<--- 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=z9hG4bKd2c.ff6fdd01.0 To: From: "Owner of number";tag=551EA12D-474C10630003E348-5202BBB0 CSeq: 10 INVITE Call-ID: 2BA92085-474C10630003E358-5202BBB0@5.6.7.8 Content-Length: 843 Contact: Content-Type: application/sdp Max-Forwards: 64 User-Agent: SemsSys v=0 o=- 1673340270 0 IN IP4 11.22.33.44 s=- c=IN IP4 11.22.33.44 t=0 0 m=audio 4040 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 - 2BA92085-474C10630003E358-5202BBB0@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 11.22.33.44:4040 Found unknown media description format G726-16 for ID 97 Found unknown media description format G726-24 for ID 98 Found unknown media description format G726-40 for ID 99 Found unknown media description format X-G727-16 for ID 105 Found unknown media description format X-G727-24-16 for ID 106 Found unknown media description format X-G727-24 for ID 107 Found unknown media description format X-G727-32-16 for ID 108 Found unknown media description format X-G727-32-24 for ID 109 Found unknown media description format X-G727-32 for ID 110 Found unknown media description format X-G727-40-16 for ID 111 Found unknown media description format X-G727-40-24 for ID 112 Found unknown media description format X-G727-40-32 for ID 113 Found audio description format G723 for ID 4 Found audio description format G723 for ID 80 Found unknown media description format X-CCD for ID 116 Found audio description format telephone-event for ID 101 Found unknown media description format RED for ID 104 Found unknown media description format no-op for ID 120 Capabilities: us - 0x408 (alaw|ilbc), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 11.22.33.44:4040 Looking for +46123456789 in inbound (domain 1.2.3.4) list_route: hop: <--- Transmitting (no NAT) to 5.6.7.8:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKd2c.ff6fdd01.0;received=5.6.7.8 From: "Owner of number";tag=551EA12D-474C10630003E348-5202BBB0 To: Call-ID: 2BA92085-474C10630003E358-5202BBB0@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/5.6.7.8-08a6b920", "0?15") in new stack -- Executing [+46123456789@inbound:2] GotoIf("SIP/5.6.7.8-08a6b920", "1?10") in new stack -- Goto (inbound,+46123456789,10) -- Executing [+46123456789@inbound:10] GotoIf("SIP/5.6.7.8-08a6b920", "1?20") in new stack -- Goto (inbound,+46123456789,20) -- Executing [+46123456789@inbound:20] GotoIf("SIP/5.6.7.8-08a6b920", "1?23") in new stack -- Goto (inbound,+46123456789,23) -- Executing [+46123456789@inbound:23] Goto("SIP/5.6.7.8-08a6b920", "fixed-route|+46123456789|1") in new stack -- Goto (fixed-route,+46123456789,1) -- Executing [+46123456789@fixed-route:1] NoOp("SIP/5.6.7.8-08a6b920", " Set(CALLERID(name)=+46987654321)") in new stack -- Executing [+46123456789@fixed-route:2] Set("SIP/5.6.7.8-08a6b920", "DN=+46123456789") in new stack -- Executing [+46123456789@fixed-route:3] Goto("SIP/5.6.7.8-08a6b920", "s|3") in new stack -- Goto (fixed-route,s,3) -- Executing [s@fixed-route:3] Dial("SIP/5.6.7.8-08a6b920", "SIP/openser/46123456789") in new stack Audio is at 1.2.3.4 port 57626 Adding codec 0x8 (alaw) to SDP Adding codec 0x400 (ilbc) 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=z9hG4bK72b1f3c6;rport From: "Owner of number" ;tag=as71e61215 To: Contact: Call-ID: 42708ed63a7377921a6b63cd6ca2bebf@1.2.3.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 27 Nov 2007 12:41:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 8902 8902 IN IP4 1.2.3.4 s=session c=IN IP4 1.2.3.4 t=0 0 m=audio 57626 RTP/AVP 8 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called openser/46123456789 <--- SIP read from 1.2.3.4:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK72b1f3c6;rport=5070 From: "Owner of number" ;tag=as71e61215 To: Call-ID: 42708ed63a7377921a6b63cd6ca2bebf@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=z9hG4bK72b1f3c6;rport=5070 From: "Owner of number" ;tag=as71e61215 To: Call-ID: 42708ed63a7377921a6b63cd6ca2bebf@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=z9hG4bK72b1f3c6;rport=5070 From: "Owner of number" ;tag=as71e61215 To: "unknown" ;tag=3212654e622 Contact: Call-ID: 42708ed63a7377921a6b63cd6ca2bebf@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-08a70108 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=z9hG4bKd2c.ff6fdd01.0;received=5.6.7.8 From: "Owner of number";tag=551EA12D-474C10630003E348-5202BBB0 To: ;tag=as17f51879 Call-ID: 2BA92085-474C10630003E358-5202BBB0@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=z9hG4bK72b1f3c6;rport=5070 From: "Owner of number" ;tag=as71e61215 To: "unknown" ;tag=3212654e622 Contact: Call-ID: 42708ed63a7377921a6b63cd6ca2bebf@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=- 3405156065 3405156065 IN IP4 10.10.4.18 s=SJphone c=IN IP4 1.2.3.4 t=0 0 m=audio 60236 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 1.2.3.4:60236 Found audio description format iLBC for ID 97 Found audio 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 1.2.3.4:60236 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=z9hG4bK46e16fae;rport Route: From: "Owner of number" ;tag=as71e61215 To: ;tag=3212654e622 Contact: Call-ID: 42708ed63a7377921a6b63cd6ca2bebf@1.2.3.4 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/openser-08a70108 answered SIP/5.6.7.8-08a6b920 Audio is at 1.2.3.4 port 59976 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- 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=z9hG4bKd2c.ff6fdd01.0;received=5.6.7.8 From: "Owner of number";tag=551EA12D-474C10630003E348-5202BBB0 To: ;tag=as17f51879 Call-ID: 2BA92085-474C10630003E358-5202BBB0@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: 240 v=0 o=root 8902 8902 IN IP4 1.2.3.4 s=session c=IN IP4 1.2.3.4 t=0 0 m=audio 59976 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 <------------> 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=z9hG4bKd2c.ff6fdd01.0 From: "Owner of number";tag=551EA12D-474C10630003E348-5202BBB0 Call-ID: 2BA92085-474C10630003E358-5202BBB0@5.6.7.8 To: ;tag=as17f51879 CSeq: 10 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- asl004*CLI> <--- SIP read from 1.2.3.4:5060 ---> BYE sip:+46987654321@1.2.3.4:5070 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK0301.2867ec3.0 Via: SIP/2.0/UDP 10.10.4.18;received=22.33.44.55;branch=z9hG4bK0a0a041200000084474c106800006f2500000040;rport=63197 From: "unknown" ;tag=3212654e622 To: ;tag=as71e61215 Contact: Call-ID: 42708ed63a7377921a6b63cd6ca2bebf@1.2.3.4 CSeq: 1 BYE Max-Forwards: 69 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 Supported: replaces,norefersub,timer <-------------> --- (13 headers 0 lines) --- Sending to 1.2.3.4 : 5060 (no NAT) asl004*CLI> <--- Transmitting (no NAT) to 1.2.3.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK0301.2867ec3.0;received=1.2.3.4 Via: SIP/2.0/UDP 10.10.4.18;received=22.33.44.55;branch=z9hG4bK0a0a041200000084474c106800006f2500000040;rport=63197 Record-Route: From: "unknown" ;tag=3212654e622 To: ;tag=as71e61215 Call-ID: 42708ed63a7377921a6b63cd6ca2bebf@1.2.3.4 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (fixed-route, s, 3) exited non-zero on 'SIP/5.6.7.8-08a6b920' Scheduling destruction of SIP dialog '2BA92085-474C10630003E358-5202BBB0@5.6.7.8' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 5.6.7.8, port 5060 Reliably Transmitting (no NAT) to 5.6.7.8:5060: BYE sip:B13934464@5.6.7.8 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK48227297;rport From: ;tag=as17f51879 To: "Owner of number";tag=551EA12D-474C10630003E348-5202BBB0 Call-ID: 2BA92085-474C10630003E358-5202BBB0@5.6.7.8 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asl004*CLI> <--- SIP read from 5.6.7.8:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.2.3.4:5070;branch=z9hG4bK48227297;rport=5070 From: ;tag=as17f51879 To: "Owner of number";tag=551EA12D-474C10630003E348-5202BBB0 Call-ID: 2BA92085-474C10630003E358-5202BBB0@5.6.7.8 CSeq: 102 BYE Supported: timer Session-Expires: 1800 Server: SemsSys Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '42708ed63a7377921a6b63cd6ca2bebf@1.2.3.4' Method: BYE Really destroying SIP dialog '2BA92085-474C10630003E358-5202BBB0@5.6.7.8' Method: ACK asl004*CLI>