Ganesh Posted July 7, 2008 Report Share Posted July 7, 2008 We are integrating pbxnsip with Avaya system on SIP trunk. We have configured SIP trunk (gateway mode) on pbxnsip and Avaya. But unable to route calls both ways between the two system. Attached is the logs and wireshark traces captured. 4229 and 4431 is extension (Avaya phones) in Avaya. 2201 and 2202 is extensions (Snom phones) on pbxnsip. Avaya uses port 5061 and supports only TLS on SIP trunk. Please let me know if you can get some information from these logs. Below is one more log taken from pbxnsip while trying to call Avaya phones. [5] 2008/07/07 04:23:17: SIP port accept from 192.168.192.28:14935 [7] 2008/07/07 04:23:20: SIP Rx udp:192.168.136.36:2051: REGISTER sip:192.168.192.50 SIP/2.0 Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-81zc3zm6vbe4;rport From: <sip:5203@192.168.192.50>;tag=3eez6gof7x To: <sip:5203@192.168.192.50> Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7 CSeq: 1033 REGISTER Max-Forwards: 70 Contact: <sip:5203@192.168.136.36:2051;line=vfooudzb>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:519aec13-5a5d-4f46-92d3-aa8451bb25aa>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/6.5.13 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.136.36 WWW-Contact: <http://192.168.136.36:80> WWW-Contact: <https://192.168.136.36:443> Expires: 3600 Content-Length: 0 [9] 2008/07/07 04:23:20: Resolve 47: aaaa udp 192.168.136.36 2051 [9] 2008/07/07 04:23:20: Resolve 47: a udp 192.168.136.36 2051 [9] 2008/07/07 04:23:20: Resolve 47: udp 192.168.136.36 2051 [7] 2008/07/07 04:23:20: SIP Tx udp:192.168.136.36:2051: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-81zc3zm6vbe4;rport=2051 From: <sip:5203@192.168.192.50>;tag=3eez6gof7x To: <sip:5203@192.168.192.50>;tag=1ee70c8e7e Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7 CSeq: 1033 REGISTER Content-Length: 0 [7] 2008/07/07 04:23:26: SIP Rx tls:192.168.25.103:2053: INVITE sip:4229@192.168.192.50;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport From: <sip:2201@192.168.192.50>;tag=p10csqj5hf To: <sip:4229@192.168.192.50;user=phone> Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/6.5.13 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 345 v=0 o=root 576733664 576733664 IN IP4 192.168.25.103 s=call c=IN IP4 192.168.25.103 t=0 0 m=audio 58152 RTP/AVP 18 4 0 8 3 9 2 101 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:9 g722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [8] 2008/07/07 04:23:26: Packet authenticated by transport layer [7] 2008/07/07 04:23:26: UDP: Opening socket on port 52908 [7] 2008/07/07 04:23:26: UDP: Opening socket on port 52909 [8] 2008/07/07 04:23:26: Could not find a trunk (1 trunks) [9] 2008/07/07 04:23:26: Using outbound proxy sip:192.168.25.103:2053;transport=tls because of flow-label [9] 2008/07/07 04:23:26: Resolve 48: tls 192.168.25.103 2053 [7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport=2053 From: <sip:2201@192.168.192.50>;tag=p10csqj5hf To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48 Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE CSeq: 1 INVITE Content-Length: 0 [7] 2008/07/07 04:23:26: Set packet length to 20 [6] 2008/07/07 04:23:26: Sending RTP for 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE#1c9ec29c48 to 192.168.25.103:58152 [9] 2008/07/07 04:23:26: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 4229@192.168.192.50 [5] 2008/07/07 04:23:26: Dialplan New: Match 4229@192.168.192.50 to <sip:4229@192.168.192.28:5061;user=phone> on trunk SIP [8] 2008/07/07 04:23:26: Play audio_moh/noise.wav [7] 2008/07/07 04:23:26: UDP: Opening socket on port 49714 [7] 2008/07/07 04:23:26: UDP: Opening socket on port 49715 [9] 2008/07/07 04:23:26: Resolve 49: url sip:192.168.192.28:5061 [9] 2008/07/07 04:23:26: Resolve 49: udp 192.168.192.28 5061 [7] 2008/07/07 04:23:26: SIP Tx udp:192.168.192.28:5061: INVITE sip:4229@192.168.192.28:5061;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.192.50:5060;branch=z9hG4bK-017e34d40401d0870149413127470191;rport From: <sip:2201@localhost>;tag=48313 To: <sip:4229@192.168.192.28:5061;user=phone> Call-ID: 910d81bb@pbx CSeq: 7463 INVITE Max-Forwards: 70 Contact: <sip:2201@192.168.192.50:5060;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/2.1.6.2450 Content-Type: application/sdp Content-Length: 294 v=0 o=- 56787 56787 IN IP4 192.168.192.50 s=- c=IN IP4 192.168.192.50 t=0 0 m=audio 49714 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/07/07 04:23:26: Set packet length to 20 [9] 2008/07/07 04:23:26: Resolve 50: tls 192.168.25.103 2053 [7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053: SIP/2.0 183 Ringing Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport=2053 From: <sip:2201@192.168.192.50>;tag=p10csqj5hf To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48 Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE CSeq: 1 INVITE Contact: <sip:2201@192.168.192.50:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2450 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 304 v=0 o=- 7292 7292 IN IP4 192.168.192.50 s=- c=IN IP4 192.168.192.50 t=0 0 m=audio 52908 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [8] 2008/07/07 04:23:26: UDP: recvfrom receives ICMP message [5] 2008/07/07 04:23:26: Connection refused on udp:192.168.192.28:5061 [6] 2008/07/07 04:23:26: Could not determine destination address on 49 [7] 2008/07/07 04:23:26: Call 910d81bb@pbx#48313: Clear last INVITE [9] 2008/07/07 04:23:26: Resolve 51: tls 192.168.25.103 2053 [7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053: SIP/2.0 500 Network Failure Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport=2053 From: <sip:2201@192.168.192.50>;tag=p10csqj5hf To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48 Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE CSeq: 1 INVITE Contact: <sip:2201@192.168.192.50:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2450 Content-Length: 0 [7] 2008/07/07 04:23:26: SIP Rx tls:192.168.25.103:2053: PRACK sip:2201@192.168.192.50:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-on4zow98h369;rport From: <sip:2201@192.168.192.50>;tag=p10csqj5hf To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48 Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 [8] 2008/07/07 04:23:26: Packet authenticated by transport layer [9] 2008/07/07 04:23:26: Resolve 52: tls 192.168.25.103 2053 [7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-on4zow98h369;rport=2053 From: <sip:2201@192.168.192.50>;tag=p10csqj5hf To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48 Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE CSeq: 2 PRACK Contact: <sip:2201@192.168.192.50:5061;transport=tls> User-Agent: pbxnsip-PBX/2.1.6.2450 Content-Length: 0 [7] 2008/07/07 04:23:26: SIP Rx tls:192.168.25.103:2053: ACK sip:4229@192.168.192.50;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport From: <sip:2201@192.168.192.50>;tag=p10csqj5hf To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48 Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1 Content-Length: 0 [8] 2008/07/07 04:23:26: Packet authenticated by transport layer [7] 2008/07/07 04:23:26: Other Ports: 1 [7] 2008/07/07 04:23:26: Call Port: 910d81bb@pbx#48313 [7] 2008/07/07 04:23:30: SIP Rx tls:192.168.25.103:2053: INVITE sip:4431@192.168.192.50;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw To: <sip:4431@192.168.192.50;user=phone> Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/6.5.13 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 347 v=0 o=root 1459444772 1459444772 IN IP4 192.168.25.103 s=call c=IN IP4 192.168.25.103 t=0 0 m=audio 58646 RTP/AVP 18 4 0 8 3 9 2 101 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:9 g722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [8] 2008/07/07 04:23:30: Packet authenticated by transport layer [7] 2008/07/07 04:23:30: UDP: Opening socket on port 52682 [7] 2008/07/07 04:23:30: UDP: Opening socket on port 52683 [8] 2008/07/07 04:23:30: Could not find a trunk (1 trunks) [9] 2008/07/07 04:23:30: Using outbound proxy sip:192.168.25.103:2053;transport=tls because of flow-label [9] 2008/07/07 04:23:30: Resolve 53: tls 192.168.25.103 2053 [7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport=2053 From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863 Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE CSeq: 1 INVITE Content-Length: 0 [7] 2008/07/07 04:23:30: Set packet length to 20 [6] 2008/07/07 04:23:30: Sending RTP for 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE#6f162b9863 to 192.168.25.103:58646 [9] 2008/07/07 04:23:30: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 4431@192.168.192.50 [5] 2008/07/07 04:23:30: Dialplan New: Match 4431@192.168.192.50 to <sip:4431@192.168.192.28:5061;user=phone> on trunk SIP [8] 2008/07/07 04:23:30: Play audio_moh/noise.wav [7] 2008/07/07 04:23:30: UDP: Opening socket on port 52286 [7] 2008/07/07 04:23:30: UDP: Opening socket on port 52287 [9] 2008/07/07 04:23:30: Resolve 54: url sip:192.168.192.28:5061 [9] 2008/07/07 04:23:30: Resolve 54: udp 192.168.192.28 5061 [7] 2008/07/07 04:23:30: SIP Tx udp:192.168.192.28:5061: INVITE sip:4431@192.168.192.28:5061;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.192.50:5060;branch=z9hG4bK-f969c5b8969691bf078c04d44f93e63f;rport From: <sip:2201@localhost>;tag=17880 To: <sip:4431@192.168.192.28:5061;user=phone> Call-ID: 63864075@pbx CSeq: 29415 INVITE Max-Forwards: 70 Contact: <sip:2201@192.168.192.50:5060;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/2.1.6.2450 Content-Type: application/sdp Content-Length: 292 v=0 o=- 2970 2970 IN IP4 192.168.192.50 s=- c=IN IP4 192.168.192.50 t=0 0 m=audio 52286 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/07/07 04:23:30: Set packet length to 20 [9] 2008/07/07 04:23:30: Resolve 55: tls 192.168.25.103 2053 [7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053: SIP/2.0 183 Ringing Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport=2053 From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863 Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE CSeq: 1 INVITE Contact: <sip:2201@192.168.192.50:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2450 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 306 v=0 o=- 28977 28977 IN IP4 192.168.192.50 s=- c=IN IP4 192.168.192.50 t=0 0 m=audio 52682 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [8] 2008/07/07 04:23:30: UDP: recvfrom receives ICMP message [5] 2008/07/07 04:23:30: Connection refused on udp:192.168.192.28:5061 [6] 2008/07/07 04:23:30: Could not determine destination address on 54 [7] 2008/07/07 04:23:30: Call 63864075@pbx#17880: Clear last INVITE [9] 2008/07/07 04:23:30: Resolve 56: tls 192.168.25.103 2053 [7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053: SIP/2.0 500 Network Failure Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport=2053 From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863 Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE CSeq: 1 INVITE Contact: <sip:2201@192.168.192.50:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2450 Content-Length: 0 [7] 2008/07/07 04:23:30: SIP Rx tls:192.168.25.103:2053: PRACK sip:2201@192.168.192.50:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-xb9k7e6khimn;rport From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863 Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 [8] 2008/07/07 04:23:30: Packet authenticated by transport layer [9] 2008/07/07 04:23:30: Resolve 57: tls 192.168.25.103 2053 [7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-xb9k7e6khimn;rport=2053 From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863 Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE CSeq: 2 PRACK Contact: <sip:2201@192.168.192.50:5061;transport=tls> User-Agent: pbxnsip-PBX/2.1.6.2450 Content-Length: 0 [7] 2008/07/07 04:23:30: SIP Rx tls:192.168.25.103:2053: ACK sip:4431@192.168.192.50;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863 Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1 Content-Length: 0 [8] 2008/07/07 04:23:30: Packet authenticated by transport layer [7] 2008/07/07 04:23:30: Other Ports: 2 [7] 2008/07/07 04:23:30: Call Port: 63864075@pbx#17880 [7] 2008/07/07 04:23:30: Call Port: 910d81bb@pbx#48313 [8] 2008/07/07 04:23:34: Hangup: Call 910d81bb@pbx#48313 not found [7] 2008/07/07 04:23:36: SIP Rx udp:192.168.136.36:2051: REGISTER sip:192.168.192.50 SIP/2.0 Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-e5flcagayes3;rport From: <sip:5203@192.168.192.50>;tag=cm2h1pdaec To: <sip:5203@192.168.192.50> Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7 CSeq: 1034 REGISTER Max-Forwards: 70 Contact: <sip:5203@192.168.136.36:2051;line=vfooudzb>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:519aec13-5a5d-4f46-92d3-aa8451bb25aa>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/6.5.13 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.136.36 WWW-Contact: <http://192.168.136.36:80> WWW-Contact: <https://192.168.136.36:443> Expires: 3600 Content-Length: 0 [9] 2008/07/07 04:23:36: Resolve 58: aaaa udp 192.168.136.36 2051 [9] 2008/07/07 04:23:36: Resolve 58: a udp 192.168.136.36 2051 [9] 2008/07/07 04:23:36: Resolve 58: udp 192.168.136.36 2051 [7] 2008/07/07 04:23:36: SIP Tx udp:192.168.136.36:2051: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-e5flcagayes3;rport=2051 From: <sip:5203@192.168.192.50>;tag=cm2h1pdaec To: <sip:5203@192.168.192.50>;tag=1ee70c8e7e Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7 CSeq: 1034 REGISTER Content-Length: 0 [8] 2008/07/07 04:23:38: Hangup: Call 63864075@pbx#17880 not found [7] 2008/07/07 04:23:45: SIP Rx tls:192.168.25.103:2053: REGISTER sip:192.168.192.50 SIP/2.0 Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-8ruwlxvmnsxh;rport From: <sip:2201@192.168.192.50>;tag=zn5jibw3c8 To: <sip:2201@192.168.192.50> Call-ID: 3c267013a604-k4l8r8hzrhkc@snom300-0004132889BE CSeq: 5 REGISTER Max-Forwards: 70 Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:5c780463-7a16-4199-bb5c-a029eae57121>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/6.5.13 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.25.103 WWW-Contact: <http://192.168.25.103:80> WWW-Contact: <https://192.168.25.103:443> Expires: 3600 Content-Length: 0 [8] 2008/07/07 04:23:45: Packet authenticated by transport layer [9] 2008/07/07 04:23:45: Resolve 59: tls 192.168.25.103 2053 [7] 2008/07/07 04:23:45: SIP Tx tls:192.168.25.103:2053: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-8ruwlxvmnsxh;rport=2053 From: <sip:2201@192.168.192.50>;tag=zn5jibw3c8 To: <sip:2201@192.168.192.50>;tag=f9ed8b9df9 Call-ID: 3c267013a604-k4l8r8hzrhkc@snom300-0004132889BE CSeq: 5 REGISTER Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;expires=178 Content-Length: 0 [5] 2008/07/07 04:23:46: SIP port accept from 192.168.192.28:14946 [7] 2008/07/07 04:23:51: SIP Rx udp:192.168.136.36:2051: REGISTER sip:192.168.192.50 SIP/2.0 Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-amoqwm4o901i;rport From: <sip:5203@192.168.192.50>;tag=g4q9ekm9mp To: <sip:5203@192.168.192.50> Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7 CSeq: 1035 REGISTER Max-Forwards: 70 Contact: <sip:5203@192.168.136.36:2051;line=vfooudzb>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:519aec13-5a5d-4f46-92d3-aa8451bb25aa>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/6.5.13 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.136.36 WWW-Contact: <http://192.168.136.36:80> WWW-Contact: <https://192.168.136.36:443> Expires: 3600 Content-Length: 0 [9] 2008/07/07 04:23:51: Resolve 60: aaaa udp 192.168.136.36 2051 [9] 2008/07/07 04:23:51: Resolve 60: a udp 192.168.136.36 2051 [9] 2008/07/07 04:23:51: Resolve 60: udp 192.168.136.36 2051 [7] 2008/07/07 04:23:51: SIP Tx udp:192.168.136.36:2051: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-amoqwm4o901i;rport=2051 From: <sip:5203@192.168.192.50>;tag=g4q9ekm9mp To: <sip:5203@192.168.192.50>;tag=1ee70c8e7e Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7 CSeq: 1035 REGISTER Content-Length: 0 Wireshark_SIP_logs1.zip SIP_logs1.txt Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted July 7, 2008 Report Share Posted July 7, 2008 Looks to me like you are talking to the wrong port (UDP 5061) on the Avaya system. Check your outbound proxy. Maybe you have to use TLS or change the port 5060. Quote Link to comment Share on other sites More sharing options...
Ganesh Posted July 17, 2008 Author Report Share Posted July 17, 2008 We configured the outbound proxy and specified to use TLS (as Avaya only supports TLS on direct SIP trunk without Avaya SES). We were still unable to route calls between the two systems. Below is the logs captured. 59999 is the Avaya phone extension. This link (http://www.avayausers.com/showthread.php?t=10700) says that "TCP Sip is supported, but UDP SIP is not supported without the Avaya Sip Server. (SES) Even then you still need a session border controller" (we are not using the Avaya SES here). It also says SBC is required while using the SIP trunk on Avaya. Since we are integrating this in the same network (LAN), do we need a SBC? Also I think SBC is inbuilt in PBXNSIP. Right? Should i try using a SBC or is there any other settings we need to do from our side. Below is the logs captured from PBXNSIP. [7] 2008/07/16 06:45:20: SIP Rx udp:192.168.38.21:12163: INVITE sip:59999@192.168.38.21 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:79999@192.168.38.21:12163> To: "59999"<sip:59999@192.168.38.21> From: "79999"<sip:79999@192.168.38.21>;tag=20694b71 Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 423 v=0 o=- 6 2 IN IP4 192.168.38.21 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.38.21 t=0 0 m=audio 3622 RTP/AVP 107 119 100 106 0 105 98 8 3 101 a=alt:1 1 : qqzBZNsZ mIyNWmYl 192.168.38.21 3622 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv [7] 2008/07/16 06:45:20: UDP: Opening socket on port 49246 [7] 2008/07/16 06:45:20: UDP: Opening socket on port 49247 [8] 2008/07/16 06:45:20: Could not find a trunk (1 trunks) [8] 2008/07/16 06:45:20: Using outbound proxy sip:192.168.38.21:12163;transport=udp because UDP packet source did not match the via header [9] 2008/07/16 06:45:20: Resolve 50: udp 192.168.38.21 12163 [7] 2008/07/16 06:45:20: SIP Tx udp:192.168.38.21:12163: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21 From: "79999" <sip:79999@192.168.38.21>;tag=20694b71 To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412 Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ. CSeq: 1 INVITE Content-Length: 0 [6] 2008/07/16 06:45:20: Sending RTP for MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.#6325fc9412 to 192.168.38.21:3622 [9] 2008/07/16 06:45:21: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 59999@192.168.38.21 [5] 2008/07/16 06:45:21: Dialplan New: Match 59999@192.168.38.21 to <sip:59999@192.168.38.20;user=phone> on trunk SIP [8] 2008/07/16 06:45:21: Play audio_moh/noise.wav [7] 2008/07/16 06:45:21: UDP: Opening socket on port 59432 [7] 2008/07/16 06:45:21: UDP: Opening socket on port 59433 [9] 2008/07/16 06:45:21: Resolve 51: url sip:192.168.38.20:5061;transport=tls [9] 2008/07/16 06:45:21: Resolve 51: a tls 192.168.38.20 5061 [9] 2008/07/16 06:45:21: Resolve 51: tls 192.168.38.20 5061 [7] 2008/07/16 06:45:21: SIP Tx tls:192.168.38.20:5061: INVITE sip:59999@192.168.38.20;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.38.21:1210;branch=z9hG4bK-48a3669a14150ffd1e6b3c48e9c5f659;rport From: <sip:79999@localhost>;tag=5447 To: <sip:59999@192.168.38.20;user=phone> Call-ID: ff5f3792@pbx CSeq: 6271 INVITE Max-Forwards: 70 Contact: <sip:79999@192.168.38.21:1210;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2450 Content-Type: application/sdp Content-Length: 423 v=0 o=- 28504 28504 IN IP4 192.168.38.21 s=- c=IN IP4 192.168.38.21 t=0 0 m=audio 59432 RTP/AVP 0 8 9 18 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:UxcnvE8+sfevfgeIrnn35dXnjcuAf3Ikos1Dnk3f a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2008/07/16 06:45:21: Resolve 52: udp 192.168.38.21 12163 [7] 2008/07/16 06:45:21: SIP Tx udp:192.168.38.21:12163: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21 From: "79999" <sip:79999@192.168.38.21>;tag=20694b71 To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412 Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ. CSeq: 1 INVITE Contact: <sip:79999@127.0.0.1:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2450 Content-Type: application/sdp Content-Length: 233 v=0 o=- 16981 16981 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 49246 RTP/AVP 0 8 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [6] 2008/07/16 06:45:21: Sending RTP for MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.#6325fc9412 to 127.0.0.1:3622 [7] 2008/07/16 06:45:21: SIP Tr udp:192.168.38.21:12163: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21 From: "79999" <sip:79999@192.168.38.21>;tag=20694b71 To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412 Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ. CSeq: 1 INVITE Contact: <sip:79999@127.0.0.1:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2450 Content-Type: application/sdp Content-Length: 233 v=0 o=- 16981 16981 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 49246 RTP/AVP 0 8 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/07/16 06:45:22: SIP Rx udp:192.168.38.21:12163: REGISTER sip:192.168.38.21 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-8e7bc46eb755e302-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14> To: "79999"<sip:79999@192.168.38.21> From: "79999"<sip:79999@192.168.38.21>;tag=0948d273 Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ. CSeq: 26 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 [9] 2008/07/16 06:45:22: Resolve 53: udp 192.168.38.21 12163 [7] 2008/07/16 06:45:22: SIP Tx udp:192.168.38.21:12163: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-8e7bc46eb755e302-1---d8754z-;rport=12163;received=192.168.38.21 From: "79999" <sip:79999@192.168.38.21>;tag=0948d273 To: "79999" <sip:79999@192.168.38.21>;tag=11399c85b8 Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ. CSeq: 26 REGISTER Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>;expires=31 Content-Length: 0 [7] 2008/07/16 06:45:22: SIP Tr udp:192.168.38.21:12163: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21 From: "79999" <sip:79999@192.168.38.21>;tag=20694b71 To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412 Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ. CSeq: 1 INVITE Contact: <sip:79999@127.0.0.1:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2450 Content-Type: application/sdp Content-Length: 233 v=0 o=- 16981 16981 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 49246 RTP/AVP 0 8 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/07/16 06:45:41: Last message repeated 4 times [5] 2008/07/16 06:45:41: SIP port accept from 192.168.38.14:24434 [7] 2008/07/16 06:45:48: SIP Rx udp:192.168.38.21:12163: REGISTER sip:192.168.38.21 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-0e56b011be31d71c-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14> To: "79999"<sip:79999@192.168.38.21> From: "79999"<sip:79999@192.168.38.21>;tag=0948d273 Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ. CSeq: 27 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 [9] 2008/07/16 06:45:48: Resolve 54: udp 192.168.38.21 12163 [7] 2008/07/16 06:45:48: SIP Tx udp:192.168.38.21:12163: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-0e56b011be31d71c-1---d8754z-;rport=12163;received=192.168.38.21 From: "79999" <sip:79999@192.168.38.21>;tag=0948d273 To: "79999" <sip:79999@192.168.38.21>;tag=11399c85b8 Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ. CSeq: 27 REGISTER Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>;expires=29 Content-Length: 0 [7] 2008/07/16 06:45:51: Call ff5f3792@pbx#5447: Clear last INVITE [9] 2008/07/16 06:45:51: Resolve 55: udp 192.168.38.21 12163 [7] 2008/07/16 06:45:51: SIP Tx udp:192.168.38.21:12163: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21 From: "79999" <sip:79999@192.168.38.21>;tag=20694b71 To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412 Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ. CSeq: 1 INVITE Contact: <sip:79999@127.0.0.1:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2450 Content-Length: 0 [7] 2008/07/16 06:45:51: SIP Rx udp:192.168.38.21:12163: ACK sip:59999@192.168.38.21 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412 From: "79999"<sip:79999@192.168.38.21>;tag=20694b71 Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ. CSeq: 1 ACK Content-Length: 0 [7] 2008/07/16 06:45:51: Other Ports: 1 [7] 2008/07/16 06:45:51: Call Port: ff5f3792@pbx#5447 [8] 2008/07/16 06:45:59: Hangup: Call ff5f3792@pbx#5447 not found Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted July 17, 2008 Report Share Posted July 17, 2008 We configured the outbound proxy and specified to use TLS (as Avaya only supports TLS on direct SIP trunk without Avaya SES). We were still unable to route calls between the two systems. Below is the logs captured. 59999 is the Avaya phone extension. This link (http://www.avayausers.com/showthread.php?t=10700) says that "TCP Sip is supported, but UDP SIP is not supported without the Avaya Sip Server. (SES) Even then you still need a session border controller" (we are not using the Avaya SES here). It also says SBC is required while using the SIP trunk on Avaya. Since we are integrating this in the same network (LAN), do we need a SBC? Also I think SBC is inbuilt in PBXNSIP. Right? Should i try using a SBC or is there any other settings we need to do from our side. Below is the logs captured from PBXNSIP. Well, obviously there is not much coming back from 192.168.38.20 on port 5061 (TLS). So I would try outbound proxy sip:192.168.38.20:5060;transport=tcp. You don't need a SBC, that job is already done by the PBX. Quote Link to comment Share on other sites More sharing options...
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