Asterisk SVN-trunk-r102700, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= NOTE: This is a development version of Asterisk, and should not be used in production installations. Connected to Asterisk SVN-trunk-r102700 currently running on laptop-asterisk (pid = 14316) laptop-asterisk*CLI> Verbosity is at least 9 laptop-asterisk*CLI> Core debug is at least 9 laptop-asterisk*CLI> <--- SIP read from UDP://192.168.0.31:5060 ---> INVITE sip:11111@192.168.0.32 SIP/2.0 Call-ID: 4547c8063349fa85320035a2ca6c94e3@192.168.0.31 CSeq: 1 INVITE From: "sipcon1@sip.omnitor.se" ;tag=1202319897693 To: Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK1202319897693 Max-Forwards: 70 Content-Type: application/sdp Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS Content-Length: 222 v=0 o=sipcon1@sip.omnitor.se 1 1 IN IP4 192.168.0.31 s=Omnitor_SDP_v1.1 c=IN IP4 192.168.0.31 t=0 0 m=audio 1032 RTP/AVP 5 4 3 0 a=rtpmap:5 DVI4/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 <-------------> --- (11 headers 10 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.0.31 : 5060 (no NAT) Using INVITE request as basis request - 4547c8063349fa85320035a2ca6c94e3@192.168.0.31 No user 'sipcon1' in SIP users list No matching peer for 'sipcon1' from '192.168.0.31:5060' Found RTP audio format 5 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 0 Peer audio RTP is at port 192.168.0.31:1032 Found audio description format DVI4 for ID 5 Found audio description format G723 for ID 4 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x27 (g723|gsm|ulaw|adpcm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.0.31:1032 Looking for 11111 in incoming-from-sip (domain 192.168.0.32) list_route: hop: <--- Transmitting (no NAT) to 192.168.0.31:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK1202319897693;received=192.168.0.31 From: "sipcon1@sip.omnitor.se" ;tag=1202319897693 To: Call-ID: 4547c8063349fa85320035a2ca6c94e3@192.168.0.31 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r102700 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [11111@incoming-from-sip:1] Dial("SIP/192.168.0.1-082296f8", "SIP/allanec@192.168.0.1") in new stack == Using SIP RTP CoS mark 5 laptop-asterisk*CLI> Audio is at 192.168.0.32 port 13086 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 192.168.0.1:5060: INVITE sip:allanec@192.168.0.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.32:5060;branch=z9hG4bK77a2f226;rport Max-Forwards: 70 From: "sipcon1@sip.omnitor.se" ;tag=as60730ae7 To: Contact: Call-ID: 15edfd7c15ed4bc4089d923d5bf07515@192.168.0.32 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r102700 Date: Wed, 06 Feb 2008 17:41:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 241 v=0 o=root 1260628979 1260628979 IN IP4 192.168.0.32 s=Asterisk PBX SVN-trunk-r102700 c=IN IP4 192.168.0.32 t=0 0 m=audio 13086 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- laptop-asterisk*CLI> -- Called allanec@192.168.0.1 laptop-asterisk*CLI> <--- SIP read from UDP://192.168.0.1:5060 ---> SIP/2.0 100 Trying From: "sipcon1@sip.omnitor.se";tag=as60730ae7 To: Call-ID: 15edfd7c15ed4bc4089d923d5bf07515@192.168.0.32 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.32:5060;received=192.168.0.32;rport=5060;branch=z9hG4bK77a2f226 Supported: 100rel,replaces Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- laptop-asterisk*CLI> <--- SIP read from UDP://192.168.0.1:5060 ---> SIP/2.0 180 Ringing From: "sipcon1@sip.omnitor.se";tag=as60730ae7 To: ;tag=22e3bc0-2100a8c0-13c4-8f25-62114de3-8f25 Call-ID: 15edfd7c15ed4bc4089d923d5bf07515@192.168.0.32 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.32:5060;received=192.168.0.32;rport=5060;branch=z9hG4bK77a2f226 Supported: 100rel,replaces Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- laptop-asterisk*CLI> -- SIP/192.168.0.1-0822da78 is ringing <--- Transmitting (no NAT) to 192.168.0.31:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK1202319897693;received=192.168.0.31 From: "sipcon1@sip.omnitor.se" ;tag=1202319897693 To: ;tag=as643633ab Call-ID: 4547c8063349fa85320035a2ca6c94e3@192.168.0.31 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r102700 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> laptop-asterisk*CLI> Audio is at 192.168.0.32 port 15716 Adding codec 0x4 (ulaw) to SDP <--- Transmitting (no NAT) to 192.168.0.31:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK1202319897693;received=192.168.0.31 From: "sipcon1@sip.omnitor.se" ;tag=1202319897693 To: ;tag=as643633ab Call-ID: 4547c8063349fa85320035a2ca6c94e3@192.168.0.31 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r102700 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 215 v=0 o=root 569198848 569198848 IN IP4 192.168.0.32 s=Asterisk PBX SVN-trunk-r102700 c=IN IP4 192.168.0.32 t=0 0 m=audio 15716 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> laptop-asterisk*CLI> <--- SIP read from UDP://192.168.0.1:5060 ---> SIP/2.0 200 OK From: "sipcon1@sip.omnitor.se";tag=as60730ae7 To: ;tag=22e3bc0-2100a8c0-13c4-8f25-62114de3-8f25 Call-ID: 15edfd7c15ed4bc4089d923d5bf07515@192.168.0.32 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.32:5060;received=192.168.0.32;rport=5060;branch=z9hG4bK77a2f226 Supported: 100rel,replaces User-Agent: FranceTelecom/eConf Accept: application/sdp,audio/telephone-event,text/plain,text/html,application/media_control+xml,application/mc+xml,application/dtmf-relay,message/sipfrag Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,PRACK,INFO,MESSAGE,SUBSCRIBE,NOTIFY Content-Type: application/SDP Content-Length: 200 v=0 o=anonymous 1202319724 1202319717 IN IP4 192.168.0.33 s=- i=eConf 4.1 c=IN IP4 192.168.0.33 b=AS:384 t=0 0 m=audio 6000 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=sendrecv <-------------> --- (13 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 8 Peer audio RTP is at port 192.168.0.33:6000 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.0.33:6000 --- set_address_from_contact host '192.168.0.33' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.33, port 5060 Transmitting (no NAT) to 192.168.0.33:5060: ACK sip:allanec@192.168.0.33 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.32:5060;branch=z9hG4bK34ad034f;rport Max-Forwards: 70 From: "sipcon1@sip.omnitor.se" ;tag=as60730ae7 To: ;tag=22e3bc0-2100a8c0-13c4-8f25-62114de3-8f25 Contact: Call-ID: 15edfd7c15ed4bc4089d923d5bf07515@192.168.0.32 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r102700 Content-Length: 0 --- laptop-asterisk*CLI> -- SIP/192.168.0.1-0822da78 answered SIP/192.168.0.1-082296f8 Audio is at 192.168.0.32 port 15716 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 192.168.0.31:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK1202319897693;received=192.168.0.31 From: "sipcon1@sip.omnitor.se" ;tag=1202319897693 To: ;tag=as643633ab Call-ID: 4547c8063349fa85320035a2ca6c94e3@192.168.0.31 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r102700 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 215 v=0 o=root 569198848 569198849 IN IP4 192.168.0.32 s=Asterisk PBX SVN-trunk-r102700 c=IN IP4 192.168.0.32 t=0 0 m=audio 15716 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> laptop-asterisk*CLI> <--- SIP read from UDP://192.168.0.31:5060 ---> ACK sip:11111@192.168.0.32:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bKd73e22a8b201fb614d8912a3194f1e93 CSeq: 1 ACK From: "sipcon1@sip.omnitor.se" ;tag=1202319897693 To: ;tag=as643633ab Call-ID: 4547c8063349fa85320035a2ca6c94e3@192.168.0.31 User-Agent: Asterisk/PBX/SVN-trunk-r102700 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Supported: replaces,timer Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- laptop-asterisk*CLI> -- Packet2Packet bridging SIP/192.168.0.1-082296f8 and SIP/192.168.0.1-0822da78 laptop-asterisk*CLI> <--- SIP read from UDP://192.168.0.31:5060 ---> BYE sip:11111@192.168.0.32:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK17703d47c9fdcbea0bc9f1e34b85af16 CSeq: 2 BYE From: "sipcon1@sip.omnitor.se" ;tag=1202319897693 To: ;tag=as643633ab Call-ID: 4547c8063349fa85320035a2ca6c94e3@192.168.0.31 User-Agent: Asterisk/PBX/SVN-trunk-r102700 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Supported: replaces,timer Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.0.31 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.0.31:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK17703d47c9fdcbea0bc9f1e34b85af16;received=192.168.0.31 From: "sipcon1@sip.omnitor.se" ;tag=1202319897693 To: ;tag=as643633ab Call-ID: 4547c8063349fa85320035a2ca6c94e3@192.168.0.31 CSeq: 2 BYE User-Agent: Asterisk PBX SVN-trunk-r102700 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> laptop-asterisk*CLI> Scheduling destruction of SIP dialog '15edfd7c15ed4bc4089d923d5bf07515@192.168.0.32' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.33, port 5060 Reliably Transmitting (no NAT) to 192.168.0.33:5060: BYE sip:allanec@192.168.0.33 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.32:5060;branch=z9hG4bK0837d5ae;rport Max-Forwards: 70 From: "sipcon1@sip.omnitor.se" ;tag=as60730ae7 To: ;tag=22e3bc0-2100a8c0-13c4-8f25-62114de3-8f25 Call-ID: 15edfd7c15ed4bc4089d923d5bf07515@192.168.0.32 CSeq: 103 BYE User-Agent: Asterisk PBX SVN-trunk-r102700 Content-Length: 0 --- laptop-asterisk*CLI> == Spawn extension (incoming-from-sip, 11111, 1) exited non-zero on 'SIP/192.168.0.1-082296f8' laptop-asterisk*CLI> <--- SIP read from UDP://192.168.0.33:5060 ---> SIP/2.0 200 OK From: "sipcon1@sip.omnitor.se";tag=as60730ae7 To: ;tag=22e3bc0-2100a8c0-13c4-8f25-62114de3-8f25 Call-ID: 15edfd7c15ed4bc4089d923d5bf07515@192.168.0.32 CSeq: 103 BYE Via: SIP/2.0/UDP 192.168.0.32:5060;rport=5060;branch=z9hG4bK0837d5ae Supported: 100rel,replaces User-Agent: FranceTelecom/eConf Accept: application/sdp,audio/telephone-event,text/plain,text/html,application/media_control+xml,application/mc+xml,application/dtmf-relay,message/sipfrag Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '15edfd7c15ed4bc4089d923d5bf07515@192.168.0.32' Method: INVITE Really destroying SIP dialog '4547c8063349fa85320035a2ca6c94e3@192.168.0.31' Method: BYE laptop-asterisk*CLI> stop now laptop-asterisk*CLI>