Asterisk SVN-trunk-r190865M, 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 SVN-trunk-r190865M currently running on debiasterisk (pid = 2054) debiasterisk*CLI> Verbosity is at least 5 debiasterisk*CLI> -- Remote UNIX connection debiasterisk*CLI> module exitsip show peerset debug peer 275 debiasterisk*CLI> SIP Debugging Enabled for IP: 172.30.0.139:35582 debiasterisk*CLI> sip set debug peer 257 debiasterisk*CLI> SIP Debugging Enabled for IP: 172.30.0.81:49630 debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:49630 ---> INVITE sip:25@172.30.0.32:5061;user=phone;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:49630;branch=z9hG4bK288b298e4BA5776D From: "PBX1" ;tag=E7342AE1-4E0EF60C To: CSeq: 1 INVITE Call-ID: 27f954c8-7f6e5f8f-36902d72@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=- 1241563282 1241563282 IN IP4 172.30.0.81 s=Polycom IP Phone c=IN IP4 172.30.0.81 t=0 0 m=audio 2232 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 <-------------> debiasterisk*CLI> --- (14 headers 11 lines) --- Sending to 172.30.0.81 : 49630 (no NAT) Using INVITE request as basis request - 27f954c8-7f6e5f8f-36902d72@172.30.0.81 Found peer '27' for '27' from 172.30.0.81:49630 <--- Reliably Transmitting (NAT) to 172.30.0.81:49630 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 172.30.0.81:49630;branch=z9hG4bK288b298e4BA5776D;received=172.30.0.81 From: "PBX1" ;tag=E7342AE1-4E0EF60C To: ;tag=as064a49b3 Call-ID: 27f954c8-7f6e5f8f-36902d72@172.30.0.81 CSeq: 1 INVITE Server: Asterisk PBX SVN-trunk-r190865M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e0cdf2b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '27f954c8-7f6e5f8f-36902d72@172.30.0.81' in 6400 ms (Method: INVITE) <--- SIP read from TLS:172.30.0.81:49630 ---> ACK sip:25@172.30.0.32:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:49630;branch=z9hG4bK288b298e4BA5776D From: "PBX1" ;tag=E7342AE1-4E0EF60C To: ;tag=as064a49b3 CSeq: 1 ACK Call-ID: 27f954c8-7f6e5f8f-36902d72@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 <-------------> debiasterisk*CLI> --- (11 headers 0 lines) --- debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:49630 ---> INVITE sip:25@172.30.0.32:5061;user=phone;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:49630;branch=z9hG4bK5725e3908B2D16 From: "PBX1" ;tag=E7342AE1-4E0EF60C To: CSeq: 2 INVITE Call-ID: 27f954c8-7f6e5f8f-36902d72@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="5e0cdf2b", uri="sip:25@172.30.0.32:5061;user=phone;transport=tls", response="b8cc3a09a74b6ca5c3700526e6ff03b6", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 1241563282 1241563282 IN IP4 172.30.0.81 s=Polycom IP Phone c=IN IP4 172.30.0.81 t=0 0 m=audio 2232 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 <-------------> debiasterisk*CLI> --- (15 headers 11 lines) --- debiasterisk*CLI> Sending to 172.30.0.81 : 49630 (NAT) debiasterisk*CLI> Using INVITE request as basis request - 27f954c8-7f6e5f8f-36902d72@172.30.0.81 debiasterisk*CLI> Found peer '27' for '27' from 172.30.0.81:49630 debiasterisk*CLI> Found RTP audio format 0 debiasterisk*CLI> Found RTP audio format 8 debiasterisk*CLI> Found RTP audio format 18 debiasterisk*CLI> Found RTP audio format 101 debiasterisk*CLI> Peer audio RTP is at port 172.30.0.81:2232 debiasterisk*CLI> Found audio description format PCMU for ID 0 debiasterisk*CLI> Found audio description format PCMA for ID 8 debiasterisk*CLI> Found audio description format G729 for ID 18 debiasterisk*CLI> Found audio description format telephone-event for ID 101 debiasterisk*CLI> 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) debiasterisk*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) debiasterisk*CLI> Peer audio RTP is at port 172.30.0.81:2232 debiasterisk*CLI> Peer video RTP is at port 172.30.0.81:33288 debiasterisk*CLI> Looking for 25 in out-27 (domain 172.30.0.32) debiasterisk*CLI> list_route: hop: debiasterisk*CLI> <--- Transmitting (NAT) to 172.30.0.81:49630 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.30.0.81:49630;branch=z9hG4bK5725e3908B2D16;received=172.30.0.81 From: "PBX1" ;tag=E7342AE1-4E0EF60C To: Call-ID: 27f954c8-7f6e5f8f-36902d72@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r190865M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> debiasterisk*CLI> -- Executing [25@out-27:1] Set("SIP/27-0881d370", "fw_ext=") in new stack debiasterisk*CLI> -- Executing [25@out-27:2] GotoIf("SIP/27-0881d370", "0?cfim") in new stack debiasterisk*CLI> -- Executing [25@out-27:3] Set("SIP/27-0881d370", "fw_ext=") in new stack debiasterisk*CLI> -- Executing [25@out-27:4] GotoIf("SIP/27-0881d370", "0?cfbs") in new stack debiasterisk*CLI> -- Executing [25@out-27:5] Set("SIP/27-0881d370", "fw_ext=") in new stack debiasterisk*CLI> -- Executing [25@out-27:6] GotoIf("SIP/27-0881d370", "0?cfto") in new stack debiasterisk*CLI> -- Executing [25@out-27:7] Goto("SIP/27-0881d370", "real-out-27,25,1") in new stack debiasterisk*CLI> -- Goto (real-out-27,25,1) debiasterisk*CLI> -- Executing [25@real-out-27:1] Set("SIP/27-0881d370", "CDR(pbx)=pbx1") in new stack debiasterisk*CLI> -- Executing [25@real-out-27:2] Set("SIP/27-0881d370", "CDR(userfield)=25") in new stack debiasterisk*CLI> -- Executing [25@real-out-27:3] Set("SIP/27-0881d370", "__PICKUPMARK=25") in new stack debiasterisk*CLI> -- Executing [25@real-out-27:4] Macro("SIP/27-0881d370", "dial-std-exten,SIP/25,out-25,25,pbx1") in new stack debiasterisk*CLI> [090506-064516] WARNING[2202]: ast_expr2.fl:434 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '>=', expecting $end; Input: >= 2 ^ debiasterisk*CLI> [090506-064516] WARNING[2202]: ast_expr2.fl:438 ast_yyerror: If you have questions, please refer to doc/tex/channelvariables.tex in the asterisk source. debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:1] GotoIf("SIP/27-0881d370", "?nocf:cfim") in new stack debiasterisk*CLI> -- Goto (macro-dial-std-exten,s,3) debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:3] Set("SIP/27-0881d370", "fw_ext=") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:4] GotoIf("SIP/27-0881d370", "0?fw") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:5] Set("SIP/27-0881d370", "vmim=") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:6] GotoIf("SIP/27-0881d370", "0?vmu") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:7] Set("SIP/27-0881d370", "dnd=") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:8] GotoIf("SIP/27-0881d370", "0?busy") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:9] Set("SIP/27-0881d370", "dsec=") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:10] GotoIf("SIP/27-0881d370", "0?dsecEnd") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:11] Set("SIP/27-0881d370", "dsec=25") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:12] NoOp("SIP/27-0881d370", "") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:13] Set("SIP/27-0881d370", "dopt=") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:14] Set("SIP/27-0881d370", "emedia=") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:15] GotoIf("SIP/27-0881d370", "1?emediaEnd") in new stack debiasterisk*CLI> -- Goto (macro-dial-std-exten,s,18) debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:18] NoOp("SIP/27-0881d370", "") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:19] Set("SIP/27-0881d370", "prng=") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:20] NoOp("SIP/27-0881d370", "") in new stack debiasterisk*CLI> -- Executing [s@macro-dial-std-exten:21] Dial("SIP/27-0881d370", "SIP/25,25,TtWwr") in new stack debiasterisk*CLI> -- Called 25 debiasterisk*CLI> <--- Transmitting (NAT) to 172.30.0.81:49630 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 172.30.0.81:49630;branch=z9hG4bK5725e3908B2D16;received=172.30.0.81 From: "PBX1" ;tag=E7342AE1-4E0EF60C To: ;tag=as5e4776c1 Call-ID: 27f954c8-7f6e5f8f-36902d72@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r190865M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> debiasterisk*CLI> -- SIP/25-088d1900 is ringing debiasterisk*CLI> <--- Transmitting (NAT) to 172.30.0.81:49630 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 172.30.0.81:49630;branch=z9hG4bK5725e3908B2D16;received=172.30.0.81 From: "PBX1" ;tag=E7342AE1-4E0EF60C To: ;tag=as5e4776c1 Call-ID: 27f954c8-7f6e5f8f-36902d72@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r190865M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> debiasterisk*CLI> -- SIP/25-088d1900 connected line has changed, passing it to SIP/27-0881d370 debiasterisk*CLI> -- SIP/25-088d1900 answered SIP/27-0881d370 debiasterisk*CLI> Audio is at 172.30.0.32 port 16474 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 172.30.0.81:49630 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.30.0.81:49630;branch=z9hG4bK5725e3908B2D16;received=172.30.0.81 From: "PBX1" ;tag=E7342AE1-4E0EF60C To: ;tag=as5e4776c1 Call-ID: 27f954c8-7f6e5f8f-36902d72@172.30.0.81 CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r190865M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 343 v=0 o=root 1986185661 1986185661 IN IP4 172.30.0.32 s=Asterisk PBX SVN-trunk-r190865M c=IN IP4 172.30.0.32 t=0 0 m=audio 16474 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:172.30.0.81:49630 ---> ACK sip:25@172.30.0.32;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.81:49630;branch=z9hG4bK4989cc7a54C18609 From: "PBX1" ;tag=E7342AE1-4E0EF60C To: ;tag=as5e4776c1 CSeq: 2 ACK Call-ID: 27f954c8-7f6e5f8f-36902d72@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="5e0cdf2b", uri="sip:25@172.30.0.32:5061;user=phone;transport=tls", response="b8cc3a09a74b6ca5c3700526e6ff03b6", 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/27-0881d370' in macro 'dial-std-exten' debiasterisk*CLI> == Spawn extension (real-out-27, 25, 4) exited non-zero on 'SIP/27-0881d370' debiasterisk*CLI> Scheduling destruction of SIP dialog '27f954c8-7f6e5f8f-36902d72@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 49630 debiasterisk*CLI> Reliably Transmitting (NAT) to 172.30.0.81:49630: BYE sip:27@172.30.0.81:49630;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK6d281552;rport Max-Forwards: 70 From: ;tag=as5e4776c1 To: "PBX1" ;tag=E7342AE1-4E0EF60C Call-ID: 27f954c8-7f6e5f8f-36902d72@172.30.0.81 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r190865M X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:49630 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK6d281552;rport From: ;tag=as5e4776c1 To: "PBX1" ;tag=E7342AE1-4E0EF60C CSeq: 102 BYE Call-ID: 27f954c8-7f6e5f8f-36902d72@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> Really destroying SIP dialog '27f954c8-7f6e5f8f-36902d72@172.30.0.81' Method: ACK debiasterisk*CLI> Reliably Transmitting (NAT) to 172.30.0.81:49630: OPTIONS sip:27@172.30.0.81:49630;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK725a4322;rport Max-Forwards: 70 From: "asterisk" ;tag=as298f4cbd To: Contact: Call-ID: 51ac82033aabbde6284b2ddd04bf78eb@172.30.0.32 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r190865M Date: Wed, 06 May 2009 11:45:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- debiasterisk*CLI> <--- SIP read from TLS:172.30.0.81:49630 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.30.0.32:5061;branch=z9hG4bK725a4322;rport From: "asterisk" ;tag=as298f4cbd To: ;tag=7EFB8CCB-DA596F9E CSeq: 102 OPTIONS Call-ID: 51ac82033aabbde6284b2ddd04bf78eb@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> Really destroying SIP dialog '51ac82033aabbde6284b2ddd04bf78eb@172.30.0.32' Method: OPTIONS debiasterisk*CLI> exit Executing last minute cleanups