Parsing /etc/asterisk/extconfig.conf Asterisk 1.2.5, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details. ========================================================================= [ Booting........................................................ ] Asterisk Ready. ]1;Asterisk]2;Asterisk Console on 'localhost.localdomain' (pid 25186)*CLI> sip debug SIP Debugging enabled *CLI> <-- SIP read from 209.247.16.1:33270: INVITE sip:+18666777930@10.30.10.50:5060 SIP/2.0 Via: SIP/2.0/UDP 209.247.16.1:5060;branch=z9hG4bK50603522629633-1152823875132 From: ;tag=VPSF50603522629633 To: Call-ID: DEN05020060714181117006219@209.244.48.214 CSeq: 1 INVITE Contact: Max-Forwards: 69 Content-Type: application/sdp Content-Length: 171 Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=- 1152900677 1152900678 IN IP4 63.215.27.94 s=- c=IN IP4 63.215.27.94 t=0 0 m=audio 60436 RTP/AVP 0 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (11 headers 8 lines)--- Transmitting (no NAT) to 209.247.16.1:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.247.16.1:5060;branch=z9hG4bK50603522629633-1152823875132;received=209.247.16.1 From: ;tag=VPSF50603522629633 To: Call-ID: DEN05020060714181117006219@209.244.48.214 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Using INVITE request as basis request - DEN05020060714181117006219@209.244.48.214 Sending to 209.247.16.1 : 5060 (non-NAT) Found no matching peer or user for '209.247.16.1:33270' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 63.215.27.94:60436 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for +18666777930 in default (domain 10.30.10.50) list_route: hop: We're at 10.30.10.50 port 13076 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 209.247.16.1:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.247.16.1:5060;branch=z9hG4bK50603522629633-1152823875132;received=209.247.16.1 From: ;tag=VPSF50603522629633 To: ;tag=as720ccf01 Call-ID: DEN05020060714181117006219@209.244.48.214 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 214 v=0 o=root 25186 25186 IN IP4 10.30.10.50 s=session c=IN IP4 10.30.10.50 t=0 0 m=audio 13076 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- <-- SIP read from 209.247.16.1:33270: ACK sip:+18666777930@10.30.10.50 SIP/2.0 Via: SIP/2.0/UDP 209.247.16.1:5060;branch=z9hG4bK50603522629633-1152823875133 From: ;tag=VPSF50603522629633 To: ;tag=as720ccf01 Call-ID: DEN05020060714181117006219@209.244.48.214 CSeq: 1 ACK Contact: Max-Forwards: 70 Content-Length: 0 --- (9 headers 0 lines)--- We're at 10.30.10.50 port 13080 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 209.247.16.1:5060: INVITE sip:+19052198209@209.247.16.1 SIP/2.0 Via: SIP/2.0/UDP 10.30.10.50:5060;branch=z9hG4bK4e1b0da3;rport From: "+19052198267" ;tag=as04a2d44e To: Contact: Call-ID: 3586b5bf3d07b7ae4b072a8e3f62f70c@10.30.10.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 14 Jul 2006 18:04:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 25186 25186 IN IP4 10.30.10.50 s=session c=IN IP4 10.30.10.50 t=0 0 m=audio 13080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- <-- SIP read from 209.247.16.1:33270: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.30.10.50:5060;rport=5060;branch=z9hG4bK4e1b0da3;received=10.30.10.50 From: "+19052198267" ;tag=as04a2d44e To: Call-ID: 3586b5bf3d07b7ae4b072a8e3f62f70c@10.30.10.50 CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines)--- <-- SIP read from 209.247.16.1:33270: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.30.10.50:5060;rport=5060;branch=z9hG4bK4e1b0da3;received=10.30.10.50 From: "+19052198267" ;tag=as04a2d44e To: ;tag=VPST50603522629633 Call-ID: 3586b5bf3d07b7ae4b072a8e3f62f70c@10.30.10.50 CSeq: 102 INVITE Contact: MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=level3-viper-boundary Content-Length: 180 --level3-viper-boundary Content-Type: application/sdp v=0 o=- 1152900684 1152900685 IN IP4 209.247.5.45 s=- c=IN IP4 209.247.5.45 t=0 0 m=audio 60026 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (10 headers 12 lines)--- <-- SIP read from 209.247.16.1:33270: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.10.50:5060;rport=5060;branch=z9hG4bK4e1b0da3;received=10.30.10.50 From: "+19052198267" ;tag=as04a2d44e To: ;tag=VPST50603522629633 Call-ID: 3586b5bf3d07b7ae4b072a8e3f62f70c@10.30.10.50 CSeq: 102 INVITE Contact: MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=level3-viper-boundary Content-Length: 180 --level3-viper-boundary Content-Type: application/sdp v=0 o=- 1152900684 1152900685 IN IP4 209.247.5.45 s=- c=IN IP4 209.247.5.45 t=0 0 m=audio 60026 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (10 headers 12 lines)--- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 209.247.16.1, port 5060 Transmitting (no NAT) to 209.247.16.1:5060: ACK sip:+19052198209@209.247.16.1:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.30.10.50:5060;branch=z9hG4bK029b2cce;rport From: "+19052198267" ;tag=as04a2d44e To: ;tag=VPST50603522629633 Contact: Call-ID: 3586b5bf3d07b7ae4b072a8e3f62f70c@10.30.10.50 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- sip no debug SIP Debugging Disabled *CLI> stop now