Asterisk 1.6.2.0-beta1, 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 1.6.2.0-beta1 currently running on debiasterisk (pid = 18875) debiasterisk*CLI> Verbosity was 4 and is now 6 debiasterisk*CLI> -- Remote UNIX connection debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:62362 ---> INVITE sip:25@172.30.0.32:5061;user=phone;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bKdc56100969F5CA7 From: "PBX1" ;tag=6140D7B-4927730E To: CSeq: 1 INVITE Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.2.0.0047 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 1241577541 1241577541 IN IP4 172.30.0.81 s=Polycom IP Phone c=IN IP4 172.30.0.81 t=0 0 m=audio 2236 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 11 lines) --- == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 Sending to 172.30.0.81 : 62362 (no NAT) Using INVITE request as basis request - cb8cdeea-b4a6239-183fcac4@172.30.0.81 Found peer '27' for '27' from 172.30.0.81:62362 <--- Reliably Transmitting (NAT) to 172.30.0.81:62362 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bKdc56100969F5CA7;received=172.30.0.81 From: "PBX1" ;tag=6140D7B-4927730E To: ;tag=as1fffddf3 Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0fac76cf" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'cb8cdeea-b4a6239-183fcac4@172.30.0.81' in 6400 ms (Method: INVITE) debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:62362 ---> ACK sip:25@172.30.0.32:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bKdc56100969F5CA7 From: "PBX1" ;tag=6140D7B-4927730E To: ;tag=as1fffddf3 CSeq: 1 ACK Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.2.0.0047 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:62362 ---> INVITE sip:25@172.30.0.32:5061;user=phone;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bKddeee6edF948248 From: "PBX1" ;tag=6140D7B-4927730E To: CSeq: 2 INVITE Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.2.0.0047 Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="27", realm="asterisk", nonce="0fac76cf", uri="sip:25@172.30.0.32:5061;user=phone;transport=tls", response="7fe759e322a809574cc8e79c14e85283", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 1241577541 1241577541 IN IP4 172.30.0.81 s=Polycom IP Phone c=IN IP4 172.30.0.81 t=0 0 m=audio 2236 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 11 lines) --- Sending to 172.30.0.81 : 62362 (NAT) Using INVITE request as basis request - cb8cdeea-b4a6239-183fcac4@172.30.0.81 Found peer '27' for '27' from 172.30.0.81:62362 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 172.30.0.81:2236 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 172.30.0.81:2236 Looking for 25 in out-27 (domain 172.30.0.32) list_route: hop: <--- Transmitting (NAT) to 172.30.0.81:62362 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bKddeee6edF948248;received=172.30.0.81 From: "PBX1" ;tag=6140D7B-4927730E To: Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [25@out-27:1] Set("SIP/27-0a2fee50", "fw_ext=") in new stack -- Executing [25@out-27:2] GotoIf("SIP/27-0a2fee50", "0?cfim") in new stack -- Executing [25@out-27:3] Set("SIP/27-0a2fee50", "fw_ext=") in new stack -- Executing [25@out-27:4] GotoIf("SIP/27-0a2fee50", "0?cfbs") in new stack -- Executing [25@out-27:5] Set("SIP/27-0a2fee50", "fw_ext=") in new stack -- Executing [25@out-27:6] GotoIf("SIP/27-0a2fee50", "0?cfto") in new stack -- Executing [25@out-27:7] Goto("SIP/27-0a2fee50", "real-out-27,25,1") in new stack -- Goto (real-out-27,25,1) -- Executing [25@real-out-27:1] Set("SIP/27-0a2fee50", "CDR(pbx)=pbx1") in new stack -- Executing [25@real-out-27:2] Set("SIP/27-0a2fee50", "CDR(userfield)=25") in new stack -- Executing [25@real-out-27:3] Set("SIP/27-0a2fee50", "__PICKUPMARK=25") in new stack -- Executing [25@real-out-27:4] Macro("SIP/27-0a2fee50", "dial-std-exten,SIP/25,out-25,25,pbx1") in new stack [090506-104305] WARNING[18909]: ast_expr2.fl:441 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '>=', expecting $end; Input: >= 2 ^ [090506-104305] WARNING[18909]: 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/27-0a2fee50", "?nocf:cfim") in new stack -- Goto (macro-dial-std-exten,s,3) debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:3] Set("SIP/27-0a2fee50", "fw_ext=") in new stack -- Executing [s@macro-dial-std-exten:4] GotoIf("SIP/27-0a2fee50", "0?fw") in new stack -- Executing [s@macro-dial-std-exten:5] Set("SIP/27-0a2fee50", "vmim=") in new stack -- Executing [s@macro-dial-std-exten:6] GotoIf("SIP/27-0a2fee50", "0?vmu") in new stack -- Executing [s@macro-dial-std-exten:7] Set("SIP/27-0a2fee50", "dnd=") in new stack -- Executing [s@macro-dial-std-exten:8] GotoIf("SIP/27-0a2fee50", "0?busy") in new stack -- Executing [s@macro-dial-std-exten:9] Set("SIP/27-0a2fee50", "dsec=") in new stack -- Executing [s@macro-dial-std-exten:10] GotoIf("SIP/27-0a2fee50", "0?dsecEnd") in new stack -- Executing [s@macro-dial-std-exten:11] Set("SIP/27-0a2fee50", "dsec=25") in new stack -- Executing [s@macro-dial-std-exten:12] NoOp("SIP/27-0a2fee50", "") in new stack -- Executing [s@macro-dial-std-exten:13] Set("SIP/27-0a2fee50", "dopt=") in new stack -- Executing [s@macro-dial-std-exten:14] Set("SIP/27-0a2fee50", "emedia=") in new stack -- Executing [s@macro-dial-std-exten:15] GotoIf("SIP/27-0a2fee50", "1?emediaEnd") in new stack -- Goto (macro-dial-std-exten,s,18) -- Executing [s@macro-dial-std-exten:18] NoOp("SIP/27-0a2fee50", "") in new stack -- Executing [s@macro-dial-std-exten:19] Set("SIP/27-0a2fee50", "prng=") in new stack -- Executing [s@macro-dial-std-exten:20] NoOp("SIP/27-0a2fee50", "") in new stack -- Executing [s@macro-dial-std-exten:21] Dial("SIP/27-0a2fee50", "SIP/25,25,TtWwr") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 25 <--- Transmitting (NAT) to 172.30.0.81:62362 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bKddeee6edF948248;received=172.30.0.81 From: "PBX1" ;tag=6140D7B-4927730E To: ;tag=as791b8816 Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> debiasterisk*CLI> -- SIP/25-0a305368 is ringing debiasterisk*CLI> <--- Transmitting (NAT) to 172.30.0.81:62362 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bKddeee6edF948248;received=172.30.0.81 From: "PBX1" ;tag=6140D7B-4927730E To: ;tag=as791b8816 Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: debiasterisk*CLI> Content-Length: 0 <------------> debiasterisk*CLI> -- SIP/25-0a305368 answered SIP/27-0a2fee50 debiasterisk*CLI> Audio is at 172.30.0.32 port 16410 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 172.30.0.81:62362 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bKddeee6edF948248;received=172.30.0.81 From: "PBX1" ;tag=6140D7B-4927730E To: ;tag=as791b8816 Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 336 v=0 o=root 175697520 175697520 IN IP4 172.30.0.32 s=Asterisk PBX 1.6.2.0-beta1 c=IN IP4 172.30.0.32 t=0 0 m=audio 16410 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:172.30.0.81:62362 ---> ACK sip:25@172.30.0.32;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bK5d990b8c43259163 From: "PBX1" ;tag=6140D7B-4927730E To: ;tag=as791b8816 CSeq: 2 ACK Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.2.0.0047 Authorization: Digest username="27", realm="asterisk", nonce="0fac76cf", uri="sip:25@172.30.0.32:5061;user=phone;transport=tls", response="7fe759e322a809574cc8e79c14e85283", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> debiasterisk*CLI> --- (12 headers 0 lines) --- debiasterisk*CLI> == Spawn extension (macro-dial-std-exten, s, 21) exited non-zero on 'SIP/27-0a2fee50' in macro 'dial-std-exten' debiasterisk*CLI> == Spawn extension (real-out-27, 25, 4) exited non-zero on 'SIP/27-0a2fee50' debiasterisk*CLI> Scheduling destruction of SIP dialog 'cb8cdeea-b4a6239-183fcac4@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 62362 debiasterisk*CLI> Reliably Transmitting (NAT) to 172.30.0.81:62362: BYE sip:27@172.30.0.81:62362;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK2111969c;rport Max-Forwards: 70 From: ;tag=as791b8816 To: "PBX1" ;tag=6140D7B-4927730E Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.2.0-beta1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:62362 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK2111969c;rport From: ;tag=as791b8816 To: "PBX1" ;tag=6140D7B-4927730E CSeq: 102 BYE Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.2.0.0047 Content-Length: 0 <-------------> debiasterisk*CLI> --- (8 headers 0 lines) --- debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:62362 ---> BYE sip:25@172.30.0.32;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bK81f46966E74895 From: "PBX1" ;tag=6140D7B-4927730E To: ;tag=as791b8816 CSeq: 3 BYE Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 Contact: User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.2.0.0047 Authorization: Digest username="27", realm="asterisk", nonce="0fac76cf", uri="sip:25@172.30.0.32:5061;user=phone;transport=tls", response="b405318f0f2efc7d52c754a74bae9493", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> debiasterisk*CLI> --- (11 headers 0 lines) --- debiasterisk*CLI> <--- Transmitting (no NAT) to 172.30.0.81:62362 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/TLS 172.30.0.81:62362;branch=z9hG4bK81f46966E74895;received=172.30.0.81 From: "PBX1" ;tag=6140D7B-4927730E To: ;tag=as791b8816 Call-ID: cb8cdeea-b4a6239-183fcac4@172.30.0.81 CSeq: 3 BYE Server: Asterisk PBX 1.6.2.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> debiasterisk*CLI> Reliably Transmitting (NAT) to 172.30.0.81:62362: OPTIONS sip:27@172.30.0.81:62362;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK1838bb12;rport Max-Forwards: 70 From: "asterisk" ;tag=as3e554d10 To: Contact: Call-ID: 222ca9f9640ae05258c918fe278fe644@172.30.0.32 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-beta1 Date: Wed, 06 May 2009 15:43:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:62362 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK1838bb12;rport From: "asterisk" ;tag=as3e554d10 To: ;tag=73124C77-B54C0DFA CSeq: 102 OPTIONS Call-ID: 222ca9f9640ae05258c918fe278fe644@172.30.0.32 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.2.0.0047 Content-Length: 0 <-------------> debiasterisk*CLI> --- (10 headers 0 lines) --- debiasterisk*CLI> exit Executing last minute cleanups