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> <--- SIP read from TLS://172.30.0.81:65246 ---> INVITE sip:33@172.30.0.32;user=phone;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bKa1885f262AB2844F From: "PBX1" ;tag=7190BC4B-A6341690 To: CSeq: 1 INVITE Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 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 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 268 v=0 o=- 1241772009 1241772009 IN IP4 172.30.0.81 s=Polycom IP Phone c=IN IP4 172.30.0.81 t=0 0 a=sendrecv m=audio 2230 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 12 lines) --- == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 Sending to 172.30.0.81 : 65246 (no NAT) Using INVITE request as basis request - f8816e44-206c4d15-46974f92@172.30.0.81 Found peer '27' for '27' from 172.30.0.81:65246 <--- Reliably Transmitting (NAT) to 172.30.0.81:65246 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bKa1885f262AB2844F;received=172.30.0.81 From: "PBX1" ;tag=7190BC4B-A6341690 To: ;tag=as0da42705 Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="57eaa7c5" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'f8816e44-206c4d15-46974f92@172.30.0.81' in 6400 ms (Method: INVITE) debiasterisk*CLI> <--- SIP read from TLS://172.30.0.81:65246 ---> ACK sip:33@172.30.0.32;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bKa1885f262AB2844F From: "PBX1" ;tag=7190BC4B-A6341690 To: ;tag=as0da42705 CSeq: 1 ACK Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 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 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- debiasterisk*CLI> <--- SIP read from TLS://172.30.0.81:65246 ---> INVITE sip:33@172.30.0.32;user=phone;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bK3e4a5f7197142BE From: "PBX1" ;tag=7190BC4B-A6341690 To: CSeq: 2 INVITE Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 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 Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="27", realm="asterisk", nonce="57eaa7c5", uri="sip:33@172.30.0.32;user=phone;transport=tls", response="f715f0b46f58b6c56c1a21a0e35845ea", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 268 v=0 o=- 1241772009 1241772009 IN IP4 172.30.0.81 s=Polycom IP Phone c=IN IP4 172.30.0.81 t=0 0 a=sendrecv m=audio 2230 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 <-------------> --- (16 headers 12 lines) --- Sending to 172.30.0.81 : 65246 (NAT) Using INVITE request as basis request - f8816e44-206c4d15-46974f92@172.30.0.81 Found peer '27' for '27' from 172.30.0.81:65246 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 172.30.0.81:2230 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x3c030e (gsm|ulaw|alaw|g729|speex|h261|h263|h263p|h264), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.30.0.81:2230 Looking for 33 in out-27 (domain 172.30.0.32) list_route: hop: <--- Transmitting (NAT) to 172.30.0.81:65246 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bK3e4a5f7197142BE;received=172.30.0.81 From: "PBX1" ;tag=7190BC4B-A6341690 To: Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [33@out-27:1] Set("SIP/27-09197880", "fw_ext=") in new stack -- Executing [33@out-27:2] GotoIf("SIP/27-09197880", "0?cfim") in new stack -- Executing [33@out-27:3] Set("SIP/27-09197880", "fw_ext=") in new stack -- Executing [33@out-27:4] GotoIf("SIP/27-09197880", "0?cfbs") in new stack -- Executing [33@out-27:5] Set("SIP/27-09197880", "fw_ext=") in new stack -- Executing [33@out-27:6] GotoIf("SIP/27-09197880", "0?cfto") in new stack -- Executing [33@out-27:7] Goto("SIP/27-09197880", "real-out-27,33,1") in new stack -- Goto (real-out-27,33,1) -- Executing [33@real-out-27:1] Set("SIP/27-09197880", "CDR(pbx)=pbx1") in new stack -- Executing [33@real-out-27:2] Set("SIP/27-09197880", "CDR(userfield)=33") in new stack -- Executing [33@real-out-27:3] Set("SIP/27-09197880", "__PICKUPMARK=33") in new stack -- Executing [33@real-out-27:4] Macro("SIP/27-09197880", "dial-std-exten,SIP/33,out-33,33,pbx1") in new stack [090507-175518] WARNING[2411]: ast_expr2.fl:441 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '>=', expecting $end; Input: >= 2 ^ [090507-175518] WARNING[2411]: 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/27-09197880", "?nocf:cfim") in new stack -- Goto (macro-dial-std-exten,s,3) -- Executing [s@macro-dial-std-exten:3] Set("SIP/27-09197880", "fw_ext=") in new stack -- Executing [s@macro-dial-std-exten:4] GotoIf("SIP/27-09197880", "0?fw") in new stack -- Executing [s@macro-dial-std-exten:5] Set("SIP/27-09197880", "vmim=") in new stack -- Executing [s@macro-dial-std-exten:6] GotoIf("SIP/27-09197880", "0?vmu") in new stack -- Executing [s@macro-dial-std-exten:7] Set("SIP/27-09197880", "dnd=") in new stack -- Executing [s@macro-dial-std-exten:8] GotoIf("SIP/27-09197880", "0?busy") in new stack -- Executing [s@macro-dial-std-exten:9] Set("SIP/27-09197880", "dsec=") in new stack -- Executing [s@macro-dial-std-exten:10] GotoIf("SIP/27-09197880", "0?dsecEnd") in new stack -- Executing [s@macro-dial-std-exten:11] Set("SIP/27-09197880", "dsec=25") in new stack -- Executing [s@macro-dial-std-exten:12] NoOp("SIP/27-09197880", "") in new stack -- Executing [s@macro-dial-std-exten:13] Set("SIP/27-09197880", "dopt=") in new stack -- Executing [s@macro-dial-std-exten:14] Set("SIP/27-09197880", "emedia=") in new stack -- Executing [s@macro-dial-std-exten:15] GotoIf("SIP/27-09197880", "1?emediaEnd") in new stack -- Goto (macro-dial-std-exten,s,18) -- Executing [s@macro-dial-std-exten:18] NoOp("SIP/27-09197880", "") in new stack -- Executing [s@macro-dial-std-exten:19] Set("SIP/27-09197880", "prng=") in new stack -- Executing [s@macro-dial-std-exten:20] NoOp("SIP/27-09197880", "") in new stack -- Executing [s@macro-dial-std-exten:21] Dial("SIP/27-09197880", "SIP/33,25,TtWwr") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 33 <--- Transmitting (NAT) to 172.30.0.81:65246 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bK3e4a5f7197142BE;received=172.30.0.81 From: "PBX1" ;tag=7190BC4B-A6341690 To: ;tag=as014d98c4 Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> debiasterisk*CLI> -- SIP/33-091a9160 is ringing <--- Transmitting (NAT) to 172.30.0.81:65246 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bK3e4a5f7197142BE;received=172.30.0.81 From: "PBX1" ;tag=7190BC4B-A6341690 To: ;tag=as014d98c4 Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> debiasterisk*CLI> -- SIP/33-091a9160 answered SIP/27-09197880 debiasterisk*CLI> Audio is at 172.30.0.32 port 16394 debiasterisk*CLI> Adding codec 0x4 (ulaw) to SDP debiasterisk*CLI> Adding codec 0x8 (alaw) to SDP debiasterisk*CLI> Adding codec 0x100 (g729) to SDP debiasterisk*CLI> Adding non-codec 0x1 (telephone-event) to SDP debiasterisk*CLI> <--- Reliably Transmitting (NAT) to 172.30.0.81:65246 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bK3e4a5f7197142BE;received=172.30.0.81 From: "PBX1" ;tag=7190BC4B-A6341690 To: ;tag=as014d98c4 Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 332 v=0 o=root 1276780521 1276780521 IN IP4 172.30.0.32 s=Asterisk PBX 1.6.1.0 c=IN IP4 172.30.0.32 t=0 0 m=audio 16394 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> debiasterisk*CLI> <--- SIP read from TLS://172.30.0.81:65246 ---> ACK sip:33@172.30.0.32:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:65246;branch=z9hG4bKbff0c38dE4D224AA From: "PBX1" ;tag=7190BC4B-A6341690 To: ;tag=as014d98c4 CSeq: 2 ACK Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 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 Max-Forwards: 70 Content-Length: 0 <-------------> debiasterisk*CLI> --- (12 headers 0 lines) --- debiasterisk*CLI> Reliably Transmitting (NAT) to 172.30.0.81:65246: OPTIONS sip:27@172.30.0.81:65246;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK2960d6f5;rport Max-Forwards: 70 From: "asterisk" ;tag=as70a0f0b2 To: Contact: Call-ID: 529f148f22ecf1334a5fcefd77c70806@172.30.0.32 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.0 Date: Thu, 07 May 2009 22:55:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- 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=z9hG4bK2960d6f5;rport From: "asterisk" ;tag=as70a0f0b2 To: ;tag=55BBD568-C135569 CSeq: 102 OPTIONS Call-ID: 529f148f22ecf1334a5fcefd77c70806@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 Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Supported: 100rel,timer,replaces Content-Length: 0 <-------------> debiasterisk*CLI> --- (14 headers 0 lines) --- debiasterisk*CLI> == Spawn extension (macro-dial-std-exten, s, 21) exited non-zero on 'SIP/27-09197880' in macro 'dial-std-exten' debiasterisk*CLI> == Spawn extension (real-out-27, 33, 4) exited non-zero on 'SIP/27-09197880' debiasterisk*CLI> Scheduling destruction of SIP dialog 'f8816e44-206c4d15-46974f92@172.30.0.81' in 6400 ms (Method: ACK) debiasterisk*CLI> set_destination: Parsing for address/port to send to debiasterisk*CLI> set_destination: set destination to 172.30.0.81, port 65246 debiasterisk*CLI> Reliably Transmitting (NAT) to 172.30.0.81:65246: BYE sip:27@172.30.0.81:65246;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK0cd0e4b4;rport Max-Forwards: 70 From: ;tag=as014d98c4 To: "PBX1" ;tag=7190BC4B-A6341690 Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- debiasterisk*CLI> <--- SIP read from TLS://172.30.0.81:65246 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK0cd0e4b4;rport From: ;tag=as014d98c4 To: "PBX1" ;tag=7190BC4B-A6341690 CSeq: 102 BYE Call-ID: f8816e44-206c4d15-46974f92@172.30.0.81 User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.1.3.0439 Accept-Language: en Content-Length: 0 <-------------> debiasterisk*CLI> --- (9 headers 0 lines) --- debiasterisk*CLI> Really destroying SIP dialog '529f148f22ecf1334a5fcefd77c70806@172.30.0.32' Method: OPTIONS debiasterisk*CLI> Really destroying SIP dialog 'f8816e44-206c4d15-46974f92@172.30.0.81' Method: ACK debiasterisk*CLI> Reliably Transmitting (NAT) to 172.30.0.81:65246: OPTIONS sip:27@172.30.0.81:65246;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK3de27a7d;rport Max-Forwards: 70 From: "asterisk" ;tag=as7cc8346a To: Contact: Call-ID: 311b512949c4f5b07fb688ff62a1ee90@172.30.0.32 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.0 Date: Thu, 07 May 2009 22:55:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- 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=z9hG4bK3de27a7d;rport From: "asterisk" ;tag=as7cc8346a To: ;tag=7A83DABF-4AA943F4 CSeq: 102 OPTIONS Call-ID: 311b512949c4f5b07fb688ff62a1ee90@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 Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Supported: 100rel,timer,replaces Content-Length: 0 <-------------> debiasterisk*CLI> --- (14 headers 0 lines) --- debiasterisk*CLI> Really destroying SIP dialog '311b512949c4f5b07fb688ff62a1ee90@172.30.0.32' Method: OPTIONS debiasterisk*CLI> exit Executing last minute cleanups