Art King Posted May 18, 2007 Report Share Posted May 18, 2007 We have an external application that does IVR functions via SIP. It works fine when we access it directly from an AudioCodes gateway. I want to be able to make a call through PBXNSIP to my external SIP application. I setup a new Trunk on the PBXNSIP and I configured a Dialplan to send calls to the external IVR. I can make a call from the PBX to the to the external IVR and the IVR application answers, and the audio comes through fine. However, I cannot get my application to recognize DTMF from snom telephone, or from a softphone. DTMF entries interrupt the voice prompts on the IVR system as they are supposed to. The IVR script is looking for four digits, and it accepts four digits correctly. However, the IVR log, shows it received the DTMF digits of ?5555? when what I entered was "5353" In all my testing, it seems like whatever the first DTMF digit I enter, this digit is repeated for all the digits requested. So if I enter ?1234? the debug shows ?1111? I changed the setting on PBXNSIP to allow ?Inband DTMF Detection? but it made no difference. I get different results when calling from the snom phone vs the X-Lite softphone, but I never have gotten it to work in any case. Below is a section of the log showing the DTMF passing across to the external IVR from an X-Lite softphone. Troubleshooting suggestions would be appreciated! Art The log below is from PBXNSIP The softphone making the call is at IP address 10.0.0.85 The PBXNSIP is at IP address 10.0.0.22 The external IVR is at 10.0.0.205 INVITE sip:321@10.0.0.22 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.85:37272;branch=z9hG4bK-d87543-ca578d2d81468133-1--d87543-;rport Max-Forwards: 70 Contact: <sip:5353@10.0.0.85:37272> To: "321"<sip:321@10.0.0.22> From: "Art King (soft)"<sip:5353@10.0.0.22>;tag=2405ad3c Call-ID: ZTk4MDUxN2VlYzkyMTM5ZTk3YmIwNzg5OGQ5NzNkZDM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 316 v=0 o=- 5 2 IN IP4 10.0.0.85 s=CounterPath X-Lite 3.0 c=IN IP4 10.0.0.85 t=0 0 m=audio 61172 RTP/AVP 107 119 0 98 8 3 101 a=alt:1 1 : X+KZtdhy XCvsEoGu 10.0.0.85 61172 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv UDP: Opening socket on port 53760 UDP: Opening socket on port 53761 Identify trunk 6 Resolve destination 50438: a udp 10.0.0.85 37272 Resolve destination 50438: udp 10.0.0.85 37272 Send Packet 100 SIP Tx udp:10.0.0.85:37272: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.85:37272;branch=z9hG4bK-d87543-ca578d2d81468133-1--d87543-;rport=37272 From: "Art King (soft)" <sip:5353@10.0.0.22>;tag=2405ad3c To: "321" <sip:321@10.0.0.22>;tag=3151257388 Call-ID: ZTk4MDUxN2VlYzkyMTM5ZTk3YmIwNzg5OGQ5NzNkZDM. CSeq: 1 INVITE Content-Length: 0 Sending RTP to 10.0.0.85:61172 Resolve destination 50439: url sip:5353@10.0.0.122 Resolve destination 50439: udp 10.0.0.122 5060 Send Packet NOTIFY Dialplan: Match 321@10.0.0.22 to <sip:321@10.0.0.205;user=phone> on trunk ExtIVR Play audio_moh/noise.wav UDP: Opening socket on port 51638 UDP: Opening socket on port 51639 Resolve destination 50440: url sip:10.0.0.205 Resolve destination 50440: udp 10.0.0.205 5060 Send Packet INVITE SIP Tx udp:10.0.0.205:5060: INVITE sip:321@10.0.0.205;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.22:5060;branch=z9hG4bK-47c500d1bc1c4e59da4c9db28bd275a3;rport From: "ExtIVR" <sip:10.0.0.205>;tag=46519 To: <sip:321@10.0.0.205;user=phone> Call-ID: 29cca7cd@pbx CSeq: 29522 INVITE Max-Forwards: 70 Contact: <sip:10.0.0.22:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: pbxnsip-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 286 v=0 o=- 26774 26774 IN IP4 10.0.0.22 s=- c=IN IP4 10.0.0.22 t=0 0 m=audio 51638 RTP/AVP 0 8 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendrecv Resolve destination 50441: a udp 10.0.0.85 37272 Resolve destination 50441: udp 10.0.0.85 37272 Send Packet 183 SIP Tx udp:10.0.0.85:37272: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.85:37272;branch=z9hG4bK-d87543-ca578d2d81468133-1--d87543-;rport=37272 From: "Art King (soft)" <sip:5353@10.0.0.22>;tag=2405ad3c To: "321" <sip:321@10.0.0.22>;tag=3151257388 Call-ID: ZTk4MDUxN2VlYzkyMTM5ZTk3YmIwNzg5OGQ5NzNkZDM. CSeq: 1 INVITE Contact: <sip:5353@10.0.0.22:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: pbxnsip-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 186 v=0 o=- 10805 10805 IN IP4 10.0.0.22 s=- c=IN IP4 10.0.0.22 t=0 0 m=audio 53760 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendrecv Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 20, 2007 Report Share Posted May 20, 2007 I don't think that the Xlite is the problem here. It would be interesting to see what the IVR answeres in the 200 Ok, especially the SDP. It should indicate what codec it is using as RFC2833. Quote Link to comment Share on other sites More sharing options...
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