satish_infy

Hi PBXnSIP, where is the NAT in this topology?

well its just not XLite.. another softphone like SJPhone using tcp had the same issue.. as soon as i used UDP (the default protocol in both softphones), the softphone hung up. I was using TCP because Speech Server insists on SIP over TCP, and the PBX's trunk has its outbound proxy setup that way, ie to force transport=tcp when talking to Speech Server. Looks like PBXnSIP very nicely converts UDP to TCP and vice versa to facilitate communication between server and client. Where exactly is the NAT in this topology btw?

Awesome support PBXnSIP.. i did not expect such a quick reply.. you just won yourselves many client recommendations You are right that PBXnSIP did transmit the BYE to the client, but the client never received it, and no ACK was seen in the pbxnsip log. I changed xlite to use UDP, and bang, it hung up! thanks!! But i'm technically curious.. why isn't tcp working.. all the other SIP commands before, like the INVITE, were in TCP right? There is a VPN connection (with a firewall) in between.. my client VPN's to my corporate network where Speech Server is installed.. could it be that the VPN is blocking the BYE as it originated from the server? But shouldn't the firewall filter UDP traffic too then..?

Pretty simple topology for demo purposes.. basically I have PBXnSIP 3.0 and Speech Server 07 on the same box. The intent is to use PBXnSIP as a SIP gatway to speech server, from my softphones like Xlite or SJPhone. I have setup two trunks, one for accepting the incoming SIP connection from my softphone, and another trunk to Speech Server (basically following these instructions http://www.selectec.co.uk/pbx.mht) I have setup a dial-plan, and the SpeechServer trunk is configured for an outbound proxy. Everything finally works.. only one problem, that PBXnSIP doesn't hangup the call, even though Speech Server already has. Notice SpeechServer sent the BYE at 00:00:15 [7] 2008/08/24 00:00:15: SIP Rx tcp:10.156.112.97:19098: BYE sip:satish@127.0.0.1:19106;transport=tcp SIP/2.0 FROM: <sip:333@localhost;user=phone>;epid=E9C4EF8C22;tag=6c6ad6a0b7 TO: <sip:satish@192.168.1.101;user=phone>;tag=41349 CSEQ: 1 BYE CALL-ID: d40833a0@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 10.156.112.97:19098;branch=z9hG4bK4a1f364 CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 And I had to manually hangup the client at 00:00:42 [7] 2008/08/24 00:00:42: SIP Rx tcp:10.110.8.81:4620: BYE sip:333@10.156.112.97:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-fe488c314e532a3d-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:satish@10.110.8.81:48800;transport=TCP> To: "welcome To Ocs"<sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484 From: "satish"<sip:satish@192.168.1.101>;tag=714c423f Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI. CSeq: 2 BYE User-Agent: X-Lite release 1100l stamp 47546 Reason: SIP;description="User Hung Up" Content-Length: 0 Looks like the client never got the BYE at 00:15.. Attached is the log: [5] 2008/08/24 00:00:08: SIP port accept from 10.110.8.81:4620 [7] 2008/08/24 00:00:08: SIP Rx tcp:10.110.8.81:4620: INVITE sip:333@10.156.112.97;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-f158cf53ab637e16-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:satish@10.110.8.81:48800;transport=TCP> To: "welcome To Ocs"<sip:333@10.156.112.97;transport=tcp> From: "satish"<sip:satish@192.168.1.101>;tag=714c423f Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI. 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: 471 v=0 o=- 0 2 IN IP4 192.168.1.101 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.101 t=0 0 m=audio 7150 RTP/AVP 107 119 100 106 0 105 98 8 3 101 a=alt:1 2 : twqBdPxb 9cMwUw1K 10.110.8.81 7150 a=alt:2 1 : azi2oQJs n+GjKaE4 192.168.1.101 7150 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] 2008/08/24 00:00:08: UDP: Opening socket on port 61328 [7] 2008/08/24 00:00:08: UDP: Opening socket on port 61329 [5] 2008/08/24 00:00:08: Identify trunk (domain name match) 3 [9] 2008/08/24 00:00:08: Resolve 167: tcp 10.110.8.81 4620 [7] 2008/08/24 00:00:08: SIP Tx tcp:10.110.8.81:4620: SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-f158cf53ab637e16-1---d8754z-;rport=4620 From: "satish" <sip:satish@192.168.1.101>;tag=714c423f To: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484 Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI. CSeq: 1 INVITE Content-Length: 0 [6] 2008/08/24 00:00:08: Sending RTP for Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484 to 192.168.1.101:7150 [8] 2008/08/24 00:00:08: Trunk: Changing the user to 999 [9] 2008/08/24 00:00:08: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 333@10.156.112.97 [5] 2008/08/24 00:00:08: Dialplan mss07: Match 333@10.156.112.97 to <sip:333@localhost;user=phone> on trunk SpeechServer [5] 2008/08/24 00:00:08: Using "satish" <sip:satish@192.168.1.101;user=phone>;tag=714c423f as redirect from [5] 2008/08/24 00:00:08: Charge user 999 for redirecting calls [8] 2008/08/24 00:00:08: Play audio_moh/noise.wav [7] 2008/08/24 00:00:08: UDP: Opening socket on port 49896 [7] 2008/08/24 00:00:08: UDP: Opening socket on port 49897 [9] 2008/08/24 00:00:08: Resolve 168: url sip:10.156.112.97:6060;transport=tcp [9] 2008/08/24 00:00:08: Resolve 168: a tcp 10.156.112.97 6060 [9] 2008/08/24 00:00:08: Resolve 168: tcp 10.156.112.97 6060 [7] 2008/08/24 00:00:08: SIP Tx tcp:10.156.112.97:6060: INVITE sip:333@localhost;user=phone SIP/2.0 Via: SIP/2.0/TCP 127.0.0.1:19105;branch=z9hG4bK-1476699eba8840bb676b659edb916c27;rport From: "satish" <sip:satish@192.168.1.101;user=phone>;tag=41349 To: <sip:333@localhost;user=phone> Call-ID: d40833a0@pbx CSeq: 22698 INVITE Max-Forwards: 70 Contact: <sip:satish@127.0.0.1:19105;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/3.0.0.2998 Diversion: <tel:999>;reason=unconditional;screen=no;privacy=off P-Asserted-Identity: <sip:999@localhost;user=phone> Content-Type: application/sdp Content-Length: 284 v=0 o=- 52411 52411 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 49896 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 [9] 2008/08/24 00:00:08: Resolve 169: tcp 10.110.8.81 4620 [7] 2008/08/24 00:00:08: SIP Tx tcp:10.110.8.81:4620: SIP/2.0 183 Ringing Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-f158cf53ab637e16-1---d8754z-;rport=4620 From: "satish" <sip:satish@192.168.1.101>;tag=714c423f To: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484 Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI. CSeq: 1 INVITE Contact: <sip:333@10.156.112.97:5060;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/3.0.0.2998 Content-Type: application/sdp Content-Length: 241 v=0 o=- 56222 56222 IN IP4 10.156.112.97 s=- c=IN IP4 10.156.112.97 t=0 0 m=audio 61328 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] 2008/08/24 00:00:08: SIP Rx tcp:10.156.112.97:6060: SIP/2.0 100 Trying FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349 TO: <sip:333@localhost;user=phone> CSEQ: 22698 INVITE CALL-ID: d40833a0@pbx VIA: SIP/2.0/TCP 127.0.0.1:19105;branch=z9hG4bK-1476699eba8840bb676b659edb916c27;rport CONTENT-LENGTH: 0 [7] 2008/08/24 00:00:08: SIP Rx tcp:10.156.112.97:6060: SIP/2.0 302 Moved Temporarily FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349 TO: <sip:333@localhost;user=phone>;tag=a8d6c6f98 CSEQ: 22698 INVITE CALL-ID: d40833a0@pbx VIA: SIP/2.0/TCP 127.0.0.1:19105;branch=z9hG4bK-1476699eba8840bb676b659edb916c27;rport CONTACT: <sip:333@localhost:19098;user=phone;transport=Tcp;maddr=10.156.112.97;x-mss-call-id=d40833a0%40pbx> CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [7] 2008/08/24 00:00:08: Call d40833a0@pbx#41349: Clear last INVITE [9] 2008/08/24 00:00:08: Resolve 170: url sip:10.156.112.97:6060;transport=tcp [9] 2008/08/24 00:00:08: Resolve 170: a tcp 10.156.112.97 6060 [9] 2008/08/24 00:00:08: Resolve 170: tcp 10.156.112.97 6060 [7] 2008/08/24 00:00:08: SIP Tx tcp:10.156.112.97:6060: ACK sip:333@localhost;user=phone SIP/2.0 Via: SIP/2.0/TCP 127.0.0.1:19105;branch=z9hG4bK-1476699eba8840bb676b659edb916c27;rport From: "satish" <sip:satish@192.168.1.101;user=phone>;tag=41349 To: <sip:333@localhost;user=phone>;tag=a8d6c6f98 Call-ID: d40833a0@pbx CSeq: 22698 ACK Max-Forwards: 70 Contact: <sip:satish@127.0.0.1:19105;transport=tcp> P-Asserted-Identity: <sip:999@localhost;user=phone> Content-Length: 0 [5] 2008/08/24 00:00:08: Redirecting call [9] 2008/08/24 00:00:08: Resolve 171: aaaa tcp 10.156.112.97 19098 [9] 2008/08/24 00:00:08: Resolve 171: a tcp 10.156.112.97 19098 [9] 2008/08/24 00:00:08: Resolve 171: tcp 10.156.112.97 19098 [7] 2008/08/24 00:00:08: SIP Tx tcp:10.156.112.97:19098: INVITE sip:333@localhost:19098;user=phone;transport=Tcp;maddr=10.156.112.97;x-mss-call-id=d40833a0%40pbx SIP/2.0 Via: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-a96f8438f6b5466df6a6c8968f199eb4;rport From: "satish" <sip:satish@192.168.1.101;user=phone>;tag=41349 To: <sip:333@localhost;user=phone> Call-ID: d40833a0@pbx CSeq: 22699 INVITE Max-Forwards: 70 Contact: <sip:satish@127.0.0.1:19106;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/3.0.0.2998 Diversion: <tel:999>;reason=unconditional;screen=no;privacy=off P-Asserted-Identity: <sip:999@localhost;user=phone> Content-Type: application/sdp Content-Length: 284 v=0 o=- 52411 52411 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 49896 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] 2008/08/24 00:00:09: SIP Rx tcp:10.156.112.97:19098: SIP/2.0 100 Trying FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349 TO: <sip:333@localhost;user=phone> CSEQ: 22699 INVITE CALL-ID: d40833a0@pbx VIA: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-a96f8438f6b5466df6a6c8968f199eb4;rport CONTENT-LENGTH: 0 [6] 2008/08/24 00:00:09: Sending RTP for Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484 to 10.110.8.81:7150 [7] 2008/08/24 00:00:09: SIP Rx tcp:10.156.112.97:19098: SIP/2.0 180 Ringing FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349 TO: <sip:333@localhost;user=phone>;epid=E9C4EF8C22;tag=6c6ad6a0b7 CSEQ: 22699 INVITE CALL-ID: d40833a0@pbx VIA: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-a96f8438f6b5466df6a6c8968f199eb4;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [8] 2008/08/24 00:00:09: Play audio_en/ringback.wav [6] 2008/08/24 00:00:09: Sending RTP for d40833a0@pbx#41349 to 10.156.112.97:41728 [7] 2008/08/24 00:00:10: SIP Rx tcp:10.156.112.97:19098: SIP/2.0 200 OK FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349 TO: <sip:333@localhost;user=phone>;epid=E9C4EF8C22;tag=6c6ad6a0b7 CSEQ: 22699 INVITE CALL-ID: d40833a0@pbx VIA: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-a96f8438f6b5466df6a6c8968f199eb4;rport CONTACT: <sip:retlarch.ad.infosys.com:19098;transport=Tcp;maddr=10.156.112.97>;automata CONTENT-LENGTH: 196 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 10.156.112.97 s=Microsoft Speech Server session c=IN IP4 10.156.112.97 t=0 0 m=audio 41728 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [7] 2008/08/24 00:00:10: Call d40833a0@pbx#41349: Clear last INVITE [7] 2008/08/24 00:00:10: Set packet length to 20 [9] 2008/08/24 00:00:10: Resolve 172: aaaa tcp 10.156.112.97 19098 [9] 2008/08/24 00:00:10: Resolve 172: a tcp 10.156.112.97 19098 [9] 2008/08/24 00:00:10: Resolve 172: tcp 10.156.112.97 19098 [7] 2008/08/24 00:00:10: SIP Tx tcp:10.156.112.97:19098: ACK sip:retlarch.ad.infosys.com:19098;transport=Tcp;maddr=10.156.112.97 SIP/2.0 Via: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-e8db414c4859d96f2949ac0c16fccf9c;rport From: "satish" <sip:satish@192.168.1.101;user=phone>;tag=41349 To: <sip:333@localhost;user=phone>;tag=6c6ad6a0b7 Call-ID: d40833a0@pbx CSeq: 22699 ACK Max-Forwards: 70 Contact: <sip:satish@127.0.0.1:19106;transport=tcp> P-Asserted-Identity: <sip:999@localhost;user=phone> Content-Length: 0 [7] 2008/08/24 00:00:10: Determine pass-through mode after receiving response [9] 2008/08/24 00:00:10: Resolve 173: tcp 10.110.8.81 4620 [7] 2008/08/24 00:00:10: SIP Tx tcp:10.110.8.81:4620: SIP/2.0 200 Ok Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-f158cf53ab637e16-1---d8754z-;rport=4620 From: "satish" <sip:satish@192.168.1.101>;tag=714c423f To: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484 Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI. CSeq: 1 INVITE Contact: <sip:333@10.156.112.97:5060;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/3.0.0.2998 Content-Type: application/sdp Content-Length: 241 v=0 o=- 56222 56222 IN IP4 10.156.112.97 s=- c=IN IP4 10.156.112.97 t=0 0 m=audio 61328 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] 2008/08/24 00:00:10: d40833a0@pbx#41349: RTP pass-through mode [7] 2008/08/24 00:00:10: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484: RTP pass-through mode [7] 2008/08/24 00:00:10: SIP Rx tcp:10.110.8.81:4620: ACK sip:333@10.156.112.97:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-5d715935b7300d60-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:satish@10.110.8.81:48800;transport=TCP> To: "welcome To Ocs"<sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484 From: "satish"<sip:satish@192.168.1.101>;tag=714c423f Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI. CSeq: 1 ACK User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 [7] 2008/08/24 00:00:15: SIP Rx tcp:10.156.112.97:19098: BYE sip:satish@127.0.0.1:19106;transport=tcp SIP/2.0 FROM: <sip:333@localhost;user=phone>;epid=E9C4EF8C22;tag=6c6ad6a0b7 TO: <sip:satish@192.168.1.101;user=phone>;tag=41349 CSEQ: 1 BYE CALL-ID: d40833a0@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 10.156.112.97:19098;branch=z9hG4bK4a1f364 CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 [9] 2008/08/24 00:00:15: Resolve 174: tcp 10.156.112.97 19098 [7] 2008/08/24 00:00:15: SIP Tx tcp:10.156.112.97:19098: SIP/2.0 200 Ok Via: SIP/2.0/TCP 10.156.112.97:19098;branch=z9hG4bK4a1f364 From: <sip:333@localhost;user=phone>;tag=6c6ad6a0b7;epid=E9C4EF8C22 To: <sip:satish@192.168.1.101;user=phone>;tag=41349 Call-ID: d40833a0@pbx CSeq: 1 BYE Contact: <sip:satish@127.0.0.1:19106;transport=tcp> User-Agent: pbxnsip-PBX/3.0.0.2998 RTP-RxStat: Dur=7,Pkt=277,Oct=47644,Underun=0 RTP-TxStat: Dur=6,Pkt=285,Oct=49020 Content-Length: 0 [7] 2008/08/24 00:00:15: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484: Media-aware pass-through mode [7] 2008/08/24 00:00:15: Other Ports: 1 [7] 2008/08/24 00:00:15: Call Port: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484 [9] 2008/08/24 00:00:15: Resolve 175: url sip:satish@10.110.8.81:48800;transport=TCP [9] 2008/08/24 00:00:15: Resolve 175: a tcp 10.110.8.81 48800 [9] 2008/08/24 00:00:15: Resolve 175: tcp 10.110.8.81 48800 [7] 2008/08/24 00:00:15: SIP Tx tcp:10.110.8.81:48800: BYE sip:satish@10.110.8.81:48800;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 10.156.112.97:19109;branch=z9hG4bK-ee49be524ff19fe16e8ab0df1fd0a7a3;rport From: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484 To: "satish" <sip:satish@192.168.1.101>;tag=714c423f Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI. CSeq: 23906 BYE Max-Forwards: 70 Contact: <sip:333@10.156.112.97:19109;transport=tcp> RTP-RxStat: Dur=7,Pkt=331,Oct=56932,Underun=0 RTP-TxStat: Dur=6,Pkt=328,Oct=56416 Content-Length: 0 [7] 2008/08/24 00:00:42: SIP Rx tcp:10.110.8.81:4620: BYE sip:333@10.156.112.97:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-fe488c314e532a3d-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:satish@10.110.8.81:48800;transport=TCP> To: "welcome To Ocs"<sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484 From: "satish"<sip:satish@192.168.1.101>;tag=714c423f Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI. CSeq: 2 BYE User-Agent: X-Lite release 1100l stamp 47546 Reason: SIP;description="User Hung Up" Content-Length: 0 [9] 2008/08/24 00:00:42: Resolve 176: tcp 10.110.8.81 4620 [7] 2008/08/24 00:00:42: SIP Tx tcp:10.110.8.81:4620: SIP/2.0 200 Ok Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-fe488c314e532a3d-1---d8754z-;rport=4620 From: "satish" <sip:satish@192.168.1.101>;tag=714c423f To: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484 Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI. CSeq: 2 BYE Contact: <sip:333@10.156.112.97:5060;transport=tcp> User-Agent: pbxnsip-PBX/3.0.0.2998 RTP-RxStat: Dur=34,Pkt=1632,Oct=280704,Underun=0 RTP-TxStat: Dur=32,Pkt=1653,Oct=284316 Content-Length: 0 [8] 2008/08/24 00:00:42: Hangup: Call Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484 not found