joso Posted September 17, 2008 Report Share Posted September 17, 2008 i have the audiocodes mp-118 fxo connected to pbxnsip and ocs 2007 and exchange 2007. When i call in, the voicemail kicks in with the exchange attendant. after the beep it seems it always disconnects after 10sec. Am i missing a setting here? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 18, 2008 Report Share Posted September 18, 2008 i have the audiocodes mp-118 fxo connected to pbxnsip and ocs 2007 and exchange 2007. When i call in, the voicemail kicks in with the exchange attendant. after the beep it seems it always disconnects after 10sec. Am i missing a setting here? Could be a problem with the ACK routing (do you have more than one IP address or a firewall somewhere?). Or could be a tone detection that is too aggressive detecting a hangup. Maybe you can get the SIP packets between the AC and the PBX and we can see if the hangup comes from there. If that does not give any insight, we can do the same thing between the PBX and Exchange. Divide and conquer. Quote Link to comment Share on other sites More sharing options...
joso Posted September 18, 2008 Author Report Share Posted September 18, 2008 I only have one firewall and ip at the moment. I have the log file, not sure if it will help: Also, what does this mean "SIP TCP/TLS timeout on 192.168.1.2:42354, closing connection " 5] 2008/09/18 15:11:12: SIP port accept from 192.168.1.2:42393 [7] 2008/09/18 15:11:21: SIP Rx udp:192.168.1.6:5060: INVITE sip:301@192.168.1.5;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019376203 Max-Forwards: 70 From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919 To: <sip:301@192.168.1.5;user=phone> Call-ID: 201937152741200005929@192.168.1.6 CSeq: 1 INVITE Contact: <sip:4032708885@192.168.1.6> Supported: em,100rel,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004 Content-Type: application/sdp Content-Disposition: session Content-Length: 287 v=0 o=AudiocodesGW 2019365073 2019364953 IN IP4 192.168.1.6 s=Phone-Call c=IN IP4 192.168.1.6 t=0 0 m=audio 6000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:6001 IN IP4 192.168.1.6 [7] 2008/09/18 15:11:21: UDP: Opening socket on port 50166 [7] 2008/09/18 15:11:21: UDP: Opening socket on port 50167 [5] 2008/09/18 15:11:21: Identify trunk (domain name match) 2 [7] 2008/09/18 15:11:21: SIP Tx udp:192.168.1.6:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019376203 From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919 To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd Call-ID: 201937152741200005929@192.168.1.6 CSeq: 1 INVITE Content-Length: 0 [7] 2008/09/18 15:11:21: Set packet length to 20 [6] 2008/09/18 15:11:21: Sending RTP for 201937152741200005929@192.168.1.6#ebaf4451bd to 192.168.1.6:6000 [5] 2008/09/18 15:11:21: Trunk AudioCodes sends call to 301 [7] 2008/09/18 15:11:21: Set packet length to 20 [7] 2008/09/18 15:11:21: SIP Tx udp:192.168.1.6:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019376203 From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919 To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd Call-ID: 201937152741200005929@192.168.1.6 CSeq: 1 INVITE Contact: <sip:josog@192.168.1.5:5060;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/3.0.0.2998 Content-Type: application/sdp Content-Length: 226 v=0 o=- 62207 62207 IN IP4 192.168.1.5 s=- c=IN IP4 192.168.1.5 t=0 0 m=audio 50166 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=sendrecv [7] 2008/09/18 15:11:21: Last message repeated 2 times [7] 2008/09/18 15:11:21: SIP Rx udp:192.168.1.6:5060: ACK sip:josog@192.168.1.5:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019562213 Max-Forwards: 70 From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919 To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd Call-ID: 201937152741200005929@192.168.1.6 CSeq: 1 ACK Contact: <sip:4032708885@192.168.1.6> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004 Content-Length: 0 [7] 2008/09/18 15:11:29: Last message repeated 2 times [6] 2008/09/18 15:11:29: Received DTMF 3 [6] 2008/09/18 15:11:29: Received DTMF 2 [6] 2008/09/18 15:11:29: Received DTMF 5 [7] 2008/09/18 15:11:32: Calling extension 325 [7] 2008/09/18 15:11:32: UDP: Opening socket on port 60722 [7] 2008/09/18 15:11:32: UDP: Opening socket on port 60723 [5] 2008/09/18 15:11:32: Dialplan jngconsulting: Match 4037105450@localhost to <sip:4037105450@192.168.1.6;user=phone> on trunk AudioCodes [5] 2008/09/18 15:11:32: Using "Subnet" <sip:4032708885@localhost;user=phone> as redirect from [5] 2008/09/18 15:11:32: Charge user 325 for redirecting calls [7] 2008/09/18 15:11:32: SIP Tx tcp:192.168.1.4:5060: INVITE sip:+4036707140325@ssi-ocsmed.jngconsulting.com;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=4673 To: "Joso Grivicic" <sip:325@localhost> Call-ID: 82331c2d@pbx CSeq: 12820 INVITE Max-Forwards: 70 Contact: <sip:325@192.168.1.5:4638;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 Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 335 v=0 o=- 60462 60462 IN IP4 192.168.1.5 s=- c=IN IP4 192.168.1.5 t=0 0 m=audio 60722 RTP/AVP 0 8 9 18 2 3 101 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 [7] 2008/09/18 15:11:32: UDP: Opening socket on port 55700 [7] 2008/09/18 15:11:32: UDP: Opening socket on port 55701 [7] 2008/09/18 15:11:32: SIP Tx udp:192.168.1.22:5060: INVITE sip:325@192.168.1.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=62862 To: "Joso Grivicic" <sip:325@localhost> Call-ID: b016bc51@pbx CSeq: 4524 INVITE Max-Forwards: 70 Contact: <sip:325@192.168.1.5:5060;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/3.0.0.2998 Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 335 v=0 o=- 23309 23309 IN IP4 192.168.1.5 s=- c=IN IP4 192.168.1.5 t=0 0 m=audio 55700 RTP/AVP 0 8 9 18 2 3 101 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 [7] 2008/09/18 15:11:32: UDP: Opening socket on port 51064 [7] 2008/09/18 15:11:32: UDP: Opening socket on port 51065 [7] 2008/09/18 15:11:32: SIP Tx udp:192.168.1.6:5060: INVITE sip:4037105450@192.168.1.6;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-1faedd9e7819005bb387250f371f51f2;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=65392 To: <sip:4037105450@192.168.1.6;user=phone> Call-ID: 5f33b786@pbx CSeq: 14343 INVITE Max-Forwards: 70 Contact: <sip:josog@192.168.1.5:5060;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/3.0.0.2998 P-Asserted-Identity: <sip:josog@192.168.1.6;user=phone> Content-Type: application/sdp Content-Length: 335 v=0 o=- 44121 44121 IN IP4 192.168.1.5 s=- c=IN IP4 192.168.1.5 t=0 0 m=audio 51064 RTP/AVP 0 8 9 18 2 3 101 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 [7] 2008/09/18 15:11:32: SIP Rx tcp:192.168.1.4:5060: SIP/2.0 100 Trying FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673 TO: "Joso Grivicic"<sip:325@localhost> CSEQ: 12820 INVITE CALL-ID: 82331c2d@pbx VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport CONTENT-LENGTH: 0 [7] 2008/09/18 15:11:32: SIP Rx udp:192.168.1.6:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-1faedd9e7819005bb387250f371f51f2;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=65392 To: <sip:4037105450@192.168.1.6;user=phone>;tag=1c2046102598 Call-ID: 5f33b786@pbx CSeq: 14343 INVITE Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004 Reason: Q.850 ;cause=3 Content-Length: 0 [7] 2008/09/18 15:11:32: Call 5f33b786@pbx#65392: Clear last INVITE [7] 2008/09/18 15:11:32: SIP Tx udp:192.168.1.6:5060: ACK sip:4037105450@192.168.1.6;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-1faedd9e7819005bb387250f371f51f2;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=65392 To: <sip:4037105450@192.168.1.6;user=phone>;tag=1c2046102598 Call-ID: 5f33b786@pbx CSeq: 14343 ACK Max-Forwards: 70 Contact: <sip:josog@192.168.1.5:5060;transport=udp> P-Asserted-Identity: <sip:josog@192.168.1.6;user=phone> Content-Length: 0 [5] 2008/09/18 15:11:32: INVITE Response: Terminate 5f33b786@pbx [7] 2008/09/18 15:11:32: Other Ports: 3 [7] 2008/09/18 15:11:32: Call Port: 201937152741200005929@192.168.1.6#ebaf4451bd [7] 2008/09/18 15:11:32: Call Port: 82331c2d@pbx#4673 [7] 2008/09/18 15:11:32: Call Port: b016bc51@pbx#62862 [7] 2008/09/18 15:11:32: SIP Rx udp:192.168.1.22:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862 To:"Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a CSeq:4524 INVITE User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504 Call-ID:b016bc51@pbx Content-Length:0 [7] 2008/09/18 15:11:32: SIP Rx udp:192.168.1.22:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862 To:"Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a CSeq:4524 INVITE User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504 Call-ID:b016bc51@pbx Content-Length:0 [7] 2008/09/18 15:11:32: SIP Rx tcp:192.168.1.4:5060: SIP/2.0 183 Session Progress FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673 TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=49601a2ec CSEQ: 12820 INVITE CALL-ID: 82331c2d@pbx VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer [7] 2008/09/18 15:11:32: SIP Rx tcp:192.168.1.4:5060: SIP/2.0 180 Ringing FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673 TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=49601a2ec CSEQ: 12820 INVITE CALL-ID: 82331c2d@pbx VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.4:5060: CANCEL sip:+4036707140325@ssi-ocsmed.jngconsulting.com;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=4673 To: "Joso Grivicic" <sip:325@localhost> Call-ID: 82331c2d@pbx CSeq: 12820 CANCEL Max-Forwards: 70 Content-Length: 0 [6] 2008/09/18 15:11:42: Redirecting to external voicemail account 325 destination sip:7325@localhost [5] 2008/09/18 15:11:42: Dialplan jngconsulting: Match 7325@localhost to <sip:325@192.168.1.2;user=phone> on trunk Exchange [5] 2008/09/18 15:11:42: Using "Subnet" <sip:4032708885@localhost;user=phone> as redirect from [7] 2008/09/18 15:11:42: SIP Tx udp:192.168.1.22:5060: CANCEL sip:325@192.168.1.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=62862 To: "Joso Grivicic" <sip:325@localhost> Call-ID: b016bc51@pbx CSeq: 4524 CANCEL Max-Forwards: 70 Content-Length: 0 [5] 2008/09/18 15:11:42: Charge user 325 for redirecting calls [7] 2008/09/18 15:11:42: UDP: Opening socket on port 57840 [7] 2008/09/18 15:11:42: UDP: Opening socket on port 57841 [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5060: INVITE sip:325@192.168.1.2;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625 To: <sip:325@192.168.1.2;user=phone> Call-ID: 26ed22a4@pbx CSeq: 19637 INVITE Max-Forwards: 70 Contact: <sip:4032708885@192.168.1.5:4649;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:325>;reason=no-answer;screen=no;privacy=off P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone> Content-Type: application/sdp Content-Length: 335 v=0 o=- 11803 11803 IN IP4 192.168.1.5 s=- c=IN IP4 192.168.1.5 t=0 0 m=audio 57840 RTP/AVP 0 8 9 18 2 3 101 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 [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5060: SIP/2.0 100 Trying FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625 TO: <sip:325@192.168.1.2;user=phone> CSEQ: 19637 INVITE CALL-ID: 26ed22a4@pbx VIA: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport CONTENT-LENGTH: 0 [7] 2008/09/18 15:11:42: SIP Rx udp:192.168.1.22:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862 To:"Joso Grivicic" <sip:325@localhost> CSeq:4524 CANCEL User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504 Call-ID:b016bc51@pbx Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY Content-Length:0 [7] 2008/09/18 15:11:42: Call b016bc51@pbx#62862: Clear last request [7] 2008/09/18 15:11:42: SIP Rx udp:192.168.1.22:5060: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862 To:"Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a CSeq:4524 INVITE User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504 Call-ID:b016bc51@pbx Content-Length:0 [7] 2008/09/18 15:11:42: Call b016bc51@pbx#62862: Clear last INVITE [7] 2008/09/18 15:11:42: SIP Tx udp:192.168.1.22:5060: ACK sip:325@192.168.1.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=62862 To: "Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a Call-ID: b016bc51@pbx CSeq: 4524 ACK Max-Forwards: 70 Contact: <sip:325@192.168.1.5:5060;transport=udp> Content-Length: 0 [5] 2008/09/18 15:11:42: INVITE Response: Terminate b016bc51@pbx [7] 2008/09/18 15:11:42: Other Ports: 3 [7] 2008/09/18 15:11:42: Call Port: 201937152741200005929@192.168.1.6#ebaf4451bd [7] 2008/09/18 15:11:42: Call Port: 26ed22a4@pbx#54625 [7] 2008/09/18 15:11:42: Call Port: 82331c2d@pbx#4673 [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.4:5060: SIP/2.0 487 Request Terminated FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673 TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=49601a2ec CSEQ: 12820 INVITE CALL-ID: 82331c2d@pbx VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer [7] 2008/09/18 15:11:42: Call 82331c2d@pbx#4673: Clear last INVITE [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.4:5060: ACK sip:+4036707140325@ssi-ocsmed.jngconsulting.com;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=4673 To: "Joso Grivicic" <sip:325@localhost>;tag=49601a2ec Call-ID: 82331c2d@pbx CSeq: 12820 ACK Max-Forwards: 70 Contact: <sip:325@192.168.1.5:4638;transport=tcp> Content-Length: 0 [5] 2008/09/18 15:11:42: INVITE Response: Terminate 82331c2d@pbx [7] 2008/09/18 15:11:42: Other Ports: 2 [7] 2008/09/18 15:11:42: Call Port: 201937152741200005929@192.168.1.6#ebaf4451bd [7] 2008/09/18 15:11:42: Call Port: 26ed22a4@pbx#54625 [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.4:5060: SIP/2.0 200 OK FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673 TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=bd82658dfc CSEQ: 12820 CANCEL CALL-ID: 82331c2d@pbx VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 MediationServer [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5060: SIP/2.0 302 Moved Temporarily FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625 TO: <sip:325@192.168.1.2;user=phone>;tag=26226ab74c CSEQ: 19637 INVITE CALL-ID: 26ed22a4@pbx VIA: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport CONTACT: <sip:325@192.168.1.2:5065;user=phone;transport=TCP> CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Diversion: <tel:325>;reason=no-answer;screen=no;privacy=off [7] 2008/09/18 15:11:42: Call 26ed22a4@pbx#54625: Clear last INVITE [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5060: ACK sip:325@192.168.1.2;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625 To: <sip:325@192.168.1.2;user=phone>;tag=26226ab74c Call-ID: 26ed22a4@pbx CSeq: 19637 ACK Max-Forwards: 70 Contact: <sip:4032708885@192.168.1.5:4649;transport=tcp> P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone> Content-Length: 0 [5] 2008/09/18 15:11:42: Redirecting call [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5065: INVITE sip:325@192.168.1.2:5065;user=phone;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625 To: <sip:325@192.168.1.2;user=phone> Call-ID: 26ed22a4@pbx CSeq: 19638 INVITE Max-Forwards: 70 Contact: <sip:4032708885@192.168.1.5:4650;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:325>;reason=no-answer;screen=no;privacy=off P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone> Content-Type: application/sdp Content-Length: 335 v=0 o=- 11803 11803 IN IP4 192.168.1.5 s=- c=IN IP4 192.168.1.5 t=0 0 m=audio 57840 RTP/AVP 0 8 9 18 2 3 101 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 [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5065: SIP/2.0 100 Trying FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625 TO: <sip:325@192.168.1.2;user=phone> CSEQ: 19638 INVITE CALL-ID: 26ed22a4@pbx VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport CONTENT-LENGTH: 0 [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5065: SIP/2.0 180 Ringing FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625 TO: <sip:325@192.168.1.2;user=phone>;epid=7EF0970BA2;tag=6b14a423e3 CSEQ: 19638 INVITE CALL-ID: 26ed22a4@pbx VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5065: SIP/2.0 200 OK FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625 TO: <sip:325@192.168.1.2;user=phone>;epid=7EF0970BA2;tag=6b14a423e3 CSEQ: 19638 INVITE CALL-ID: 26ed22a4@pbx VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport CONTACT: <sip:SSI-TESTDC.jngconsulting.com:5065;transport=Tcp;maddr=192.168.1.2>;automata CONTENT-LENGTH: 192 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 192.168.1.2 s=Microsoft Exchange Speech Engine c=IN IP4 192.168.1.2 t=0 0 m=audio 6272 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [7] 2008/09/18 15:11:42: Call 26ed22a4@pbx#54625: Clear last INVITE [7] 2008/09/18 15:11:42: Set packet length to 20 [6] 2008/09/18 15:11:42: Sending RTP for 26ed22a4@pbx#54625 to 192.168.1.2:6272 [7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5065: ACK sip:SSI-TESTDC.jngconsulting.com:5065;transport=Tcp;maddr=192.168.1.2 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9869b4983d009ddb2438347e95d0ccb0;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625 To: <sip:325@192.168.1.2;user=phone>;tag=6b14a423e3 Call-ID: 26ed22a4@pbx CSeq: 19638 ACK Max-Forwards: 70 Contact: <sip:4032708885@192.168.1.5:4650;transport=tcp> P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone> Content-Length: 0 [7] 2008/09/18 15:11:42: Determine pass-through mode after receiving response [7] 2008/09/18 15:11:42: 26ed22a4@pbx#54625: RTP pass-through mode [7] 2008/09/18 15:11:42: 201937152741200005929@192.168.1.6#ebaf4451bd: RTP pass-through mode [7] 2008/09/18 15:12:02: SIP Rx udp:192.168.1.6:5060: BYE sip:josog@192.168.1.5:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2122852159 Max-Forwards: 70 From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919 To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd Call-ID: 201937152741200005929@192.168.1.6 CSeq: 2 BYE Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004 Reason: Q.850 ;cause=31 ;text="RTP Broken Connection" Content-Length: 0 [7] 2008/09/18 15:12:02: SIP Tx udp:192.168.1.6:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2122852159 From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919 To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd Call-ID: 201937152741200005929@192.168.1.6 CSeq: 2 BYE Contact: <sip:josog@192.168.1.5:5060;transport=udp> User-Agent: pbxnsip-PBX/3.0.0.2998 RTP-RxStat: Dur=41,Pkt=2064,Oct=350952,Underun=0 RTP-TxStat: Dur=41,Pkt=1545,Oct=265740 Content-Length: 0 [7] 2008/09/18 15:12:02: 26ed22a4@pbx#54625: Media-aware pass-through mode [7] 2008/09/18 15:12:02: Other Ports: 1 [7] 2008/09/18 15:12:02: Call Port: 26ed22a4@pbx#54625 [7] 2008/09/18 15:12:02: SIP Tx tcp:192.168.1.2:5065: BYE sip:SSI-TESTDC.jngconsulting.com:5065;transport=Tcp;maddr=192.168.1.2 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-b02d89319eec09d9258589db2e1c8e79;rport From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625 To: <sip:325@192.168.1.2;user=phone>;tag=6b14a423e3 Call-ID: 26ed22a4@pbx CSeq: 19639 BYE Max-Forwards: 70 Contact: <sip:4032708885@192.168.1.5:4650;transport=tcp> RTP-RxStat: Dur=21,Pkt=495,Oct=85140,Underun=0 RTP-TxStat: Dur=20,Pkt=1017,Oct=174924 P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone> Content-Length: 0 [7] 2008/09/18 15:12:02: SIP Rx tcp:192.168.1.2:5065: SIP/2.0 200 OK FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625 TO: <sip:325@192.168.1.2;user=phone>;tag=6b14a423e3;epid=7EF0970BA2 CSEQ: 19639 BYE CALL-ID: 26ed22a4@pbx VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-b02d89319eec09d9258589db2e1c8e79;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [7] 2008/09/18 15:12:02: Call 26ed22a4@pbx#54625: Clear last request [5] 2008/09/18 15:12:02: BYE Response: Terminate 26ed22a4@pbx [6] 2008/09/18 15:12:11: SIP TCP/TLS timeout on 192.168.1.2:42354, closing connection [5] 2008/09/18 15:13:12: SIP port accept from 192.168.1.2:30667 [6] 2008/09/18 15:14:11: SIP TCP/TLS timeout on 192.168.1.2:42365, closing connection [5] 2008/09/18 15:15:12: SIP port accept from 192.168.1.2:30674 Quote Link to comment Share on other sites 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Vodia PBX Posted September 18, 2008 Report Share Posted September 18, 2008 I only have one firewall and ip at the moment. I have the log file, not sure if it will help: Also, what does this mean "SIP TCP/TLS timeout on 192.168.1.2:42354, closing connection " Well, that timeout happens after the call is already over. The gateway initiates the hangup, and it says it does that because of RTP problems. You can see in the BYE response that the PBX sends far less packets than it receives. So it is understandable why the gateway hangs up. Maybe you have silence suppression turned on? With media related problems, it makes sense to get a Wireshark trace. Then we will be able to see why the PBX does not send RTP data. Maybe it is because of the Speech Engine, maybe there is a option to disable silence suppression. Also, check if there is a upgrade availale. 3.0.0 sounds like there is... Quote Link to comment Share on other sites More sharing options...
joso Posted September 19, 2008 Author Report Share Posted September 19, 2008 I think i found the problem. In the audiocodes mp-118 under coders. I had it set at G.711U-Law, when i changed this to G.711A-Law it works fine. Silence SUpression has always been disabled. What setting should this be set at, since there are many to choose from. Joso Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 19, 2008 Report Share Posted September 19, 2008 I think i found the problem. In the audiocodes mp-118 under coders. I had it set at G.711U-Law, when i changed this to G.711A-Law it works fine. Silence SUpression has always been disabled. What setting should this be set at, since there are many to choose from. Whow, that means the AC has a problem with the codec negotiation? The PBX answers with it's own priority (which is obviously Ulaw, then Alaw), and maybe the AC cannot deal with that... Anyway, if it works then keep it this way. Quote Link to comment Share on other sites More sharing options...
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