TomH

I am on 2.1.8 and you are correct it keeps the entry. The text "Device Expires Reboot" through me.

Passthrough log records when it does not work [7] 2008/04/30 08:30:00: Determine pass-through mode after receiving response [7] 2008/04/30 08:30:00: 62be9fb0@pbx#34346: RTP pass-through mode [7] 2008/04/30 08:30:00: 8609-3418547751-802139@NXT01.broadvox.net#945f1258df: RTP pass-through mode [7] 2008/04/30 08:30:02: Determine pass-through mode after receiving response [8] 2008/04/30 08:30:02: Passthrough: Changing destination to 172.26.1.81:8545 Passthrough log records when it Works [7] 2008/04/30 08:39:19: Determine pass-through mode after receiving response [7] 2008/04/30 08:39:19: 3297304d@pbx#64201: RTP pass-through mode [7] 2008/04/30 08:39:19: ac205a6825c71d48@192.168.6.102#74ebbabf04: RTP pass-through mode [7] 2008/04/30 08:39:21: Determine pass-through mode after receiving response [7] 2008/04/30 08:39:44: ac205a6825c71d48@192.168.6.102#74ebbabf04: Media-aware pass-through mode **************When it does not work you retrieve the RPT ports twice************ [7] 2008/04/30 08:30:02: SIP Rx tcp:172.26.1.81:5067: INVITE sip:103@172.26.1.75:2594;transport=tcp SIP/2.0 FROM: <sip:103@be7.ezoutlook.com;user=phone>;epid=BE122FA941;tag=887c2779c4 TO: <sip:103@ezoutlook.com>;tag=34346 CSEQ: 1 INVITE CALL-ID: 62be9fb0@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.26.1.81:5067;branch=z9hG4bKb7c3d736 CONTACT: <sip:be7.ezoutlook.com:5067;transport=Tcp;maddr=172.26.1.81;ms-opaque=0d02c8b9260b7733>;automata CONTENT-LENGTH: 283 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp v=0 o=- 0 1 IN IP4 172.26.1.81 s=session c=IN IP4 172.26.1.81 t=0 0 m=audio 0 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=image 8545 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy ****************first time************************* [7] 2008/04/30 08:30:02: UDP: Opening socket on port 9098 [7] 2008/04/30 08:30:02: UDP: Opening socket on port 9050 . . . [7] 2008/04/30 08:30:02: SIP Rx udp:209.249.3.59:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.26.1.75:5060;branch=z9hG4bK-605578d4313fe553d739d96e6861ef95;rport To: "EZ OUTLOOK WEB " <sip:5024255328@209.249.3.59>;tag=3418547751-802148 From: <sip:10000555024102925@209.249.3.56:5060>;tag=945f1258df Call-ID: 8609-3418547751-802139@NXT01.broadvox.net CSeq: 23081 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: <sip:5024255328@209.249.3.59:5060> Call-Info: <sip:209.249.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 315 v=0 o=NXT01 0 1 IN IP4 209.249.3.59 s=sip call c=IN IP4 209.249.3.60 t=0 0 m=audio 44164 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 m=image 44170 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv [7] 2008/04/30 08:30:02: Call 8609-3418547751-802139@NXT01.broadvox.net#945f1258df: Clear last INVITE ****************second time************************* [7] 2008/04/30 08:30:02: UDP: Opening socket on port 9006 [7] 2008/04/30 08:30:02: UDP: Opening socket on port 9064 When it does work you get them once [7] 2008/04/30 08:39:19: SIP Rx tcp:172.26.1.81:5067: INVITE sip:103@172.26.1.75:2594;transport=tcp SIP/2.0 FROM: <sip:103@be7.ezoutlook.com;user=phone>;epid=BE122FA941;tag=7f988c487 TO: <sip:103@ezoutlook.com>;tag=64201 CSEQ: 1 INVITE CALL-ID: 3297304d@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.26.1.81:5067;branch=z9hG4bK44a2a66f CONTACT: <sip:be7.ezoutlook.com:5067;transport=Tcp;maddr=172.26.1.81;ms-opaque=0d02c8b9260b7733>;automata CONTENT-LENGTH: 283 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp v=0 o=- 0 1 IN IP4 172.26.1.81 s=session c=IN IP4 172.26.1.81 t=0 0 m=audio 0 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=image 8630 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy ****************first time and only time************************* [7] 2008/04/30 08:39:19: UDP: Opening socket on port 9080 [7] 2008/04/30 08:39:19: UDP: Opening socket on port 9012 . . . [7] 2008/04/30 08:39:21: SIP Rx udp:74.143.31.154:14062: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.26.1.75:5060;branch=z9hG4bK-3b7e804129e942888eb09361ddde3b82;rport From: <sip:103@pbx.ezoutlook.com>;tag=74ebbabf04 To: "Fax Machine" <sip:105@pbx.ezoutlook.com>;tag=a68eb379542a755d Call-ID: ac205a6825c71d48@192.168.6.102 CSeq: 28130 INVITE User-Agent: Grandstream HT287 1.1.0.3 Warning: 399 74.143.31.154 "detected NAT type is full cone" Contact: <sip:105@74.143.31.154:14062> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Supported: replaces, timer Content-Length: 249 v=0 o=105 8000 8004 IN IP4 74.143.31.154 s=SIP Call c=IN IP4 74.143.31.154 t=0 0 m=audio 0 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 m=image 50168 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy a=ptime:20 [7] 2008/04/30 08:39:21: Call ac205a6825c71d48@192.168.6.102#74ebbabf04: Clear last INVITE *************************you use the original RTP port pair******************** [9] 2008/04/30 08:39:21: Resolve 7467: tcp 172.26.1.81 5067 [7] 2008/04/30 08:39:21: SIP Tx tcp:172.26.1.81:5067:

I agree the Bind is fine. In a test situation with plenty of ports. The issue is on the faxes that do not work PBXnSIP retrieves a pair of open ports for RTP/T38 twice. Once correctly and once after after the recieved OK from BroadVox which invalidates the port number originally given to BroadVox. In the example I gave you: [7] 2008/04/23 13:42:28: UDP: Opening socket on port 9048 [7] 2008/04/23 13:42:28: UDP: Opening socket on port 9008 is the original pair and 9008 was given to BroadVox Then you retrieved a second pair this time [7] 2008/04/23 13:42:29: UDP: Opening socket on port 9064 [7] 2008/04/23 13:42:29: UDP: Opening socket on port 9030 and used 9064 to go to broadvox. i.e. all goes in the bit bucket. When you dont retrieve the secnd pair it always works. The question is why do you retrieve a second pair sometimes and not others. You are consistantly doing the same thing between devices. If its going for device A to device B it always has the problem or always works depending if you retrieve the second pair. I am guessing that the issue may be PBXnSIP: IF IT WORKS - returns from end to end understanding it is a return and remembers and uses the original pair. understands FAX > PBXnSIP > BroadVox > PBXnSIP IF IT FAILS - returns from end to end does not understanding it is a return and does not remember and get a new pair pair. thinks BroadVox > PBXnSIP Totally a guess though. Thanks Tom

Configuration Exchange inbound fax <> PBXnSIP <> BroadVox HT286 Fax Kapaga fax phone Issue After review we find that we consistantly work or don't work. The difference being that the inbound fax device changes to T38 with a reinvite using PORT 9001, PBXnSIP gets two ports for RTP (10001, 10002) tells Broadvox he will use port 10001, BroadVox says he will use port 20002 in an OK, If it works PBXnSIP OK's the client invite and uses 10002 so that fax to pbxnsip RTP is 9001 <> 10002 PBXnSIP to broadvox RTP is 10001 <> 20002 If it does not work PBXnSIP gets two new ports (10003, 10004) and OK's the client invite with 10003 so that: fax to pbxnsip RTP is 9001 <> 10003 PBXnSIP to broadvox RTP is 10004 <> 20002 The issue is I believe that a state/route has been setup on Broadvox linking 10001 <> 20002 when messages come in 10004 <> 20002 they go in the bit bucket. My questions is why is there an inconsistancy? Some times you use the original ports (this works), and sometimes you get new ports (this fails). Thanks for any guidance. I have included and commented the relevant part of the log. Tom Invite from fax client requesting T38 [7] 2008/04/23 13:42:28: SIP Rx tcp:172.1.1.81:5065: INVITE sip:103@172.1.1.75:4957;transport=tcp SIP/2.0 FROM: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1 TO: <sip:103@ezoutlook.com>;tag=5392 CSEQ: 1 INVITE CALL-ID: 46269039@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK12926ec CONTACT: <sip:be7.mydomain.com:5065;transport=Tcp;maddr=172.1.1.81;ms-opaque=2eb6a8402dbb9419>;automata CONTENT-LENGTH: 284 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp v=0 o=- 0 1 IN IP4 172.1.1.81 s=session c=IN IP4 172.1.1.81 t=0 0 m=audio 0 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=image 16861 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy [0] 2008/04/23 13:42:28: UDP: bind() to port 9066 failed Gets two ports, above bind failure does not always happen [7] 2008/04/23 13:42:28: UDP: Opening socket on port 9048 [7] 2008/04/23 13:42:28: UDP: Opening socket on port 9008 [9] 2008/04/23 13:42:28: Resolve 72699: url sip:5025555555@200.200.3.59:5060 [9] 2008/04/23 13:42:28: Resolve 72699: udp 200.200.3.59 5060 Invites broadvox [7] 2008/04/23 13:42:28: SIP Tx udp:200.200.3.59:5060: INVITE sip:5025555555@200.200.3.59:5060 SIP/2.0 Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-48ad4751e9b09902de83fe8ca58bdf61;rport From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334 To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590 Call-ID: 1560-3417961711-711581@NXT01.broadvox.net CSeq: 13831 INVITE Max-Forwards: 70 Contact: <sip:Anonymous@172.1.1.75:5060;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/2.1.8.2463 Content-Type: application/sdp Content-Length: 283 v=0 o=- 17827 17828 IN IP4 172.1.1.75 s=- c=IN IP4 172.1.1.75 t=0 0 m=audio 0 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=image 9008 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy [9] 2008/04/23 13:42:28: Resolve 72700: tcp 172.1.1.81 5065 [7] 2008/04/23 13:42:28: SIP Tx tcp:172.1.1.81:5065: SIP/2.0 100 Trying Via: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK12926ec From: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1 To: <sip:103@ezoutlook.com>;tag=5392 Call-ID: 46269039@pbx CSeq: 1 INVITE Content-Length: 0 [7] 2008/04/23 13:42:29: SIP Rx udp:200.200.3.59:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-48ad4751e9b09902de83fe8ca58bdf61;rport From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334 To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590 Call-ID: 1560-3417961711-711581@NXT01.broadvox.net CSeq: 13831 INVITE Content-Length: 0 Broadvox OK [7] 2008/04/23 13:42:29: SIP Rx udp:200.200.3.59:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-48ad4751e9b09902de83fe8ca58bdf61;rport To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590 From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334 Call-ID: 1560-3417961711-711581@NXT01.broadvox.net CSeq: 13831 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: <sip:5025555555@200.200.3.59:5060> Call-Info: <sip:200.200.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 315 v=0 o=NXT01 0 1 IN IP4 200.200.3.59 s=sip call c=IN IP4 200.200.3.60 t=0 0 m=audio 16222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 m=image 16228 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv [7] 2008/04/23 13:42:29: Call 1560-3417961711-711581@NXT01.broadvox.net#bfe14f6334: Clear last INVITE HERE IS THE ISSUE IF YOU GET TWO NEW PORTS IT FAILS IF YOU KEEP THE ORIGINALS IT WORKS [7] 2008/04/23 13:42:29: UDP: Opening socket on port 9064 [7] 2008/04/23 13:42:29: UDP: Opening socket on port 9030 Now you are using a port 9064 for RTP to Broadvox instead of the stated port 9008 [9] 2008/04/23 13:42:29: Resolve 72701: tcp 172.1.1.81 5065 [7] 2008/04/23 13:42:29: SIP Tx tcp:172.1.1.81:5065: SIP/2.0 200 Ok Via: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK12926ec From: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1 To: <sip:103@ezoutlook.com>;tag=5392 Call-ID: 46269039@pbx CSeq: 1 INVITE Contact: <sip:103@172.1.1.75:4957;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.8.2463 Content-Type: application/sdp Content-Length: 282 v=0 o=- 1981 1982 IN IP4 172.1.1.75 s=- c=IN IP4 172.1.1.75 t=0 0 m=audio 9048 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 9030 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy [9] 2008/04/23 13:42:29: Resolve 72702: url sip:5025555555@200.200.3.59:5060 [9] 2008/04/23 13:42:29: Resolve 72702: udp 200.200.3.59 5060 [7] 2008/04/23 13:42:29: SIP Tx udp:200.200.3.59:5060: ACK sip:5025555555@200.200.3.59:5060 SIP/2.0 Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-41742d16b815e8fef8a65e845f1a72cc;rport From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334 To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590 Call-ID: 1560-3417961711-711581@NXT01.broadvox.net CSeq: 13831 ACK Max-Forwards: 70 Contact: <sip:Anonymous@172.1.1.75:5060;transport=udp> Content-Length: 0 [7] 2008/04/23 13:42:29: Determine pass-through mode after receiving response [7] 2008/04/23 13:42:29: SIP Rx tcp:172.1.1.81:5065: ACK sip:103@172.1.1.75:4957;transport=tcp SIP/2.0 FROM: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1 TO: <sip:103@ezoutlook.com>;tag=5392 CSEQ: 1 ACK CALL-ID: 46269039@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK28915536 CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 [8] 2008/04/23 13:42:29: Passthrough: Changing destination to 172.1.1.81:16861 [5] 2008/04/23 13:43:08: SIP port accept from 172.1.1.81:16868 [7] 2008/04/23 13:43:08: SIP Rx tcp:172.1.1.81:16868: OPTIONS sip:172.1.1.75:5060 SIP/2.0 FROM: <sip:be7.mydomain.com:5060;transport=Tcp;ms-opaque=932002b23e9bfaae>;epid=6618ACCBF4;tag=b56e3cb8e9 TO: <sip:172.1.1.75:5060> CSEQ: 6449 OPTIONS CALL-ID: d6996cf8fab742ee91d089bf12794b6b MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.1.1.81:16868;branch=z9hG4bK13c7d83c ACCEPT: application/sdp CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 [9] 2008/04/23 13:43:08: Resolve 72703: tcp 172.1.1.81 16868 [7] 2008/04/23 13:43:08: SIP Tx tcp:172.1.1.81:16868: SIP/2.0 200 Ok Via: SIP/2.0/TCP 172.1.1.81:16868;branch=z9hG4bK13c7d83c From: <sip:be7.mydomain.com:5060;transport=Tcp;ms-opaque=932002b23e9bfaae>;epid=6618ACCBF4;tag=b56e3cb8e9 To: <sip:172.1.1.75:5060>;tag=7905a6d1f3 Call-ID: d6996cf8fab742ee91d089bf12794b6b CSeq: 6449 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [7] 2008/04/23 13:43:15: SIP Rx tcp:172.1.1.81:5065: BYE sip:103@172.1.1.75:4957;transport=tcp SIP/2.0 FROM: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1 TO: <sip:103@ezoutlook.com>;tag=5392 CSEQ: 2 BYE CALL-ID: 46269039@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK1643ff5 CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 [9] 2008/04/23 13:43:15: Resolve 72704: tcp 172.1.1.81 5065 [7] 2008/04/23 13:43:15: SIP Tx tcp:172.1.1.81:5065: SIP/2.0 200 Ok Via: SIP/2.0/TCP 172.1.1.81:5065;branch=z9hG4bK1643ff5 From: <sip:103@be7.mydomain.com;user=phone>;epid=34AEC95009;tag=54acd35ba1 To: <sip:103@ezoutlook.com>;tag=5392 Call-ID: 46269039@pbx CSeq: 2 BYE Contact: <sip:103@172.1.1.75:4957;transport=tcp> User-Agent: pbxnsip-PBX/2.1.8.2463 RTP-RxStat: Dur=49,Pkt=129,Oct=11466,Underun=0 RTP-TxStat: Dur=49,Pkt=62,Oct=10664 Content-Length: 0 [7] 2008/04/23 13:43:15: 1560-3417961711-711581@NXT01.broadvox.net#bfe14f6334: Media-aware pass-through mode [7] 2008/04/23 13:43:15: Other Ports: 1 [7] 2008/04/23 13:43:15: Call Port: 1560-3417961711-711581@NXT01.broadvox.net#bfe14f6334 [9] 2008/04/23 13:43:15: Resolve 72705: url sip:5025555555@200.200.3.59:5060 [9] 2008/04/23 13:43:15: Resolve 72705: udp 200.200.3.59 5060 [7] 2008/04/23 13:43:15: SIP Tx udp:200.200.3.59:5060: BYE sip:5025555555@200.200.3.59:5060 SIP/2.0 Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-9965f8d5a9dd9483b8b0fadae730f1ec;rport From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334 To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590 Call-ID: 1560-3417961711-711581@NXT01.broadvox.net CSeq: 13832 BYE Max-Forwards: 70 Contact: <sip:Anonymous@172.1.1.75:5060;transport=udp> RTP-RxStat: Dur=49,Pkt=89,Oct=15308,Underun=0 RTP-TxStat: Dur=47,Pkt=170,Oct=18518 Content-Length: 0 [7] 2008/04/23 13:43:16: SIP Rx udp:200.200.3.59:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.1.1.75:5060;branch=z9hG4bK-9965f8d5a9dd9483b8b0fadae730f1ec;rport To: "EZ OUTLOOK WEB " <sip:5025555555@200.200.3.59>;tag=3417961711-711590 From: <sip:10000555024444444@200.200.3.56:5060>;tag=bfe14f6334 Call-ID: 1560-3417961711-711581@NXT01.broadvox.net CSeq: 13832 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: <sip:5025555555@200.200.3.59:5060> Content-Length: 0

Here is where it stands Broadvox is not going to fix the issue with multiple m records in the sdp any time soon. I need to move on. I am going to look for a SIP trunk vendor that does support this. At the same time does PBXnSIP have the abilility to recieve and forward inbound faxes? this is another alternative for us.

I also found this it appears CISCO call manager is having the same issue can you help? http://forums.microsoft.com/TechNet/ShowPo...1&SiteID=17 Following is exerpt from microsoft forum. "We've been down this path, and unfortunately were not able to reach any sort of solution via CUCM 6.01 & Ex07 SP1 (regardless of inband detection enabled). Neither or Cisco or Microsoft are really able to help on this unless they come to some sort of agreement. The details are basically as follows: Microsoft and Cisco use two different standards for handling t.38 calls over SIP. The SIP signaling used by Microsoft Exchange to switch a call from audio to T.38 fax is not understood by CallManager, causing fax calls to fail. Specifically, Exchange sends SDP with two "m=" lines, one to terminate audio and one to enable fax session. However, CallManager interprets it as audio requests that terminate the audio channels. Currently CallManager only supports send/receive SIP INVITE signal to switch an audio call to T.38 call by a single image m=line in SIP SDP portion. This SDP signals the endpoint to replace the existing channels (audio in our case) and establish a new channel for T.38. Both implementations by Cisco CallManager and Microsoft Exchange conform to the standards (RFC3264 (MS) and ITU T.38 (Cisco)). Support for both standards is in development for CallManager, This functionality will be fully supported in CallManager 7.0. It is currently in a 'resolved' state meaning this functionality is tested and confirmed in development builds of 7.0. This is tracked via BugID CSCsg60357, but was private the last I checked. Hope this helps. If anyone can use this information to get this functional, I'd love to see it."

Well, just heard back from BroadVox. They are saying that the issue is that there are two codex in the SDP: m=audio 0 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=image 9200 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPFEC I need to get rid of the audio which does have port 0. I know you all are just passing this through from exchange but can you offer any help on correcting it on exchange or correcting it in the pass through. I believe they are correct and if I can remove it the fax will pass through correctly. Also, are there other PBXNSIP users sucessfully doing inbound faxing. If so maybe I need to reinstall. Tom

BroadVox has two questions. 1) see below we recieve an "INVITE" from exchange but passs it through to broadvox as a "UPDATE" can it be passed through as an "INVITE"? 2) and more important can we limit the protocols to just T38 so there is no choise? Thanks Receives INVITE from 172.26.1.81 (Exchange) to go to T38 [7] 2008/02/28 16:45:39: SIP Rx tcp:172.26.1.81:5065: INVITE sip:103@172.26.1.75:3006;transport=tcp SIP/2.0 FROM: <sip:103@be7.ezoutlook.com;user=phone>;epid=AA79D0A609;tag=a9fab924e2 TO: <sip:103@ezoutlook.com>;tag=59403 CSEQ: 1 INVITE CALL-ID: a0b0509d@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.26.1.81:5065;branch=z9hG4bK2dfacb75 CONTACT: <sip:be7.ezoutlook.com:5065;transport=Tcp;maddr=172.26.1.81;ms-opaque=c9e23a1203e9a49b>;automata CONTENT-LENGTH: 276 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp v=0 o=- 0 1 IN IP4 172.26.1.81 s=session c=IN IP4 172.26.1.81 t=0 0 m=audio 0 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=image 9200 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPFEC Sends UPDATE to BroadVox [7] 2008/02/28 16:45:39: SIP Tx udp:64.152.60.75:5060: UPDATE sip:2163736227@64.152.60.75:5060 SIP/2.0 Via: SIP/2.0/UDP 172.26.1.75:5060;branch=z9hG4bK-7dcdaaf32ca71adac0facc4140d0d153;rport From: <sip:5024102925@172.26.1.75>;tag=c98233afea To: "JERGENS INC " <sip:2163736227@64.152.60.75>;tag=gK0229cd68 Call-ID: 1661104583_128358926@64.152.60.75 CSeq: 2771 UPDATE Max-Forwards: 70 Contact: <sip:Anonymous@172.26.1.75:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Content-Type: application/sdp Content-Length: 276 v=0 o=- 43466 43467 IN IP4 172.26.1.75 s=- c=IN IP4 172.26.1.75 t=0 0 m=audio 0 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=image 9098 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPFEC

Yes, I provided them the same info and will let you know what they say. From your standpoint this is something that should work and the issues appears to be on their side?

My issue is T.38 pass through to Exchange 2007. Any help on missing pieces, trouble shooting, instructions, or examples would be appreciated. Thanks to anyone for any help. The flow is as follows Fax Machine >> Broadvox >> PBXnSIP >> Exchange 2007 Some facts are: • If called on a telephone line pbxnsip rings the phone and then passes the call on to exchange which answers and forward to exchange voice mail if no answer. Everything works great. • Exchange is 172.x.x.81 • PBXnSIP is 172.x.x.75 When a fax machine rings: *PBXnSIP rings phones INVITE sip:103@be7.domain.com;user=phone SIP/2.0 . . . *PBXnSIP Transfers to Exchange [7] 2008/02/24 16:57:47: SIP Tx tcp:172.x.x.81:5065: INVITE sip:103@be7.domain.com:5065;user=phone;transport=TCP SIP/2.0 . . . *Exchange pickups *Exchange recognizes fax *Exchange transitions to T38 [7] 2008/02/24 16:57:58: SIP Rx tcp:172.x.x.81:5065: INVITE sip:103@172.x.x.75:3444;transport=tcp SIP/2.0 FROM: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832 TO: <sip:103@domain.com>;tag=20019 . . m=image 9200 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPFEC *PBXnSIP transition to T38 [7] 2008/02/24 16:57:58: SIP Tx udp:64.152.60.75:5060: UPDATE sip:5552907492@64.152.60.75:5060 SIP/2.0 Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 . . m=image 9086 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPFEC *BroadVox transitions to T38 [7] 2008/02/24 16:57:58: SIP Rx udp:64.152.60.75:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport=5060 From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 Call-ID: 1459988973_58241008@64.152.60.75 . . m=image 0 udptl t38 a=T38FaxRateManagement:transferredTCF a= T38FaxUdpEC:t38UDPFEC a=sendrecv Issues occurs now, it appear when Broadvox starts pinging port 0 instead of port given. In the m parameter of the SDP protocol I see a zero instead of a valid port# Could this be the issue. Thanks again. Full PBXnSIP Log follow: [9] 2008/02/24 16:57:47: Resolve 1975: aaaa udp 64.152.60.75 5060 [9] 2008/02/24 16:57:47: Resolve 1975: a udp 64.152.60.75 5060 [9] 2008/02/24 16:57:47: Resolve 1975: udp 64.152.60.75 5060 [7] 2008/02/24 16:57:47: SIP Tx udp:64.152.60.75:5060: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B04d42fb1611ec6f9 From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 Call-ID: 1459988973_58241008@64.152.60.75 CSeq: 18297 INVITE Contact: <sip:Anonymous@172.x.x.75:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 201 v=0 o=- 51427 51427 IN IP4 172.x.x.75 s=- c=IN IP4 172.x.x.75 t=0 0 m=audio 9036 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [8] 2008/02/24 16:57:47: DNS: Add dns_a be7.domain.com 172.x.x.81 (ttl=1200) [9] 2008/02/24 16:57:47: Resolve 1974: a tcp be7.domain.com 5060 [9] 2008/02/24 16:57:47: Resolve 1974: tcp 172.x.x.81 5060 [7] 2008/02/24 16:57:47: SIP Tx tcp:172.x.x.81:5060: INVITE sip:103@be7.domain.com;user=phone SIP/2.0 Via: SIP/2.0/TCP 172.x.x.75:3443;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport From: "Tom Haselden" <sip:103@domain.com>;tag=20019 To: <sip:103@be7.domain.com;user=phone> Call-ID: 708d4bca@pbx CSeq: 8815 INVITE Max-Forwards: 70 Contact: <sip:103@172.x.x.75:3443;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Diversion: <tel:103>;reason=no-answer;screen=no;privacy=off Content-Type: application/sdp Content-Length: 287 v=0 o=- 62722 62722 IN IP4 172.x.x.75 s=- c=IN IP4 172.x.x.75 t=0 0 m=audio 9042 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] 2008/02/24 16:57:47: SIP Rx udp:64.152.60.75:5060: PRACK sip:Anonymous@216.x.x.75:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05037402bed8a5c3 From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 Call-ID: 1459988973_58241008@64.152.60.75 CSeq: 18298 PRACK Max-Forwards: 70 RAck: 1 18297 INVITE Content-Length: 0 [9] 2008/02/24 16:57:47: Resolve 1976: aaaa udp 64.152.60.75 5060 [9] 2008/02/24 16:57:47: Resolve 1976: a udp 64.152.60.75 5060 [9] 2008/02/24 16:57:47: Resolve 1976: udp 64.152.60.75 5060 [7] 2008/02/24 16:57:47: SIP Tx udp:64.152.60.75:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05037402bed8a5c3 From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 Call-ID: 1459988973_58241008@64.152.60.75 CSeq: 18298 PRACK Contact: <sip:Anonymous@172.x.x.75:5060;transport=udp> User-Agent: pbxnsip-PBX/2.1.6.2448 Content-Length: 0 [7] 2008/02/24 16:57:47: SIP Rx tcp:172.x.x.81:5060: SIP/2.0 100 Trying FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019 TO: <sip:103@be7.domain.com;user=phone> CSEQ: 8815 INVITE CALL-ID: 708d4bca@pbx VIA: SIP/2.0/TCP 172.x.x.75:3443;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport CONTENT-LENGTH: 0 [7] 2008/02/24 16:57:47: SIP Rx tcp:172.x.x.81:5060: SIP/2.0 302 Moved Temporarily FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019 TO: <sip:103@be7.domain.com;user=phone>;tag=60116e5511 CSEQ: 8815 INVITE CALL-ID: 708d4bca@pbx VIA: SIP/2.0/TCP 172.x.x.75:3443;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport CONTACT: <sip:103@be7.domain.com:5065;user=phone;transport=TCP> CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 Diversion: <tel:103>;reason=no-answer;screen=no;privacy=off [7] 2008/02/24 16:57:47: Call 708d4bca@pbx#20019: Clear last INVITE [9] 2008/02/24 16:57:47: Resolve 1977: url sip:103@be7.domain.com;user=phone [9] 2008/02/24 16:57:47: Resolve 1977: naptr be7.domain.com [5] 2008/02/24 16:57:47: Redirecting call [9] 2008/02/24 16:57:47: Resolve 1978: url sip:103@be7.domain.com:5065;user=phone;transport=TCP [9] 2008/02/24 16:57:47: Resolve 1978: a tcp be7.domain.com 5065 [9] 2008/02/24 16:57:47: Resolve 1978: tcp 172.x.x.81 5065 [7] 2008/02/24 16:57:47: SIP Tx tcp:172.x.x.81:5065: INVITE sip:103@be7.domain.com:5065;user=phone;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport From: "Tom Haselden" <sip:103@domain.com>;tag=20019 To: <sip:103@be7.domain.com;user=phone> Call-ID: 708d4bca@pbx CSeq: 8816 INVITE Max-Forwards: 70 Contact: <sip:103@172.x.x.75:3444;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Diversion: <tel:103>;reason=no-answer;screen=no;privacy=off Content-Type: application/sdp Content-Length: 287 v=0 o=- 62722 62722 IN IP4 172.x.x.75 s=- c=IN IP4 172.x.x.75 t=0 0 m=audio 9042 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 [8] 2008/02/24 16:57:47: DNS: Add dns_naptr be7.domain.com (ttl=3600) [9] 2008/02/24 16:57:47: Resolve 1977: naptr be7.domain.com [9] 2008/02/24 16:57:47: Resolve 1977: srv tls _sips._tcp.be7.domain.com [8] 2008/02/24 16:57:47: DNS: Add dns_srv _sips._tcp.be7.domain.com (ttl=3600) [9] 2008/02/24 16:57:47: Resolve 1977: srv tls _sips._tcp.be7.domain.com [9] 2008/02/24 16:57:47: Resolve 1977: srv tcp _sip._tcp.be7.domain.com [8] 2008/02/24 16:57:47: DNS: Add dns_srv _sip._tcp.be7.domain.com (ttl=3600) [9] 2008/02/24 16:57:47: Resolve 1977: srv tcp _sip._tcp.be7.domain.com [9] 2008/02/24 16:57:47: Resolve 1977: srv udp _sip._udp.be7.domain.com [8] 2008/02/24 16:57:47: DNS: Add dns_srv _sip._udp.be7.domain.com (ttl=3600) [9] 2008/02/24 16:57:47: Resolve 1977: srv udp _sip._udp.be7.domain.com [9] 2008/02/24 16:57:47: Resolve 1977: a udp be7.domain.com 5060 [9] 2008/02/24 16:57:47: Resolve 1977: udp 172.x.x.81 5060 [7] 2008/02/24 16:57:47: SIP Tx udp:172.x.x.81:5060: ACK sip:103@be7.domain.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport From: "Tom Haselden" <sip:103@domain.com>;tag=20019 To: <sip:103@be7.domain.com;user=phone>;tag=60116e5511 Call-ID: 708d4bca@pbx CSeq: 8815 ACK Max-Forwards: 70 Contact: <sip:103@172.x.x.75:5060;transport=udp> Content-Length: 0 [8] 2008/02/24 16:57:47: UDP: recvfrom receives ICMP message [5] 2008/02/24 16:57:47: Connection refused on udp:172.x.x.81:5060 [6] 2008/02/24 16:57:47: Could not determine destination address on 1977 [7] 2008/02/24 16:57:47: SIP Rx tcp:172.x.x.81:5065: SIP/2.0 100 Trying FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019 TO: <sip:103@be7.domain.com;user=phone> CSEQ: 8816 INVITE CALL-ID: 708d4bca@pbx VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport CONTENT-LENGTH: 0 [7] 2008/02/24 16:57:49: SIP Rx tcp:172.x.x.81:5065: SIP/2.0 180 Ringing FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019 TO: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832 CSEQ: 8816 INVITE CALL-ID: 708d4bca@pbx VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [8] 2008/02/24 16:57:49: Play audio_en/ringback.wav [7] 2008/02/24 16:57:52: SIP Rx tcp:172.x.x.81:5065: SIP/2.0 200 OK FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019 TO: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832 CSEQ: 8816 INVITE CALL-ID: 708d4bca@pbx VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport CONTACT: <sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81>;automata CONTENT-LENGTH: 193 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 172.x.x.81 s=Microsoft Exchange Speech Engine c=IN IP4 172.x.x.81 t=0 0 m=audio 35840 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [7] 2008/02/24 16:57:52: Call 708d4bca@pbx#20019: Clear last INVITE [7] 2008/02/24 16:57:52: Set packet length to 20 [6] 2008/02/24 16:57:52: Sending RTP for 708d4bca@pbx#20019 to 172.x.x.81:35840 [9] 2008/02/24 16:57:52: Resolve 1979: aaaa tcp 172.x.x.81 5065 [9] 2008/02/24 16:57:52: Resolve 1979: a tcp 172.x.x.81 5065 [9] 2008/02/24 16:57:52: Resolve 1979: tcp 172.x.x.81 5065 [7] 2008/02/24 16:57:52: SIP Tx tcp:172.x.x.81:5065: ACK sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81 SIP/2.0 Via: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-abf7641b93baa986f47ec986f0906a1b;rport From: "Tom Haselden" <sip:103@domain.com>;tag=20019 To: <sip:103@be7.domain.com;user=phone>;tag=f1f4aa832 Call-ID: 708d4bca@pbx CSeq: 8816 ACK Max-Forwards: 70 Contact: <sip:103@172.x.x.75:3444;transport=tcp> Content-Length: 0 [7] 2008/02/24 16:57:52: Determine pass-through mode after receiving response [9] 2008/02/24 16:57:52: Resolve 1980: aaaa udp 64.152.60.75 5060 [9] 2008/02/24 16:57:52: Resolve 1980: a udp 64.152.60.75 5060 [9] 2008/02/24 16:57:52: Resolve 1980: udp 64.152.60.75 5060 [7] 2008/02/24 16:57:52: SIP Tx udp:64.152.60.75:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B04d42fb1611ec6f9 From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 Call-ID: 1459988973_58241008@64.152.60.75 CSeq: 18297 INVITE Contact: <sip:Anonymous@172.x.x.75:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Content-Type: application/sdp Content-Length: 201 v=0 o=- 51427 51427 IN IP4 172.x.x.75 s=- c=IN IP4 172.x.x.75 t=0 0 m=audio 9036 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [7] 2008/02/24 16:57:52: 708d4bca@pbx#20019: RTP pass-through mode [7] 2008/02/24 16:57:52: 1459988973_58241008@64.152.60.75#a9a6181f02: RTP pass-through mode [7] 2008/02/24 16:57:52: SIP Rx udp:64.152.60.75:5060: ACK sip:Anonymous@172.x.x.75:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B051200e6bed8a5c3 From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 Call-ID: 1459988973_58241008@64.152.60.75 CSeq: 18297 ACK Max-Forwards: 70 Content-Length: 0 [5] 2008/02/24 16:57:55: Call bf515e7b@pbx#19778: Last request not finished [9] 2008/02/24 16:57:55: Resolve 1981: tcp 172.x.x.130 5060 [7] 2008/02/24 16:57:55: SIP Tx tcp:172.x.x.130:5060: CANCEL sip:55552907492@sip:172.x.x.130:5060;transport=tcp;user=phone SIP/2.0 Via: SIP/2.0/TCP 172.x.x.75:3442;branch=z9hG4bK-3c718048209c673690214a2681b04258;rport From: "Tom Haselden" <sip:103@domain.com>;tag=19778 To: <sip:55552907492@sip:172.x.x.130:5060;transport=tcp;user=phone> Call-ID: bf515e7b@pbx CSeq: 26279 CANCEL Max-Forwards: 70 Content-Length: 0 [8] 2008/02/24 16:57:55: Hangup: Call bf515e7b@pbx#19778 not found [7] 2008/02/24 16:57:58: SIP Rx tcp:172.x.x.81:5065: INVITE sip:103@172.x.x.75:3444;transport=tcp SIP/2.0 FROM: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832 TO: <sip:103@domain.com>;tag=20019 CSEQ: 1 INVITE CALL-ID: 708d4bca@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK2e2c2c66 CONTACT: <sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81;ms-opaque=c9e23a1203e9a49b>;automata CONTENT-LENGTH: 276 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp v=0 o=- 0 1 IN IP4 172.x.x.81 s=session c=IN IP4 172.x.x.81 t=0 0 m=audio 0 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=image 9200 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPFEC [7] 2008/02/24 16:57:58: UDP: Opening socket on port 9060 [7] 2008/02/24 16:57:58: UDP: Opening socket on port 9086 [9] 2008/02/24 16:57:58: Resolve 1982: url sip:5552907492@64.152.60.75:5060 [9] 2008/02/24 16:57:58: Resolve 1982: udp 64.152.60.75 5060 [7] 2008/02/24 16:57:58: SIP Tx udp:64.152.60.75:5060: UPDATE sip:5552907492@64.152.60.75:5060 SIP/2.0 Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 Call-ID: 1459988973_58241008@64.152.60.75 CSeq: 707 UPDATE Max-Forwards: 70 Contact: <sip:Anonymous@172.x.x.75:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Content-Type: application/sdp Content-Length: 276 v=0 o=- 51427 51428 IN IP4 172.x.x.75 s=- c=IN IP4 172.x.x.75 t=0 0 m=audio 0 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 m=image 9086 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPFEC [9] 2008/02/24 16:57:58: Resolve 1983: tcp 172.x.x.81 5065 [7] 2008/02/24 16:57:58: SIP Tx tcp:172.x.x.81:5065: SIP/2.0 100 Trying Via: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK2e2c2c66 From: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832 To: <sip:103@domain.com>;tag=20019 Call-ID: 708d4bca@pbx CSeq: 1 INVITE Content-Length: 0 [7] 2008/02/24 16:57:58: SIP Rx udp:64.152.60.75:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport=5060 From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 Call-ID: 1459988973_58241008@64.152.60.75 CSeq: 707 UPDATE Contact: "Haselden Tom " <sip:5552907492@64.152.60.75:5060> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 1800;refresher=uas Content-Length: 329 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 825 15704 IN IP4 64.152.60.75 s=SIP Media Capabilities c=IN IP4 64.152.60.71 t=0 0 m=audio 32370 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 m=image 0 udptl t38 a=T38FaxRateManagement:transferredTCF a= T38FaxUdpEC:t38UDPFEC a=sendrecv [7] 2008/02/24 16:57:58: Call 1459988973_58241008@64.152.60.75#a9a6181f02: Clear last request [9] 2008/02/24 16:57:58: Resolve 1984: tcp 172.x.x.81 5065 [7] 2008/02/24 16:57:58: SIP Tx tcp:172.x.x.81:5065: SIP/2.0 200 Ok Via: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK2e2c2c66 From: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832 To: <sip:103@domain.com>;tag=20019 Call-ID: 708d4bca@pbx CSeq: 1 INVITE Contact: <sip:103@172.x.x.75:3444;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.6.2448 Content-Type: application/sdp Content-Length: 275 v=0 o=- 62722 62723 IN IP4 172.x.x.75 s=- c=IN IP4 172.x.x.75 t=0 0 m=audio 9060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 0 udptl t38 a=T38FaxRateManagement:transferredTCF a= T38FaxUdpEC:t38UDPFEC [7] 2008/02/24 16:57:58: SIP Rx tcp:172.x.x.81:5065: ACK sip:103@172.x.x.75:3444;transport=tcp SIP/2.0 FROM: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832 TO: <sip:103@domain.com>;tag=20019 CSEQ: 1 ACK CALL-ID: 708d4bca@pbx MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK68d5d4f CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 [5] 2008/02/24 16:58:02: SIP port accept from 172.x.x.81:18364 [7] 2008/02/24 16:58:02: SIP Rx tcp:172.x.x.81:18364: OPTIONS sip:172.x.x.75:5060 SIP/2.0 FROM: <sip:be7.domain.com:5060;transport=Tcp;ms-opaque=95944f3af7203520>;epid=7BE54968BB;tag=e59d6c85f0 TO: <sip:172.x.x.75:5060> CSEQ: 6 OPTIONS CALL-ID: ff3f5d4ff10b4315a6865919470a673f MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.x.x.81:18364;branch=z9hG4bK8948f21 ACCEPT: application/sdp CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.0.0.0 [9] 2008/02/24 16:58:02: Resolve 1985: tcp 172.x.x.81 18364 [7] 2008/02/24 16:58:02: SIP Tx tcp:172.x.x.81:18364: SIP/2.0 200 Ok Via: SIP/2.0/TCP 172.x.x.81:18364;branch=z9hG4bK8948f21 From: <sip:be7.domain.com:5060;transport=Tcp;ms-opaque=95944f3af7203520>;epid=7BE54968BB;tag=e59d6c85f0 To: <sip:172.x.x.75:5060>;tag=b3edb24a21 Call-ID: ff3f5d4ff10b4315a6865919470a673f CSeq: 6 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [7] 2008/02/24 16:58:31: SIP Rx udp:64.152.60.75:5060: BYE sip:Anonymous@172.x.x.75:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05813609bed8a5c3 From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 Call-ID: 1459988973_58241008@64.152.60.75 CSeq: 18299 BYE Max-Forwards: 70 Supported: 100rel Content-Length: 0 [9] 2008/02/24 16:58:31: Resolve 1986: aaaa udp 64.152.60.75 5060 [9] 2008/02/24 16:58:31: Resolve 1986: a udp 64.152.60.75 5060 [9] 2008/02/24 16:58:31: Resolve 1986: udp 64.152.60.75 5060 [7] 2008/02/24 16:58:31: SIP Tx udp:64.152.60.75:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05813609bed8a5c3 From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4 To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02 Call-ID: 1459988973_58241008@64.152.60.75 CSeq: 18299 BYE Contact: <sip:Anonymous@172.x.x.75:5060;transport=udp> User-Agent: pbxnsip-PBX/2.1.6.2448 RTP-RxStat: Dur=88,Pkt=272,Oct=46784,Underun=0 RTP-TxStat: Dur=63,Pkt=272,Oct=46784 Content-Length: 0 [7] 2008/02/24 16:58:31: 708d4bca@pbx#20019: Media-aware pass-through mode [7] 2008/02/24 16:58:31: Other Ports: 1 [7] 2008/02/24 16:58:31: Call Port: 708d4bca@pbx#20019 [8] 2008/02/24 16:58:31: UDP: recvfrom receives ICMP message [8] 2008/02/24 16:58:31: Last message repeated 13 times [9] 2008/02/24 16:58:31: Resolve 1987: aaaa tcp 172.x.x.81 5065 [9] 2008/02/24 16:58:31: Resolve 1987: a tcp 172.x.x.81 5065 [9] 2008/02/24 16:58:31: Resolve 1987: tcp 172.x.x.81 5065 [7] 2008/02/24 16:58:31: SIP Tx tcp:172.x.x.81:5065: BYE sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81 SIP/2.0 Via: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-846f98ba102960c739a4a8f2c078531a;rport From: "Tom Haselden" <sip:103@domain.com>;tag=20019 To: <sip:103@be7.domain.com;user=phone>;tag=f1f4aa832 Call-ID: 708d4bca@pbx CSeq: 8817 BYE Max-Forwards: 70 Contact: <sip:103@172.x.x.75:3444;transport=tcp> RTP-RxStat: Dur=68,Pkt=4,Oct=688,Underun=0 RTP-TxStat: Dur=63,Pkt=285,Oct=49020 Content-Length: 0 [8] 2008/02/24 16:58:31: UDP: recvfrom receives ICMP message [8] 2008/02/24 16:58:31: Last message repeated 3 times [7] 2008/02/24 16:58:31: SIP Rx tcp:172.x.x.81:5065: SIP/2.0 200 OK FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019 TO: <sip:103@be7.domain.com;user=phone>;tag=f1f4aa832;epid=AA79D0A609 CSEQ: 8817 BYE CALL-ID: 708d4bca@pbx VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-846f98ba102960c739a4a8f2c078531a;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [7] 2008/02/24 16:58:31: Call 708d4bca@pbx#20019: Clear last request [5] 2008/02/24 16:58:31: BYE Response: Terminate 708d4bca@pbx [3] 2008/02/24 16:58:32: SMTP: Cannot resolve mail.domain.com

Thanks, the problem is the phone on the other side can not register on the PBX. I'll see if I can find another way.

That did the trick. Thanks so much I am server literate put PBX challenged. Good to have the help.

I have a pbx account "1000" that I can sucessfully call, this is in the domain and PBX I have an external extension "71000" that I can sucessfully call, it goes through the dial plan locates the trunk and forwards the call. I can add the extension as a static registration (add contact) but the PBX always looks to resolve it as a URI instead of an extension. Is there a way to make a static registration for this external extension? Thanks

I have external calls for a single sip gateway coming into a single trunk. Right now I can make the trunk visible to all domains and set up seperate dial plans for each domain. The issue is the calls always go to the domain the trunk is in and the service cannot resolve the extension of the other domain. So if domain primary.com wants all calls to 1111xxxx, where 1111 is the domain extension prefix, it works if the trunk is in primary.com. But if a call comes through for 2222xxxx where 2222 is the domain prefix for domain alternate.com the service cannot resolve it. Even though there is a dial plan for 2222 in the alternate.com domain using the primary.com domain's visible trunk. I am trying to avoid having a SIP gateway for each domain is there a way to route these external calls to the proper domain based stricly on the extension number prefix.