<--- SIP read from A.A.A.A:5060 ---> INVITE sip:123456789@D.D.D.D SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK5521.d23b8943.0 Via: SIP/2.0/UDP E.E.E.E;branch=z9hG4bK5521.46993d06.0 Via: SIP/2.0/UDP F.F.F.F:5060;branch=z9hG4bK60bjgo109000bcg146c0.1 From: ;tag=4B718C43-21AEAF8-3E86ED92 To: Call-ID: 4B718C43-00240BEA@DDUS0_PCU-004 CSeq: 1 INVITE Accept: application/sdp,application/isup,multipart/mixed,application/vnd.siemens.key-event,application/vnd.siemens.surpass,application/dtmf-relay Contact: MIME-Version: 1.0 Supported: timer,100rel Max-Forwards: 26 P-Asserted-Identity: Session-Expires: 1800 Allow: ACK,INFO,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,PRACK,UPDATE Content-Type: application/sdp Content-Length: 350 v=0 o=hiQ9200 1008620100109172435 1011613779 IN IP4 62.180.246.11 s=Phone Call via hiQ9200 SIPCA c=IN IP4 62.180.246.11 t=0 0 m=audio 20696 RTP/AVP 8 0 111 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 image udptl t38 a=sendrecv a=ptime:20 <-------------> --- (20 headers 15 lines) --- Sending to A.A.A.A : 5060 (no NAT) Using INVITE request as basis request - 4B718C43-00240BEA@DDUS0_PCU-004 Found peer 'BTNGN3' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 62.180.246.11:20696 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 111 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x80c (ulaw|alaw|g726)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 62.180.246.11:20696 Looking for 123456789 in from-inbound (domain D.D.D.D) list_route: hop: list_route: hop: estpbx01*CLI> <--- Transmitting (no NAT) to A.A.A.A:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK5521.d23b8943.0;received=A.A.A.A Via: SIP/2.0/UDP E.E.E.E;branch=z9hG4bK5521.46993d06.0 Via: SIP/2.0/UDP F.F.F.F:5060;branch=z9hG4bK60bjgo109000bcg146c0.1 Record-Route: Record-Route: From: ;tag=4B718C43-21AEAF8-3E86ED92 To: Call-ID: 4B718C43-00240BEA@DDUS0_PCU-004 CSeq: 1 INVITE User-Agent: asterisk 1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [123456789@from-inbound:1] Dial("SIP/bt.com-08238b28", "SIP/faxserver/K0001123456789") in new stack Audio is at D.D.D.D port 13326 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to C.C.C.C:5060: INVITE sip:K0001123456789@C.C.C.C SIP/2.0 Via: SIP/2.0/UDP D.D.D.D:5060;branch=z9hG4bK10492583;rport From: "123456789" ;tag=as31e2cce0 To: Contact: Call-ID: 03f0bc4832e47e87403a6ee47bf704f2@D.D.D.D CSeq: 102 INVITE User-Agent: asterisk 1.4 Max-Forwards: 70 Date: Tue, 09 Feb 2010 16:24:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 436 436 IN IP4 D.D.D.D s=session c=IN IP4 D.D.D.D t=0 0 m=audio 13326 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called faxserver/K0001123456789 estpbx01*CLI> <--- SIP read from C.C.C.C:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP D.D.D.D:5060;branch=z9hG4bK10492583;received=D.D.D.D;rport=5060 From: "123456789" ;tag=as31e2cce0 To: Call-ID: 03f0bc4832e47e87403a6ee47bf704f2@D.D.D.D CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.1-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- estpbx01*CLI> <--- SIP read from C.C.C.C:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP D.D.D.D:5060;branch=z9hG4bK10492583;received=D.D.D.D;rport=5060 From: "123456789" ;tag=as31e2cce0 To: ;tag=as774ab834 Call-ID: 03f0bc4832e47e87403a6ee47bf704f2@D.D.D.D CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.1-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 1235114045 1235114045 IN IP4 C.C.C.C s=Asterisk PBX 1.6.2.1-1 c=IN IP4 C.C.C.C t=0 0 m=audio 16348 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (12 headers 12 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port C.C.C.C:16348 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port C.C.C.C:16348 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to C.C.C.C, port 5060 Transmitting (no NAT) to C.C.C.C:5060: ACK sip:K0001123456789@C.C.C.C SIP/2.0 Via: SIP/2.0/UDP D.D.D.D:5060;branch=z9hG4bK04588125;rport From: "123456789" ;tag=as31e2cce0 To: ;tag=as774ab834 Contact: Call-ID: 03f0bc4832e47e87403a6ee47bf704f2@D.D.D.D CSeq: 102 ACK User-Agent: asterisk 1.4 Max-Forwards: 70 Content-Length: 0 --- -- SIP/faxserver-0823df78 answered SIP/bt.com-08238b28 Audio is at D.D.D.D port 13240 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP estpbx01*CLI> <--- Reliably Transmitting (no NAT) to A.A.A.A:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK5521.d23b8943.0;received=A.A.A.A Via: SIP/2.0/UDP E.E.E.E;branch=z9hG4bK5521.46993d06.0 Via: SIP/2.0/UDP F.F.F.F:5060;branch=z9hG4bK60bjgo109000bcg146c0.1 Record-Route: Record-Route: From: ;tag=4B718C43-21AEAF8-3E86ED92 To: ;tag=as4e3d3282 Call-ID: 4B718C43-00240BEA@DDUS0_PCU-004 CSeq: 1 INVITE User-Agent: asterisk 1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 436 436 IN IP4 D.D.D.D s=session c=IN IP4 D.D.D.D t=0 0 m=audio 13240 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/bt.com-08238b28 and SIP/faxserver-0823df78 estpbx01*CLI> <--- SIP read from A.A.A.A:5060 ---> ACK sip:123456789@D.D.D.D;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK5521.d23b8943.2 Via: SIP/2.0/UDP E.E.E.E;branch=z9hG4bK5521.46993d06.2 Via: SIP/2.0/UDP F.F.F.F:5060;branch=z9hG4bK7ghki120b04ggc09t4g1.1 From: ;tag=4B718C43-21AEAF8-3E86ED92 To: ;tag=as4e3d3282 Call-ID: 4B718C43-00240BEA@DDUS0_PCU-004 CSeq: 1 ACK Max-Forwards: 26 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- estpbx01*CLI> <--- SIP read from C.C.C.C:5060 ---> INVITE sip:123456789@D.D.D.D SIP/2.0 Via: SIP/2.0/UDP C.C.C.C:5060;branch=z9hG4bK1e1e6698;rport Max-Forwards: 70 From: ;tag=as774ab834 To: "123456789" ;tag=as31e2cce0 Contact: Call-ID: 03f0bc4832e47e87403a6ee47bf704f2@D.D.D.D CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.1-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 347 v=0 o=root 1235114045 1235114046 IN IP4 C.C.C.C s=Asterisk PBX 1.6.2.1-1 c=IN IP4 C.C.C.C t=0 0 m=image 4125 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (14 headers 14 lines) --- Sending to C.C.C.C : 5060 (NAT) Got T.38 offer in SDP in dialog 03f0bc4832e47e87403a6ee47bf704f2@D.D.D.D Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 03f0bc4832e47e87403a6ee47bf704f2@D.D.D.D Capabilities: us - 0x8 (alaw), 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 (NAT) to C.C.C.C:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP C.C.C.C:5060;branch=z9hG4bK1e1e6698;received=C.C.C.C;rport=5060 From: ;tag=as774ab834 To: "123456789" ;tag=as31e2cce0 Call-ID: 03f0bc4832e47e87403a6ee47bf704f2@D.D.D.D CSeq: 102 INVITE User-Agent: asterisk 1.4 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 A.A.A.A, port 5060 Reliably Transmitting (no NAT) to A.A.A.A:5060: INVITE sip:F.F.F.F:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP D.D.D.D:5060;branch=z9hG4bK0d127125;rport Route: , From: ;tag=as4e3d3282 To: ;tag=4B718C43-21AEAF8-3E86ED92 Contact: Call-ID: 4B718C43-00240BEA@DDUS0_PCU-004 CSeq: 102 INVITE User-Agent: asterisk 1.4 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: 354 v=0 o=root 436 437 IN IP4 D.D.D.D s=session c=IN IP4 D.D.D.D t=0 0 m=image 4356 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1400 a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy --- estpbx01*CLI> <--- SIP read from A.A.A.A:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP D.D.D.D:5060;branch=z9hG4bK0d127125;rport=5060 From: ;tag=as4e3d3282 To: ;tag=4B718C43-21AEAF8-3E86ED92 Call-ID: 4B718C43-00240BEA@DDUS0_PCU-004 CSeq: 102 INVITE Server: OpenSIPS (1.4.5-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- estpbx01*CLI> <--- SIP read from A.A.A.A:5060 ---> SIP/2.0 606 Not Acceptable Via: SIP/2.0/UDP D.D.D.D:5060;received=D.D.D.D;branch=z9hG4bK0d127125;rport=5060 From: ;tag=as4e3d3282 To: ;tag=4B718C43-21AEAF8-3E86ED92 Call-ID: 4B718C43-00240BEA@DDUS0_PCU-004 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- -- Got SIP response 606 "Not Acceptable" back from A.A.A.A set_destination: Parsing for address/port to send to set_destination: set destination to A.A.A.A, port 5060 Transmitting (no NAT) to A.A.A.A:5060: ACK sip:F.F.F.F:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP D.D.D.D:5060;branch=z9hG4bK0d127125;rport Route: , From: ;tag=as4e3d3282 To: ;tag=4B718C43-21AEAF8-3E86ED92 Contact: Call-ID: 4B718C43-00240BEA@DDUS0_PCU-004 CSeq: 102 ACK User-Agent: asterisk 1.4 Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '03f0bc4832e47e87403a6ee47bf704f2@D.D.D.D' in 32000 ms (Method: INVITE) == Spawn extension (from-inbound, 123456789, 1) exited non-zero on 'SIP/bt.com-08238b28'