================================================================================== ======================= FAX ROUTED TO ASTERISK GATEWAY =========================== ================================================================================== <--- SIP read from 10.10.20.130:5060 ---> INVITE sip:1616#0220005622457495@10.10.20.133:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKab9e.32c90ad1.0 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK7c471192a4200 From: ;tag=7c471192a4 To: Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 200 INVITE Supported: timer, replaces Min-SE: 1800 Date: Thu, 10 Apr 2008 15:04:28 GMT User-Agent: AddPac SIP Gateway Contact: Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 307 Max-Forwards: 16 v=0 o=7772001003 1207839868 1207839868 IN IP4 10.10.10.233 s=AddPac Gateway SDP c=IN IP4 10.10.10.233 t=1207839868 0 m=audio 23416 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (18 headers 13 lines) --- Sending to 10.10.20.130 : 5060 (no NAT) Using INVITE request as basis request - 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 Found peer 'EDGE_LCR' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.233:23416 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x101 (g723|g729), peer - audio=0x101 (g723|g729)/video=0x0 (nothing), combined - 0x101 (g723|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.10.233:23416 Looking for 1616#0220005622457495 in INC_EXT_LCR (domain 10.10.20.133) list_route: hop: <--- Transmitting (no NAT) to 10.10.20.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKab9e.32c90ad1.0;received=10.10.20.130 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK7c471192a4200 Record-Route: From: ;tag=7c471192a4 To: Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 200 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [1616#0220005622457495@INC_EXT_LCR:1] Set("SIP/10.10.10.233-0947f8d0", "GROUP()=1616#") in new stack -- Executing [1616#0220005622457495@INC_EXT_LCR:2] GotoIf("SIP/10.10.10.233-0947f8d0", "0?BLOCK") in new stack -- Executing [1616#0220005622457495@INC_EXT_LCR:3] Set("SIP/10.10.10.233-0947f8d0", "CALLERID(all)=5555848119 <5555848119>") in new stack -- Executing [1616#0220005622457495@INC_EXT_LCR:4] Dial("SIP/10.10.10.233-0947f8d0", "SIP/0220005622457495@SIP_LCR") in new stack Audio is at 10.10.20.133 port 16678 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.10.20.36:5060: INVITE sip:0220005622457495@siplcr.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK13b57887 From: "5555848119" ;tag=as534bbcc7 To: Contact: Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 INVITE User-Agent: B2BUA06 Max-Forwards: 70 Date: Thu, 10 Apr 2008 19:04:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 309 v=0 o=root 29000 29000 IN IP4 10.10.20.133 s=session c=IN IP4 10.10.20.133 t=0 0 m=audio 16678 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 0220005622457495@SIP_LCR <--- SIP read from 10.10.20.36:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK13b57887 From: "5555848119" ;tag=as534bbcc7 To: Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 INVITE Server: Sip EXpress router (0.9.3 (i386/linux)) Content-Length: 0 Warning: 392 10.10.20.36:5060 "Noisy feedback tells: pid=5993 req_src_ip=10.10.20.133 req_src_port=5060 in_uri=sip:0220005622457495@siplcr.mydomain.com out_uri=sip:0220005622457495@10.10.20.126:5060 via_cnt==1" <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.10.20.36:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK13b57887 Record-Route: From: "5555848119" ;tag=as534bbcc7 To: ;tag=as17762969 Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 INVITE User-Agent: B2BUA01 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from 10.10.20.36:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK13b57887 Record-Route: From: "5555848119" ;tag=as534bbcc7 To: ;tag=as17762969 Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 INVITE User-Agent: B2BUA01 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 288 v=0 o=root 31588 31588 IN IP4 10.10.20.126 s=session c=IN IP4 10.10.20.126 t=0 0 m=audio 11844 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- -- SIP/SIP_LCR-0946ce30 is ringing <--- Transmitting (no NAT) to 10.10.20.130:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKab9e.32c90ad1.0;received=10.10.20.130 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK7c471192a4200 Record-Route: From: ;tag=7c471192a4 To: ;tag=as28c07e16 Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 200 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 10.10.20.126:11844 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x101 (g723|g729), peer - audio=0x101 (g723|g729)/video=0x0 (nothing), combined - 0x101 (g723|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.20.126:11844 -- SIP/SIP_LCR-0946ce30 is making progress passing it to SIP/10.10.10.233-0947f8d0 Audio is at 10.10.20.133 port 16918 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.10.20.130:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKab9e.32c90ad1.0;received=10.10.20.130 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK7c471192a4200 Record-Route: From: ;tag=7c471192a4 To: ;tag=as28c07e16 Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 200 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 309 v=0 o=root 29000 29000 IN IP4 10.10.20.133 s=session c=IN IP4 10.10.20.133 t=0 0 m=audio 16918 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 10.10.20.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK13b57887 Record-Route: From: "5555848119" ;tag=as534bbcc7 To: ;tag=as17762969 Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 INVITE User-Agent: B2BUA01 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 288 v=0 o=root 31588 31589 IN IP4 10.10.20.126 s=session c=IN IP4 10.10.20.126 t=0 0 m=audio 11844 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (13 headers 14 lines) --- Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 10.10.20.126:11844 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x101 (g723|g729), peer - audio=0x101 (g723|g729)/video=0x0 (nothing), combined - 0x101 (g723|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.20.126:11844 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.20.36, port 5060 Transmitting (no NAT) to 10.10.20.36:5060: ACK sip:0220005622457495@10.10.20.126 SIP/2.0 Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK56d8fe8a Route: From: "5555848119" ;tag=as534bbcc7 To: ;tag=as17762969 Contact: Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 ACK User-Agent: B2BUA06 Max-Forwards: 70 Content-Length: 0 --- -- SIP/SIP_LCR-0946ce30 answered SIP/10.10.10.233-0947f8d0 Audio is at 10.10.20.133 port 16918 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.10.20.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKab9e.32c90ad1.0;received=10.10.20.130 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK7c471192a4200 Record-Route: From: ;tag=7c471192a4 To: ;tag=as28c07e16 Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 200 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 309 v=0 o=root 29000 29001 IN IP4 10.10.20.133 s=session c=IN IP4 10.10.20.133 t=0 0 m=audio 16918 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 10.10.20.130:5060 ---> ACK sip:1616#0220005622457495@10.10.20.133 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.10.20.130;branch=0 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK7c471192a4200 From: ;tag=7c471192a4 To: ;tag=as28c07e16 Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 200 ACK Content-Length: 0 Max-Forwards: 16 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.10.20.36:5060 ---> INVITE sip:5555848119@10.10.20.133 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.10.20.36;branch=z9hG4bK774a.d0b9c882.0 Via: SIP/2.0/UDP 10.10.20.126:5060;branch=z9hG4bK4a2d5a68;rport=5060 From: ;tag=as17762969 To: "5555848119" ;tag=as534bbcc7 Contact: Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 INVITE User-Agent: B2BUA01 Max-Forwards: 16 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 351 v=0 o=root 31588 31590 IN IP4 10.10.20.126 s=session c=IN IP4 10.10.20.126 t=0 0 m=image 4208 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (15 headers 15 lines) --- Sending to 10.10.20.36 : 5060 (no NAT) Got T.38 offer in SDP in dialog 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com Capabilities: us - 0x101 (g723|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) <--- Transmitting (no NAT) to 10.10.20.36:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.20.36;branch=z9hG4bK774a.d0b9c882.0;received=10.10.20.36 Via: SIP/2.0/UDP 10.10.20.126:5060;branch=z9hG4bK4a2d5a68;rport=5060 Record-Route: From: ;tag=as17762969 To: "5555848119" ;tag=as534bbcc7 Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.20.130, port 5060 Reliably Transmitting (no NAT) to 10.10.20.130:5060: INVITE sip:7772001003@10.10.10.233 SIP/2.0 Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK071ed6a3 Route: From: ;tag=as28c07e16 To: ;tag=7c471192a4 Contact: Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 102 INVITE User-Agent: B2BUA06 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-info: SIP re-invite (T38 switchover) Content-Type: application/sdp Content-Length: 351 v=0 o=root 29000 29002 IN IP4 10.10.20.133 s=session c=IN IP4 10.10.20.133 t=0 0 m=image 4832 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy --- <--- SIP read from 10.10.20.130:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK071ed6a3 From: ;tag=as28c07e16 To: ;tag=7c471192a4 Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.10.20.130:5060 "Noisy feedback tells: pid=7916 req_src_ip=10.10.20.133 req_src_port=5060 in_uri=sip:7772001003@10.10.10.233 out_uri=sip:7772001003@10.10.10.233 via_cnt==1" <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.10.20.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK071ed6a3 From: ;tag=as28c07e16 To: ;tag=7c471192a4 Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 102 INVITE User-Agent: AddPac SIP Gateway Contact: sip:7772001003@10.10.10.233 Content-Type: application/sdp Content-Length: 149 Record-Route: v=0 o=7772001003 1207839897 1207839897 IN IP4 10.10.10.233 s=AddPac Gateway SDP c=IN IP4 10.10.10.233 t=1207839897 0 m=image 23416 udptl t38 <-------------> --- (11 headers 6 lines) --- Got T.38 offer in SDP in dialog 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 Capabilities: us - 0x101 (g723|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) <--- Reliably Transmitting (no NAT) to 10.10.20.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.36;branch=z9hG4bK774a.d0b9c882.0;received=10.10.20.36 Via: SIP/2.0/UDP 10.10.20.126:5060;branch=z9hG4bK4a2d5a68;rport=5060 Record-Route: From: ;tag=as17762969 To: "5555848119" ;tag=as534bbcc7 Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 351 v=0 o=root 29000 29001 IN IP4 10.10.20.133 s=session c=IN IP4 10.10.20.133 t=0 0 m=image 4447 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.20.130, port 5060 Transmitting (no NAT) to 10.10.20.130:5060: ACK sip:7772001003@10.10.10.233 SIP/2.0 Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK072c7929 Route: From: ;tag=as28c07e16 To: ;tag=7c471192a4 Contact: Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 102 ACK User-Agent: B2BUA06 Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 10.10.20.36:5060 ---> ACK sip:5555848119@10.10.20.133 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.10.20.36;branch=0 Via: SIP/2.0/UDP 10.10.20.126:5060;branch=z9hG4bK29e19457;rport=5060 From: ;tag=as17762969 To: "5555848119" ;tag=as534bbcc7 Contact: Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 102 ACK User-Agent: B2BUA01 Max-Forwards: 16 Content-Length: 0 <--- SIP read from 10.10.20.130:5060 ---> BYE sip:1616#0220005622457495@10.10.20.133 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKbb9e.382c8702.0 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK7c471192a4201 From: ;tag=7c471192a4 To: ;tag=as28c07e16 Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 201 BYE Date: Thu, 10 Apr 2008 15:05:10 GMT User-Agent: AddPac SIP Gateway Contact: Content-Length: 0 Max-Forwards: 16 <-------------> --- (13 headers 0 lines) --- Sending to 10.10.20.130 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.10.20.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKbb9e.382c8702.0;received=10.10.20.130 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK7c471192a4201 Record-Route: From: ;tag=7c471192a4 To: ;tag=as28c07e16 Call-ID: 7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233 CSeq: 201 BYE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.20.36, port 5060 Reliably Transmitting (no NAT) to 10.10.20.36:5060: BYE sip:0220005622457495@10.10.20.126 SIP/2.0 Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK02869c27 Route: From: "5555848119" ;tag=as534bbcc7 To: ;tag=as17762969 Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 103 BYE User-Agent: B2BUA06 Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (INC_EXT_LCR, 1616#0220005622457495, 4) exited non-zero on 'SIP/10.10.10.233-0947f8d0' <--- SIP read from 10.10.20.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK02869c27 Record-Route: From: "5555848119" ;tag=as534bbcc7 To: ;tag=as17762969 Call-ID: 00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com CSeq: 103 BYE User-Agent: B2BUA01 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '00f74fdc33721ada18c024da3bcf42f5@sipproxy.mydomain.com' Method: ACK Really destroying SIP dialog '7c2cfe47-03a9-11d3-8492-0002a400248b@10.10.10.233' Method: BYE ================================================================================== ======================== FAX ROUTED TO ADDPAC GATEWAY ============================ ================================================================================== <--- SIP read from 10.10.20.130:5060 ---> INVITE sip:1616#0212005622457495@10.10.20.133:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKbea6.fbe579f3.0 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK2b476f8da4199 From: ;tag=2b476f8da4 To: Call-ID: 2b29fe47-1214-6fc0-848d-0002a400248b@10.10.10.233 CSeq: 199 INVITE Supported: timer, replaces Min-SE: 1800 Date: Thu, 10 Apr 2008 14:50:19 GMT User-Agent: AddPac SIP Gateway Contact: Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 307 Max-Forwards: 16 v=0 o=7772001003 1207839019 1207839019 IN IP4 10.10.10.233 s=AddPac Gateway SDP c=IN IP4 10.10.10.233 t=1207839019 0 m=audio 23414 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (18 headers 13 lines) --- Sending to 10.10.20.130 : 5060 (no NAT) Using INVITE request as basis request - 2b29fe47-1214-6fc0-848d-0002a400248b@10.10.10.233 Found peer 'EDGE_LCR' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.233:23414 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x101 (g723|g729), peer - audio=0x101 (g723|g729)/video=0x0 (nothing), combined - 0x101 (g723|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.10.233:23414 Looking for 1616#0212005622457495 in INC_EXT_LCR (domain 10.10.20.133) list_route: hop: <--- Transmitting (no NAT) to 10.10.20.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKbea6.fbe579f3.0;received=10.10.20.130 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK2b476f8da4199 Record-Route: From: ;tag=2b476f8da4 To: Call-ID: 2b29fe47-1214-6fc0-848d-0002a400248b@10.10.10.233 CSeq: 199 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [1616#0212005622457495@INC_EXT_LCR:1] Set("SIP/10.10.10.233-0945a470", "GROUP()=1616#") in new stack -- Executing [1616#0212005622457495@INC_EXT_LCR:2] GotoIf("SIP/10.10.10.233-0945a470", "0?BLOCK") in new stack -- Executing [1616#0212005622457495@INC_EXT_LCR:3] Set("SIP/10.10.10.233-0945a470", "CALLERID(all)=5555848119 <5555848119>") in new stack -- Executing [1616#0212005622457495@INC_EXT_LCR:4] Dial("SIP/10.10.10.233-0945a470", "SIP/0212005622457495@SIP_LCR") in new stack Audio is at 10.10.20.133 port 13682 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.10.20.36:5060: INVITE sip:0212005622457495@siplcr.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK460c26de From: "5555848119" ;tag=as1fe2a9f6 To: Contact: Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com CSeq: 102 INVITE User-Agent: B2BUA06 Max-Forwards: 70 Date: Thu, 10 Apr 2008 18:50:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 309 v=0 o=root 29000 29000 IN IP4 10.10.20.133 s=session c=IN IP4 10.10.20.133 t=0 0 m=audio 13682 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 0212005622457495@SIP_LCR <--- SIP read from 10.10.20.36:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK460c26de From: "5555848119" ;tag=as1fe2a9f6 To: Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com CSeq: 102 INVITE Server: Sip EXpress router (0.9.3 (i386/linux)) Content-Length: 0 Warning: 392 10.10.20.36:5060 "Noisy feedback tells: pid=5975 req_src_ip=10.10.20.133 req_src_port=5060 in_uri=sip:0212005622457495@siplcr.mydomain.com out_uri=sip:005622457495@10.10.20.83:5060 via_cnt==1" <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.10.20.36:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK460c26de From: "5555848119" ;tag=as1fe2a9f6 To: ;tag=2a4719b9a4 Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com CSeq: 102 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:005622457495@10.10.20.83 Content-Type: application/sdp Content-Length: 257 Record-Route: v=0 o=005622457495 1207839019 1207839019 IN IP4 10.10.20.83 s=AddPac Gateway SDP c=IN IP4 10.10.20.83 t=1207839019 0 m=audio 24086 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (12 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.10.20.83:24086 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x101 (g723|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.20.83:24086 -- SIP/SIP_LCR-0946d220 is making progress passing it to SIP/10.10.10.233-0945a470 Audio is at 10.10.20.133 port 14068 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.10.20.130:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKbea6.fbe579f3.0;received=10.10.20.130 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK2b476f8da4199 Record-Route: From: ;tag=2b476f8da4 To: ;tag=as5b327f2d Call-ID: 2b29fe47-1214-6fc0-848d-0002a400248b@10.10.10.233 CSeq: 199 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 309 v=0 o=root 29000 29000 IN IP4 10.10.20.133 s=session c=IN IP4 10.10.20.133 t=0 0 m=audio 14068 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 10.10.20.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK460c26de From: "5555848119" ;tag=as1fe2a9f6 To: ;tag=2a4719b9a4 Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com CSeq: 102 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:005622457495@10.10.20.83 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 257 Record-Route: v=0 o=005622457495 1207839020 1207839020 IN IP4 10.10.20.83 s=AddPac Gateway SDP c=IN IP4 10.10.20.83 t=1207839020 0 m=audio 24086 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (13 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.10.20.83:24086 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x101 (g723|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.20.83:24086 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.20.36, port 5060 Transmitting (no NAT) to 10.10.20.36:5060: ACK sip:005622457495@10.10.20.83 SIP/2.0 Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK271efad0 Route: From: "5555848119" ;tag=as1fe2a9f6 To: ;tag=2a4719b9a4 Contact: Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com CSeq: 102 ACK User-Agent: B2BUA06 Max-Forwards: 70 Content-Length: 0 --- -- SIP/SIP_LCR-0946d220 answered SIP/10.10.10.233-0945a470 Audio is at 10.10.20.133 port 14068 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.10.20.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.130;branch=z9hG4bKbea6.fbe579f3.0;received=10.10.20.130 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK2b476f8da4199 Record-Route: From: ;tag=2b476f8da4 To: ;tag=as5b327f2d Call-ID: 2b29fe47-1214-6fc0-848d-0002a400248b@10.10.10.233 CSeq: 199 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 309 v=0 o=root 29000 29001 IN IP4 10.10.20.133 s=session c=IN IP4 10.10.20.133 t=0 0 m=audio 14068 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 10.10.20.130:5060 ---> ACK sip:1616#0212005622457495@10.10.20.133 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.10.20.130;branch=0 Via: SIP/2.0/UDP 10.10.10.233:5060;branch=z9hG4bK2b476f8da4199 From: ;tag=2b476f8da4 To: ;tag=as5b327f2d Call-ID: 2b29fe47-1214-6fc0-848d-0002a400248b@10.10.10.233 CSeq: 199 ACK Content-Length: 0 Max-Forwards: 16 <-------------> --- (10 headers 0 lines) --- *CLI> *CLI> <--- SIP read from 10.10.20.36:5060 ---> INVITE sip:5555848119@10.10.20.133 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.10.20.36;branch=z9hG4bK7dc7.2d9698a1.0 Via: SIP/2.0/UDP 10.10.20.83:5060;branch=z9hG4bK2a4719b9a4215 From: ;tag=2a4719b9a4 To: "5555848119" ;tag=as1fe2a9f6 Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com CSeq: 215 INVITE Supported: timer, replaces Min-SE: 1800 Date: Thu, 10 Apr 2008 14:50:41 GMT User-Agent: AddPac SIP Gateway Contact: Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 149 Max-Forwards: 16 v=0 o=005622457495 1207839041 1207839041 IN IP4 10.10.20.83 s=AddPac Gateway SDP c=IN IP4 10.10.20.83 t=1207839041 0 m=image 24086 udptl t38 <-------------> --- (18 headers 6 lines) --- Sending to 10.10.20.36 : 5060 (no NAT) Got T.38 offer in SDP in dialog 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com Capabilities: us - 0x101 (g723|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Apr 10 14:50:42] NOTICE[29023]: chan_sip.c:5493 process_sdp: No compatible codecs, not accepting this offer! <--- Transmitting (no NAT) to 10.10.20.36:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 10.10.20.36;branch=z9hG4bK7dc7.2d9698a1.0;received=10.10.20.36 Via: SIP/2.0/UDP 10.10.20.83:5060;branch=z9hG4bK2a4719b9a4215 From: ;tag=2a4719b9a4 To: "5555848119" ;tag=as1fe2a9f6 Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com CSeq: 215 INVITE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> <--- SIP read from 10.10.20.36:5060 ---> ACK sip:5555848119@10.10.20.133 SIP/2.0 Via: SIP/2.0/UDP 10.10.20.36;branch=z9hG4bK7dc7.2d9698a1.0 From: ;tag=2a4719b9a4 Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com To: "5555848119" ;tag=as1fe2a9f6 CSeq: 215 ACK Route: User-Agent: Sip EXpress router(0.9.3 (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.10.20.36:5060 ---> BYE sip:5555848119@10.10.20.133 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.10.20.36;branch=z9hG4bK4dc7.69b3ccd1.0 Via: SIP/2.0/UDP 10.10.20.83:5060;branch=z9hG4bK2a4719b9a4216 From: ;tag=2a4719b9a4 To: "5555848119" ;tag=as1fe2a9f6 Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com CSeq: 216 BYE Date: Thu, 10 Apr 2008 14:50:41 GMT User-Agent: AddPac SIP Gateway Contact: Content-Length: 0 Max-Forwards: 16 <-------------> --- (13 headers 0 lines) --- Sending to 10.10.20.36 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.10.20.36:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.36;branch=z9hG4bK4dc7.69b3ccd1.0;received=10.10.20.36 Via: SIP/2.0/UDP 10.10.20.83:5060;branch=z9hG4bK2a4719b9a4216 Record-Route: From: ;tag=2a4719b9a4 To: "5555848119" ;tag=as1fe2a9f6 Call-ID: 7e4cb7471b6c1dcf1f4ba8a42b306f7d@sipproxy.mydomain.com CSeq: 216 BYE User-Agent: B2BUA06 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (INC_EXT_LCR, 1616#0212005622457495, 4) exited non-zero on 'SIP/10.10.10.233-0945a470' Scheduling destruction of SIP dialog '2b29fe47-1214-6fc0-848d-0002a400248b@10.10.10.233' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.20.130, port 5060 Reliably Transmitting (no NAT) to 10.10.20.130:5060: BYE sip:7772001003@10.10.10.233 SIP/2.0 Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK33a46f51 Route: From: ;tag=as5b327f2d To: ;tag=2b476f8da4 Call-ID: 2b29fe47-1214-6fc0-848d-0002a400248b@10.10.10.233 CSeq: 102 BYE User-Agent: B2BUA06 Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 10.10.20.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.20.133:5060;branch=z9hG4bK33a46f51 From: ;tag=as5b327f2d To: ;tag=2b476f8da4 Call-ID: 2b29fe47-1214-6fc0-848d-0002a400248b@10.10.10.233 CSeq: 102 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 Record-Route: