<--- SIP read from 5.6.7.8:5060 ---> INVITE sip:+46123456789@1.2.3.4:5070 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKeb0c.69f789a4.0 To: From: "Owner of number";tag=2291338A-474BD4500001C934-4DC2ABB0 CSeq: 10 INVITE Call-ID: 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 Content-Length: 843 Contact: Content-Type: application/sdp Max-Forwards: 64 User-Agent: SemsSys v=0 o=- 2023717616 0 IN IP4 11.22.33.44 s=- c=IN IP4 11.22.33.44 t=0 0 m=audio 4210 RTP/AVP 0 8 97 98 2 99 105 106 107 108 109 110 111 112 113 4 80 18 3 116 101 104 13 120 a=rtpmap:97 G726-16/8000/1 a=rtpmap:98 G726-24/8000/1 a=rtpmap:99 G726-40/8000/1 a=rtpmap:105 X-G727-16/8000/1 a=rtpmap:106 X-G727-24-16/8000/1 a=rtpmap:107 X-G727-24/8000/1 a=rtpmap:108 X-G727-32-16/8000/1 a=rtpmap:109 X-G727-32-24/8000/1 a=rtpmap:110 X-G727-32/8000/1 a=rtpmap:111 X-G727-40-16/8000/1 a=rtpmap:112 X-G727-40-24/8000/1 a=rtpmap:113 X-G727-40-32/8000/1 a=rtpmap:4 G723/8000/1 a=fmtp:4 bitrate=6.3;annexa=yes a=rtpmap:80 G723/8000/1 a=fmtp:80 bitrate=5.3;annexa=yes a=fmtp:18 annexb=yes a=rtpmap:116 X-CCD/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:104 RED/8000 a=fmtp:104 a=rtpmap:120 no-op/8000 <-------------> --- (11 headers 29 lines) --- Sending to 5.6.7.8 : 5060 (no NAT) Using INVITE request as basis request - 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 Found peer 'pstn' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 2 Found RTP audio format 99 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 108 Found RTP audio format 109 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 113 Found RTP audio format 4 Found RTP audio format 80 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 116 Found RTP audio format 101 Found RTP audio format 104 Found RTP audio format 13 Found RTP audio format 120 Peer audio RTP is at port 11.22.33.44:4210 Found description format G726-16 for ID 97 Found description format G726-24 for ID 98 Found description format G726-40 for ID 99 Found description format X-G727-16 for ID 105 Found description format X-G727-24-16 for ID 106 Found description format X-G727-24 for ID 107 Found description format X-G727-32-16 for ID 108 Found description format X-G727-32-24 for ID 109 Found description format X-G727-32 for ID 110 Found description format X-G727-40-16 for ID 111 Found description format X-G727-40-24 for ID 112 Found description format X-G727-40-32 for ID 113 Found unknown description format G723 for ID 4 Found unknown description format G723 for ID 80 Found description format X-CCD for ID 116 Found unknown description format telephone-event for ID 101 Found description format RED for ID 104 Found description format no-op for ID 120 Capabilities: us - 0x408 (alaw|ilbc), peer - audio=0x200f1f (g723|gsm|ulaw|alaw|g726|g729|speex|ilbc|g726aal2|h264)/video=0x0 (nothing), combined - 0x408 (alaw|ilbc) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 11.22.33.44:4210 Looking for +46123456789 in inbound (domain 1.2.3.4) list_route: hop: asl004*CLI> <--- Transmitting (no NAT) to 5.6.7.8:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKeb0c.69f789a4.0;received=5.6.7.8 From: "Owner of number";tag=2291338A-474BD4500001C934-4DC2ABB0 To: Call-ID: 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 CSeq: 10 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asl004*CLI> <--- SIP read from 5.6.7.8:5060 ---> INVITE sip:+46123456789@1.2.3.4:5070 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKeb0c.69f789a4.0 To: From: "Owner of number";tag=2291338A-474BD4500001C934-4DC2ABB0 CSeq: 10 INVITE Call-ID: 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 Content-Length: 843 Contact: Content-Type: application/sdp Max-Forwards: 64 User-Agent: SemsSys v=0 o=- 2023717616 0 IN IP4 11.22.33.44 s=- c=IN IP4 11.22.33.44 t=0 0 m=audio 4210 RTP/AVP 0 8 97 98 2 99 105 106 107 108 109 110 111 112 113 4 80 18 3 116 101 104 13 120 a=rtpmap:97 G726-16/8000/1 a=rtpmap:98 G726-24/8000/1 a=rtpmap:99 G726-40/8000/1 a=rtpmap:105 X-G727-16/8000/1 a=rtpmap:106 X-G727-24-16/8000/1 a=rtpmap:107 X-G727-24/8000/1 a=rtpmap:108 X-G727-32-16/8000/1 a=rtpmap:109 X-G727-32-24/8000/1 a=rtpmap:110 X-G727-32/8000/1 a=rtpmap:111 X-G727-40-16/8000/1 a=rtpmap:112 X-G727-40-24/8000/1 a=rtpmap:113 X-G727-40-32/8000/1 a=rtpmap:4 G723/8000/1 a=fmtp:4 bitrate=6.3;annexa=yes a=rtpmap:80 G723/8000/1 a=fmtp:80 bitrate=5.3;annexa=yes a=fmtp:18 annexb=yes a=rtpmap:116 X-CCD/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:104 RED/8000 a=fmtp:104 a=rtpmap:120 no-op/8000 <-------------> --- (11 headers 29 lines) --- Ignoring this INVITE request asl004*CLI> <--- Transmitting (no NAT) to 5.6.7.8:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKeb0c.69f789a4.0;received=5.6.7.8 From: "Owner of number";tag=2291338A-474BD4500001C934-4DC2ABB0 To: Call-ID: 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 CSeq: 10 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [+46123456789@inbound:1] GotoIf("SIP/5.6.7.8-08d03ea8", "0?15") in new stack -- Executing [+46123456789@inbound:2] GotoIf("SIP/5.6.7.8-08d03ea8", "1?10") in new stack -- Goto (inbound,+46123456789,10) -- Executing [+46123456789@inbound:10] GotoIf("SIP/5.6.7.8-08d03ea8", "1?20") in new stack -- Goto (inbound,+46123456789,20) -- Executing [+46123456789@inbound:20] GotoIf("SIP/5.6.7.8-08d03ea8", "1?23") in new stack -- Goto (inbound,+46123456789,23) -- Executing [+46123456789@inbound:23] Goto("SIP/5.6.7.8-08d03ea8", "fixed-route|+46123456789|1") in new stack -- Goto (fixed-route,+46123456789,1) -- Executing [+46123456789@fixed-route:1] NoOp("SIP/5.6.7.8-08d03ea8", " Set(CALLERID(name)=+46987654321)") in new stack -- Executing [+46123456789@fixed-route:2] Set("SIP/5.6.7.8-08d03ea8", "DN=+46123456789") in new stack -- Executing [+46123456789@fixed-route:3] Goto("SIP/5.6.7.8-08d03ea8", "s|3") in new stack -- Goto (fixed-route,s,3) -- Executing [s@fixed-route:3] Dial("SIP/5.6.7.8-08d03ea8", "SIP/openser/46123456789") in new stack Audio is at 1.2.3.4 port 58956 Adding codec 0x400 (ilbc) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 1.2.3.4:5060: INVITE sip:46123456789@1.2.3.4 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK1aa768ec;rport From: "Owner of number" ;tag=as6d0f838d To: Contact: Call-ID: 690ca473176e690d3c01844e435c8ca4@1.2.3.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 27 Nov 2007 08:24:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 28521 28521 IN IP4 1.2.3.4 s=session c=IN IP4 1.2.3.4 t=0 0 m=audio 58956 RTP/AVP 97 8 101 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 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 openser/46123456789 asl004*CLI> <--- SIP read from 1.2.3.4:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK1aa768ec;rport=5070 From: "Owner of number" ;tag=as6d0f838d To: Call-ID: 690ca473176e690d3c01844e435c8ca4@1.2.3.4 CSeq: 102 INVITE Server: OpenSER (1.2.0-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asl004*CLI> <--- SIP read from 1.2.3.4:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK1aa768ec;rport=5070 From: "Owner of number" ;tag=as6d0f838d To: Call-ID: 690ca473176e690d3c01844e435c8ca4@1.2.3.4 CSeq: 102 INVITE Server: OpenSER (1.2.0-notls (i386/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asl004*CLI> <--- SIP read from 1.2.3.4:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK1aa768ec;rport=5070 From: "Owner of number" ;tag=as6d0f838d To: "unknown" ;tag=603856a3c57 Contact: Call-ID: 690ca473176e690d3c01844e435c8ca4@1.2.3.4 CSeq: 102 INVITE Content-Length: 0 Record-Route: Server: SJphone/1.65.377a (SJ Labs) <-------------> --- (10 headers 0 lines) --- -- SIP/openser-08d09af0 is ringing asl004*CLI> <--- Transmitting (no NAT) to 5.6.7.8:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKeb0c.69f789a4.0;received=5.6.7.8 From: "Owner of number";tag=2291338A-474BD4500001C934-4DC2ABB0 To: ;tag=as69919f11 Call-ID: 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 CSeq: 10 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asl004*CLI> <--- SIP read from 1.2.3.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK1aa768ec;rport=5070 From: "Owner of number" ;tag=as6d0f838d To: "unknown" ;tag=603856a3c57 Contact: Call-ID: 690ca473176e690d3c01844e435c8ca4@1.2.3.4 CSeq: 102 INVITE Content-Length: 249 Content-Type: application/sdp Record-Route: Server: SJphone/1.65.377a (SJ Labs) Supported: replaces,norefersub,timer v=0 o=- 3405140686 3405140686 IN IP4 10.10.4.18 s=SJphone c=IN IP4 1.2.3.4 t=0 0 m=audio 60220 RTP/AVP 97 101 c=IN IP4 1.2.3.4 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=setup:active a=sendrecv <-------------> --- (12 headers 12 lines) --- Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 1.2.3.4:60220 Found unknown description format iLBC for ID 97 Found unknown description format telephone-event for ID 101 Capabilities: us - 0x408 (alaw|ilbc), peer - audio=0x400 (ilbc)/video=0x0 (nothing), combined - 0x400 (ilbc) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 1.2.3.4:60220 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 1.2.3.4, port 5060 Transmitting (no NAT) to 1.2.3.4:5060: ACK sip:46123456789@22.33.44.55:63197 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK466c4f9d;rport Route: From: "Owner of number" ;tag=as6d0f838d To: ;tag=603856a3c57 Contact: Call-ID: 690ca473176e690d3c01844e435c8ca4@1.2.3.4 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/openser-08d09af0 answered SIP/5.6.7.8-08d03ea8 Audio is at 1.2.3.4 port 57272 Adding codec 0x400 (ilbc) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP asl004*CLI> <--- Reliably Transmitting (no NAT) to 5.6.7.8:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKeb0c.69f789a4.0;received=5.6.7.8 From: "Owner of number";tag=2291338A-474BD4500001C934-4DC2ABB0 To: ;tag=as69919f11 Call-ID: 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 CSeq: 10 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 28521 28521 IN IP4 1.2.3.4 s=session c=IN IP4 1.2.3.4 t=0 0 m=audio 57272 RTP/AVP 97 8 101 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 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/5.6.7.8-08d03ea8 and SIP/openser-08d09af0 asl004*CLI> <--- SIP read from 5.6.7.8:5060 ---> ACK sip:+46123456789@1.2.3.4:5070 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKeb0c.69f789a4.0 From: "Owner of number";tag=2291338A-474BD4500001C934-4DC2ABB0 Call-ID: 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 To: ;tag=as69919f11 CSeq: 10 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- asl004*CLI> <--- SIP read from 5.6.7.8:5060 ---> BYE sip:+46123456789@1.2.3.4:5070 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKfb0c.a0f64d44.0 To: ;tag=as69919f11 From: "Owner of number";tag=2291338A-474BD4500001C934-4DC2ABB0 CSeq: 11 BYE Call-ID: 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 Content-Length: 0 Max-Forwards: 64 User-Agent: SemsSys <-------------> --- (9 headers 0 lines) --- Sending to 5.6.7.8 : 5060 (no NAT) <--- Transmitting (no NAT) to 5.6.7.8:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKfb0c.a0f64d44.0;received=5.6.7.8 From: "Owner of number";tag=2291338A-474BD4500001C934-4DC2ABB0 To: ;tag=as69919f11 Call-ID: 21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8 CSeq: 11 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '690ca473176e690d3c01844e435c8ca4@1.2.3.4' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 1.2.3.4, port 5060 Reliably Transmitting (no NAT) to 1.2.3.4:5060: BYE sip:46123456789@22.33.44.55:63197 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK78ad3a0a;rport Route: From: "Owner of number" ;tag=as6d0f838d To: ;tag=603856a3c57 Call-ID: 690ca473176e690d3c01844e435c8ca4@1.2.3.4 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --l004*CLI> == Spawn extension (fixed-route, s, 3) exited non-zero on 'SIP/5.6.7.8-08d03ea8' asl004*CLI> <--- SIP read from 1.2.3.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK78ad3a0a;rport=5070 From: "Owner of number" ;tag=as6d0f838d To: "unknown" ;tag=603856a3c57 Contact: Call-ID: 690ca473176e690d3c01844e435c8ca4@1.2.3.4 CSeq: 103 BYE Content-Length: 0 Record-Route: Server: SJphone/1.65.377a (SJ Labs) Supported: replaces,norefersub,timer <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '690ca473176e690d3c01844e435c8ca4@1.2.3.4' Method: INVITE Really destroying SIP dialog '21B8285C-474BD4500001C942-4DC2ABB0@5.6.7.8' Method: BYE asl004*CLI>