joso

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

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?

is there a link or instructions of some sort on how to set one up non pnp.

Hi there, i have been testing out pbxnsip with Exchange UM and OCS, and things are going pretty good. I want to be able to test out dual forking with the use of office communicator and a desk phone. I currently only have a mitel 5220 SIP Phone. Can this phone work with pbxnsip? How do i go about this. Thanks Joso