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dlynton

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  1. It used to always play music on hold. This is a supervised transfer, where the caller hears hold music while the server dials a new call, waits for confirmation from the called party, then connects the two parties. so it's not strictly a redirect. i was a little surprised that we were getting pbxnsip music on hold from a MSS supervised transfer, but that is how it used to work.
  2. After upgrading to the latest version of PBXNSIP, we are no longer getting hold music when we do a supervised transfer. As you can see in the call log below, pbxnsip is not attempting to play moh.wav, but it does say "call hold from trunk". We haven't changed anything in our application, just upgraded to the latest version of pbxnsip. Any ideas? REGISTER sip:callcentric.com SIP/2.0 Via: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-2fbc56286f9c2e41e5b5a6024332fdf6;rport From: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990 To: "CallCentricSanta" <sip:17772557741@callcentric.com> Call-ID: 9enn0ykw@pbx CSeq: 27448 REGISTER Max-Forwards: 70 Contact: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>" User-Agent: pbxnsip-PBX/2.1.6.2448 Expires: 3600 Content-Length: 0 [9] 2008/03/05 11:31:17: SIP Rx udp:204.11.192.23:5080: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-2fbc56286f9c2e41e5b5a6024332fdf6;rport=5060 f: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990 t: "CallCentricSanta" <sip:17772557741@callcentric.com> i: 9enn0ykw@pbx CSeq: 27448 REGISTER m: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>";expires=97 l: 0 [8] 2008/03/05 11:31:33: Found interface on 192.168.54.103 with netmask 255.255.255.0 [8] 2008/03/05 11:31:33: Found interface on 127.0.0.1 with netmask 255.0.0.0 [9] 2008/03/05 11:32:05: Resolve 163: url sip:callcentric.com [9] 2008/03/05 11:32:05: Resolve 163: naptr callcentric.com [9] 2008/03/05 11:32:05: Resolve 163: srv tls _sips._tcp.callcentric.com [9] 2008/03/05 11:32:05: Resolve 163: srv tcp _sip._tcp.callcentric.com [9] 2008/03/05 11:32:05: Resolve 163: srv udp _sip._udp.callcentric.com [9] 2008/03/05 11:32:05: Resolve 163: a udp alpha1.callcentric.com 5080 [9] 2008/03/05 11:32:05: Resolve 163: udp 204.11.192.22 5080 [9] 2008/03/05 11:32:05: Resolve 163: a udp alpha2.callcentric.com 5080 [9] 2008/03/05 11:32:05: Resolve 163: udp 204.11.192.23 5080 [9] 2008/03/05 11:32:05: Resolve 163: a udp callcentric.com 5060 [9] 2008/03/05 11:32:05: Resolve 163: udp 204.11.192.23 5060 [8] 2008/03/05 11:32:05: Trunk 6 (callcentric) has outbound proxy udp:204.11.192.22:5080 udp:204.11.192.23:5060 udp:204.11.192.23:5080 [9] 2008/03/05 11:32:05: Resolve 164: udp 204.11.192.23 5080 [9] 2008/03/05 11:32:05: SIP Tx udp:204.11.192.23:5080: REGISTER sip:callcentric.com SIP/2.0 Via: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-5c7846da9d261398d53f498c45f07a0e;rport From: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990 To: "CallCentricSanta" <sip:17772557741@callcentric.com> Call-ID: 9enn0ykw@pbx CSeq: 27450 REGISTER Max-Forwards: 70 Contact: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>" User-Agent: pbxnsip-PBX/2.1.6.2448 Expires: 3600 Content-Length: 0 [9] 2008/03/05 11:32:05: SIP Rx udp:204.11.192.23:5080: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-5c7846da9d261398d53f498c45f07a0e;rport=5060 f: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990 t: "CallCentricSanta" <sip:17772557741@callcentric.com> i: 9enn0ykw@pbx CSeq: 27450 REGISTER m: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>";expires=98 l: 0 [9] 2008/03/05 11:32:12: SIP Rx udp:192.168.54.100:5060: INVITE sip:7135599056@192.168.54.103 SIP/2.0 Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224 To: <sip:7135599056@192.168.54.103> Contact: <sip:2814610506@192.168.54.100> Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 05 Mar 2008 17:49:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 265 v=0 o=root 5779 5779 IN IP4 192.168.54.100 s=session c=IN IP4 192.168.54.100 t=0 0 m=audio 10858 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - [7] 2008/03/05 11:32:12: UDP: Opening socket on port 56516 [7] 2008/03/05 11:32:12: UDP: Opening socket on port 56517 [5] 2008/03/05 11:32:12: Identify trunk (IP address/port and domain match) 5 [9] 2008/03/05 11:32:12: Resolve 165: aaaa udp 192.168.54.100 5060 [9] 2008/03/05 11:32:12: Resolve 165: a udp 192.168.54.100 5060 [9] 2008/03/05 11:32:12: Resolve 165: udp 192.168.54.100 5060 [9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.100:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport=5060 From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224 To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100 CSeq: 102 INVITE Content-Length: 0 [6] 2008/03/05 11:32:12: Sending RTP for 1a92846a480027bf329359bc5a096045@192.168.54.100#00b1880f9c to 192.168.54.100:10858 [5] 2008/03/05 11:32:12: Trunk asterisk outbound sends call to 7135599056 [7] 2008/03/05 11:32:12: Calling extension 7135599056 [6] 2008/03/05 11:32:12: Redirecting to external voicemail account 7135599056 destination sip:011997135599056@192.168.54.103 [9] 2008/03/05 11:32:12: Dialplan: Evaluating !^(7132234676)@.*!sip:\1@\r;user=phone!i against 011997135599056@192.168.54.103 [9] 2008/03/05 11:32:12: Dialplan: Evaluating !^01199([0-9]*)@.*!sip:\1@\r;user=phone!i against 011997135599056@192.168.54.103 [5] 2008/03/05 11:32:12: Dialplan default: Match 011997135599056@192.168.54.103 to <sip:7135599056@192.168.54.200;user=phone> on trunk MS Speech Server [5] 2008/03/05 11:32:12: Using "DELESANDRI T " <sip:2814610506@192.168.54.103> as redirect from [5] 2008/03/05 11:32:12: Charge user 7135599056 for redirecting calls [8] 2008/03/05 11:32:12: Play audio_moh/noise.wav [7] 2008/03/05 11:32:12: UDP: Opening socket on port 58934 [7] 2008/03/05 11:32:12: UDP: Opening socket on port 58935 [9] 2008/03/05 11:32:12: Resolve 166: url sip:192.168.54.200:5060;transport=tcp [9] 2008/03/05 11:32:12: Resolve 166: a tcp 192.168.54.200 5060 [9] 2008/03/05 11:32:12: Resolve 166: tcp 192.168.54.200 5060 [9] 2008/03/05 11:32:12: SIP Tx tcp:192.168.54.200:5060: INVITE sip:7135599056@192.168.54.200;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.54.103:2335;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140 To: <sip:7135599056@192.168.54.200;user=phone> Call-ID: f89e1cc4@pbx CSeq: 883 INVITE Max-Forwards: 70 Contact: <sip:2814610506@192.168.54.103:2335;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Diversion: <tel:7135599056>;reason=unavailable;screen=no;privacy=off P-Preferred-Identity: <sip:7135599056@192.168.54.103> Content-Type: application/sdp Content-Length: 220 v=0 o=- 35388 35388 IN IP4 192.168.54.103 s=- c=IN IP4 192.168.54.103 t=0 0 m=audio 58934 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2008/03/05 11:32:12: Resolve 167: aaaa udp 192.168.54.100 5060 [9] 2008/03/05 11:32:12: Resolve 167: a udp 192.168.54.100 5060 [9] 2008/03/05 11:32:12: Resolve 167: udp 192.168.54.100 5060 [9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.100:5060: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport=5060 From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224 To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100 CSeq: 102 INVITE Contact: <sip:7135599056@192.168.54.103:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Content-Type: application/sdp Content-Length: 220 v=0 o=- 63166 63166 IN IP4 192.168.54.103 s=- c=IN IP4 192.168.54.103 t=0 0 m=audio 56516 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:5060: SIP/2.0 100 Trying FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140 TO: <sip:7135599056@192.168.54.200;user=phone> CSEQ: 883 INVITE CALL-ID: f89e1cc4@pbx VIA: SIP/2.0/TCP 192.168.54.103:2335;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport CONTENT-LENGTH: 0 [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:5060: SIP/2.0 302 Moved Temporarily FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140 TO: <sip:7135599056@192.168.54.200;user=phone>;tag=f4d3cd3c6c CSEQ: 883 INVITE CALL-ID: f89e1cc4@pbx VIA: SIP/2.0/TCP 192.168.54.103:2335;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport CONTACT: <sip:7135599056@192.168.54.200:2257;user=phone;transport=Tcp;maddr=192.168.54.200;x-mss-call-id=f89e1cc4%40pbx> CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [7] 2008/03/05 11:32:12: Call f89e1cc4@pbx#5140: Clear last INVITE [9] 2008/03/05 11:32:12: Resolve 168: url sip:7135599056@192.168.54.200;user=phone [9] 2008/03/05 11:32:12: Resolve 168: udp 192.168.54.200 5060 [9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.200:5060: ACK sip:7135599056@192.168.54.200;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140 To: <sip:7135599056@192.168.54.200;user=phone>;tag=f4d3cd3c6c Call-ID: f89e1cc4@pbx CSeq: 883 ACK Max-Forwards: 70 Contact: <sip:2814610506@192.168.54.103:5060;transport=udp> P-Preferred-Identity: <sip:7135599056@192.168.54.103> Content-Length: 0 [5] 2008/03/05 11:32:12: Redirecting call [9] 2008/03/05 11:32:12: Resolve 169: aaaa tcp 192.168.54.200 2257 [9] 2008/03/05 11:32:12: Resolve 169: a tcp 192.168.54.200 2257 [9] 2008/03/05 11:32:12: Resolve 169: tcp 192.168.54.200 2257 [9] 2008/03/05 11:32:12: SIP Tx tcp:192.168.54.200:2257: INVITE sip:7135599056@192.168.54.200:2257;user=phone;transport=Tcp;maddr=192.168.54.200;x-mss-call-id=f89e1cc4%40pbx SIP/2.0 Via: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140 To: <sip:7135599056@192.168.54.200;user=phone> Call-ID: f89e1cc4@pbx CSeq: 884 INVITE Max-Forwards: 70 Contact: <sip:2814610506@192.168.54.103:2336;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Diversion: <tel:7135599056>;reason=unavailable;screen=no;privacy=off P-Preferred-Identity: <sip:7135599056@192.168.54.103> Content-Type: application/sdp Content-Length: 220 v=0 o=- 35388 35388 IN IP4 192.168.54.103 s=- c=IN IP4 192.168.54.103 t=0 0 m=audio 58934 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [8] 2008/03/05 11:32:12: UDP: recvfrom receives ICMP message [5] 2008/03/05 11:32:12: Connection refused on udp:192.168.54.200:5060 [6] 2008/03/05 11:32:12: Could not determine destination address on 168 [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:2257: SIP/2.0 100 Trying FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140 TO: <sip:7135599056@192.168.54.200;user=phone> CSEQ: 884 INVITE CALL-ID: f89e1cc4@pbx VIA: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport CONTENT-LENGTH: 0 [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:2257: SIP/2.0 180 Ringing FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140 TO: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf CSEQ: 884 INVITE CALL-ID: f89e1cc4@pbx VIA: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [8] 2008/03/05 11:32:12: Play audio_en/ringback.wav [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:2257: SIP/2.0 200 OK FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140 TO: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf CSEQ: 884 INVITE CALL-ID: f89e1cc4@pbx VIA: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport CONTACT: <sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200>;automata CONTENT-LENGTH: 198 CONTENT-TYPE: application/sdp ALLOW: UPDATE SERVER: RTCC/3.0.0.0 ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify v=0 o=- 0 0 IN IP4 192.168.54.200 s=Microsoft Speech Server session c=IN IP4 192.168.54.200 t=0 0 m=audio 13440 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [7] 2008/03/05 11:32:12: Call f89e1cc4@pbx#5140: Clear last INVITE [7] 2008/03/05 11:32:12: Set packet length to 20 [6] 2008/03/05 11:32:12: Sending RTP for f89e1cc4@pbx#5140 to 192.168.54.200:13440 [9] 2008/03/05 11:32:12: Resolve 170: aaaa tcp 192.168.54.200 2257 [9] 2008/03/05 11:32:12: Resolve 170: a tcp 192.168.54.200 2257 [9] 2008/03/05 11:32:12: Resolve 170: tcp 192.168.54.200 2257 [9] 2008/03/05 11:32:12: SIP Tx tcp:192.168.54.200:2257: ACK sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200 SIP/2.0 Via: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-a0f9d3e52482a7033ffd2563d08a6970;rport From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140 To: <sip:7135599056@192.168.54.200;user=phone>;tag=73d97a1bbf Call-ID: f89e1cc4@pbx CSeq: 884 ACK Max-Forwards: 70 Contact: <sip:2814610506@192.168.54.103:2336;transport=tcp> P-Preferred-Identity: <sip:7135599056@192.168.54.103> Content-Length: 0 [7] 2008/03/05 11:32:12: Determine pass-through mode after receiving response [9] 2008/03/05 11:32:12: Resolve 171: aaaa udp 192.168.54.100 5060 [9] 2008/03/05 11:32:12: Resolve 171: a udp 192.168.54.100 5060 [9] 2008/03/05 11:32:12: Resolve 171: udp 192.168.54.100 5060 [9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.100:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport=5060 From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224 To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100 CSeq: 102 INVITE Contact: <sip:7135599056@192.168.54.103:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Content-Type: application/sdp Content-Length: 220 v=0 o=- 63166 63166 IN IP4 192.168.54.103 s=- c=IN IP4 192.168.54.103 t=0 0 m=audio 56516 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/03/05 11:32:12: f89e1cc4@pbx#5140: RTP pass-through mode [7] 2008/03/05 11:32:12: 1a92846a480027bf329359bc5a096045@192.168.54.100#00b1880f9c: RTP pass-through mode [9] 2008/03/05 11:32:12: SIP Rx udp:192.168.54.100:5060: ACK sip:7135599056@192.168.54.103:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK5784477d;rport From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224 To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c Contact: <sip:2814610506@192.168.54.100> Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 [9] 2008/03/05 11:32:39: SIP Rx tcp:192.168.54.200:2257: INVITE sip:2814610506@192.168.54.103:2336;transport=tcp SIP/2.0 FROM: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf TO: <sip:2814610506@192.168.54.103>;tag=5140 CSEQ: 1 INVITE CALL-ID: f89e1cc4@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 192.168.54.200:2257;branch=z9hG4bK2ae9f2 CONTACT: <sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200;ms-opaque=6b1c435266dc81e2>;automata CONTENT-LENGTH: 210 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp v=0 o=- 0 0 IN IP4 192.168.54.200 s=Microsoft Speech Server session c=IN IP4 192.168.54.200 t=0 0 m=audio 13440 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendonly a=ptime:20 [7] 2008/03/05 11:32:39: Set packet length to 20 [9] 2008/03/05 11:32:39: Resolve 172: tcp 192.168.54.200 2257 [9] 2008/03/05 11:32:39: SIP Tx tcp:192.168.54.200:2257: SIP/2.0 200 Ok Via: SIP/2.0/TCP 192.168.54.200:2257;branch=z9hG4bK2ae9f2 From: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf To: <sip:2814610506@192.168.54.103>;tag=5140 Call-ID: f89e1cc4@pbx CSeq: 1 INVITE Contact: <sip:2814610506@192.168.54.103:2336;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Content-Type: application/sdp Content-Length: 232 v=0 o=- 35388 35388 IN IP4 192.168.54.103 s=- c=IN IP4 192.168.54.103 t=0 0 m=audio 58934 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly [7] 2008/03/05 11:32:39: f89e1cc4@pbx#5140: Media-aware pass-through mode [6] 2008/03/05 11:32:39: Call hold from trunk [9] 2008/03/05 11:32:39: SIP Rx tcp:192.168.54.200:2257: ACK sip:2814610506@192.168.54.103:2336;transport=tcp SIP/2.0 FROM: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf TO: <sip:2814610506@192.168.54.103>;tag=5140 CSEQ: 1 ACK CALL-ID: f89e1cc4@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 192.168.54.200:2257;branch=z9hG4bKa050b1ce CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 [5] 2008/03/05 11:32:39: SIP port accept from 192.168.54.200:2728 [9] 2008/03/05 11:32:39: SIP Rx tcp:192.168.54.200:2728: INVITE sip:7132234676@192.168.54.103:5060;user=phone SIP/2.0 FROM: <sip:2814610506@vtspeech07.dc.voicetarget.com:2257;user=phone>;epid=8D9F696238;tag=891bce9f3f TO: <sip:7132234676@192.168.54.103:5060;user=phone> CSEQ: 5 INVITE CALL-ID: a8066fed-b09c-4d54-9b9d-bca36ec8415c MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 192.168.54.200:2728;branch=z9hG4bKa3f789a CONTACT: <sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200;ms-opaque=6b1c435266dc81e2>;automata CONTENT-LENGTH: 340 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp ALLOW: UPDATE ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify v=0 o=- 0 0 IN IP4 192.168.54.200 s=Microsoft Speech Server session c=IN IP4 192.168.54.200 t=0 0 m=audio 63872 RTP/AVP 114 115 4 0 8 97 101 a=rtpmap:114 x-msrta/16000 a=fmtp:114 bitrate=29000 a=rtpmap:115 x-msrta/8000 a=fmtp:115 bitrate=11800 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [7] 2008/03/05 11:32:39: UDP: Opening socket on port 55368 [7] 2008/03/05 11:32:39: UDP: Opening socket on port 55369 [5] 2008/03/05 11:32:39: Identify trunk (IP address and domain match) 4 [9] 2008/03/05 11:32:39: Resolve 173: tcp 192.168.54.200 2728 [9] 2008/03/05 11:32:39: SIP Tx tcp:192.168.54.200:2728: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.54.200:2728;branch=z9hG4bKa3f789a From: <sip:2814610506@vtspeech07.dc.voicetarget.com:2257;user=phone>;epid=8D9F696238;tag=891bce9f3f To: <sip:7132234676@192.168.54.103:5060;user=phone>;tag=f73b3fc676 Call-ID: a8066fed-b09c-4d54-9b9d-bca36ec8415c CSeq: 5 INVITE Content-Length: 0
  3. Is there a codec preference or anything else I can try changing in the trunk settings? Anything else you can recommend to troublshoot this problem? I assume that a supervised transfer / conference call would typically work fine in your pbx? i.e. DTMF tones are usually heard by all parties? If you recall, I'm the one you helped set up for connectivity with Speech Server & Asterisk. Do you know anything about Asterisk that we should be checking to make sure the DTMF gets passed through? I noticed in the Asterisk call log "getdtmf on channel 41: Operation now in progress" only appeard once during the two test calls I made. It's quite frustrating to be so close, yet so far! We've come a long way to get everything working between MSS, Asterisk and PBXNSIP. It will be such a relief to finally get this behind us, so any other help you can offer is greatly appreciated!
  4. I searched for replace in my log and found the sip header as well as pbxnsip reporting "Supported: 100rel, replaces, norefersub". I did *not* see any instance of "Replaces: Call-ID...not found". I see PBXNSIP assign a call id of ced43e0d-b8a6-46b1-bfb0-3da0b8e9d40a and then that is used later in the REPLACES= of the sip header. Do you know what the to-tag and from-tag are for? I'm attaching the log from my call (I have the log level set to 10--what can I set it to while being confident I'll get all the relevant messages?). The SIP REFER looks like it's doing everything right, and the supervised transfer works great except for the DTMF tones. Any other ideas? log.txt
  5. When performing a wireshark trace there isn't any traffic when I send DTMF tones. The trace at the time of the transfer is almost identical. After this point the call should be only between the caller and the called party. I see a SIP REFER, and then a BYE response on both the supervised and blind transfer calls. Here is the only difference I've found between the blind transfer (DTMF can be heard) and the supervised transfer (DTMF tones sound like clicks). Blind transfer: REFER-TO: <sip:7132234676@192.168.54.103:5060;transport=tcp;user=phone> Supervised: REFER-TO: <sip:7132234676@192.168.54.103:5060;transport=tcp;user=phone?REPLACES=2d886047-461f-483c-b59e-179845662721%3Bto-tag%3D7b97da7e0c%3Bfrom-tag%3D264b596bbf> Do you have any idea what the REPLACES querystring is doing? Thanks for your help! We are so close to being able to deploy our new IVR application on MS Speech Server + PBXNSIP! This is our last issue after a barrage of problems.
  6. I'm using Microsoft Speech Server to initiate a supervised transfer. After connecting the two callers, neither party can hear DTMF tones that they punch in. The tones come through sounding like a click. This doesn't happen when I do a blind transfer. Can you speculate if this is being caused by PBXNSIP and if there is a way to troubleshoot it? Thanks, Daniel
  7. Has anyone else gotten pbxnsip to work with Microsoft Speech Server 2007? I am trying to set it up as an extension but the pbx is not even attempting to forward the call. It just rings, rings, and eventually hangs up. Can anyone help me with this? I will be an extremely loyal customer to pbxnsip if we get this working! I haven't found anything else that will work for me. Thanks! E-mail Daniel@lyntonweb.com
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