Asterisk 1.6.1.0, 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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found == Binding queue_log to mysql/asterisk/queue_log Connected to Asterisk 1.6.1.0 currently running on debiasterisk (pid = 2000) debiasterisk*CLI> Verbosity is at least 5 debiasterisk*CLI> -- Remote UNIX connection debiasterisk*CLI> == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 debiasterisk*CLI> -- Executing [27@out-33:1] Set("SIP/33-090cd510", "fw_ext=") in new stack -- Executing [27@out-33:2] GotoIf("SIP/33-090cd510", "0?cfim") in new stack -- Executing [27@out-33:3] Set("SIP/33-090cd510", "fw_ext=") in new stack -- Executing [27@out-33:4] GotoIf("SIP/33-090cd510", "0?cfbs") in new stack -- Executing [27@out-33:5] Set("SIP/33-090cd510", "fw_ext=") in new stack -- Executing [27@out-33:6] GotoIf("SIP/33-090cd510", "0?cfto") in new stack -- Executing [27@out-33:7] Goto("SIP/33-090cd510", "real-out-33,27,1") in new stack -- Goto (real-out-33,27,1) -- Executing [27@real-out-33:1] Set("SIP/33-090cd510", "CDR(pbx)=pbx1") in new stack -- Executing [27@real-out-33:2] Set("SIP/33-090cd510", "CDR(userfield)=27") in new stack -- Executing [27@real-out-33:3] Set("SIP/33-090cd510", "__PICKUPMARK=27") in new stack -- Executing [27@real-out-33:4] Macro("SIP/33-090cd510", "dial-std-exten,SIP/27,out-27,27,pbx1") in new stack [090507-175449] WARNING[2407]: ast_expr2.fl:441 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '>=', expecting $end; Input: >= 2 ^ [090507-175449] WARNING[2407]: ast_expr2.fl:445 ast_yyerror: If you have questions, please refer to doc/tex/channelvariables.tex in the asterisk source. -- Executing [s@macro-dial-std-exten:1] GotoIf("SIP/33-090cd510", "?nocf:cfim") in new stack -- Goto (macro-dial-std-exten,s,3) -- Executing [s@macro-dial-std-exten:3] Set("SIP/33-090cd510", "fw_ext=") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:4] GotoIf("SIP/33-090cd510", "0?fw") in new stack -- Executing [s@macro-dial-std-exten:5] Set("SIP/33-090cd510", "vmim=") in new stack -- Executing [s@macro-dial-std-exten:6] GotoIf("SIP/33-090cd510", "0?vmu") in new stack -- Executing [s@macro-dial-std-exten:7] Set("SIP/33-090cd510", "dnd=") in new stack -- Executing [s@macro-dial-std-exten:8] GotoIf("SIP/33-090cd510", "0?busy") in new stack -- Executing [s@macro-dial-std-exten:9] Set("SIP/33-090cd510", "dsec=") in new stack -- Executing [s@macro-dial-std-exten:10] GotoIf("SIP/33-090cd510", "0?dsecEnd") in new stack -- Executing [s@macro-dial-std-exten:11] Set("SIP/33-090cd510", "dsec=25") in new stack -- Executing [s@macro-dial-std-exten:12] NoOp("SIP/33-090cd510", "") in new stack -- Executing [s@macro-dial-std-exten:13] Set("SIP/33-090cd510", "dopt=") in new stack -- Executing [s@macro-dial-std-exten:14] Set("SIP/33-090cd510", "emedia=") in new stack -- Executing [s@macro-dial-std-exten:15] GotoIf("SIP/33-090cd510", "1?emediaEnd") in new stack -- Goto (macro-dial-std-exten,s,18) -- Executing [s@macro-dial-std-exten:18] NoOp("SIP/33-090cd510", "") in new stack -- Executing [s@macro-dial-std-exten:19] Set("SIP/33-090cd510", "prng=") in new stack -- Executing [s@macro-dial-std-exten:20] NoOp("SIP/33-090cd510", "") in new stack -- Executing [s@macro-dial-std-exten:21] Dial("SIP/33-090cd510", "SIP/27,25,TtWwr") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 Audio is at 172.30.0.32 port 16454 Video is at 172.30.0.32 port 16444 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 172.30.0.81:65246: INVITE sip:27@172.30.0.81:65246;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK3a21876a;rport Max-Forwards: 70 From: "Ext 33" ;tag=as272d05d3 To: Contact: Call-ID: 20916dca25589b4335c0c4db3d54bcd3@172.30.0.32 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0 Date: Thu, 07 May 2009 22:54:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 376 v=0 o=root 1368880843 1368880843 IN IP4 172.30.0.32 s=Asterisk PBX 1.6.1.0 c=IN IP4 172.30.0.32 b=CT:384 t=0 0 m=audio 16454 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 0 RTP/AVP 34 a=rtpmap:34 H263/90000 a=sendrecv --- -- Called 27 debiasterisk*CLI> <--- SIP read from TLS://172.30.0.81:65246 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK3a21876a;rport From: "Ext 33" ;tag=as272d05d3 To: ;tag=F04AF3AD-9FDA124A CSeq: 102 INVITE Call-ID: 20916dca25589b4335c0c4db3d54bcd3@172.30.0.32 Contact: User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.1.3.0439 Accept-Language: en Content-Length: 0 <-------------> --- (10 headers 0 lines) --- debiasterisk*CLI> <--- SIP read from TLS://172.30.0.81:65246 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK3a21876a;rport From: "Ext 33" ;tag=as272d05d3 To: ;tag=F04AF3AD-9FDA124A CSeq: 102 INVITE Call-ID: 20916dca25589b4335c0c4db3d54bcd3@172.30.0.32 Contact: User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.1.3.0439 Allow-Events: talk,hold,conference Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- SIP/27-091b12b8 is ringing debiasterisk*CLI> <--- SIP read from TLS://172.30.0.81:65246 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK3a21876a;rport From: "Ext 33" ;tag=as272d05d3 To: ;tag=F04AF3AD-9FDA124A CSeq: 102 INVITE Call-ID: 20916dca25589b4335c0c4db3d54bcd3@172.30.0.32 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.1.3.0439 Accept-Language: en Content-Type: application/sdp Content-Length: 255 v=0 o=- 1241771981 1241771981 IN IP4 172.30.0.81 s=Polycom IP Phone c=IN IP4 172.30.0.81 t=0 0 m=audio 2226 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 34 a=sendrecv a=rtpmap:34 H263/90000 -event/8000 <-------------> --- (12 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found RTP video format 34 Peer audio RTP is at port 172.30.0.81:2226 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Found video description format H263 for ID 34 Capabilities: us - 0x3c030e (gsm|ulaw|alaw|g729|speex|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80004 (ulaw|h263) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.30.0.81:2226 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 172.30.0.81, port 65246 Transmitting (NAT) to 172.30.0.81:65246: ACK sip:27@172.30.0.81:65246;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK73562967;rport Max-Forwards: 70 From: "Ext 33" ;tag=as272d05d3 To: ;tag=F04AF3AD-9FDA124A Contact: Call-ID: 20916dca25589b4335c0c4db3d54bcd3@172.30.0.32 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.0 Content-Length: 0 --- -- SIP/27-091b12b8 answered SIP/33-090cd510 debiasterisk*CLI> <--- SIP read from TLS://172.30.0.81:65246 ---> BYE sip:33@172.30.0.32:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bKb62ee8a34F14EC08 From: ;tag=F04AF3AD-9FDA124A To: "Ext 33" ;tag=as272d05d3 CSeq: 1 BYE Call-ID: 20916dca25589b4335c0c4db3d54bcd3@172.30.0.32 Contact: User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.1.3.0439 Accept-Language: en Max-Forwards: 70 debiasterisk*CLI> Content-Length: 0 <-------------> debiasterisk*CLI> --- (11 headers 0 lines) --- debiasterisk*CLI> Sending to 172.30.0.81 : 65246 (NAT) debiasterisk*CLI> <--- Transmitting (NAT) to 172.30.0.81:65246 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bKb62ee8a34F14EC08;received=172.30.0.81 From: ;tag=F04AF3AD-9FDA124A To: "Ext 33" ;tag=as272d05d3 Call-ID: 20916dca25589b4335c0c4db3d54bcd3@172.30.0.32 CSeq: 1 BYE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> debiasterisk*CLI> == Spawn extension (macro-dial-std-exten, s, 21) exited non-zero on 'SIP/33-090cd510' in macro 'dial-std-exten' debiasterisk*CLI> == Spawn extension (real-out-33, 27, 4) exited non-zero on 'SIP/33-090cd510' debiasterisk*CLI> Really destroying SIP dialog '20916dca25589b4335c0c4db3d54bcd3@172.30.0.32' Method: BYE debiasterisk*CLI> exit Executing last minute cleanups