DJBufOptFactor gt lt IpDiffServ gt lt ControllPDiffServ gt lt EnableSilenceCompression gt lt RTPRedundancyDepth gt Preference 1 10 The preference option is used to determine the priority of the Profile If both IP and Tel profiles apply to the same call the coders and other common parameters of the pref
21. SIP Responses continues on pages 29 to 31 4xx Response 404 Not Found 405 Method Not Allowed 406 Not Acceptable 407 Proxy Authentication Required 408 Request Timeout 409 Conflict 410 Gone 411 Length Required 413 Request Entity Too Large 414 Request URL Too Long 415 Unsupported Media 420 Bad Extension 480 Temporarily Unavailable 481 Call Leg Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops 484 Address Incomplete Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Comments The SIP gateway generates this response if it is unable to locate the callee On receiving this response the gateway notifies the User with a Reorder Tone The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call Authentication support for Basic and Digest On receiving this message the GW issues a new request according to the scheme received on this response The gateway generates this response if no answer timeout expired On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call
22. The gateway can now play a specific Call Waiting Tone from the Call Progress Tones file This option enables the called party to distinguish between four different call origins e g external vs internal calls This feature is relevant only to Broadsoft s application servers the tone is played using INFO message Relevant parameter FirstCallWaitingTonelD DNS Service Record SRV queries can now also be used to resolve domain names of the Registrar server and any domain name that appears in the Contact and Record Route headers SRV can be used to resolve all domain names or just for Proxy servers If a port is part of the domain name i e lt FQDN gt lt port gt SRV isn t used Relevant parameters EnableSRVQuery EnableProxySRVQuery RegistrarIP Full support was added for the following coders gt G 729 Annex B with no correlation to EnableSilenceCompression G 729 support for Silence Suppression is negotiated between both sides by using the annex b parameter in the SDP body of the SIP messages Relevant parameters CoderName CoderName_ID If the coder G 723 is used when silence suppression is enabled the gateway now includes the string annex a in the SDP Fax fallback If T 38 negotiation fails the gateway can now re initiate a fax session using the coder G 711 A law u law with adaptations Relevant parameter IsFaxUsed 9 June 2005 x A 26 WH 3 ett AudioCodes MediaPack and Mediant 1000
23. The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call If the gateway receives a 415 Unsupported Media response it notifies the User with a Reorder Tone The gateway generates this response in case of SDP mismatch The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call If the gateway receives a 480 Temporarily Unavailable response it notifies the User with a Reorder Tone This response is issued if there is no response from remote The gateway does not generate this response On reception of this message before a 2000K has been received the gateway r
24. What s New Release in 4 6 Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name Description SerialFlowControl Determines the value of the RS 232 flow control 0 None default 1 Hardware Secure Hypertext Transport Protocol HTTPS Parameters MP 11x and Mediant 1000 only HTTPSOnly Determines the protocol types used to access the Embedded Web Server Secured Web Connection 0 HTTP and HTTPS default 1 HTTPS only unencrypted HTTP packets are blocked HTTPSPort Determine 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 HTTPSRequireClientCertific Requires client certificates for HTTPS connection The client certificate must be ate preloaded to the gateway and its matching private key must be installed on the managing PC Time and date must be correctly set on the gateway for the client certificate to be verified 0 Client certificates are not required default 1 Client certificates are required HTTPSRootFileName 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 gateway is loaded via BootP TFTP For information on loading this file via the Embedded
25. and basic configuration such as to modify most of the ini file parameters and to change the network settings IP address subnet mask and default gateway IP address of the gateway Relevant Parameters TelnetServerEnable TelnetServerldieDisconnect TelnetServerPort 1 4 Resolved Constraints Version 4 6 Domain names are now resolved using DNS A Record mechanism if the SRV process fails In addition DNS resolution of the Proxy domain name is now applicable even when the Keep Alive mechanism isn t used A new allocation mechanism protects the existing configuration files e g CPT logo from being deleted during a software upgrade When upgrading the cmp file and burning it to the non volatile memory the cmp is burned independently Several Web messages that were blocked by popup blocking Web browsers are now available when java script is enabled 11 June 2005 Ca AudioCodes MediaPack and Mediant 1000 SIP 1 5 New and Modified Parameters Most new parameters described in Table 1 1 can be configured with the ini file and via the Embedded Web Server Note that only those parameters contained within square brackets are configurable via the Embedded Web Server Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name CoderName_ID Description Coder list for Profiles up to five coders in each group The CoderName_ID parameter ID from 1 to 4
26. is equal to RegistrarlP either FQDN or numerical IP address if configured gt Otherwise the servername is equal to ProxyName if configured The ProxyName can be any string gt Otherwise the servername is equal to ProxyIP either FQDN or numerical IP address Version 4 6 23 x aA e wt AudioCodes MediaPack and Mediant 1000 SIP The sipgatewayname parameter defined in the ini file or set from the Web browser 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 REGISTER message is sent to the Registrar s IP address if configured or to the Proxy s IP address The message is sent per gateway or per gateway endpoint according to the AuthenticationMode parameter Usually the FXS gateways are registered per gateway port while FXO gateways send a single registration message where Username is used instead of phone number in From To headers The registration request is resent according to the parameter RegistrartionTimeDivider For example if RegistrationTimeDivider 70 and Registration Expires time 3600 the gateway resends its registration request after 3600 x 70 2520 sec The default value of RegistrartionTimeDivider is 50 e Proxy and Registrar Authentication handling 401 and 407 responses using Basic or Digest methods e Single gatew
27. 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 Note Applicable only to MP 11x and Mediant 1000 Determines the Bell modem transport method 0 Transparent default 2 Bypass 3 Transparent with events 0 Disable IP address translation 1 Enable IP address translation for RTP and T 38 packets default When enabled the gateway compares the source IP address of the first incoming packet to the remote IP address stated in the opening of the channel If the two IP addresses don t match the NAT mechanism is activated Consequently the remote IP address of the outgoing stream is replaced by the source IP address of the first incoming packet Note The NAT mechanism must be enabled for this parameter to take effect DisableNAT 0 Enables to view changes made on the fly to parameters via Web or SNMP 0 Deactivate default 1 Activate Voice Mail Parameters MP 1xx FXO and Mediant 1000 FXO only VoiceMaillnterface Voice Mail Interface SMDI Enable SMDI SMDITimeOut SMDI Timeout LineTransferMode Line Transfer Mode Version 4 6 Enables the VM application on the MP 1xx and determines the communication method used between the PBX and the gateway 0 None default 1 DTMF 2 SMDI MP 1xx FXO only Enables the Simplified Message Desk Interface
28. parameter in the SDP attributes the gateway now keeps sending DTMF digits using transparent mode as part of the voice RTP If the coder G 729 is used with silence suppression enabled the gateway now includes the string annex b in the SDP Support for Alert Info header for Distinctive Ringing for FXS gateways IP gt Tel calls was added To use this feature define several Distinctive Ringing tones in the Call Progress Tones definition file Can now correctly handle Subscription to DTMF events according to DTMF SUBSCRIBE NOTIFY IETF draft draft mahy sipping signaled digits 01 Can now configure the sip URI host part in the OPTIONS message to be either the gateway s IP address or the gatewayname parameter Relevant parameter UseGatewayNameForOPTIONS 4 1 4 SNMP and Web Server New Features 59 60 61 After changing at least one of the networking parameters IP address subnet mask or the default gateway s IP address in the Network Settings screen and pressing the button Submit a prompt appears indicating that for the change s to take effect the gateway will reset and the current configuration will be burned to flash memory The gateway s Web Interface appearance was updated and enhanced A SIP Channel Status screen was added to the Embedded Web Server This screen can be accessed via the Channel Status screen It contains SIP static information and associated calls information of the
29. recommended not to select both G 729 and G 729 Annex B coders at the same time RFC 2198 redundancy mode with RFC 2833 is not supported that is if a complete DTMF digit was lost it is not reconstructed The current RFC 2833 implementation does support redundancy for inter digit information lost Date and Time should be set after each gateway power reset unless NTP Network Time Protocol is used 33 June 2005 x a 14 15 16 17 18 19 20 21 22 23 24 25 26 WH 3 wt AudioCodes MediaPack and Mediant 1000 SIP After resetting the Web password using the ini file parameter ResetWebPassword and defining a new password the user must load an ini file with ResetWebPassword set to 0 Channel parameters such as Voice DTMF gain silence suppression except for G 729 and Jitter buffer are collectively configured in the ini file on a per gateway usage not on a per call basis By using Profiles this limitation can be overcome Two versions of the DSP template firmware are available DSP Template Versions 0 and 2 default The DSP template number 2 supports the silence detection feature that is used for FXO disconnect supervision FXS and FXO gateways use different configuration Coeff dat files The polarity reversal detection option on FXO gateways isn t functional when using a 12 kHz coefficient file MP1xx12 1 12khz fxo The gateway only supports symmetrical coders the s
30. selected according to the parameter FaxModemBypassBasicRtpPacketinterval NSE payload type for Cisco Bypass compatible mode The valid range is 96 127 The default value is 105 Note Cisco gateways usually use NSE payload type of 100 The name and path of the file containing the Prerecorded Tones Defines the value of the DiffServ field in the IP header for the signaling session The valid range is 0 to 63 The default value is 0 Defines the time period in seconds after which a Registration request is resent if registration fails with 4xx or there is no response from the Proxy Registrar The default is 30 seconds The range is 10 to 3600 0 None default 1 P asserted 2 P preferred The Asserted ID mode defines the header that is used in the generated INVITE request The header also depends on the calling Privacy allowed or restricted The P asserted or P preferred headers are used if the originating party has a Caller ID name The Caller ID name is presented as a display name in the P asserted or P preferred headers P asserted or P preferred headers are used together with the Privacy header If Caller ID is restricted the Privacy id will be included Otherwise for allowed Caller ID the Privacy none will be used If Caller ID is restricted received from Tel or configured in the gateway the From header is set to lt anonymous anonymous invalid gt 0 The SIP Proxy is not Trusted 1 SIP
31. selected port MediaPack and Mediant 1000 SIP Release Notes 42 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History 62 63 64 65 66 67 68 69 70 Previous Version A new Web wizard guides the user through the process of software upgrade selection of files and loading them to the gateway The wizard also enables the user to upgrade the software and to maintain the existing configuration A radio button was added alerting the user whether to burn or not to burn changes to flash during reset New SNMP MIB for collection and monitoring system performance Introduction of a carrier grade alarm system with the following characteristics 1 Allows an Element Manager EM to determine which alarms are currently active active alarm table 2 Allows an EM to detect lost alarm raise and clear traps 3 Allows an EM to recover lost alarm raise and clear traps alarm history table Enable private labeling of the Web browser s title when a graphical logo is used The FXO gateway can now detect unconnected analog ports These ports are marked using a color indication on the Web channel status page Adding the capability to provision the table of authorized SNMP managers In addition to acBoard MIB a new set of MIBs for configuration and status is introduced The new MIBs are divided by functionality Media Analog Control System Users can now configure th
32. the rport value of the response to the actual port from which the request was received This method is used for example to enable the gateway to identify its port mapping outside a NAT Registration gt An option was added to configure the gateway s registration name that is used in REGISTER messages gt A registrar domain name can now be used instead of an IP address gt Users can now determine the registration timing in percentage of the re register timing that is set by the Registrar Relevant parameters RegistrationTimeDivider GWRegistrationName RegistrarName Registration retry time can now be configured Relevant parameter RegistrationRetryTime On the fly Registration Unregistration to Proxy Registrar using the Embedded Web Server s Re Register button Users can now unregister and reregister when an endpoint s phone number s or other authentication parameters e g username password were modified 41 June 2005 x A 49 50 51 52 53 54 55 56 57 58 WH 3 et AudioCodes MediaPack and Mediant 1000 SIP Message Waiting Indication MWI according to IETF lt draft ietf sipping mwi 04 txt gt including SUBSCRIBE to MWI server is now implemented MP 1xx FXS gateways 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 tone followed by the playing of a continuous dial tone I
33. the gateway generates an MWI FSK message that is displayed on the MWI display This parameter is applicable only to FXS gateways Duration in msec of the played stutter dial tone that indicates waiting message s The default is 2000 2 seconds The range is 1000 to 60000 Operation modes of the Alternative Routing mechanism 0 Disabled default 1 Enabled 2 Enabled for status only not for routing decisions Defines the destination IP address for CDR logs The default value is a null string that causes the CDR messages to be sent with all Syslog messages IP address in dotted format notation of the NTP server The default IP address is 0 0 0 0 the internal NTP client is disabled Defines the UTC Universal Time Coordinate offset in seconds from the NTP server The default offset is 0 The offset range is 43200 to 43200 seconds Defines the time interval in seconds the NTP client requests for a time update The default interval is 86400 seconds 24 hours The range is 0 to 214783647 seconds Note It isn t recommended to be set beyond one month 2592000 seconds 57 June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name Description PolarityReversalType Defines the voltage change slope during polarity reversal or wink 0 Soft default 1 Hard Note 1 S
34. their phone it can also be configured in the Embedded Web Server For example KeyHotLine 83 KeyHotLineDeact 84 To activate the delayed hotline option from the telephone e Press the preconfigured sequence number on the keypad a dial tone is heard e Press the telephone number to which the phone automatically dials after a configurable delay a confirmation tone is heard To deactivate the hotline option press the KeyHotLineDeact sequence after the sequence is pressed a confirmation tone is heard Note This option is applicable only to FXS gateways Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History Previous Version Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name KeyCLIR KeyCLIRDeact Keypad Features EnableMWI Enable MWI EnableMWISubscription Subscribe for MWI MWIExpirationTime MWI Subscribe Expiration Time SubscribeRetryTime MWI Subscribe Retry Time MwiServerlIP MWI Server IP Address MW1IAnalogLamp MWI Analog Lamp MWIDisplay MWI Display StutterToneDuration Stutter Tone Duration AltRoutingTel2IPEnable Enable Alt Routing Tel to IP CDRSyslogServerIP CDR Server IP Address NTPServerlIP NTPServerUTCOffset NTPUpdatelinterval Version 4 6 Description Keypad sequence that activates the Caller ID restriction CLIR Users can enable
35. 3 0 0 Not configured the default value of the parameter is used Common parameter used in both IP and Tel profiles Note 1 The Tel ProfilelD can be used in the Hunt group table TrunkGroup_x parameter Note 2 Profile Name assigned to a ProfilelD enabling User s to identify it intuitively and easily Note 3 This parameter can appear up to 4 times 5i June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name oe cripuon CoderName_ID Coder list for Profiles up to five coders in each group Coder Group Settings The CoderName_ID parameter ID from 1 to 4 provides groups of coders that can be associated with IP or Tel profiles You can select the following coders g711Alaw64k G 711 A law g711Ulaw64k G 711 p law g7231 G 723 1 6 3 kbps default g7231r53 G 723 1 5 3 kbps g726 G 726 ADPCM 32 kbps Payload Type 35 g729 G 729A The RTP packetization period ptime in msec depends on the selected Coder name and can have the following values g711 family 10 20 30 40 50 60 80 100 120 default 20 9729 10 20 30 40 50 60 default 20 9723 family 30 60 90 default 30 G 726 family 10 20 30 40 50 60 80 100 120 default 20 Note 1 If not specified the ptime gets a default value
36. AudioCodes CPE amp Access Gateway Products Analog VoIP Gateways MediaPack amp Analog Mediant 1000 SIP Release Notes Version 4 6 Document LTRT 65606 oC TET bel TIT leo TEE jees Ci i Tisseeeeee eee l a wi AudioCodes Notice This document describes the release of the AudioCodes analog Mediant 1000 and MediaPack Series MP 124 24 port MP 108 8 port MP 104 4 port MP 102 2 port MP 118 8 port MP 114 4 port and MP 112 2 port 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 the accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions Updates to this document and other documents can be viewed by registered Technical Support customers at www audiocodes com under Support Product Documentation Copyright 2005 AudioCodes Ltd All rights reserved This document is subject to change without notice Date Published Jul 13 2005 Date Printed Jul 31 2005 Table of Contents 1 Whats s specie in lionna TE o AEE E EN EAE EE A AEA EEOAE EA E ENA AEAEE i 3 5 Previous Releases ELLELE TI POPP 59 Version 4 6 3 June 2005 MediaPack and Mediant 1000 SIP List of Tables Browser Parameter Name continues on pages 12 to 22 ble 2 10 6xx SIP Responses Table 4 1 Release 4
37. Auto 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 IsDTMFUsed 1 This enables the sending of DTMF digits in band transparent of RFC 2833 in addition to out of band DTMF messages Note Usually this mode is not recommended MediaPack and Mediant 1000 SIP Release Notes 52 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History Previous Version Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name DNS2IP Internal DNS Table AltRouteCauseTel2IP Reasons for Alternative Routing Table AltRouteCauselP2Tel Reasons for Alternative Routing Table Prefix Tel to IP Routing Table PSTNPrefix IP to Hunt Group Routing Table Version 4 6 Description Internal DNS table used to resolve host names to IP addresses Two different IP addresses in dotted format notation can be assigned to a hostname DNS2IP lt Hostname gt lt first IP address gt lt second IP address gt Note 1 If the internal DNS table is configured the gateway first tries to resolve a domain name using this table If the domain name isn t found the gateway performs a DNS resolution using an external DNS server N
38. FXO coefficients file and the location of the server IP address or FQDN from which it is loaded http server_nameffile https server_name file FXSCoeffFileURL Specifies the name of the FXS coefficients file and the location of the server IP address or FQDN from which it is loaded http server_nameffile https server_name file AutoUpdateCmpFile 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 AutoUpdateFrequency Determines the number of minutes the gateway waits between automatic updates The default value is 0 the update at fixed intervals mechanism is disabled AutoUpdatePredefinedTime 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 ResetNow Invokes an immediate restart of the gateway This option can be used to activate offline not on the fly parameters that are loaded via IniFileUrl 0 The immediate restart mechanism is disabled default 1 The gateway immediately restarts after an ini file with this parameter set to 1 is loaded MediaPack and Mediant 1000 SIP Release Notes 22 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 2 SIP Compatibility 2 SIP Compatibility 2 1
39. IP to the corresponding FXS gateway The FXS port generates the 12 16KHz metering tone according to the configured metering tone type Relevant parameter SendMetering2IP FXO Only MeteringType If calling party name is not defined CallerDisplayInfoX lt name gt is not specified per gateway s x port the calling number can be used instead Applicable to Tel gt IP calls Relevant parameter UseSourceNumberAsDisplayName The Hotline and Warmline feature immediate or with delay was added Each gateway port can now be configured to automatically dial a pre configured number if no digits are entered after handset offhook after specified timer for playing the dial tone expires Relevant parameters TargetOfChannelX HotLineDialToneDuration An additional column was added to the Caller ID table This column Presentation determines whether a specific Caller ID is restricted or not The Caller ID string isn t sent when a Call is initiated by a restricted port To maintain backward compatibility when a Caller ID name is private the Caller ID is restricted and the Presentation value is ignored Relevant parameter CallerlDInfo Generation and detection of Indian Danish Brazilian British and Swedish Type 1 DTMF based Caller ID signals is now supported Relevant parameter CallerIDType Support for Caller ID generation during Call waiting was added If an incoming IP call is designated to a busy port the called party c
40. Mediant 1000 SIP Release Notes 58 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 5 Previous Releases 5 Previous Releases Details of previous releases can be found in the Release Notes of Version 4 4 published by AudioCodes on Jan 12 2005 printed on recycled paper nd available on CD or Web site THE STANDARDS INSTITUTION OF ISRAEL Version 4 6 59 June 2005 AudioCodes CPE amp Access Gateway Products Analog VoIP Gateways wi AudioCodes www audiocodes com
41. Proxy is found the gateway either continues working with it until the next failure occurs or reverts to the primary Proxy refer to the Redundancy Mode parameter If none of the Proxy servers respond the gateway goes over the list again The gateway also provides real time switching hotswap mode between the primary and redundant proxies IsProxyHotSwap 1 If the first Proxy doesn t respond to INVITE message the same INVITE message is immediately sent to the second Proxy Note 1 If EnableProxyKeepAlive 1 the gateway monitors the connection with the Proxies by using keep alive messages OPTIONS Note 2 To use Proxy Redundancy you must specify one or more redundant Proxies using multiple ProxylIP lt IP address gt definitions Note 3 When port number is specified DNS SRV queries aren t performed even if EnableProxySRVQuery is set to 1 IP addresses of the redundant Proxies you are using Enter the IP address as FQDN or in dotted format notation for example 192 10 1 255 You can also specify the selected port in the format lt IP Address gt lt port gt Note 1 This parameter is available only if you select Yes in the Is Proxy Used field Note 2 When port number is specified DNS SRV queries aren t performed even if EnableProxySRVQuery is set to 1 ini file note The IP addresses of the redundant Proxies are defined by the second third and forth repetition of the ini file
42. S capabilities are enhanced to support more complex tones and additional tones frequencies Tones with AM modulation Up to four cadences per tone 32 Call Progress Tones Up to 64 different frequencies Generation of voice during off time of the tone cadence for Call Waiting Tones Y VV VV WV Burst tones The Automatic Update mechanism was improved The gateway can now periodically check for updated software cmp or ini files on a remote server In addition new parameters that enable the configuration of a separate URL for each configuration file e g CPT are introduced This mechanism can be used even for Customer Premise s Equipment CPE devices that are installed behind NAT and firewalls For detailed information on the Automatic Update mechanism refer to the MediaPack User s Manual Relevant Parameters CmpFileURL IniFileURL IniFileTemplateURL PrtFileURL CptFileURL FXOCoeffFileURL FXSCoeffFileURL AutoUpdateCmpFile AutoUpdateFrequency AutoUpdatePredefinedTime ResetNow ra June 2005 x a 7 10 11 12 13 14 15 16 WH 3 et AudioCodes MediaPack and Mediant 1000 SIP Media Control and Management OAM traffic in the MediaPack can now be separated into three dedicated networks Instead of a single IP address the MediaPack can be assigned three IP addresses and subnet masks each relates to a different traffic type This architecture enables users to integrate the MediaPack into a thr
43. SIP It is now possible to configure the gateway to route a call according to its Called Number received in the To header of the SIP INVITE message instead of the user part of the Request URI When configured the gateway also uses the Username parameter as the user part of the Contact header Relevant parameters IsUseToHeaderAsCalledNumber Username 1 3 Web SNMP and Command Line New Features 27 28 29 30 31 32 MP 11x and Mediant 1000 only SSL Secure Socket Layer and TLS Transport Layer Security protocols can now be used to secure access to the Embedded Web HTTPS and Telnet Servers Relevant Parameters HTTPSOnly HTTPSPort HTTPSRequireClientCertificate HTTPSRootFileName HTTPSCertFileName TelnetServerEnable Up to 10 authorized client IP addresses that are permitted to access the gateway via Web or Telnet interface can now be defined This security feature is inactive the gateway can be accessed from any IP address by default Relevant parameter WebAccessList_x An IP routing table that is used by the gateway to determine IP routing rules is now available Before sending an IP packet the gateway searches this table for an entry that matches the requested destination host network If such entry is found the gateway sends the packet to the indicated router If no explicit entry is found the packet is sent to the default gateway Relevant parameters RoutingTableDestinationsColumn RoutingTableDes
44. Server Use ini file configuration instead The Caller ID Name column in the Caller ID table in the Embedded Web Server can t contain the inverted commas character For example entering John is not allowed In the ini file this string can be used SNMP Constraints 33 34 35 36 37 Configuration alarm does not clear The following RTP MIB objects are not supported rtpRcvrSRCSSRC rtpRevrSSRC rtpSenderSSRC rtpRcvrLostPackets rtpRcvrPackets rtpSenderPackets rtpRcvrOctets rtpSenderOctets The range of the faxModemRelayVolume MIB object is wrong Instead of 0 to 15 it should be 18 to 3 corresponding to an actual volume of 18 5 dBm to 3 5 dBm Cold start trap doesn t appear after soft reset for MediaPack Only one SNMP manager can access the device simultaneously 35 June 2005 MediaPack nd Mediant 1000 SIP Reader s Notes MediaPack and Mediant 1000 SIP Release Notes 36 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History Previous Version 4 Recent Revision History 4 1 Revision 4 4 4 1 1 General Gateway New Features 1 Version 4 6 Extensive Profiles support was added Different Profiles can now be assigned on a per call basis using the Tel to IP and IP to Tel routing tables or by assigning different Profiles to the gateway s endpoint s The Profiles contain parameters such as Coders T 38 relay Voice an
45. Supported SIP Features The MediaPack SIP main features are e Reliable User Datagram Protocol UDP transport with retransmissions e Transmission Control Protocol TCP Transport layer e SIPS using TLS MP 11x and Mediant 1000 only e 1 38 real time Fax using SIP Note If the remote side includes the fax maximum rate parameter in the SDP body of the INVITE message the gateway returns the same rate in the response SDP e Works with Proxy or without Proxy using an internal routing table e Fallback to internal routing table if Proxy is not responding e Supports up to four Proxy servers If the primary Proxy fails the MediaPack automatically switches to a redundant Proxy e Supports domain name resolving using DNS SRV records for Proxy Registrar and domain names that appear in the Contact and Record Route headers e Proxy or Registrar Registration per gateway or per gateway endpoint such as REGISTER sip servername SIP 2 0 VIA SIP 2 0 UDP 212 179 22 229 branch z 9hG4bRaC7AU234 From lt sip 101 sipgatewayname gt tag 1c29347 To lt sip 101 sipgatewayname gt Call ID 104532712 1 79 22 2229 Seq 1 REGISTER Expires 3600 Contact sip 101 212 179 22 229 Content Length 0 The servername string is defined according to the following rules gt The servername is equal to RegistrarName if configured The RegistrarName can be any string gt Otherwise the servername
46. T 65606 MediaPack and Mediant 1000 SIP Release Notes 2 SIP Compatibility gt INFO method compatible with Cisco gateways gt NOTIFY method lt draft mahy sipping signaled digits 01 txt gt e SIP URL sip phone number IP address such as 122 10 1 2 4 where 122 is the phone number of the source or destination phone number or sip phone_number domain name such as 122 myproxy com Note that the SIP URI host name can be configured differently per called number e Can negotiate coder from a list of given coders e Supported coders gt G 711 A law 64 kbps 10 20 30 40 50 60 80 100 120 msec gt G 711 u law 64 kbps 10 20 30 40 50 60 80 100 120 msec gt G 723 1 5 3 6 3 kbps 30 60 90 msec gt G 726 32 kbps 10 20 30 40 50 60 80 100 120 msec gt G 729A B 8 kbps 10 20 30 40 50 60 msec e Implementation of MWI IETF lt draft ietf sipping mwi 04 txt gt including SUBSCRIBE to the MWI server The MediaPack FXS gateways can accept an MWI NOTIFY message that indicates waiting messages or indicates that the MWI is cleared 2 2 Unsupported SIP Features The following SIP features are NOT supported e MESSAGE method e Preconditions RFC 3312 e SDP Simple Capability Declaration RFC 3407 e Proxy discovery using NAPTR DNS records e Multicast e GRUU Version 4 6 25 June 2005 7a Ta wi AudioCodes MediaPack and Mediant 1000 SIP 2 3 SIP Compliance Tables The MediaPack SIP
47. Web Server refer to the Security section in the User s Manual HTTPSCertFileName 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 gateway is loaded via BootP TFTP For information on loading this file via the Embedded Web Server refer to the Security section in the User s Manual Telnet Parameters TelnetServerEnable Enables or disables the embedded Telnet server Telnet is disabled by default for Embedded Telnet Server security reasons 0 Disable default 1 Enable Unsecured 2 Enable Secured SSL Applicable only to MP 11x and Mediant 1000 TelnetServerPort Defines the port number for the embedded Telnet server Telnet Server TCP Port The valid range valid port numbers The default port is 23 TelnetServerldleDisconnect Sets the timeout for disconnection of an idle Telnet session in minutes When set to Telnet Server Idle Timeout zero idle sessions are not disconnected The valid range is any value The default value is 0 IP Routing Table parameters The IP routing ini file parameters are array parameters Each parameter configures a specific column in the IP routing table The first entry in 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 conf
48. address 301 Moved Yes The gateway responds with an ACK and resends the request to new Permanently address 302 Moved Yes The SIP gateway generates this response when call forward is used to Temporarily redirect the call to another destination If such response is received the calling gateway initiates an INVITE message to the new destination 305 Use Proxy Yes The gateway responds with an ACK and resends the request to new address 380 Alternate Yes Service 2 3 5 4 4xx Response Request Failure Responses Table 2 8 4xx SIP Responses continues on pages 29 to 31 4xx Response Supported Comments 400 Bad Request Yes The gateway does not generate this response On reception of this message before a 200OK has been received the gateway responds with an ACK and disconnects the call 401 Unauthorized Yes Authentication support for Basic and Digest On receiving this message the GW issues a new request according to the scheme received on this response 402 Payment Yes The gateway does not generate this response On reception of this Required message before a 2000K has been received the gateway responds with an ACK and disconnects the call 403 Forbidden Yes The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call Version 4 6 29 June 2005 a Ce AudioCodes MediaPack and Mediant 1000 SIP Table 2 8 4xx
49. ame coder is used for transmit and for receive though different ptime is supported Coder names in ini file are case sensitive The RFC2833RxPayloadType and RFC2833TxPayloadType parameters in the Embedded Web Server s Channel Settings screen or in the ini file should not be used Use the parameter Rfc2833PayloadT ype instead Configuring the board to auto negotiate mode while the opposite port is set manually to full duplex either 10 Base T or 100 Base TX is invalid It is also invalid to set the board to one of the manual modes while the opposite port is configured differently It is recommended to use full duplex connections instead of half duplex and 100 Base TX instead of 10 Base T due to the larger bandwidth It is strongly recommended to use 100 Base T switches Use of 10 Base T LAN hubs should be avoided In some cases when the spanning tree algorithm is enabled on the external Ethernet switch port connected to the gateway the external switch blocks traffic entering and exiting the gateway for some time after the gateway is reset This may cause the loss of important packets such as BootP and TFTP requests which in turn may cause the board to fail to start up A possible workaround for this issue is to set the parameter BootPRetries to 5 forcing the gateway to issue 20 BootP requests for 60 seconds A second workaround is to disable the spanning tree algorithm on the port of the external switch tha
50. ameter Once defined and saved in the flash memory it is used even if it doesn t appear in the ini file SaveConfiguration Set to 1 to store the Call Progress Tones and Coefficient files in the non volatile memory Note The parameters BurnCallProgressToneFile and BurnCoeffFile are no longer supported IsCiscoSCEMode 0 There isn t a Cisco gateway at the remote side default 1 There is a Cisco gateway at the remote side When there is a Cisco gateway at the remote side the local gateway must set the value of the annexb parameter of the fmtp attribute in the SDP to no This logic should be used if EnableSilenceCompression 2 enable without adaptation In this case Silence Suppression should be used on the channel but not declared in the SDP SNMP Parameters SNMPTrustedMGR_x Up to five IP addresses of remote trusted SNMP managers from which the SNMP agent accepts and processes get and set requests Note 1 If no values are assigned to these parameters any manager can access the device Note 2 Trusted managers can work with a community strings SNMPReadOnlyCommunity Read only community string up to 19 chars String_x The default string is public SNMPReadWriteCommunity Read write community string up to 19 chars String_x The default string is private SNMPTrapCommunityString Community string used in traps up to 19 chars x The default string is trapuser MediaPack and
51. an now view the Caller ID string The feature is supported for the following Caller ID types Bellcore and ETSI Applicable only to FXS gateways MediaPack and Mediant 1000 SIP Release Notes 38 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History 21 22 23 24 25 26 Previous Version Users can now configure the gateway to receive T 38 fax relay packets into the same port used by the RTP packets instead of the RTP port 2 This solves compatibility issues with certain NATs and Firewalls Relevant parameter T38UseRTPPort Generation of date and time with Caller ID is now supported The date and time are obtained from the internal gateway clock or from NTP Network Time Protocol if enabled Relevant parameters NTPServerlP NTPServerUTCOffset and NTPUpdatelnterval Users can now configure the duration of the current disconnect signal for FXS gateways and the detection range of the current disconnect signal for FXO gateways Relevant parameter CurrentDisconnectDuration Supports the generation of Caller ID with distinctive ringing Relevant parameter and value AnalogCaller DTimingMode 1 T 38 Redundancy Enhancement The redundancy of the low speed data is now determined according to the enhanced redundancy parameter Optimization of channel parameters when detecting fax or modem signals applicable only if the channel was opened with the G 711 coder When detecting a f
52. applicable only to FXS gateways Enables the use of DNS Service Record SRV queries to discover Proxy servers 0 Disabled default 1 Enabled If enabled and the Proxy IP address parameter contains a domain name without port definition e g ProxyIP domain com an SRV query is performed The SRV query returns up to four Proxy host names and their weights The gateway 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 2 IP addresses each no more searches are performed If the Proxy IP address parameter contains a domain name with port definition e g ProxylP domain com 5080 the gateway performs a regular DNS A record query Note This mechanism is applicable only if EnableProxyKeepAlive 1 IP address of the primary Proxy server you are using Enter the IP address as FQDN or in dotted format notation for example 201 10 8 1 You can also specify the selected port in the format lt IP Address gt lt port gt This parameter is applicable only if you select Yes in the Is Proxy Used field If you enable Proxy Redundancy by setting EnableProxyKeepAlive 1 the gateway can function with up to three Proxy servers If there is no response from the primary Proxy the gateway tries to communicate with the redundant Proxies When a redundant
53. ax or modem signal on the terminating or originating sides the gateway modifies the channel s settings to work with voice band data signals such as disable NLP disable or enable Echo Canceler EC is enabled for fax calls and disabled for modem calls disable silence suppression and setting optimized Jitter Buffer mode Relevant parameters and values FaxTransportType 3 and VxxModemTransportType 3 Transparent with events 4 1 2 Routing and Manipulation New Features 27 28 29 Version 4 6 An option was added to the Tel to IP Routing table to take precedence over a Proxy for routing calls When this option is enabled the gateway checks the Destination IP Address field in the Tel to IP Routing table for a match with the outgoing call Only if a match is not found a Proxy is used Relevant parameters PreferRouteTable AlwaysSendToProxy SendinviteToProxy Alternative routing for released calls for both Tel to IP and IP to Tel calls Users can now define several call release reasons to be used for alternative routing If a new call is released as a result of one of these reasons the gateway tries to find an alternative routing rule to that call If such a rule is found the gateway immediately performs a new call according to that rule In the current release only one alternative rule can be defined Note that if there is no response from the remote party the call is released internally with a 408 reason T
54. ay Registration or multiple Registration of all gateway endpoints e Configuration of authentication username and password per each gateway endpoint or single username and password per gateway e Supported methods INVITE CANCEL BYE ACK REGISTER OPTIONS INFO REFER NOTIFY PRACK UPDATE and SUBSCRIBE e Modifying connection parameters for an already established call re INVITE e Working with Redirect server and handling 3xx responses e Early media Supporting 183 Session Progress e PRACK reliable provisional responses RFC 3262 e Call Hold and Transfer Supplementary services using REFER Refer To Referred By Replaces and NOTIFY e Call Forward using 302 response Immediate Busy No reply Busy or No reply Do Not Disturb e Supports RFC 3327 Adding Path to Supported header e Supports RFC 3581 Symmetric Response Routing e Supports RFC 4028 Session Timers in SIP e Supports network asserted identity and privacy RFC 3325 and RFC 3323 e Supports Tel URI Uniform Resource Identifier according to RFC 2806 bis e Remote party ID lt draft ietf sip privacy 04 txt gt e Supports obtaining Proxy Domain Name s from DHCP Dynamic Host Control Protocol according to RFC 3361 e RFC 2833 Relay for DTMF Digits including payload type negotiation e DTMF out of band transfer using gt INFO method lt draft choudhuri sip info digit 00 txt gt MediaPack and Mediant 1000 SIP Release Notes 24 Document LTR
55. caller to hang up before disturbing the called party with Call Waiting Indications Applicable only to FXS gateways Relevant parameter TimeBeforeWaitingIndication Max call duration Users can now limit the maximum duration of a call When this time expires the call is released from both sides IP and Tel Relevant parameter MaxCallDuration Hotline Dial Tone Duration Users can now define the dial tone duration after which a port acts as a Hotline If the gateway received digits during this time period the call process continues as usual and the Hotline feature isn t used Relevant parameter HotLineDialToneDuration Cut Through feature An option to receive incoming IP calls on a port in an offhooked state was added Applicable only to MP 1xx FXS Relevant parameter CutThrough Additional fields were added to CDR reports Call Setup Time Call Connect Time Call Release Time RTP Delay and Jitter RTP SSRC of local and remote sides Redirect number Redirect TON NPI and Redirect reason Note The Call Time parameters are included in the CDR only if NTP is used or if the gateway s local time and date were configured An option to configure a separate destination IP address for CDR Syslog reports was added in order to work smoothly with third party billing servers Relevant parameter CDRSyslogServerIP Metering Tones Relay When an FXO gateway detects a 12 16 kHz metering tone it now sends an INFO message over
56. ctory defaults 3 2 SIP Constraints The Netcoder coder is no longer supported When using out of band DTMF transport IsDTMFUsed 1 the DTMFTransportType parameter should be set to 0 erase digits from voice stream If the first incoming INVITE message contains both audio and T 38 coders the gateway will reply with the first media in SDP and not with an audio coder as was in 4 21 version G 726 16 kbps 24 kbps and 40 kbps coders are not supported Only G 726 32 kbps is supported Only the ptime packetization time of the first coder in the defined coder list is declared in the SDP section of INVITE 200 OK messages even if multiple coders are defined Therefore in the Coders screen in the Web Interface only the ptime of the first coder in the list is relevant For example if G 711 and G 723 coders are used the ptime is set to 30 msec The number of RTP payloads packed in a single G 729 packet M channel parameter is limited to 5 In the current Voice Mail VM implementation Supervised Transfer isn t supported Supervised Transfer notifies the VM application if transfer fails because the transferred extension is busy 3 3 Gateway Constraints 10 11 12 13 Version 4 6 When upgrading the MediaPack loading new software onto the gateway from version 4 4 to version 4 6 using the BootP TFTP configuration utility the device s auxiliary files CPT logo etc are erased It is highly
57. d DTMF gains Silence suppression Echo Canceler RTP DiffServ current disconnect reverse polarity and more The Profiles feature allows the user to tune these parameters or turn them on or off per source or destination routing and or the specific gateway or its ports For example analog ports can be designated for Fax only by having a profile which always uses G 711 For more detailed information on the Profiles feature refer to the MP 1xx SIP User s Manual Users can now monitor SIP real time activity such as call details and call statistics including the number of call attempts failed calls fax calls etc The accumulated data can be viewed in the Embedded Web Server Status and Diagnostics menu and via SNMP Cisco NSE mode is now supported for fax pass through in addition to the existing support for modem Relevant parameters NSEMode NSEPayloadType The following two additional Call Forward modes are now supported gt Busy or No Reply In this mode calls are forwarded either when the gateway s port is busy or when the call is not answered after a configurable period of time gt Do Not Disturb In this mode incoming calls are immediately released This feature is applicable only to MP 1xx FXS Relevant parameter FWDInfo_x FXS gateways now support subscriber activation and deactivation of the Call Forward Caller ID Restriction CLIR and Hotline features directly from the connected telephone s k
58. dServiceClassPriori ty Gold Priority VlanBronzeServiceClassPri ority Bronze Priority EnableDNSasOAM EnableNTPasOAM Multiple IPs Parameters EnableMultiplelPs IP Networking Mode LocalMedialPAddress IP Address LocalMediaSubnetMask Subnet Mask LocalMediaDefaultGW Default Gateway Address LocalControllPAddress IP Address LocalControlSubnetMask Subnet Mask LocalControlDefaultGW Default Gateway Address LocalOAMIPAddress IP Address LocalOAMSubnetMask Subnet Mask LocalOAMDefaultGW Default Gateway Address Description Sets the priority for the Premium service class content and control traffic The valid range is 0 to 7 The default value is 6 Sets the priority for the Gold service class content The valid range is 0 to 7 The default value is 4 Sets the priority for the Bronze service class content The valid range is 0 to 7 The default value is 2 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 OAM default 0 Control 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 OAM default 0 Control Enables disables the Multiple IPs mechanism 0 Disabled default 1 Enabled The gateway s source IP addr
59. de wildcards IP addresses in the Source IP Address column of the IP to Hunt Group Routing table and the Source IP column in the Destination Phone Number Manipulation Table for IP to Tel Calls can include the x wildcard that represents single digits For example 10 8 8 x 10 8 8 0 10 8 8 9 10 8 8 xx 10 8 8 10 10 8 8 99 10 8 xx xxx 10 8 10 100 10 8 99 255 Relevant parameters PSTNPrefix NumberMapIP2Tel Supports digit delivery to the IP side Using the manipulation tables the gateway can now be configured to play pre configured DTMF digits per call after the call is answered Relevant parameter EnableDigitDelivery2 IP IP DiffServ code can now be configured for SIP signaling protocol in addition to RTP Diffserv Relevant parameter ControllPDiffServ The Called Number Manipulation table was increased to 50 rows The Calling Number Manipulation table was increased to 20 rows MediaPack and Mediant 1000 SIP Release Notes 40 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History Previous Version 4 1 3 SIP New Features 40 41 42 43 44 45 46 47 48 Version 4 6 Locating SIP Proxy servers The gateway can now use DNS Service Record SRV queries to discover Proxy servers If the Proxy IP address parameter contains a domain name without port definition e g ProxylP domain com an SRV query is performed if enabled T
60. e detection range of a Flash Hook signal for FXO ports via the Channel Settings screen in the Embedded Web Server 4 1 5 Miscellaneous New Features 71 72 73 74 75 Version 4 6 Support for prerecorded Call Progress Tones was added Using the TrunkPack Downloadable Conversion Utility users can now create a file that contains prerecorded tones Each tone is assigned with a tone type After loading it to the device the prerecorded tones are played as regular Call Progress Tones according to the tone types No detection is supported for these tones The prerecorded tones file can be burned to the non volatile memory Relevant parameter PrerecordedTonesFileName filename Users can now instruct the gateway to load a new software cmp file and or configuration files from a preconfigured TFTP server after a Web SNMP reset Therefore the gateway can now obtain its networking parameters from BootP or DHCP servers and its software and configuration files from a different TFTP server preconfigured in ini file The ini file can be loaded according to a specific gateway s MAC address enabling easy configuration for different gateways Relevant parameters IniFileURL CmpFileURL An external utility CPTWizard simplifies the MP 10x FXO configuration task by automatically detecting the local set of Call Progress Tones generated by the switch PBX The utility creates a CPT ini configuration file NTP support The
61. ediant 1000 SIP Fast Track Installation Guide Version 4 6 5 June 2005 x aA r K AudioCodes MediaPack and Mediant 1000 SIP MP 1xx refers to the MP 124 24 port MP 108 8 port MP 104 4 port and MP 102 2 port VoIP gateways having similar functionality except for the number of channels the MP 124 and MP 102 support only FXS MP 10x refers to MP 108 8 port MP 104 4 port and MP 102 2 port gateways MP 1xx FXS refers only to the MP 124 FXS MP 108 FXS MP 104 FXS and MP 102 FXS gateways MP 10x FXO refers only to MP 108 FXO and MP 104 FXO gateways MP 11x refers to the MP 118 8 port MP 114 4 port and MP 112 2 port FXS VoIP gateways having similar functionality except for the number of channels These Release Notes describe the MP 1xx SIP VoIP gateways the MP 11x SIP VoIP gateways and the analog Mediant 1000 VoIP gateway Unless otherwise specified whenever reference is made to the MediaPack in these Release Notes it automatically includes the MP 11x and the analog Mediant 1000 MediaPack and Mediant 1000 SIP Release Notes 6 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 1 What s New Release in 4 6 1 1 Version 4 6 What s New in Release 4 6 General Gateway New Features 1 MP 1xx FXO and Mediant 1000 FXO only Line Disconnection The status of the analog phone line is now examined before proceeding with a new IP to Tel call If the line is disconnected the call is released wit
62. ee network environment that is focused on security and segregation Each entity in the MediaPack e g Web RTP is mapped to a single traffic type in which it operates Relevant Parameters EnableMultiplelPs LocalMedialPAddress LocalMediaSubnetMask LocalMediaDefaultGW LocalControllPAddress LocalControlSubnetMask LocalControlDefaultGW LocalOAMIPAddress LocalOAMSubnetMask LocalOAMDefaultGW EnableDNSasOAM EnableNTPasOAM Support for 802 1p Q VLANs and priority was added The MediaPack can now be integrated into a VLAN aware environment that includes switches routers and endpoints Relevant Parameters VianMode VilanNativeVianID VianOamVlanID VianControlVlanID VianMediaVlanID VlanNetworkServiceClassPriority VlanPremiumServiceClassMediaPriority VianPremiumServiceClassControIPriority VianGoldServiceClassPriority VianBronzeServiceClassPriority EnableDNSasOAM EnableNTPasOAM MP 11x and Mediant 1000 only Silence Indicator SID packets that are sent and received according to RFC 3389 can now contain spectral coefficients information The number of coefficients that are added to the SID packets can be determined using the parameter RTPS IDCoeffNum Relevant parameters RTPSIDCoeffNum The IP address translation mechanism used for far end NAT traversal now supports T 38 in addition to RTP Relevant parameters EnablelPAddrTranslation DisableNAT Support for injection and detection of NTT Caller ID type 2 offhook was added
63. ens cannot be accessed Reset Save Configuration Software Upgrade Wizard Load Auxiliary Files Configuration File and Regional Settings The Change Password screen can only be used to change the monitoring password Default username User Default password User MediaPack and Mediant 1000 SIP Release Notes 10 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 1 What s New Release in 4 6 33 34 35 36 37 38 A new Calls Routing Status screen was added This screen provides information on the current routing method used by the gateway This information includes the IP address and FQDN if used of the Proxy server the gateway currently operates with DateAndTime VarBind Variable Binding was added to all AC traps One of the five available SNMP managers can now be defined using a FQDN The resolved IP address appears in the bottom row of the trap managers table Relevant parameter SNMPTrapManagerHostName A new Performance Monitoring infrastructure enables collecting and retrieving current and historical performance data via SNMP Changes made on the fly to parameters via Web or SNMP can now be viewed in the Syslog server Relevant parameter EnableParametersMonitoring An embedded Command Line Interface CLI is now available on the MediaPack The CLI can be accessed via Telnet RS 232 and the Embedded Web Server You can use the CLI for diagnostics
64. eral SNMP managers can now be configured to access the gateway concurrently Caller ID can also be generated for Distinctive Ringing signals if AnalogCallerIDTimingMode 1 MediaPack and Mediant 1000 SIP Release Notes 44 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History Previous Version 10 DHCP now supports limited IP leasing time The gateway performs lease renewal and initiates a new DHCP request when the lease time expires 11 All request URI s for mid dialog requests issued by the gateway contains all URI parameters received in contact record route 12 Send an immediate NOTIFY with 100 trying as a result of a received REFER request 13 Requests URI s for INVITE request issued as a result of REFER 3xx will contain all URI parameters and new headers received in the REFER to contact headers 14 Up to four Proxies are now supported 4 1 7 New and Modified Parameters Most new parameters described in Table 4 1 can be configured with the ini file and via the Embedded Web Server Note that only those parameters contained within square brackets are configurable via the Embedded Web Server Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name PreferRouteTable Prefer Routing Table EnablePtime GWAppDelayTime Delay After Reset CurrentDisconnectDefaultT hreshold TimeToSampleAnalogLineV
65. erred Profile will be applied to that call If the Preference of the Tel and IP Profiles is identical the Tel Profile parameters will be applied For example IPProfile_1 name1 2 1 0 10 13 15 44 1 1 IPProfile_2 name2 1 Not configured the default value of the parameter is used Common parameter used in both IP and Tel profiles Note 1 The IP ProfilelD can be used in the Tel2IP and IP2Tel routing tables Prefix and PSTNPrefix parameters Note 2 Profile Name assigned to a ProfilelD enabling User s to identify it intuitively and easily Note 3 This parameter can appear up to 4 times TelProfile_ lt Profile ID gt lt Profile Name gt lt Preference gt lt Coder Group ID gt lt IsFaxUsed gt lt DJBufMinDelay gt lt DJBufOptFactor gt lt IPDiffServ gt lt ControllPDiffServ gt lt DtmfVolume gt lt InputGain gt lt VoiceVolume gt lt EnableReversePolarity gt lt EnableCurrentDisconnect gt lt EnableDigitDelivery gt lt ECE gt Preference 1 10 The preference option is used to determine the priority of the Profile If both IP and Tel profiles apply to the same call the coders and other common parameters of the preferred Profile will be applied to that call If the Preference of the Tel and IP Profiles is identical the Tel Profile parameters will be applied For examples TelProfile_1 FaxProfile 1 2 0 10 5 22 33 2 22 34 1 0 1 1 TelProfile_2 ModemProfile 0 10 1
66. ers p 1 5 seconds pause and d detection of dial tone If the character d is used it must be the first digit in the called number The character p can be used several times For example the called number can be as follows d1005 dpp699 p9p300 To add the d and p digits use the usual number manipulation rules Note 2 To use this feature with FXO gateways configure the gateway to work in one stage dialing mode Note 3 If the parameter EnableDigitDelivery is enabled it is possible to configure the gateway to wait for dial tone per destination phone number before or during dialing of destination phone number therefore the parameter IsWaitForDialTone that is configurable for the entire gateway is ignored Note 4 The FXS gateway sends 200 OK messages only after it finishes playing the DTMF digits to the phone line SendMetering2IP 0 Disabled default Send Metering Message to FXO gateways send a metering tone message to IP on detection of 12 16 kHz IP metering pulse FXS gateways generate the 12 16 kHz metering tone on reception of a metering message Note Suitable 12 kHz or 16 kHz coeff file must be used for both FXS and FXO gateways The MeteringType parameter must be defined in both FXS FXO gateways MeteringType Defines the metering tone 12 kHz or 16 kHz that is detected by FXO gateways and generated by FXS gateways 0 12 kHz metering tone default
67. es separately This method is usually used for FXS gateways default 1 Per gateway Single subscription for the entire gateway This method is usually used for FXO gateways 45 June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name Description FWDInfo_X Forward incoming IP calls using 302 response based on the gateway port to which the Call Forward Table call is routed FwdlInfo_ lt Gateway Port Number 0 to 23 gt lt Forward Type gt lt Forwarded SIP User Identification gt lt Timeout in seconds for No Reply gt 0 Not in use 1 On busy forward incoming calls when the port is busy 2 Immediate always forward any incoming call 3 No reply forward incoming calls that are not answered after a configurable period of time 4 On busy or No reply forward incoming calls when the port is busy or when calls are not answered after a configurable period of time 5 Do Not Disturb immediately reject incoming calls Note 1 Applicable only to MP 1xx FXS gateways Note 2 When a Proxy isn t used the Forward to Phone Number must be specified in the Tel to IP Routing table of the forwarding gateway EnableDID_X Enables generation of Japan NTT Modem DID signal per port EnableDID_ lt Port gt lt Modem DID gt Modem DID 0 Disabled default 1 Enabled I
68. esponds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call The gateway does not generate this response On reception of this message before a 2000K has been received the gateway responds with an ACK and disconnects the call MediaPack and Mediant 1000 SIP Release Notes 30 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 2 SIP Compatibility 4xx Response 485 486 487 488 Table 2 8 4xx SIP Responses continues on pages 29 to 31 Supported Ambiguous Yes Busy Here Yes Request Canceled Yes Not Acceptable Yes Comments The gateway does not generate this response On reception of this message before a 200OK has been received the gateway responds with an ACK and disconnects the call The SIP gateway generates this response if the called party is off hook and the call cannot be presented as a call waiting call On receiving this response the gateway notifies the User and generates a busy tone This response indicates that the initial request is terminated with a BYE or CANCEL request The gateway does not generate this response On reception of this message before a 2000K ha
69. ess in the Media network The default value is 0 0 0 0 The gateway s subnet mask in the Media network The default subnet mask is 0 0 0 0 The gateway s default gateway IP address in the Media network The default value is 0 0 0 0 The gateway s source IP address in the Control network The default value is 0 0 0 0 The gateway s subnet mask in the Control network The default subnet mask is 0 0 0 0 N A Use the IP Routing table instead Advanced Configuration gt Network Settings The gateway s source IP address in the OAM network The default value is 0 0 0 0 The gateway s subnet mask in the OAM network The default subnet mask is 0 0 0 0 N A Use the IP Routing table instead Advanced Configuration gt Network Settings Automatic Update Parameters CmpFileURL Version 4 6 Specifies the name of the cmp file and the location of the server IP address or FQDN from which the gateway loads a new cmp file and updates itself The cmp file can be loaded using TFTP HTTP or HTTPS For example tftp 192 168 0 1 filename Note 1 When this parameter is set in the ini file the gateway always loads the cmp file after it is reset Note 2 The cmp file is validated before it is burned to flash The checksum of the cmp file is also compared to the previously burnt checksum to avoid unnecessary resets 21 June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP Table 1 1 Release 4 6 ini File Web
70. eypad Activation deactivation is invoked by dialing a pre configured sequence Successful configuration of these features is followed by a confirmation tone Relevant parameters KeyCFUncond KeyCFNoAnswer KeyCFBusy KeyCFBusyOrNoAnswer KeyCFDoNotDisturb KeyCFDeact KeyCLIR KeyCLIRDeact KeyHotLine KeyHotLineDeact Japan NTT Modem DID support FXS gateways can now be connected to Japan s NTT PBX using Modem DID lines These DID lines are used to deliver a called number to the PBX Applicable for FXS gateways The DID signal can be sent alone or combined with a NTT Caller ID signal This feature can be enabled disabled per port currently can only be configured via the ini file Relevant parameters EnableDID with NTT CallerlDType EnableDID_X Caller ID generation for FXS gateways and detection for FXO gateways can now be enabled or disabled per port and not only for the entire gateway Relevant parameter EnableCallerlD_X An option was added to configure the number of rings after which the gateway detects Caller ID Applicable only to FXO gateways Relevant parameter RingsBeforeCallerID 37 June 2005 A 9 10 11 12 13 14 15 16 17 18 19 20 WH 2 wt AudioCodes MediaPack and Mediant 1000 SIP Call Waiting Indication delay Users can now configure a delay interval before a Call Waiting Indication is played to the currently busy port This enables the
71. f not configured use the global parameter EnableDID Note Applicable only to MP 1xx FXS gateways EnableCallerlID_X Enables Caller ID generation FXS or detection FXO per port Generate Caller ID to Tel Detect Caller ID from Tel EnableCallerlID_ lt Port gt lt Caller ID gt Caller ID 0 Disabled default 1 Enabled If not configured use the global parameter EnableCaller D Note 1 The numbering of ports starts with 0 Note 2 This parameter can appear up to eight times for MP 108 and up to 24 times for MP 124 RingsBeforeCallerlD Sets the number of rings before the gateway starts detection of Caller ID FXO only Rings before Detecting Caller 0 Before first ring ID 1 After first ring default 2 After second ring Timeserorewatingindi catio Defines the interval in seconds before a call waiting indication is played to the port that n i is currently in a call FXS only er Waiting The valid range is 0 to 100 The default time is 0 seconds MaxCallDuration Defines the maximum call duration in minutes If this time expires both sides of the call Max Call Duration are released IP and Tel The valid range is 0 to 120 The default time is 0 no limitation HotLineDialToneDuration Duration in seconds of the Hotline dial tone Hot Line Dial Tone Duration If no digits are received during the Hotline dial tone duration the gateway initiates a call to a preconfigured numbe
72. f 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 illuminated The gateway can subscribe to this service per port usually used on FXS or per gateway used on FXO Relevant parameters EnableMWI MWIServerlP MWIAnalogLamp MWIDisplay StutterToneDuration SubscriptionMode Subscription unsubscription to the MWI service can now be controlled via the Supplementary Services screen in the Embedded Web Server Relevant parameters EnableMWISubscription MWIExpirationTime SubscribeRetryTime Support for Path Extension Header according to RFC 3327 was added The gateway adds a Path parameter to the Supported header field of REGISTER messages This field allows to accumulate the list of Proxies IP addresses between the gateway and the Registrar The gateway can also receive the Path header in a response IP Alert Timeout Users can now define a timer for the gateway to wait for a 200 OK response from the called party IP side If the timer expires the call is released Relevant parameter IPAlertTimeout Users can now use the SDP attribute a sendonly to place the remote party on hold in addition to the use of the IP address of 0 0 0 0 and the attribute a inactive Relevant parameter HoldFormat RFC 2833 Negotiation If the remote side doesn t include the telephone event
73. from which the gateway loads a new cmp file and updates itself For example tftp 192 168 0 1 filename Note 1 When this parameter is set in the ini file the gateway always loads the cmp file after it is reset Note 2 The version of the loaded cmp file isn t checked Defines the user name that is used in From and To headers of REGISTER messages Applicable only to single registration per gateway AuthenticationMode 1 If GWRegistrationName isn t specified default the Username parameter is used instead Note If AuthenticationMode 0 all the gateway s endpoints are registered with a user name that equals to the endpoint s phone number Registrar Domain Name If specified the name is used as Request URI in REGISTER messages If isn t specified default the Registrar IP address or Proxy name or Proxy IP address is used instead Defines the re registration timing in percentage The timing is a percentage of the re register timing set by the Registration server The valid range is 50 to 100 The default value is 50 For example If RegistrationTimeDivider 70 and Registration Expires time 3600 the gateway resends its registration request after 3600 x 70 2520 sec Defines the time in seconds the gateway 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
74. gateways comply with RFC 3261 as shown in the following sections 2 3 1 SIP Functions Function User Agent Client UAC User Agent Server UAS Proxy Server Redirect Server Registrar Server Table 2 1 SIP Functions Supported Yes Yes Third party only Checked with Ubiquity Delta3 Microsoft 3Com Snom and Cisco Proxies Third party Third party 2 3 2 SIP Methods Method INVITE ACK BYE CANCEL REGISTER REFER NOTIFY INFO OPTIONS PRACK UPDATE Table 2 2 SIP Methods Supported Comments Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Send only Receive only 2 3 3 SIP Headers The following SIP Headers are supported by the MediaPack SIP gateway Header Field Accept Accept Encoding Alert Info Allow Also Asserted ldentity Authorization Call ID Table 2 3 SIP Headers continues on pages 26 to 27 Supported Yes Yes Yes Yes Yes Yes Yes Yes MediaPack and Mediant 1000 SIP Release Notes 26 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes Header Field Call Info Contact Content Encoding Content Length Content Type Cseq Date Diversion Encryption Expires Fax From Max Forwards Messages Waiting MIN SE Organization Priority Proxy Authenticate Proxy Authorization Proxy Require Prack Record Route Refer To Referred By Remote Party ID Replaces Require Remote Party ID Response Key Retry After Route Rseq Session Expires Serve
75. h a 4xx response If the line is disconnected during a call the call is released immediately The gateway now supports the ThroughPacket mechanism 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 ThroughPacket can be applied to the entire gateway or using IP Profile to specific IP addresses Relevant parameters BaseUDPPort RemoteBaseUDPPort L1L1ComplexTxUDPPort L1L1ComplexRxUDPPort IPProfile_ID Support was added for generation of the following Caller ID type 1 standards ETSI before ring DT AS ETSI before ring RP AS ETSI before ring LR DT AS ETSI not ring related DT AS ETSI not ring related RP AS ETSI not ring related LR DT AS and Bellcore not ring related Relevant parameters BellcoreCallerlID TypeOneSubStandard ETS ICallerlDTypeOneSubStandard Support was added for generation of the following Message Waiting Indication type 1 standards ETSI before ring DT AS ETSI before ring RP AS ETSI before ring LR DT AS ETSI not ring related DT AS ETSI not ring related RP AS ETSI not ring related LR DT AS Bellcore not ring related Relevant parameters ETSIVMWITypeOneStandard BellcoreVMWITypeOneStandard MP 11x and Mediant 1000 only The In Band Signaling IB
76. he SRV query returns up to four Proxy host names and their weights The gateway 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 2 IP addresses each no more searches are performed If the Proxy IP address parameter contains a domain name with port definition e g ProxylP domain com 5080 the gateway performs a regular DNS A record query Note This mechanism is applicable only if EnableProxyKeepAlive 1 Relevant parameter EnableProxySRVQuery Support for SIP UPDATE method according to RFC 3311 was added the gateway doesn t initiate UPDATE messages but responds to them Network Asserted Identity RFC 3325 supporting both P Asserted and P Preferred Identity headers Relevant parameters AssertedildMode IsTrustedProxy Support for the Privacy header RFC 3323 and RFC 3325 was added If Caller ID is restricted the INVITE message will include a Privacy header with id parameter privacy id The privacy header is used together with P asserted or P preferred headers Proxy Domain Name s can now obtained from a DHCP server according to RFC 3361 Symmetric Response Routing according to RFC 3581 is now supported The gateway adds a rport parameter to the Via header field of each SIP message The first Proxy that receives this message sets
77. he called and calling numbers match the a and g conditions The manipulation rules are executed in the following order b d and c Parameters can be skipped by using the sign for example SourceNumberMapTel2IP 01 2 972 0 0 1 SourceNumberMapTel2IP 03 2 667 0 0 22 Note 1 Presentation is set to Restricted only if Asserted Identity Mode is set to P Asserted Note 2 Number Plan amp Type can optionally be used in Remote Party ID RPID header by using the EnableRPlHeader and AddTON2RPI parameters Manipulate the destination number for IP to Tel calls NumberMapIP2Tel a b c d e f g a Source number prefix b Number of stripped digits from the left or if brackets are used from the right A combination of both options is allowed c String to add as prefix or if brackets are used as suffix A combination of both options is allowed d Number of remaining digits from the right e Notin use should be set to f Not in use should be set to g Destination number prefix The b to d manipulation rules are applied if the called and calling numbers match the a and g conditions The manipulation rules are executed in the following order b d and c Parameters can be skipped by using the sign for example NumberMapIP2Tel 01 2 972 034 NumberMapIP2Tel 03 2 667 22 Defines
78. he called party to distinguish between four different call origins e g external vs internal calls The gateway 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 1 It is assumed that all Call Waiting Tones are defined in sequence in the CPT file Note 2 This feature is relevant only to Broadsoft s application servers the tone is played using INFO message EnableProxySRVQuery Enables the use of DNS Service Record SRV queries to discover Proxy servers Enable Proxy SRV Queries 0 Disabled default 1 Enabled If enabled 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 gateway 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 2 IP addresses each no more searches are performed If the Proxy IP address parameter contains a domain name with port definition e g ProxylP domain com 5080 the gateway performs a regular DNS A record query Note When enabled SRV queries are used to discover Proxy servers even if the parameter EnableSRVQuery
79. his internal reason can be also used to initiate an alternative call The timeout for no response decision depends on the alternative IP addresses a Ifthe resolution of the called domain name results with two IP addresses the no response timeout will be according to the number of Hot Swap retransmissions using the parameter ProxyHotSwapRitx default 3 retransmissions b Otherwise the no response timeout will be according to the usual number of the SIP retransmissions 7 default For Tel to IP calls this feature is relevant only if the internal Tel to IP routing table is used to route the calls This feature isn t applicable when Proxy is used to route Tel to IP calls Relevant parameters AltRouteCauselP2Tel AltRouteCauseTel2IP PSTNPrefix A new Status Only mode was added to the Alternative Routing feature The new IP Connectivity screen can be used to display the status of IP address connections using Ping and QoS results without enabling disabling the routing rules Relevant parameter AltRoutingTel2IPEnable 39 June 2005 x A 30 31 32 33 34 35 36 37 38 39 WH 3 wt AudioCodes MediaPack and Mediant 1000 SIP Internal DNS table was added Similar to a DNS resolution translates hostnames into IP addresses This table is used when hostname translation is required e g Tel to IP Routing table Two different IP addresses can be assigned t
80. igured when the gateway 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 RoutingTableDestinationsC Specifies the IP address of the destination host network olumn RoutingTableDestinationMa Specifies the subnet mask of the destination host network sksColumn RoutingTableGatewaysColu Specifies the IP address of the router to which the packets are sent if their destination mn matches the rules in the adjacent columns RoutingTableHopsCountCo The maximum number of allowed routers between the gateway and destination lumn Version 4 6 19 June 2005 fal AudioCodes MediaPack and Mediant 1000 SIP Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name Description RoutingTablelnterfacesColu Specifies the network type the routing rule is applied to mn 0 OAM default 1 Control 2 Media RADIUS Login Authentication Parameters MP 11x and Mediant 1000 only EnableRADIUS Enable RADIUS Access Control WebRADIUSLogin Use RADIUS for Web Telnet Login RADIUSAuthServerIP RADIUS Authentication Server IP Address RADIUSAuthPort RADIUS Authentication Server Port SharedSecret RADIUS Shared Secret
81. is disabled MediaPack and Mediant 1000 SIP Release Notes 14 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 1 What s New Release in 4 6 Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name EnableSRVQuery Enable SRV Queries IsUseToHeaderAsCalledNu mber UseDisplayNameAsSource Number Use Display Name as Source Number RemoteBaseUDPPort Remote RTP Base UDP Port LiL1ComplexTxUDPPort RTP Multiplexing Local UDP Port LiL1ComplexRxUDPPort RTP Multiplexing Remote UDP Port Version 4 6 Description Enables the use of DNS 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 Disable default 1 Enable If enabled and the Proxy Registrar IP address parameter or the domain name in the Contact Record Route headers contains a domain name without port definition an SRV query is performed The gateway uses the first host name received from the SRV query The gateway then performs DNS A record query for the host name to locate an IP address If the Proxy Registrar IP address parameter or the domain name in the Contact Record Route headers contains a domain name with port definition the gateway performs a regular DNS A record query To enable SRV queries only for Proxy servers set the parameter EnablePr
82. l tone IsWaitForDialTone is disabled 2 For call transfer The delay after hook flash is generated and dialing is begun The valid range in milliseconds is 0 to 20000 20 seconds The default value is 1000 1 second MW1IOnCode Determines a digit code used by the gateway to notify the PBX of messages waiting for MWI On Digit Pattern a specific extension This code is added as prefix to the dialed number The valid range is a 25 character string MWwWIOffCode Determines a digit code used by the gateway to notify the PBX that there aren t any MWI Off Digit Pattern messages waiting for a specific extension This code is added as prefix to the dialed number The valid range is a 25 character string TelDisconnectCode Determines a digit pattern that when received from the Tel side indicates the gateway Disconnect Call Digit Pattern to disconnect the call The valid range is a 25 character string The following digit pattern parameters apply only to VM applications that use the DTMF communication method For the available patterns syntaxes refer to the User s Manual DigitPatternForwardOnBus Determines the digit pattern used by the PBX to indicate call forward on busy y The valid range is a 120 character string Forward on Busy Digit Pattern DigitPatternForwardOnNoA Determines the digit pattern used by the PBX to indicate call forward on no answer nswer The valid range is a 120 character string Forward
83. lel is now supported Relevant parameters If DisableAutoDTMFMute 1 in band DTMF transmission is set according to the DIMFTransportT ype parameter When DHCP is enabled the gateway includes its product name e g MP 108 FXS or MP 104 FXO in the DHCP option 60 Vendor Class Identifier The DHCP server can use this product name to assign an IP address accordingly Note After power up the gateway issues two DHCP requests Only in the second request the DHCP option 60 is contained If the gateway is reset from the Web SNMP only a single DHCP request containing option 60 is sent The error message that indicates an invalid ini file configuration now contains the line number of the invalid parameter in the ini file 4 1 6 Resolved Constraints Can now handle 401 407 authentication required responses for all SIP requests Passes the called display name to INVITE messages if it appears in the Refer To header in a REFER request Session timer is now supported also for T 38 faxes and for Held calls Enables SIP destination port configuration for the entire UDP range Reliable sending of DTMF digits using INFO messages The gateway now waits for 2000K before sending new DTMF digits SIPDestinationPort if used only affects the destination of the INVITE requests unless IsAlwaysUseProxy 1 forcing all SIP messages to be sent to this port Static NAT is now supported for local IP calls Sev
84. o the same hostname If the hostname isn t found in this table the gateway communicates with an external DNS server Up to 10 hostnames can be configured Relevant parameter Dns2lP Enhanced Tel to IP routing selection Selection of destination IP address and IP Profiles optional can now be performed according to both Destination and Source numbers Relevant parameter Prefix Enhanced IP to Tel routing selection Selection of hunt groups and IP Profiles optional can now be performed according to Destination number Source Number and Source IP address Relevant parameter PSTNPrefix Enhanced Number Manipulation support In all four manipulation tables the following functionalities were added gt Can now select an entry according to both destination and source numbers gt Can now apply the Digits to add and Digits to remove manipulation rules also on number suffixes in addition to number prefixes Relevant parameters NumberMapTel2IP NumberMapIP2Tel SourceNumberMapTel2IP SourceNumberMapIP2Tel An option to allow or restrict sending of Caller ID information on a per call basis was added using the Tel to IP number manipulation table A Source IP column was added to the Destination Phone Number Manipulation Table for IP to Tel Calls This field enables to manipulate the destination number also according to the source IP address of the call Relevant parameter NumberMapIP2Tel IP addresses can now inclu
85. ome Caller ID signals uses reversal polarity and or wink In these cases it is recommended to set PolarityReversalType to 1 Hard Note 2 Applicable only to FXS gateways CurrentDisconnectDuration Duration of the current disconnect pulse in msec The default is 900 msec The range is 200 to 1500 msec Applicable for both FXS and FXO gateways Note The FXO gateways detection range is 200 msec of the parameter s value 100 For example if CurrentDisconnectDuration 200 the detection range is 100 to 500 msec AnalogCallerIDTimimgMode 0 Caller ID is generated between the first two rings default 1 The gateway attempts to find an optimized timing to generate the Caller ID according to the selected Caller ID type Note that when used with distinctive ringing the Caller ID signal will not change the distinctive ringing timing Note Applicable only to FXS gateways BootPSelectiveEnable Enables the Selective BootP mechanism 1 Enabled 0 Disabled default The Selective BootP mechanism enables the gateway 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 DHCP servers respond to gateway BootP requests Note1 When working with DHCP EnableDHCP 1 the selective BootP feature must be disabled Note 2 The BootPSelectiveEnable is a special Hidden par
86. on No Answer Digit Pattern DigitPatternForwardOnDND Determines the digit pattern used by the PBX to indicate call forward on do not disturb Forward on Do Not Disturb The valid range is a 120 character string Digit Pattern DigitPatternForwardNoReas Determines the digit pattern used by the PBX to indicate call forward with no reason on The valid range is a 120 character string Forward on No Reason Digit Pattern DigitPatternInternalCall Determines the digit pattern used by the PBX to indicate an internal call Internal Call Digit Pattern The valid range is a 120 character string DigitPatternExternalCall Determines the digit pattern used by the PBX to indicate an external call External Call Digit Pattern The valid range is a 120 character string Serial parameters applicable only to the SMDI application SerialBaudRate Determines the value of the RS 232 baud rate The valid range is any value It is recommended to use the following standard values 1200 2400 9600 default 14400 19200 38400 57600 115200 SerialData Determines the value of the RS 232 data bit 7 7 bit 8 8 bit default SerialParity Determines the value of the RS 232 polarity 0 None default 1 Odd 2 Even SerialStop Determines the value of the RS 232 stop bit 1 1 bit default 2 2 bit MediaPack and Mediant 1000 SIP Release Notes 18 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 1
87. or IP to Tel calls is according to destination number source number and source IP address Note 1 To support the in call alternative routing feature Users can use two entries that support the same call but assigned it with a different hunt groups The second entree functions as an alternative selection if the first rule fails as a result of one of the release reasons listed in the AltRouteCauselP2Tel table Note 2 An optional IP ProfilelD 1 to 5 can be applied to each routing rule Note 3 The Source IP Address can include the x wildcard to represent single digits For example 10 8 8 xx represents all IP addresses between 10 8 8 10 to 10 8 8 99 53 June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name Description NumberMapIP2Tel Manipulate the destination number for IP to Tel calls Destination Phone Number NumberMapIP2Tel a b c d e f g h i Manipulation Table for IP gt Tel calls a Destination number prefix b Number of stripped digits from the left or if brackets are used from the right A combination of both options is allowed c String to add as prefix or if brackets are used as suffix A combination of both options is allowed d Number of remaining digits from the right e Not applicable set to f Not applicable set to g Source number prefix
88. ote 2 This parameter can appear up to 10 times Table of call failure reason values received from the IP side If a call is released as a result of one of these reasons the gateway tries to find an alternative route to that call in the Tel to IP Routing table For example AltRouteCauseTel2IP 408 Response timeout AltRouteCauseTel2IP 486 User is busy Note 1 The 408 reason can be used to specify that there was no response from the remote party to the INVITE request Note 2 This parameter can appear up to 4 times Table of call failure reason values received from the telephony side in SIP presentation If a call is released as a result of one of these reasons the gateway tries to find an alternative hunt group to that call in the IP to Hunt Group Routing table For example AltRouteCauselP2Tel 3 No route to destination AltRouteCauselP2Tel 17 Busy here Note This parameter can appear up to 4 times Prefix lt Destination Phone Prefix gt lt IP Address gt lt Sre Phone Prefix gt lt IP Profile ID gt Selection of IP address for Tel To IP calls is according to destination and source prefixes Note An optional IP ProfilelD 1 to 5 can be applied to each routing rule PSTNPrefix a b c d e a Destination Number Prefix b Hunt Group ID c Source Number Prefix d Source IP address obtained from the Contact header in the INVITE message e IP Profile ID Selection of hunt groups f
89. oxySRVQuery to 1 0 Sets the destination number to the user part of the Request URI for IP gt Tel calls and sets the Contact header to the source number for Tel gt IP calls default 1 Sets the destination number to the user part of the To header for IP gt Tel calls and sets the Contact header to the username parameter for Tel gt IP calls 0 Interworks the IP Source Number to the Tel Source Number default 1 Sets the Tel Source Number to IP Display Name Applicable to IP gt Tel calls If enabled the outgoing Source Number is set to the IP Display Name and Presentation is set to Allowed If there isn t a Display Name the user part of the SIP URI is used as the Source Number and the Presentation is set to Restricted For example When the following is received from 100 lt sip 200 201 202 203 204 gt the outgoing Source Number is set to 100 the Display Name is set to 100 and the Presentation is set to Allowed 0 When the following is received from lt sip 100 101 102 103 104 gt the outgoing Source Number is set to 100 and the Presentation is set to Restricted 1 Determines the lower boundary of UDP ports used for RTP RTCP and T 38 by a remote gateway If this parameter is set to a non zero value ThroughPacket is enabled Note that the value of RemoteBaseUDPPort on the local gateway must equal the value of BaseUDPPort of the remote gateway The gateway use
90. parameter ProxyIP A7 June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name EnableDigitDelivery2IP 0 Disabled default Enable Digit Delivery to IP 1 Enable digit delivery to IP The digit delivery feature enables sending of DTMF digits to the destination IP address after the Tel gt IP call was answered To enable this feature modify the called number to include at least one p character The gateway uses the digits before the p character in the initial INVITE message After the call was answered the gateway waits for the required time of p 1 5 seconds and then sends the rest of the DTMF digits using the method chosen in band out of band Description Note The called number can include several p characters 1 5 seconds pause For example the called number can be as follows pp699 p9p300 EnableDigitDelivery 0 Disabled default Enable Digit Delivery to Tel 1 Enable Digit Delivery feature for MP 1xx FXO amp FXS The digit delivery feature enables sending of DTMF digits to the gateway s port after the line is Off Hooked FXS or seized FXO For IP gt Tel calls after the line is Off Hooked seized the MP 1xx plays the DTMF digits of the called number towards the phone line Note 1 The called number can also include the charact
91. provides groups of coders that can be associated with IP or Tel profiles You can select the following coders g711Alaw64k G 711 A law g711Ulaw64k G 711 p law g7231 G 723 1 6 3 kbps default g7231r53 G 723 1 5 3 kbps g726 G 726 ADPCM 32 kbps Payload Type 2 g729 G 729A g729_AnnexB G 729 Annex B The RTP packetization period ptime in msec depends on the selected Coder name and can have the following values G 711 family 10 20 30 40 50 60 80 100 120 default 20 G 729 family 10 20 30 40 50 60 default 20 G 723 family 30 60 90 default 30 G 726 family 10 20 30 40 50 60 80 100 120 default 20 Note If the coder G 729 is selected the gateway includes annexb no in the SDP of the relevant SIP messages If G 729 Annex B is selected annexb yes is included An exception to this logic is when the remote gateway is a Cisco device IsCiscoSCEMode ini file note 1 This parameter CoderName_ID can appear up to 20 times five coders in four coder groups ini file note 2 The coder name is case sensitive ini file note 3 Enter in the format Coder ptime For example the following three coders belong to coder group with ID 1 CoderName_1 g711Alaw64k 20 CoderName_1 g711Ulaw64k 40 CoderName_1 g7231 90 MediaPack and Mediant 1000 SIP Release Notes 12 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 1 What s Ne
92. r Subject Supported Timestamp To Unsupported User Agent Via Voicemail Warning WWW Authenticate Version 4 6 Table 2 3 SIP Headers continues on pages 26 to 27 Supported Yes Yes Yes Yes Yes Yes Yes Yes No Yes Yes Yes Yes Yes Yes No No Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes 27 2 SIP Compatibility June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP 2 3 4 SDP Headers The following SDP Headers are supported by the MediaPack SIP gateway Table 2 4 SDP Headers SDP Header Element Supported v Protocol version Yes o Owner creator and session identifier Yes a Attribute information Yes c Connection information Yes d Digit Yes m Media name and transport address Yes s Session information Yes t Time alive header Yes b Bandwidth header Yes u Uri Description Header Yes e Email Address header Yes i Session Info Header Yes p Phone number header Yes y Year Yes 2 3 5 SIP Responses The 2 3 5 1 1XX following SIP responses are supported by the MediaPack SIP gateway 1xx Response Information Responses 2xx Response Successful Responses 3xx Response Redirection Responses 4xx Response Request Failure Responses 5xx Response Server Failure Responses 6xx Response Global Responses Response Information Responses Table 2 5 1xx SIP Responses
93. r set in the automatic dialing table The valid range is 0 to 60 The default time is 16 seconds Applicable to FXS and FXO gateways HoldFormat Determines the format of the hold request Hold Format 0 The connection IP address in SDP is 0 0 0 0 default 1 The last attribute of the SDP contains the following a sendonly MediaPack and Mediant 1000 SIP Release Notes 46 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 4 Recent Revision History Previous Version Table 4 1 Release 4 4 ini File Web Browser Parameter Name continues on pages 45 to 58 ini File Web Interface Parameter Name CutThrough Enable Calls Cut Through EnableProxySRVQuery Enable Proxy SRV Queries ProxyIP Proxy IP Address ProxyIP Redundant Proxy IP Address Version 4 6 Description Enables users to receive incoming IP calls while the port is in an off hooked state 0 Disabled default 1 Enabled If enabled FXS gateways answer the call and cut through the voice channel if there is no other active call on that port even if the port is in off hooked state When the call is terminated by the remote party the gateway plays a reorder tone for TimeForReorderTone seconds and is then ready to answer the next incoming call without on hooking the phone The waiting call is automatically answered by the gateway when the current call is terminated EnableCallWaiting 1 Note This option is
94. r Name continues on pages 45 to 58 ini File Web Interface Parameter Name NSEMode NSEPayloadType PrerecordedTonesFileName ControllPDiffServ Signaling DiffServ RegistrationRetryTime Registration Retry Time AssertedidMode Asserted Identity Mode IsTrustedProxy Ils Proxy Trusted AddTON2RPI Add Number Plan and Type to Remote Party ID Header T38UseRTPPort MediaPack and Mediant 1000 SIP Release Notes 50 Description Cisco compatible fax and modem bypass mode 0 NSE disabled default 1 NSE enabled Note 1 This feature can be used only if VxxModemTransportT ype 2 Bypass Note 2 If NSE mode is enabled the SDP contains the following line a rtpmap 100 X NSE 8000 Note 3 To use this feature e The Cisco gateway must include the following definition modem passthrough nse payload type 100 codec g711alaw e Set the Modem transport type to Bypass mode VxxModemTransportType 2 for all modems e Configure the gateway parameter NSEPayloadType 100 In NSE bypass mode the gateway 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 p 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
95. s been received the gateway responds with an ACK and disconnects the call 2 3 5 5 5xx Response Server Failure Responses 5xx Response 500 501 502 503 504 505 Internal Server Error Not Implemented Bad gateway Service Unavailable Gateway Timeout Version Not Supported Table 2 9 5xx SIP Responses Comments On reception of any of these Responses the GW releases the call sending appropriate release cause to PSTN side The GW generates 5xx response according to PSTN release cause coming from PSTN 2 3 5 6 6xx Response Global Responses 6XX Response 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable Version 4 6 Table 2 10 6xx SIP Responses Comments On reception of any of these Responses the GW releases the call sending appropriate release cause to PSTN side 31 June 2005 MediaPack nd Mediant 1000 SIP Reader s Notes MediaPack and Mediant 1000 SIP Release Notes 32 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 3 Known Constraints 3 Known Constraints 3 1 Hardware Constraints Mediant 1000 Only specific combinations of FXS and FXO modules are currently supported For detailed information contact AudioCodes MP 11x After running the procedure for restoring the networking parameters to their initial state the gateway must be reset again using a hardware reset If a software reset is issued the gateway reverts to its fa
96. s these parameters 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 4000 to 64000 The default value is 0 ThroughPacket is disabled Note To enable ThroughPacket the parameters L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort must be set to a non zero value Determines the local UDP port used for outgoing multiplexed RTP packets applies to the ThroughPacket mechanism The valid range is the range of possible UDP ports 4000 to 64000 The default value is 0 ThroughPacket is disabled This parameter cannot be changed on the fly and requires a gateway reset Determines the remote UDP port the multiplexed RTP packets are sent to and the local UDP port used for incoming multiplexed RTP packets applies to the ThroughPacket mechanism The valid range is the range of possible UDP ports 4000 to 64000 The default value is 0 ThroughPacket is disabled This parameter cannot be changed on the fly and requires a gateway reset Note All gateways that participate in the same ThroughPacket session must use the same L1L1ComplexRxUDPPort 15 June 2005 7a Ta wt AudioCodes MediaPack and Mediant 1000 SIP Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name Pacts Upson IPProfile_ID IPProfile_ lt Profile ID gt
97. t is connected to the gateway When RTP packets are received after a sudden large network delay 200 to 300 msec the drift correction could take about 5 seconds During this period voice towards the TDM side is silent Static NAT is not supported for local IP calls 3 4 Web Constraints 27 Not all parameters can be changed on the fly from the Web browser Parameters that can t be changed on the fly are noted with To change these parameters reset the gateway using the Web browser reset button MediaPack and Mediant 1000 SIP Release Notes 34 Document LTRT 65606 MediaPack and Mediant 1000 SIP Release Notes 3 Known Constraints 3 5 Version 4 6 28 29 30 31 32 When changing gateway parameters from Web Browser the new parameters are permanently stored in flash memory only after the gateway is reset from the Web or after Save Configuration button is pressed The number of fax calls indicated by the fields Attempted Fax Calls Counter and Successful Fax Calls Counter in the Calls Count screens isn t accurate In the screens Coders and Coder Group Settings When G 729 is used with ptimes 80 100 and 120 and G 723 is used with ptimes 120 and 150 the voice quality is reduced Therefore using these ptimes isn t recommended In the current version the option to save changes to the IP Routing table so they are available after power fail isn t available via the Embedded Web
98. time of day can now be obtained from a standard NTP server Relevant parameters NTPServerlP NTPServerUTCOffset NTPUpdateinterval When NTP is enabled a timestamp string hour minutes seconds is added to all Syslog messages 43 June 2005 x A 76 TZ 78 79 80 81 82 WH 3 wt AudioCodes MediaPack and Mediant 1000 SIP DHCP client improvements The DHCP client now supports limited IP leasing time and performs lease renewal In addition the time server and SIP DHCP options are now supported Operation in a multiple routers network was improved The gateway now learns the network topology by responding to ICMP redirections and caching them as routing rules with expiration time Support was added for loading and retrieving encoded ini files from the gateway instead of clear text files Files are encoded decoded using the TrunkPack Downloadable Conversion utility The mechanism for burning configuration files in non volatile memory was improved The new mechanism enables users to maintain their configuration when upgrading the software version Users should note the following changes gt Saving the entire configuration parameters and files in non volatile memory is now controlled by a single parameter SaveConfiguration default 1 gt BurnCallProgressToneFile and BurnCoeffFile parameters are no longer supported Sending of in band and out of band DTMF digits RFC 2833 in paral
99. tinationMasksColumn RoutingTableGatewaysColumn RoutingTableHopsCountColumn RoutingTablelnterfacesColumn The maximum length of the administrator s username and password was increased to 19 characters Note that if after a long password is set the user goes back to version 4 4 or earlier the username and password are deleted changed to blank MP 11x and Mediant 1000 only Users can now enhance the security and capabilities of logging to the gateway s Web and Telnet embedded servers by using a Remote Authentication Dial In User Service RADIUS to store numerous usernames and passwords allowing multiple user management on a centralized platform RADIUS RFC 2865 is a standard authentication protocol that defines a method for contacting a predefined server and verifying a given name and password pair against a remote database in a secure manner Relevant parameters EnableRADIUS WebRADIUSLogin RADIUSAuthServerlP RADIUSAuthPort SharedSecret To prevent unauthorized access to the Embedded Web Server two levels of security are now available Administrator also used for Telnet access and Monitoring Each employs a different username and password Users can access the Embedded Web Server as either gt Administrator all Web screens are read write and can be modified Default username Admin Default password Admin gt Monitoring all Web screens are read only and cannot be modified In addition the following scre
100. tricted For example CallerDisplayInfoO John 0 CallerDisplayInfo7 David 1 Note 1 The numbering of channels starts with 0 Note 2 This parameter can appear up to eight times for MP 108 and up to 24 times for MP 124 Keypad sequence that activates the call forward features KeyCFUncond For unconditional call forward KeyCFNoAnswer For call forward on no answer KeyCFBusy For call forward on busy KeyCFBusyOrNoAnswer For call forward on busy or no answer KeyCFDoNotDisturb For call forward on Do Not Disturb configuration Users can configure the call forward reason and forwarding number directly from their phone it can also be configured in the Embedded Web Server For example KeyCFUncond 73 KeyCFDeact 75 To activate the required forward method from the telephone e Press the preconfigured sequence number on the keypad a dial tone is heard e Press the telephone number to which the call is forwarded a confirmation tone is heard To deactivate call forward press the KeyCFDeact sequence after the sequence is pressed a confirmation tone is heard Note This option is applicable only to FXS gateways Keypad sequence that activates the hotline feature The hotline feature directs the FXS gateway to dial a preconfigured hotline number if no digits were collected during a dial tone duration about 15 seconds Users can enable disable the hotline feature and enter the hotline number directly from
101. value is 180 Defines the time in seconds that is used in the Min SE header field This field defines the minimum time that the user agent supports for session refresh The valid range is 10 to 100000 The default value is 90 Defines the maximum number of calls that the gateway can have active at the same time If the maximum number of calls is reached new calls are not established The default value is max available channels no restriction on the maximum number of calls The valid range is 0 to max number of channels 0 Doesn t use user phone string in From header default 1 user phone string is part of the From header 0 Interworks the Tel calling name to SIP Display Name default 1 Set Display Name to Source Number if not available from Tel Applicable to Tel gt IP calls If enabled and calling party name is not defined CallerDisplaylnfoX lt name gt is not specified per gateway s x port the calling number is used instead 0 Use the gateway s IP address in keep alive OPTIONS messages default 1 Use GatewayName in keep alive OPTIONS messages The OPTIONS Request URI host part contains either the gateway s IP address or a string defined by the parameter Gatewayname The gateway uses the OPTIONS request as a keep alive message to its primary and redundant Proxies 49 June 2005 Ce AudioCodes MediaPack and Mediant 1000 SIP Table 4 1 Release 4 4 ini File Web Browser Paramete
102. w Release in 4 6 Table 1 1 Release 4 6 ini File Web Browser Parameter Name continues on pages 12 to 22 ini File Web Interface Parameter Name CoderName SIPTransportType SIP Transport Layer TCPLocalS IPPort SIP TCP Local Port EnableSIPS Enable SIPS TLSLocalS PPort SIP TLS Local Port SIPSRequireClientCertificat e Version 4 6 Description Enter the coders in the format CoderName lt Coder gt lt ptime gt For example CoderName g711Alaw64k 20 CoderName g711Ulaw64k 40 CoderName g7231 90 Note 1 This parameter CoderName can appear up to 10 times Note 2 The coder name is case sensitive You can select the following coders g711Alaw64k G 711 A law g711Ulaw64k G 711 p law g7231 G 723 1 6 3 kbps default g7231r53 G 723 1 5 3 kbps g726 G 726 ADPCM 32 kbps Payload Type 2 g729 G 729A g729_AnnexB G 729 Annex B Note If the coder G 729 is selected the gateway includes annexb no in the SDP of the relevant SIP messages If G 729 Annex B is selected annexb yes is included An exception to this logic is when the remote gateway is a Cisco device IsCiscoSCEMode The RTP packetization period ptime in msec depends on the selected coder name and can have the following values G 711 10 20 30 40 50 60 80 100 120 default 20 G 729 10 20 30 40 50 60 default 20 G 723 30 60 90 default 30 G 726 10
103. x relay If the T 38 negotiation fails the gateway re initiates a fax session using the coder G 711 A law p law with adaptations see note 1 Note 1 Fax adaptations 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 Note 2 If the gateway initiates a fax session using G 711 option 2 and possibly 3 a gpmd attribute is added to the SDP in the following format For A law a gomd 0 vbd yes ecan on For p law a gomd 8 vod yes ecan on Note 3 When IsFaxUsed is set to 1 2 or 3 the parameter FaxTransportMode is ignored BellcoreCallerlDTypeOneSu_ Selects the Bellcore Caller ID sub standard bStandard 0 Between rings default 1 Not ring related ETSICallerIDTypeOneSubSt Selects the ETSI Caller ID Type 1 sub standard FXS only andard 0 ETSI between rings default 1 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 ETSIVMWITypeOneStandar Selects the ETSI Visual Message Waiting Indication VMWI Type 1 sub standard d 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

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LTRT-65606 MediaPack &amp; Mediant 1000 SIP Analog

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LTRT-65606 MediaPack & Mediant 1000 SIP Analog