Use 'exit' when done Asterisk 1.4.2, Copyright (C) 1999 - 2006 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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found  == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.2 currently running on demobox (pid = 7519) demobox*CLI> Verbosity was 0 and is now 5 demobox*CLI> sip set debug demobox*CLI> SIP Debugging enabled demobox*CLI> set verbose 9 demobox*CLI> Verbosity was 5 and is now 9 demobox*CLI> The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead. demobox*CLI> core set verbose 9 demobox*CLI> Verbosity is at least 9 demobox*CLI> [May 2 07:59:26] DEBUG[7558]: manager.c:1984 process_message: Manager received command 'Login' demobox*CLI> == Parsing '/etc/asterisk/manager.conf': Found demobox*CLI> == Manager 'MessageNet' logged on from 127.0.0.1 demobox*CLI> [May 2 07:59:26] DEBUG[7558]: manager.c:1984 process_message: Manager received command 'Command' demobox*CLI> [May 2 07:59:26] DEBUG[7558]: manager.c:1984 process_message: Manager received command 'Command' demobox*CLI> [May 2 07:59:28] DEBUG[7558]: manager.c:1984 process_message: Manager received command 'Logoff' == Manager 'MessageNet' logged off from 127.0.0.1 demobox*CLI> -- Attempting call on SIP/522 for smvoice_callprogress@smvoice-dialout:1 (Retry 1) demobox*CLI> Video is at 192.168.1.150 port 18060 demobox*CLI> Audio is at 192.168.1.150 port 16714 demobox*CLI> Adding codec 0x4 (ulaw) to SDP demobox*CLI> Adding codec 0x8 (alaw) to SDP demobox*CLI> Adding codec 0x80000 (h263) to SDP demobox*CLI> Adding codec 0x200000 (h264) to SDP demobox*CLI> Adding non-codec 0x1 (telephone-event) to SDP demobox*CLI> Reliably Transmitting (no NAT) to 192.168.1.207:5060: INVITE sip:522@192.168.1.207:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK5402cdf8 From: "Kevin Brown 801" ;tag=as4935c00e To: Contact: Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 May 2007 11:59:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 363 v=0 o=root 7519 7519 IN IP4 192.168.1.150 s=session c=IN IP4 192.168.1.150 b=CT:384 t=0 0 m=audio 16714 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 18060 RTP/AVP 34 99 a=rtpmap:34 H263/90000 a=rtpmap:99 H264/90000 a=sendrecv --- demobox*CLI> <--- SIP read from 192.168.1.207:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK5402cdf8 From: "Kevin Brown 801" ;tag=as4935c00e To: Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 102 INVITE User-Agent: Grandstream GXV3000 1.0.0.27 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- demobox*CLI> <--- SIP read from 192.168.1.207:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK5402cdf8 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 102 INVITE User-Agent: Grandstream GXV3000 1.0.0.27 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- demobox*CLI> <--- SIP read from 192.168.1.207:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK5402cdf8 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 102 INVITE User-Agent: Grandstream GXV3000 1.0.0.27 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Supported: replaces, timer, 100rel Content-Length: 390 v=0 o=522 8000 8000 IN IP4 192.168.1.207 s=SIP Call c=IN IP4 192.168.1.207 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 5006 RTP/AVP 99 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4fII== a=framerate:15 <-------------> --- (12 headers 16 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found RTP video format 99 Peer audio RTP is at port 192.168.1.207:5004 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Found description format H264 for ID 99 Capabilities: us - 0x28000c (ulaw|alaw|h263|h264), peer - audio=0x200004 (ulaw|h264)/video=0x200000 (h264), combined - 0x200004 (ulaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.207:5004 Peer video RTP is at port 192.168.1.207:5006 list_route: hop: [May 2 07:59:34] DEBUG[7543]: chan_sip.c:5643 reqprep: Strict routing enforced for session 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.207, port 5060 Transmitting (no NAT) to 192.168.1.207:5060: ACK sip:522@192.168.1.207:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK001513fe From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Contact: Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- demobox*CLI> > Channel SIP/522-0986a310 was answered. -- Executing [smvoice_callprogress@smvoice-dialout:1] GotoIf("SIP/522-0986a310", "1?smvoice_callprogress|3:smvoice_callprogress|2") in new stack -- Goto (smvoice-dialout,smvoice_callprogress,3) -- Executing [smvoice_callprogress@smvoice-dialout:3] AGI("SIP/522-0986a310", "smvoice|-digium_asterisk") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice demobox*CLI> -- Playing '/tmp/smvoice.7565_0' (escape_digits=0123456789#) (sample_offset 0) demobox*CLI> [May 2 07:59:36] DEBUG[7544]: pbx_spool.c:391 scan_service: Delaying retry since we're currently running '/var/spool/asterisk/outgoing/agi.12.862559971' demobox*CLI> -- AGI Script smvoice completed, returning 0 -- Executing [smvoice_dial_no_extension@smvoice-dialout:1] Dial("SIP/522-0986a310", "SIP/532||tT") in new stack Video is at 192.168.1.150 port 13320 Audio is at 192.168.1.150 port 11828 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.206:5060: INVITE sip:532@192.168.1.206:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK00c08e0f From: "Kevin Brown 801" ;tag=as2d005538 To: Contact: Call-ID: 7f6fea193be0e8b05c89f155642fbfe1@192.168.1.150 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 May 2007 11:59:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 336 v=0 o=root 7519 7519 IN IP4 192.168.1.150 s=session c=IN IP4 192.168.1.150 b=CT:384 t=0 0 m=audio 11828 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 13320 RTP/AVP 99 a=rtpmap:99 H264/90000 a=sendrecv --- -- Called 532 demobox*CLI> <--- SIP read from 192.168.1.206:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK00c08e0f From: "Kevin Brown 801" ;tag=as2d005538 To: Call-ID: 7f6fea193be0e8b05c89f155642fbfe1@192.168.1.150 CSeq: 102 INVITE User-Agent: Grandstream GXV3000 1.0.0.27 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 192.168.1.206:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK00c08e0f From: "Kevin Brown 801" ;tag=as2d005538 To: ;tag=eaa756e5b543ad36 Call-ID: 7f6fea193be0e8b05c89f155642fbfe1@192.168.1.150 CSeq: 102 INVITE User-Agent: Grandstream GXV3000 1.0.0.27 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- demobox*CLI> -- SIP/532-09878b00 is ringing [May 2 07:59:38] DEBUG[7563]: chan_sip.c:5643 reqprep: Strict routing enforced for session 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.207, port 5060 Video is at 192.168.1.150 port 18060 Audio is at 192.168.1.150 port 16714 Adding codec 0x4 (ulaw) to SDP Adding codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.207:5060: INVITE sip:522@192.168.1.207:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK1b50c390 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Contact: Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 312 v=0 o=root 7519 7520 IN IP4 192.168.1.150 s=session c=IN IP4 192.168.1.150 b=CT:384 t=0 0 m=audio 16714 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 18060 RTP/AVP 99 a=rtpmap:99 H264/90000 a=sendrecv --- <--- SIP read from 192.168.1.207:5060 ---> SIP/2.0 200 OK V demobox*CLI> ia: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK1b50c390 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 103 INVITE User-Agent: Grandstream GXV3000 1.0.0.27 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Supported: replaces, timer, 100rel Content-Length: 390 v demobox*CLI> =0 o=522 8000 8001 IN IP4 192.168.1.207 s=SIP Call c=IN IP4 192.168.1.207 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 5006 RTP/AVP 99 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFClaBQfk,KClOBPI= a=framerate:15 <-------------> --- (12 headers 16 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found RTP video format 99 Peer audio RTP is at port 192.168.1.207:5004 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Found description format H264 for ID 99 Capabilities: us - 0x28000c (ulaw|alaw|h263|h264), peer - audio=0x200004 (ulaw|h264)/video=0x200000 (h264), combined - 0x200004 (ulaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.207:5004 Peer video RTP is at port 192.168.1.207:5006 [May 2 07:59:38] DEBUG[7543]: chan_sip.c:5643 reqprep: Strict routing enforced for session 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.207, port 5060 Transmitting (no NAT) to 192.168.1.207:5060: ACK sip:522@192.168.1.207:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK00e5eea0 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Contact: Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- demobox*CLI> -- SIP/522-0986a310 requested special control 17, passing it to SIP/532-09878b00 demobox*CLI> [May 2 07:59:39] DEBUG[7544]: pbx_spool.c:391 scan_service: Delaying retry since we're currently running '/var/spool/asterisk/outgoing/agi.12.862559971' demobox*CLI> <--- SIP read from 192.168.1.206:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK00c08e0f From: "Kevin Brown 801" ;tag=as2d005538 To: ;tag=eaa756e5b543ad36 Call-ID: 7f6fea193be0e8b05c89f155642fbfe1@192.168.1.150 CSeq: 102 INVITE User-Agent: Grandstream GXV3000 1.0.0.27 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Supported: replaces, timer, 100rel Content-Length: 390 v=0 o=532 8000 8000 IN IP4 192.168.1.206 s=SIP Call c=IN IP4 192.168.1.206 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 5006 RTP/AVP 99 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4CHIA= a=framerate:15 <-------------> --- (12 headers 16 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found RTP video format 99 Peer audio RTP is at port 192.168.1.206:5004 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Found description format H264 for ID 99 Capabilities: us - 0x20000c (ulaw|alaw|h264), peer - audio=0x200004 (ulaw|h264)/video=0x200000 (h264), combined - 0x200004 (ulaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.206:5004 Peer video RTP is at port 192.168.1.206:5006 list_route: hop: [May 2 07:59:40] DEBUG[7543]: chan_sip.c:5643 reqprep: Strict routing enforced for session 7f6fea193be0e8b05c89f155642fbfe1@192.168.1.150 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.206, port 5060 Transmitting (no NAT) to 192.168.1.206:5060: ACK sip:532@192.168.1.206:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK72f85ad3 From: "Kevin Brown 801" ;tag=as2d005538 To: ;tag=eaa756e5b543ad36 Contact: Call-ID: 7f6fea193be0e8b05c89f155642fbfe1@192.168.1.150 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/532-09878b00 left from hold -- SIP/532-09878b00 answered SIP/522-0986a310 [May 2 07:59:40] DEBUG[7563]: chan_sip.c:5643 reqprep: Strict routing enforced for session 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.207, port 5060 Video is at 192.168.1.150 port 18060 Audio is at 192.168.1.150 port 16714 Adding codec 0x4 (ulaw) to SDP Adding codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.207:5060: INVITE sip:522@192.168.1.207:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK3d58d233 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Contact: Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 310 v=0 o=root 7519 7521 IN IP4 192.168.1.206 s=session c=IN IP4 192.168.1.206 b=CT:384 t=0 0 m=audio 5004 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 5006 RTP/AVP 99 a=rtpmap:99 H264/90000 a=sendrecv --- demobox*CLI> <--- SIP read from 192.168.1.207:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK3d58d233 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 104 INVITE User-Agent: Grandstream GXV3000 1.0.0.27 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Supported: replaces, timer, 100rel Content-Length: 390 v=0 o=522 8000 8002 IN IP4 192.168.1.207 s=SIP Call c=IN IP4 192.168.1.207 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 5006 RTP/AVP 99 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4LyM== a=framerate:15 <-------------> --- (12 headers 16 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found RTP video format 99 Peer audio RTP is at port 192.168.1.207:5004 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Found description format H264 for ID 99 Capabilities: us - 0x28000c (ulaw|alaw|h263|h264), peer - audio=0x200004 (ulaw|h264)/video=0x200000 (h264), combined - 0x200004 (ulaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.207:5004 Peer video RTP is at port 192.168.1.207:5006 [May 2 07:59:40] DEBUG[7543]: chan_sip.c:5643 reqprep: Strict routing enforced for session 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.207, port 5060 Transmitting (no NAT) to 192.168.1.207:5060: ACK sip:522@192.168.1.207:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK2eb2b997 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Contact: Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- demobox*CLI> [May 2 07:59:42] DEBUG[7544]: pbx_spool.c:391 scan_service: Delaying retry since we're currently running '/var/spool/asterisk/outgoing/agi.12.862559971' demobox*CLI> <--- SIP read from 192.168.1.206:5060 ---> BYE sip:3175661677@192.168.1.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.206:5060;branch=z9hG4bK67b4e850f5a79d82 From: ;tag=eaa756e5b543ad36 To: "Kevin Brown 801" ;tag=as2d005538 Call-ID: 7f6fea193be0e8b05c89f155642fbfe1@192.168.1.150 CSeq: 6967 BYE User-Agent: Grandstream GXV3000 1.0.0.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 demobox*CLI> <-------------> --- (10 headers 0 lines) --- Sending to 192.168.1.206 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.206:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.206:5060;branch=z9hG4bK67b4e850f5a79d82;received=192.168.1.206 From: ;tag=eaa756e5b543ad36 To: "Kevin Brown 801" ;tag=as2d005538 Call-ID: 7f6fea193be0e8b05c89f155642fbfe1@192.168.1.150 CSeq: 6967 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> demobox*CLI> == Spawn extension (smvoice-dialout, smvoice_dial_no_extension, 1) exited non-zero on 'SIP/522-0986a310' Scheduling destruction of SIP dialog '58ac08f061f4637a0ee60d112c3f487d@192.168.1.150' in 32000 ms (Method: INVITE) [May 2 07:59:44] DEBUG[7563]: chan_sip.c:5643 reqprep: Strict routing enforced for session 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.207, port 5060 Reliably Transmitting (no NAT) to 192.168.1.207:5060: BYE sip:522@192.168.1.207:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK725faf18 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [May 2 07:59:44] NOTICE[7563]: pbx_spool.c:351 attempt_thread: Call completed to SIP/522 demobox*CLI> <--- SIP read from 192.168.1.207:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK725faf18 From: "Kevin Brown 801" ;tag=as4935c00e To: ;tag=f0c407c295217c36 Call-ID: 58ac08f061f4637a0ee60d112c3f487d@192.168.1.150 CSeq: 105 BYE User-Agent: Grandstream GXV3000 1.0.0.27 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '7f6fea193be0e8b05c89f155642fbfe1@192.168.1.150' Method: BYE Really destroying SIP dialog '58ac08f061f4637a0ee60d112c3f487d@192.168.1.150' Method: INVITE demobox*CLI> uit demobox*CLI> No such command 'uit' (type 'help' for help) demobox*CLI> uiquit Executing last minute cleanups Asterisk cleanly ending (0).