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. ========================================================================= Connected to Asterisk 1.4.2 currently running on browan_ast (pid = 29449) welles*CLI> -- Remote UNIX connection Verbosity is at least 3 [welles*CLI> <--- SIP read from 192.168.1.10:5060 ---> INVITE sip:1000@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;rport;branch=z9hG4bK453922887 Route: From: welles ;tag=1879699136 To: Call-ID: 3510083655@192.168.1.10 CSeq: 20 INVITE Contact: Max-Forwards: 70 User-Agent: eXosip/0.1 Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=userX 20000001 20000001 IN IP4 192.168.1.10 s=A call c=IN IP4 192.168.1.10 t=1175220371 1175223971 m=audio 10600 RTP/AVP 124 111 8 0 123 3 101 a=rtpmap:124 AMR-WB/16000/1 a=rtpmap:111 ILBC/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:123 AMR/8000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:101 telephone-event/8000/1 a=AS:111 20 m=video 10700 RTP/AVP 34 a=rtpmap:34 H263/90000/1 <-------------> --- (14 headers 16 lines) --- Sending to 192.168.1.10 : 5060 (NAT) Using INVITE request as basis request - 3510083655@192.168.1.10 <--- Reliably Transmitting (no NAT) to 192.168.1.10:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK453922887;received=192.168.1.10;rport=5060 From: welles ;tag=1879699136 To: ;tag=as71642bc1 Call-ID: 3510083655@192.168.1.10 CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7b2745ad" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3510083655@192.168.1.10' in 32000 ms (Method: INVITE) Found user 'welles' [welles*CLI> <--- SIP read from 192.168.1.10:5060 ---> ACK sip:1000@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;rport;branch=z9hG4bK453922887 Route: From: welles ;tag=1879699136 To: ;tag=as71642bc1 Call-ID: 3510083655@192.168.1.10 CSeq: 20 ACK Content-Length: 0 <-------------> [[welles*CLI> --- (8 headers 0 lines) --- [[welles*CLI> <--- SIP read from 192.168.1.10:5060 ---> INVITE sip:1000@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;rport;branch=z9hG4bK3694283843 Route: From: welles ;tag=1879699136 To: Call-ID: 3510083655@192.168.1.10 CSeq: 21 INVITE Contact: Proxy-Authorization: Digest username="welles", realm="asterisk", nonce="7b2745ad", uri="sip:1000@192.168.1.102", response="88d858703d684015ecd52397b3270512", algorithm=MD5 Max-Forwards: 70 User-Agent: eXosip/0.1 Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 408 v=0 o=userX 20000001 20000001 IN IP4 192.168.1.10 s=A call c=IN IP4 192.168.1.10 t=1175220371 1175223971 m=audio 10600 RTP/AVP 124 111 8 0 123 3 101 a=rtpmap:124 AMR-WB/16000/1 a=rtpmap:111 ILBC/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:123 AMR/8000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:101 telephone-event/8000/1 a=AS:111 20 m=video 10700 RTP/AVP 34 a=rtpmap:34 H263/90000/1 <-------------> [[welles*CLI> --- (15 headers 16 lines) --- [[welles*CLI> Sending to 192.168.1.10 : 5060 (NAT) [[welles*CLI> Using INVITE request as basis request - 3510083655@192.168.1.10 [[welles*CLI> Found user 'welles' [[welles*CLI> Found RTP audio format 124 [welles*CLI> Found RTP audio format 111 Found RTP audio format 8 [[welles*CLI> Found RTP audio format 0 Found RTP audio format 123 Found RTP audio format 3 [[welles*CLI>Found RTP audio format 101 Found RTP video format 34 [[welles*CLI> Peer audio RTP is at port 192.168.1.10:10600 Found description format AMR-WB for ID 124 [[welles*CLI> Found description format ILBC for ID 111 Found description format PCMA for ID 8 [[welles*CLI> Found description format PCMU for ID 0 Found description format AMR for ID 123 [[welles*CLI> Found description format GSM for ID 3 Found description format telephone-event for ID 101 [[welles*CLI> Found description format H263 for ID 34 [[welles*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8040e (gsm|ulaw|alaw|ilbc|h263)/video=0x80000 (h263), combined - 0x8000e (gsm|ulaw|alaw|h263) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [[welles*CLI> Peer audio RTP is at port 192.168.1.10:10600 Looking for 1000 in test (domain 192.168.1.102) [[welles*CLI> list_route: hop: [[welles*CLI> <--- Transmitting (no NAT) to 192.168.1.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3694283843;received=192.168.1.10;rport=5060 From: welles ;tag=1879699136 To: Call-ID: 3510083655@192.168.1.10 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [[welles*CLI> -- Executing [1000@test:1] ^[[1;36;40mAnswer^[[0;37;40m("^[[1;35;40mSIP/welles-08944710^[[0;37;40m", "^[[1;35;40m^[[0;37;40m") in new stack [[welles*CLI> Audio is at 192.168.1.102 port 15332 [[welles*CLI> Adding codec 0x2 (gsm) to SDP [[welles*CLI> Adding codec 0x4 (ulaw) to SDP [[welles*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [[welles*CLI> <--- Reliably Transmitting (no NAT) to 192.168.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3694283843;received=192.168.1.10;rport=5060 From: welles ;tag=1879699136 To: ;tag=as668a2d77 Call-ID: 3510083655@192.168.1.10 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp C^[[welles*CLI> ontent-Length: 289 v=0 o=root 29449 29449 IN IP4 192.168.1.102 s=session c=IN IP4 192.168.1.102 t=0 0 m=audio 15332 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 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 <------------> [[welles*CLI> -- Executing [1000@test:2] ^[[1;36;40mBackGround^[[0;37;40m("^[[1;35;40mSIP/welles-08944710^[[0;37;40m", "^[[1;35;40mdemo-congrats^[[0;37;40m") in new stack [[welles*CLI> -- Playing 'demo-congrats' (language 'en') [[welles*CLI> <--- SIP read from 192.168.1.10:5060 ---> ACK sip:1000@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;rport;branch=z9hG4bK2515317780 From: welles ;tag=1879699136 To: ;tag=as668a2d77 Call-ID: 3510083655@192.168.1.10 CSeq: 21 ACK Contact: Max-Forwards: 70 User-Agent: eXosip/0.1 Content-Length: 0 <-------------> [[welles*CLI> --- (10 headers 0 lines) --- [[welles*CLI> -- Executing [1000@test:3] ^[[1;36;40mFestival^[[0;37;40m("^[[1;35;40mSIP/welles-08944710^[[0;37;40m", "^[[1;35;40mhow are you^[[0;37;40m") in new stack == Parsing '/etc/asterisk/festival.conf': Found [[welles*CLI> -- Executing [1000@test:4] ^[[1;36;40mFestival^[[0;37;40m("^[[1;35;40mSIP/welles-08944710^[[0;37;40m", "^[[1;35;40masterisk^[[0;37;40m") in new stack [[welles*CLI> == Parsing '/etc/asterisk/festival.conf': Found [[welles*CLI> -- Executing [1000@test:5] ^[[1;36;40mFestival^[[0;37;40m("^[[1;35;40mSIP/welles-08944710^[[0;37;40m", "^[[1;35;40mthank you^[[0;37;40m") in new stack == Parsing '/etc/asterisk/festival.conf': Found [[welles*CLI> -- Executing [1000@test:6] ^[[1;36;40mFestival^[[0;37;40m("^[[1;35;40mSIP/welles-08944710^[[0;37;40m", "^[[1;35;40mgood bye^[[0;37;40m") in new stack == Parsing '/etc/asterisk/festival.conf': Found [[welles*CLI> -- Executing [1000@test:7] ^[[1;36;40mHangup^[[0;37;40m("^[[1;35;40mSIP/welles-08944710^[[0;37;40m", "^[[1;35;40m^[[0;37;40m") in new stack [[welles*CLI> == Spawn extension (test, 1000, 7) exited non-zero on 'SIP/welles-08944710' [[welles*CLI> Scheduling destruction of SIP dialog '3510083655@192.168.1.10' in 32000 ms (Method: ACK) [[welles*CLI> set_destination: Parsing for address/port to send to [[welles*CLI> set_destination: set destination to 192.168.1.10, port 5060 [[welles*CLI> Reliably Transmitting (no NAT) to 192.168.1.10:5060: BYE sip:welles@192.168.1.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3771dcdb;rport From: ;tag=as668a2d77 To: welles ;tag=1879699136 Call-ID: 3510083655@192.168.1.10 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [[welles*CLI> <--- SIP read from 192.168.1.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3771dcdb;rport=5060 From: ;tag=as668a2d77 To: welles ;tag=1879699136 Call-ID: 3510083655@192.168.1.10 CSeq: 102 BYE Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER Content-Length: 0 <-------------> [[welles*CLI> --- (8 headers 0 lines) --- [[welles*CLI> SIP Response message for INCOMING dialog BYE arrived [[welles*CLI> <--- SIP read from 192.168.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3771dcdb From: ;tag=as668a2d77 To: welles ;tag=1879699136 Call-ID: 3510083655@192.168.1.10 CSeq: 102 BYE Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER Content-Length: 0 <-------------> [[welles*CLI> --- (8 headers 0 lines) --- [[welles*CLI> SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '3510083655@192.168.1.10' Method: ACK [[welles*CLI>