== Parsing '/etc/asterisk/asterisk.conf': Found Connected to Asterisk SVN-branch-1.4-r211807 currently running on asterisk (pid = 2398) Verbosity is at least 3 Core debug is at least 3 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:34] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Retransmitting #1 (NAT) to 192.168.30.254:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK64bf09d6;received=192.168.30.254;rport=5060 From: "1039" ;tag=as6e13e2a6 To: ;tag=as21465626 Call-ID: 127bff2d14fbc66011facd964102c57d@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 14958 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Aug 14 14:14:34] DEBUG[24487]: rtp.c:2712 ast_rtp_raw_write: Difference is 968, ms is 141 [Aug 14 14:14:34] DEBUG[24486]: rtp.c:2712 ast_rtp_raw_write: Difference is 976, ms is 142 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[24312]: rtp.c:2712 ast_rtp_raw_write: Difference is 984, ms is 143 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '323de31c7f8315ec40a0683726f1f420@127.0.0.1' [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 323de31c7f8315ec40a0683726f1f420@127.0.0.1 Really destroying SIP dialog '323de31c7f8315ec40a0683726f1f420@127.0.0.1' Method: REGISTER [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[24312]: rtp.c:2712 ast_rtp_raw_write: Difference is 1456, ms is 202 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[24637]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b58ba2e0", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:34] DEBUG[24637]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '51' [Aug 14 14:14:34] DEBUG[24637]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:34] DEBUG[24637]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b58ba2e0", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:34] DEBUG[24637]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b58ba2e0", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:34] DEBUG[24637]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b58ba2e0", "30") in new stack [Aug 14 14:14:34] DEBUG[24637]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b58ba2e0' does not support indication 8, emulating it [Aug 14 14:14:34] DEBUG[24637]: channel.c:3090 set_format: Set channel SIP/siptrunk-b58ba2e0 to write format slin [Aug 14 14:14:34] WARNING[24216]: pbx.c:2492 __ast_pbx_run: Timeout, but no rule 't' in context 'default-application-acd-customer-new-english' [Aug 14 14:14:34] DEBUG[24216]: channel.c:1453 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/siptrunk-b465bf20' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [h@default-application-acd-customer-new-english:1] Goto("SIP/siptrunk-b465bf20", "all-hangup|s|1") in new stack -- Goto (all-hangup,s,1) [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '2' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [s@all-hangup:1] GotoIf("SIP/siptrunk-b465bf20", "0?all-faxnotify|s|1:2") in new stack -- Goto (all-hangup,s,2) [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1843 pbx_extension_helper: Launching 'ResetCDR' -- Executing [s@all-hangup:2] ResetCDR("SIP/siptrunk-b465bf20", "w") in new stack [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '"customer-E:1034" <1034>' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1034' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '5000' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'private-siptrunk-incoming' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'SIP/siptrunk-b465bf20' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '(null)' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'NoOp' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '"=== END QUEUE (default-customer-new-english) ==="' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:13:33' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:13:33' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:34' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '61' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '61' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'siptrunk' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1250273613.6257' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '5000' [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1843 pbx_extension_helper: Launching 'NoCDR' -- Executing [s@all-hangup:3] NoCDR("SIP/siptrunk-b465bf20", "") in new stack [Aug 14 14:14:34] DEBUG[24216]: pbx.c:1843 pbx_extension_helper: Launching 'System' -- Executing [s@all-hangup:4] System("SIP/siptrunk-b465bf20", "/var/www/scopserv/telephony/scripts/billing/cdr.sh 1250273613.6257") in new stack [Aug 14 14:14:34] DEBUG[24637]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:34] DEBUG[24637]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[24639]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b5155518", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '127bff2d14fbc66011facd964102c57d@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:16786 do_monitor: chan_sip: ast_sched_runq ran 44 all at once [Aug 14 14:14:34] DEBUG[24639]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b5155518", "CDR(userfield)=5000") in new stack [Aug 14 14:14:34] DEBUG[24639]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b5155518", "10") in new stack -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:34] DEBUG[24639]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b5155518", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '453a64de2a1ebbb923c393712cd310b4@127.0.0.1' [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 453a64de2a1ebbb923c393712cd310b4@127.0.0.1 Really destroying SIP dialog '453a64de2a1ebbb923c393712cd310b4@127.0.0.1' Method: REGISTER Really destroying SIP dialog '5fa70adc50265f1552b6022e3e0a5be1@192.168.30.254' Method: ACK Really destroying SIP dialog '7cd6e5fb12104ee068c4b6993a83b590@192.168.30.254' Method: ACK Really destroying SIP dialog '6e22f39401502ea0771fdd617ef924d4@192.168.30.254' Method: ACK Really destroying SIP dialog '560106e256b25ba96877374579b4ddb2@192.168.30.254' Method: ACK [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[24317]: channel.c:3090 set_format: Set channel SIP/siptrunk-b50fe470 to write format ulaw [Aug 14 14:14:34] DEBUG[24317]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b50fe470' [Aug 14 14:14:34] DEBUG[24317]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b50fe470, SIP callid 5576cad60f3002355f5eaf3c406678ea@192.168.30.254) Scheduling destruction of SIP dialog '5576cad60f3002355f5eaf3c406678ea@192.168.30.254' in 32000 ms (Method: ACK) [Aug 14 14:14:34] DEBUG[24317]: chan_sip.c:6283 reqprep: Strict routing enforced for session 5576cad60f3002355f5eaf3c406678ea@192.168.30.254 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.254, port 5060 Reliably Transmitting (NAT) to 192.168.30.254:5060: BYE sip:1040@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK044ca1f0;rport From: ;tag=as3a895e0f To: "1040" ;tag=as26ad7239 Call-ID: 5576cad60f3002355f5eaf3c406678ea@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Aug 14 14:14:34] DEBUG[24317]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:34] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:34] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:34] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[24634]: rtp.c:2712 ast_rtp_raw_write: Difference is 736, ms is 112 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[24312]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes <--- SIP read from 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK044ca1f0;received=192.168.30.165;rport=5060 From: ;tag=as3a895e0f To: "1040" ;tag=as26ad7239 Call-ID: 5576cad60f3002355f5eaf3c406678ea@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '5576cad60f3002355f5eaf3c406678ea@192.168.30.254' of Request 102: Match Found SIP Response message for INCOMING dialog BYE arrived [Aug 14 14:14:34] DEBUG[24312]: rtp.c:2712 ast_rtp_raw_write: Difference is 888, ms is 131 <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK72023591;rport From: "1004" ;tag=as7dd33234 To: Contact: Call-ID: 7a1949af07bcb4995662e5015da30fff@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 12476 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 7a1949af07bcb4995662e5015da30fff@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 7a1949af07bcb4995662e5015da30fff@192.168.30.254 [Aug 14 14:14:34] DEBUG[24487]: rtp.c:2712 ast_rtp_raw_write: Difference is 2016, ms is 272 Found peer 'siptrunk' [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK72023591;received=192.168.30.254;rport=5060 From: "1004" ;tag=as7dd33234 To: ;tag=as7e76ad35 Call-ID: 7a1949af07bcb4995662e5015da30fff@192.168.30.254 CSeq: 102 INVITE ser-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3bfa4079" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '7a1949af07bcb4995662e5015da30fff@192.168.30.254' in 32000 ms (Method: INVITE) [Aug 14 14:14:34] DEBUG[24634]: rtp.c:2712 ast_rtp_raw_write: Difference is 744, ms is 113 <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK159d3d8b;rport From: "1032" ;tag=as07d70209 To: Contact: Call-ID: 17d1844f4a52341a3d9409326da7c500@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 17788 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 17d1844f4a52341a3d9409326da7c500@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 17d1844f4a52341a3d9409326da7c500@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK159d3d8b;received=192.168.30.254;rport=5060 From: "1032" ;tag=as07d70209 To: ;tag=as7d73dc80 Call-ID: 17d1844f4a52341a3d9409326da7c500@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d9e9ce9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '17d1844f4a52341a3d9409326da7c500@192.168.30.254' in 32000 ms (Method: INVITE) [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK72023591;rport From: "1004" ;tag=as7dd33234 To: ;tag=as7e76ad35 Contact: Call-ID: 7a1949af07bcb4995662e5015da30fff@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '7a1949af07bcb4995662e5015da30fff@192.168.30.254' of Response 102: Match Found [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:34] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6b801144;rport From: "1004" ;tag=as7dd33234 To: Contact: Call-ID: 7a1949af07bcb4995662e5015da30fff@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="3bfa4079", response="30eb8e893c9fd442da199fb25aea97fb" Date: Fri, 14 Aug 2009 18:14:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 12476 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 7a1949af07bcb4995662e5015da30fff@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:12476 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:12476 [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b4823260: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6b801144;received=192.168.30.254;rport=5060 From: "1004" ;tag=as7dd33234 To: Call-ID: 7a1949af07bcb4995662e5015da30fff@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:34] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK159d3d8b;rport From: "1032" ;tag=as07d70209 To: ;tag=as7d73dc80 Contact: Call-ID: 17d1844f4a52341a3d9409326da7c500@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:34] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '17d1844f4a52341a3d9409326da7c500@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK43aaca7e;rport From: "1032" ;tag=as07d70209 To: Contact: Call-ID: 17d1844f4a52341a3d9409326da7c500@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="3d9e9ce9", response="e19b3284becc08c82b327389db9ef88a" Date: Fri, 14 Aug 2009 18:14:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 17788 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:35] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24639]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b5155518", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:35] DEBUG[24639]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '51' [Aug 14 14:14:35] DEBUG[24639]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:35] DEBUG[24639]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b5155518", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:35] DEBUG[24639]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b5155518", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:35] DEBUG[24639]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b5155518", "30") in new stack [Aug 14 14:14:35] DEBUG[24639]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b5155518' does not support indication 8, emulating it [Aug 14 14:14:35] DEBUG[24639]: channel.c:3090 set_format: Set channel SIP/siptrunk-b5155518 to write format slin [Aug 14 14:14:35] DEBUG[24639]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:35] DEBUG[24639]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK2b59dca2;rport From: "1000" ;tag=as37d1bbf9 To: Contact: Call-ID: 497406a7474738db5022f0ee681dae86@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 19734 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 497406a7474738db5022f0ee681dae86@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 497406a7474738db5022f0ee681dae86@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK2b59dca2;received=192.168.30.254;rport=5060 From: "1000" ;tag=as37d1bbf9 To: ;tag=as08b64fa9 Call-ID: 497406a7474738db5022f0ee681dae86@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06946b7d" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '497406a7474738db5022f0ee681dae86@192.168.30.254' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK2b59dca2;rport From: "1000" ;tag=as37d1bbf9 To: ;tag=as08b64fa9 Contact: Call-ID: 497406a7474738db5022f0ee681dae86@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '497406a7474738db5022f0ee681dae86@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0c4dc749;rport From: "1000" ;tag=as37d1bbf9 To: Contact: Call-ID: 497406a7474738db5022f0ee681dae86@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="06946b7d", response="ad03fb7c711b08c665ea2eb328daf1ae" Date: Fri, 14 Aug 2009 18:14:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 19734 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 497406a7474738db5022f0ee681dae86@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:19734 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:19734 [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b4872d88: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0c4dc749;received=192.168.30.254;rport=5060 From: "1000" ;tag=as37d1bbf9 To: Call-ID: 497406a7474738db5022f0ee681dae86@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:35] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:35] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:35] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:35] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b4872d88", "") in new stack [Aug 14 14:14:35] DEBUG[24663]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:35] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:35] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:35] DEBUG[24663]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b4872d88 [Aug 14 14:14:35] DEBUG[24663]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:35] DEBUG[24663]: chan_sip.c:6761 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Aug 14 14:14:35] DEBUG[24663]: chan_sip.c:6762 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.30.165 port 12770 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Aug 14 14:14:35] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:35] DEBUG[24663]: chan_sip.c:6879 add_sdp: -- Done with adding codecs to SDP [Aug 14 14:14:35] DEBUG[24663]: chan_sip.c:6988 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) <--- Reliably Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0c4dc749;received=192.168.30.254;rport=5060 From: "1000" ;tag=as37d1bbf9 To: ;tag=as4d614e20 Call-ID: 497406a7474738db5022f0ee681dae86@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 12770 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b4872d88", "CHANNEL(musicclass)=default") in new stack [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1000' [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1000' [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b4872d88", ""INCOMING CALL FROM CALLER ID: 1000 (1000)"") in new stack [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:4] Set("SIP/siptrunk-b4872d88", "__INCOMING_DNIS=5000") in new stack [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b4872d88", "CALLERID(dnid)=5000") in new stack <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3ec6bab6;rport From: "1000" ;tag=as37d1bbf9 To: ;tag=as4d614e20 Contact: Call-ID: 497406a7474738db5022f0ee681dae86@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '497406a7474738db5022f0ee681dae86@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b4872d88", "CDR(userfield)=5000") in new stack [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b4872d88", "10") in new stack -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:35] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b4872d88", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24638]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b58bbbc0", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:35] DEBUG[24638]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '52' [Aug 14 14:14:35] DEBUG[24638]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:35] DEBUG[24638]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b58bbbc0", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:35] DEBUG[24638]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b58bbbc0", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:35] DEBUG[24638]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b58bbbc0", "30") in new stack [Aug 14 14:14:35] DEBUG[24638]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b58bbbc0' does not support indication 8, emulating it [Aug 14 14:14:35] DEBUG[24638]: channel.c:3090 set_format: Set channel SIP/siptrunk-b58bbbc0 to write format slin [Aug 14 14:14:35] DEBUG[24638]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:35] DEBUG[24638]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3303c270;rport From: "1046" ;tag=as2b157408 To: Contact: Call-ID: 668633161fd37aaf700502546588a0fa@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 19394 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 668633161fd37aaf700502546588a0fa@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 668633161fd37aaf700502546588a0fa@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3303c270;received=192.168.30.254;rport=5060 From: "1046" ;tag=as2b157408 To: ;tag=as389750d6 Call-ID: 668633161fd37aaf700502546588a0fa@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="64f5058c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '668633161fd37aaf700502546588a0fa@192.168.30.254' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3303c270;rport From: "1046" ;tag=as2b157408 To: ;tag=as389750d6 Contact: Call-ID: 668633161fd37aaf700502546588a0fa@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '668633161fd37aaf700502546588a0fa@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK094514c5;rport From: "1046" ;tag=as2b157408 To: Contact: Call-ID: 668633161fd37aaf700502546588a0fa@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="64f5058c", response="26adeb3896eb009fb4969c52465dff0b" Date: Fri, 14 Aug 2009 18:14:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 19394 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 668633161fd37aaf700502546588a0fa@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:19394 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:19394 [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:35] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b4827668: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK094514c5;received=192.168.30.254;rport=5060 From: "1046" ;tag=as2b157408 To: Call-ID: 668633161fd37aaf700502546588a0fa@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:35] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:35] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:35] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:35] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b4827668", "") in new stack [Aug 14 14:14:35] DEBUG[23948]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:35] DEBUG[24665]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:35] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:35] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:35] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:35] DEBUG[24665]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b4827668 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24665]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:35] DEBUG[24665]: chan_sip.c:6761 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Aug 14 14:14:35] DEBUG[24665]: chan_sip.c:6762 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.30.165 port 15930 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Aug 14 14:14:35] DEBUG[24665]: chan_sip.c:6879 add_sdp: -- Done with adding codecs to SDP [Aug 14 14:14:35] DEBUG[24665]: chan_sip.c:6988 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) <--- Reliably Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK094514c5;received=192.168.30.254;rport=5060 From: "1046" ;tag=as2b157408 To: ;tag=as2a23017b Call-ID: 668633161fd37aaf700502546588a0fa@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 15930 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b4827668", "CHANNEL(musicclass)=default") in new stack <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK16824d2d;rport From: "1046" ;tag=as2b157408 To: ;tag=as2a23017b Contact: Call-ID: 668633161fd37aaf700502546588a0fa@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '668633161fd37aaf700502546588a0fa@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1046' [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1046' [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b4827668", ""INCOMING CALL FROM CALLER ID: 1046 (1046)"") in new stack [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:4] Set("SIP/siptrunk-b4827668", "__INCOMING_DNIS=5000") in new stack [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b4827668", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b4827668", "CDR(userfield)=5000") in new stack [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b4827668", "10") in new stack [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:35] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b4827668", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '037ff4d41ecfe1786ebf0cbc00ecbb63@127.0.0.1' [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 037ff4d41ecfe1786ebf0cbc00ecbb63@127.0.0.1 Really destroying SIP dialog '037ff4d41ecfe1786ebf0cbc00ecbb63@127.0.0.1' Method: REGISTER [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '3eb67c7c709f297e261f6f892f646bef@127.0.0.1' [Aug 14 14:14:35] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 3eb67c7c709f297e261f6f892f646bef@127.0.0.1 Really destroying SIP dialog '3eb67c7c709f297e261f6f892f646bef@127.0.0.1' Method: REGISTER [Aug 14 14:14:35] DEBUG[24312]: pbx.c:2392 __ast_pbx_run: Spawn extension (private-siptrunk-incoming,5000,501) exited non-zero on 'SIP/siptrunk-b50fa068' == Spawn extension (private-siptrunk-incoming, 5000, 501) exited non-zero on 'SIP/siptrunk-b50fa068' [Aug 14 14:14:35] DEBUG[24312]: channel.c:1453 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/siptrunk-b50fa068' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [h@private-siptrunk-incoming:1] Goto("SIP/siptrunk-b50fa068", "all-hangup|s|1") in new stack -- Goto (all-hangup,s,1) [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '2' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [s@all-hangup:1] GotoIf("SIP/siptrunk-b50fa068", "0?all-faxnotify|s|1:2") in new stack -- Goto (all-hangup,s,2) [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1843 pbx_extension_helper: Launching 'ResetCDR' -- Executing [s@all-hangup:2] ResetCDR("SIP/siptrunk-b50fa068", "w") in new stack [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '"1008" <1008>' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1008' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '5000' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'private-siptrunk-incoming' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'SIP/siptrunk-b50fa068' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '(null)' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'Congestion' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '30' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:13:59' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:13:59' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:35' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '36' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '36' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'siptrunk' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1250273639.6286' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '5000' [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1843 pbx_extension_helper: Launching 'NoCDR' -- Executing [s@all-hangup:3] NoCDR("SIP/siptrunk-b50fa068", "") in new stack [Aug 14 14:14:35] DEBUG[24312]: pbx.c:1843 pbx_extension_helper: Launching 'System' -- Executing [s@all-hangup:4] System("SIP/siptrunk-b50fa068", "/var/www/scopserv/telephony/scripts/billing/cdr.sh 1250273639.6286") in new stack [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24487]: rtp.c:2712 ast_rtp_raw_write: Difference is 1104, ms is 158 [Aug 14 14:14:35] DEBUG[24486]: rtp.c:2712 ast_rtp_raw_write: Difference is 1112, ms is 159 [Aug 14 14:14:35] DEBUG[24634]: rtp.c:2712 ast_rtp_raw_write: Difference is 1016, ms is 147 [Aug 14 14:14:35] DEBUG[24487]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:35] DEBUG[24635]: rtp.c:2712 ast_rtp_raw_write: Difference is 1160, ms is 165 [Aug 14 14:14:35] DEBUG[24639]: rtp.c:2712 ast_rtp_raw_write: Difference is 1160, ms is 165 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] DEBUG[24638]: rtp.c:2712 ast_rtp_raw_write: Difference is 1232, ms is 174 [Aug 14 14:14:35] DEBUG[24637]: rtp.c:2712 ast_rtp_raw_write: Difference is 1168, ms is 166 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:35] DEBUG[24659]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b4823260", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:35] DEBUG[24659]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '53' [Aug 14 14:14:35] DEBUG[24659]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:35] DEBUG[24659]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b4823260", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:35] DEBUG[24659]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b4823260", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:35] DEBUG[24659]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b4823260", "30") in new stack [Aug 14 14:14:35] DEBUG[24659]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b4823260' does not support indication 8, emulating it [Aug 14 14:14:35] DEBUG[24659]: channel.c:3090 set_format: Set channel SIP/siptrunk-b4823260 to write format slin [Aug 14 14:14:35] DEBUG[24659]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:35] DEBUG[24659]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:35] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24311]: channel.c:3090 set_format: Set channel SIP/siptrunk-b58dc998 to write format ulaw [Aug 14 14:14:36] DEBUG[24311]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b58dc998' [Aug 14 14:14:36] DEBUG[24311]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b58dc998, SIP callid 79ed0423747cb7ae47d834ad40f70c8d@192.168.30.254) Scheduling destruction of SIP dialog '79ed0423747cb7ae47d834ad40f70c8d@192.168.30.254' in 32000 ms (Method: ACK) [Aug 14 14:14:36] DEBUG[24311]: chan_sip.c:6283 reqprep: Strict routing enforced for session 79ed0423747cb7ae47d834ad40f70c8d@192.168.30.254 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.254, port 5060 Reliably Transmitting (NAT) to 192.168.30.254:5060: BYE sip:1009@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK05c1f6a6;rport From: ;tag=as11c2a4d5 To: "1009" ;tag=as099f9640 Call-ID: 79ed0423747cb7ae47d834ad40f70c8d@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Aug 14 14:14:36] DEBUG[24311]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK05c1f6a6;received=192.168.30.165;rport=5060 From: ;tag=as11c2a4d5 To: "1009" ;tag=as099f9640 Call-ID: 79ed0423747cb7ae47d834ad40f70c8d@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '79ed0423747cb7ae47d834ad40f70c8d@192.168.30.254' of Request 102: Match Found SIP Response message for INCOMING dialog BYE arrived [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24323]: channel.c:3090 set_format: Set channel SIP/siptrunk-b5160890 to write format ulaw [Aug 14 14:14:36] DEBUG[24323]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b5160890' [Aug 14 14:14:36] DEBUG[24323]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b5160890, SIP callid 5ab6e60f016458485828448a79e42c4b@192.168.30.254) Scheduling destruction of SIP dialog '5ab6e60f016458485828448a79e42c4b@192.168.30.254' in 32000 ms (Method: ACK) [Aug 14 14:14:36] DEBUG[24323]: chan_sip.c:6283 reqprep: Strict routing enforced for session 5ab6e60f016458485828448a79e42c4b@192.168.30.254 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.254, port 5060 Reliably Transmitting (NAT) to 192.168.30.254:5060: BYE sip:1049@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK70a38588;rport From: ;tag=as204d1e57 To: "1049" ;tag=as5fb7dcb5 Call-ID: 5ab6e60f016458485828448a79e42c4b@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Aug 14 14:14:36] DEBUG[24323]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:36] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). <--- SIP read from 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK70a38588;received=192.168.30.165;rport=5060 From: ;tag=as204d1e57 To: "1049" ;tag=as5fb7dcb5 Call-ID: 5ab6e60f016458485828448a79e42c4b@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '5ab6e60f016458485828448a79e42c4b@192.168.30.254' of Request 102: Match Found SIP Response message for INCOMING dialog BYE arrived [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '5ab6e60f016458485828448a79e42c4b@192.168.30.254' Method: ACK Really destroying SIP dialog '79ed0423747cb7ae47d834ad40f70c8d@192.168.30.254' Method: ACK [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24659]: rtp.c:2712 ast_rtp_raw_write: Difference is 1504, ms is 208 [Aug 14 14:14:36] DEBUG[24635]: rtp.c:2712 ast_rtp_raw_write: Difference is 1440, ms is 200 [Aug 14 14:14:36] DEBUG[24487]: rtp.c:2712 ast_rtp_raw_write: Difference is 1456, ms is 202 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24638]: rtp.c:2712 ast_rtp_raw_write: Difference is 1440, ms is 200 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:36] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b4827668", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:36] DEBUG[24665]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '52' [Aug 14 14:14:36] DEBUG[24665]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:36] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b4827668", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:36] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b4827668", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:36] DEBUG[24665]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b4827668", "30") in new stack [Aug 14 14:14:36] DEBUG[24665]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b4827668' does not support indication 8, emulating it [Aug 14 14:14:36] DEBUG[24665]: channel.c:3090 set_format: Set channel SIP/siptrunk-b4827668 to write format slin [Aug 14 14:14:36] DEBUG[24665]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:36] DEBUG[24665]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24660]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b482a398", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:36] DEBUG[24660]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '53' [Aug 14 14:14:36] DEBUG[24660]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:36] DEBUG[24660]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b482a398", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:36] DEBUG[24660]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b482a398", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:36] DEBUG[24660]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b482a398", "30") in new stack [Aug 14 14:14:36] DEBUG[24660]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b482a398' does not support indication 8, emulating it [Aug 14 14:14:36] DEBUG[24660]: channel.c:3090 set_format: Set channel SIP/siptrunk-b482a398 to write format slin [Aug 14 14:14:36] DEBUG[24660]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:36] DEBUG[24660]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:36] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b4872d88", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:36] DEBUG[24663]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '54' [Aug 14 14:14:36] DEBUG[24663]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:36] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b4872d88", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:36] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b4872d88", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:36] DEBUG[24663]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b4872d88", "30") in new stack [Aug 14 14:14:36] DEBUG[24663]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b4872d88' does not support indication 8, emulating it [Aug 14 14:14:36] DEBUG[24663]: channel.c:3090 set_format: Set channel SIP/siptrunk-b4872d88 to write format slin [Aug 14 14:14:36] DEBUG[24663]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:36] DEBUG[24663]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1751f074;rport From: "1028" ;tag=as6d417735 To: Contact: Call-ID: 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:36 GMT llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 16254 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1751f074;received=192.168.30.254;rport=5060 From: "1028" ;tag=as6d417735 To: ;tag=as48c7addf Call-ID: 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59b857b8" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '21d5341121e98cb873bf78c74b4fb26d@192.168.30.254' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1751f074;rport From: "1028" ;tag=as6d417735 To: ;tag=as48c7addf Contact: Call-ID: 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '21d5341121e98cb873bf78c74b4fb26d@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK5352f073;rport From: "1028" ;tag=as6d417735 To: Contact: Call-ID: 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="59b857b8", response="5632c70151b371645f5f3059aca14a63" Date: Fri, 14 Aug 2009 18:14:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 16254 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:16254 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:16254 [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b58dc998: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK5352f073;received=192.168.30.254;rport=5060 From: "1028" ;tag=as6d417735 To: Call-ID: 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:36] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:36] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Ping' [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b58dc998", "") in new stack [Aug 14 14:14:36] DEBUG[24671]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:36] DEBUG[24671]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b58dc998 [Aug 14 14:14:36] DEBUG[24671]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:36] DEBUG[24671]: chan_sip.c:6761 add_sdp: <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK08b2f94e;rport From: "1023" ;tag=as0474b639 To: Contact: Call-ID: 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 10752 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- ** Our capability: 0x4 (ulaw) Video flag: True [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24332]: channel.c:3090 set_format: Set channel SIP/siptrunk-b482bc78 to write format ulaw [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:36] DEBUG[24332]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b482bc78' [Aug 14 14:14:36] DEBUG[24332]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b482bc78, SIP callid 4fcfb2520532c6bd465f34ab55044546@192.168.30.254) Scheduling destruction of SIP dialog '4fcfb2520532c6bd465f34ab55044546@192.168.30.254' in 32000 ms (Method: ACK) [Aug 14 14:14:36] DEBUG[24332]: chan_sip.c:6283 reqprep: Strict routing enforced for session 4fcfb2520532c6bd465f34ab55044546@192.168.30.254 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.254, port 5060 Reliably Transmitting (NAT) to 192.168.30.254:5060: BYE sip:1000@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK38bb1784;rport From: ;tag=as59674dff To: "1000" ;tag=as6bd0ba46 Call-ID: 4fcfb2520532c6bd465f34ab55044546@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Aug 14 14:14:36] DEBUG[24332]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK08b2f94e;received=192.168.30.254;rport=5060 From: "1023" ;tag=as0474b639 To: ;tag=as51af1d61 Call-ID: 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0971acab" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK38bb1784;received=192.168.30.165;rport=5060 From: ;tag=as59674dff To: "1000" ;tag=as6bd0ba46 Call-ID: 4fcfb2520532c6bd465f34ab55044546@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 -------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '4fcfb2520532c6bd465f34ab55044546@192.168.30.254' of Request 102: Match Found SIP Response message for INCOMING dialog BYE arrived [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK08b2f94e;rport From: "1023" ;tag=as0474b639 To: ;tag=as51af1d61 Contact: Call-ID: 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4ef5dfaa;rport From: "1023" ;tag=as0474b639 To: Contact: Call-ID: 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="0971acab", response="a2cda014e9b87113a383ec5723f00c7c" Date: Fri, 14 Aug 2009 18:14:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 10752 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Found peer 'siptrunk' Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:10752 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:10752 [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b4d1d670: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4ef5dfaa;received=192.168.30.254;rport=5060 From: "1023" ;tag=as0474b639 To: Call-ID: 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:36] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:36] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24671]: chan_sip.c:6762 add_sdp: ** Our prefcodec: 0x0 (nothing) [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Audio is at 192.168.30.165 port 12876 [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b4d1d670", "") in new stack [Aug 14 14:14:36] DEBUG[24672]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24672]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b4d1d670 [Aug 14 14:14:36] DEBUG[24672]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:36] DEBUG[24672]: chan_sip.c:6761 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Aug 14 14:14:36] DEBUG[24672]: chan_sip.c:6762 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.30.165 port 12870 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Aug 14 14:14:36] DEBUG[24672]: chan_sip.c:6879 add_sdp: -- Done with adding codecs to SDP [Aug 14 14:14:36] DEBUG[24672]: chan_sip.c:6988 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) <--- Reliably Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4ef5dfaa;received=192.168.30.254;rport=5060 From: "1023" ;tag=as0474b639 To: ;tag=as058c4965 Call-ID: 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 12870 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b4d1d670", "CHANNEL(musicclass)=default") in new stack [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1023' [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1023' [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b4d1d670", ""INCOMING CALL FROM CALLER ID: 1023 (1023)"") in new stack [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:4] Set("SIP/siptrunk-b4d1d670", "__INCOMING_DNIS=5000") in new stack [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b4d1d670", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b4d1d670", "CDR(userfield)=5000") in new stack [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b4d1d670", "10") in new stack -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b4d1d670", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24128]: rtp.c:923 ast_rtcp_read: Got RTCP report of 44 bytes [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK398e41d1;rport From: "1023" ;tag=as0474b639 To: ;tag=as058c4965 Contact: Call-ID: 09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '09a680aa60db2ba57ba2d43e2e2e523d@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Aug 14 14:14:36] DEBUG[24671]: chan_sip.c:6879 add_sdp: -- Done with adding codecs to SDP [Aug 14 14:14:36] DEBUG[24671]: chan_sip.c:6988 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) <--- Reliably Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK5352f073;received=192.168.30.254;rport=5060 From: "1028" ;tag=as6d417735 To: ;tag=as3215cbe1 Call-ID: 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 12876 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b58dc998", "CHANNEL(musicclass)=default") in new stack <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK14ddd87c;rport From: "1028" ;tag=as6d417735 To: ;tag=as3215cbe1 Contact: Call-ID: 21d5341121e98cb873bf78c74b4fb26d@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '21d5341121e98cb873bf78c74b4fb26d@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1028' [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1028' [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b58dc998", ""INCOMING CALL FROM CALLER ID: 1028 (1028)"") in new stack [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:4] Set("SIP/siptrunk-b58dc998", "__INCOMING_DNIS=5000") in new stack [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b58dc998", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b58dc998", "CDR(userfield)=5000") in new stack Really destroying SIP dialog '4fcfb2520532c6bd465f34ab55044546@192.168.30.254' Method: ACK [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b58dc998", "10") in new stack -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:36] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b58dc998", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24319]: channel.c:3090 set_format: Set channel SIP/siptrunk-b58c5f08 to write format ulaw [Aug 14 14:14:36] DEBUG[24319]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b58c5f08' [Aug 14 14:14:36] DEBUG[24319]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b58c5f08, SIP callid 1aaca22e2f0450ac490c9ca82e5d06b8@192.168.30.254) Scheduling destruction of SIP dialog '1aaca22e2f0450ac490c9ca82e5d06b8@192.168.30.254' in 32000 ms (Method: ACK) [Aug 14 14:14:36] DEBUG[24304]: channel.c:3090 set_format: Set channel SIP/siptrunk-b4687fb8 to write format ulaw [Aug 14 14:14:36] DEBUG[24319]: chan_sip.c:6283 reqprep: Strict routing enforced for session 1aaca22e2f0450ac490c9ca82e5d06b8@192.168.30.254 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.254, port 5060 Reliably Transmitting (NAT) to 192.168.30.254:5060: BYE sip:1043@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK61be78b2;rport From: ;tag=as5446ef45 To: "1043" ;tag=as3563146b Call-ID: 1aaca22e2f0450ac490c9ca82e5d06b8@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Aug 14 14:14:36] DEBUG[24191]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b46ae400' <--- SIP read from 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK61be78b2;received=192.168.30.165;rport=5060 From: ;tag=as5446ef45 To: "1043" ;tag=as3563146b Call-ID: 1aaca22e2f0450ac490c9ca82e5d06b8@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[24191]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b46ae400, SIP callid 59b2f6733453108222addeb974567fd8@192.168.30.254) Scheduling destruction of SIP dialog '59b2f6733453108222addeb974567fd8@192.168.30.254' in 32000 ms (Method: ACK) [Aug 14 14:14:36] DEBUG[24319]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:36] DEBUG[24304]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b4687fb8' [Aug 14 14:14:36] DEBUG[24304]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b4687fb8, SIP callid 153968e556b467b951befef63e8a0714@192.168.30.254) Scheduling destruction of SIP dialog '153968e556b467b951befef63e8a0714@192.168.30.254' in 32000 ms (Method: ACK) [Aug 14 14:14:36] DEBUG[24191]: chan_sip.c:6283 reqprep: Strict routing enforced for session 59b2f6733453108222addeb974567fd8@192.168.30.254 set_destination: Parsing for address/port to send to [Aug 14 14:14:36] DEBUG[24304]: chan_sip.c:6283 reqprep: Strict routing enforced for session 153968e556b467b951befef63e8a0714@192.168.30.254 [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '1aaca22e2f0450ac490c9ca82e5d06b8@192.168.30.254' of Request 102: Match Found [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. set_destination: set destination to 192.168.30.254, port 5060 Reliably Transmitting (NAT) to 192.168.30.254:5060: BYE sip:1014@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK4dc26968;rport From: ;tag=as0b4e0bfb To: "1014" ;tag=as27018265 Call-ID: 59b2f6733453108222addeb974567fd8@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Aug 14 14:14:36] DEBUG[24191]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.254, port 5060 Reliably Transmitting (NAT) to 192.168.30.254:5060: BYE sip:1038@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK1df304dd;rport From: ;tag=as4ff0320c To: "1038" ;tag=as1c4ab69a Call-ID: 153968e556b467b951befef63e8a0714@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Aug 14 14:14:36] DEBUG[24304]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) SIP Response message for INCOMING dialog BYE arrived <--- SIP read from 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK4dc26968;received=192.168.30.165;rport=5060 From: ;tag=as0b4e0bfb To: "1014" ;tag=as27018265 Call-ID: 59b2f6733453108222addeb974567fd8@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '59b2f6733453108222addeb974567fd8@192.168.30.254' of Request 102: Match Found SIP Response message for INCOMING dialog BYE arrived <--- SIP read from 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK1df304dd;received=192.168.30.165;rport=5060 From: ;tag=as4ff0320c To: "1038" ;tag=as1c4ab69a Call-ID: 153968e556b467b951befef63e8a0714@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '153968e556b467b951befef63e8a0714@192.168.30.254' of Request 102: Match Found SIP Response message for INCOMING dialog BYE arrived [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b4d1d670", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '51' [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b4d1d670", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b4d1d670", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:36] DEBUG[24672]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b4d1d670", "30") in new stack [Aug 14 14:14:36] DEBUG[24672]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b4d1d670' does not support indication 8, emulating it [Aug 14 14:14:36] DEBUG[24672]: channel.c:3090 set_format: Set channel SIP/siptrunk-b4d1d670 to write format slin [Aug 14 14:14:36] DEBUG[24672]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:36] DEBUG[24672]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] DEBUG[24216]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b465bf20' [Aug 14 14:14:36] DEBUG[24216]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b465bf20, SIP callid 4d6d5c76159d07bf59c07a4a44abbca2@192.168.30.254) Scheduling destruction of SIP dialog '4d6d5c76159d07bf59c07a4a44abbca2@192.168.30.254' in 32000 ms (Method: ACK) [Aug 14 14:14:36] DEBUG[24216]: chan_sip.c:6283 reqprep: Strict routing enforced for session 4d6d5c76159d07bf59c07a4a44abbca2@192.168.30.254 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.254, port 5060 Reliably Transmitting (NAT) to 192.168.30.254:5060: BYE sip:1034@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK0a73ccb0;rport From: ;tag=as71491d08 To: "1034" ;tag=as2269d9a7 Call-ID: 4d6d5c76159d07bf59c07a4a44abbca2@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Aug 14 14:14:36] DEBUG[24216]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk <--- SIP read from 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK0a73ccb0;received=192.168.30.165;rport=5060 From: ;tag=as71491d08 To: "1034" ;tag=as2269d9a7 Call-ID: 4d6d5c76159d07bf59c07a4a44abbca2@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:36] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '4d6d5c76159d07bf59c07a4a44abbca2@192.168.30.254' of Request 102: Match Found SIP Response message for INCOMING dialog BYE arrived [Aug 14 14:14:36] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:36] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:36] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '1aaca22e2f0450ac490c9ca82e5d06b8@192.168.30.254' Method: ACK Really destroying SIP dialog '153968e556b467b951befef63e8a0714@192.168.30.254' Method: ACK [Aug 14 14:14:36] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '4d6d5c76159d07bf59c07a4a44abbca2@192.168.30.254' Method: ACK Really destroying SIP dialog '59b2f6733453108222addeb974567fd8@192.168.30.254' Method: ACK [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[24486]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:37] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[24663]: rtp.c:2712 ast_rtp_raw_write: Difference is 1432, ms is 199 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[24665]: rtp.c:2712 ast_rtp_raw_write: Difference is 1536, ms is 212 [Aug 14 14:14:37] DEBUG[24637]: rtp.c:2712 ast_rtp_raw_write: Difference is 1592, ms is 219 [Aug 14 14:14:37] DEBUG[24635]: rtp.c:2712 ast_rtp_raw_write: Difference is 1560, ms is 215 [Aug 14 14:14:37] DEBUG[24639]: rtp.c:2712 ast_rtp_raw_write: Difference is 1576, ms is 217 [Aug 14 14:14:37] DEBUG[24638]: rtp.c:2712 ast_rtp_raw_write: Difference is 1584, ms is 218 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:37] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[24312]: channel.c:3090 set_format: Set channel SIP/siptrunk-b50fa068 to write format ulaw [Aug 14 14:14:37] DEBUG[24312]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b50fa068' [Aug 14 14:14:37] DEBUG[24312]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b50fa068, SIP callid 1ee5cbe75822d4cf558e285d35018961@192.168.30.254) Scheduling destruction of SIP dialog '1ee5cbe75822d4cf558e285d35018961@192.168.30.254' in 32000 ms (Method: ACK) [Aug 14 14:14:37] DEBUG[24312]: chan_sip.c:6283 reqprep: Strict routing enforced for session 1ee5cbe75822d4cf558e285d35018961@192.168.30.254 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.254, port 5060 Reliably Transmitting (NAT) to 192.168.30.254:5060: BYE sip:1008@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK5257ea58;rport From: ;tag=as1571513d To: "1008" ;tag=as075e6b90 Call-ID: 1ee5cbe75822d4cf558e285d35018961@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Aug 14 14:14:37] DEBUG[24312]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:37] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:37] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK5257ea58;received=192.168.30.165;rport=5060 From: ;tag=as1571513d To: "1008" ;tag=as075e6b90 Call-ID: 1ee5cbe75822d4cf558e285d35018961@192.168.30.254 CSeq: 102 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:37] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '1ee5cbe75822d4cf558e285d35018961@192.168.30.254' of Request 102: Match Found SIP Response message for INCOMING dialog BYE arrived [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '1ee5cbe75822d4cf558e285d35018961@192.168.30.254' Method: ACK [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[23891]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[24647]: channel.c:3090 set_format: Set channel SIP/siptrunk-b4830b88 to write format slin [Aug 14 14:14:37] DEBUG[24647]: res_musiconhold.c:261 ast_moh_files_next: SIP/siptrunk-b4830b88 Opened file 2 '/var/lib/asterisk/moh//fpm-calm-river' [Aug 14 14:14:37] DEBUG[24647]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:37] DEBUG[24647]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:37] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '29d7461e3172050b492f23294e373ed6@127.0.0.1' [Aug 14 14:14:37] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 29d7461e3172050b492f23294e373ed6@127.0.0.1 Really destroying SIP dialog '29d7461e3172050b492f23294e373ed6@127.0.0.1' Method: REGISTER [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '538f66564779c29f1c5aa7de5d146fdb@127.0.0.1' [Aug 14 14:14:37] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 538f66564779c29f1c5aa7de5d146fdb@127.0.0.1 Really destroying SIP dialog '538f66564779c29f1c5aa7de5d146fdb@127.0.0.1' Method: REGISTER [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '3b0514690c1d105e03cacc3e0765226a@127.0.0.1' [Aug 14 14:14:37] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 3b0514690c1d105e03cacc3e0765226a@127.0.0.1 Really destroying SIP dialog '3b0514690c1d105e03cacc3e0765226a@127.0.0.1' Method: REGISTER [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[24161]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:37] DEBUG[24160]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:37] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[24337]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:38] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[23779]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b58dc998", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '50' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b58dc998", "0?500") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:6065 pbx_builtin_gotoif: Not taking any branch [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:13] Set("SIP/siptrunk-b58dc998", "FAXNUMBER=5000") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:14] Set("SIP/siptrunk-b58dc998", "__INCOMINGLINE=1") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1028' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:15] GotoIf("SIP/siptrunk-b58dc998", "0?16:17") in new stack -- Goto (private-siptrunk-incoming,5000,17) [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1028' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:17] GotoIf("SIP/siptrunk-b58dc998", "0?18:19") in new stack -- Goto (private-siptrunk-incoming,5000,19) [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1028' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:19] GotoIf("SIP/siptrunk-b58dc998", "0?20:21") in new stack -- Goto (private-siptrunk-incoming,5000,21) [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1028' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:21] GotoIf("SIP/siptrunk-b58dc998", "0?22:23") in new stack [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Goto (private-siptrunk-incoming,5000,23) [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:23] Set("SIP/siptrunk-b58dc998", "CHANNEL(language)=en") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:24] Set("SIP/siptrunk-b58dc998", "QUEUE_PRIO=0") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Gosub' -- Executing [5000@private-siptrunk-incoming:25] Gosub("SIP/siptrunk-b58dc998", "default-application-acd-customer-new-english|s|1") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [s@default-application-acd-customer-new-english:1] NoOp("SIP/siptrunk-b58dc998", ""=== START QUEUE (default-customer-new-english) ==="") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [s@default-application-acd-customer-new-english:2] Set("SIP/siptrunk-b58dc998", "__QUEUENAME=default-customer-new-english") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '6' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [s@default-application-acd-customer-new-english:3] GotoIf("SIP/siptrunk-b58dc998", "0?6") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:6065 pbx_builtin_gotoif: Not taking any branch [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [s@default-application-acd-customer-new-english:4] Set("SIP/siptrunk-b58dc998", "MONITOR_FILENAME=1250273676.6307") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '7' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [s@default-application-acd-customer-new-english:5] Goto("SIP/siptrunk-b58dc998", "7") in new stack -- Goto (default-application-acd-customer-new-english,s,7) [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [s@default-application-acd-customer-new-english:7] Set("SIP/siptrunk-b58dc998", "TIMEOUT(digit)=1") in new stack -- Digit timeout set to 1 [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [s@default-application-acd-customer-new-english:8] Set("SIP/siptrunk-b58dc998", "TIMEOUT(response)=1") in new stack -- Response timeout set to 1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [s@default-application-acd-customer-new-english:9] Set("SIP/siptrunk-b58dc998", "__ACD_TO_OUTGOING=1") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [s@default-application-acd-customer-new-english:10] Set("SIP/siptrunk-b58dc998", "CHANNEL(musicclass)=default") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [s@default-application-acd-customer-new-english:11] Set("SIP/siptrunk-b58dc998", "__ALLOW_TRANSFER=twk") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1028' [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [s@default-application-acd-customer-new-english:12] Set("SIP/siptrunk-b58dc998", "CALLERID(name)=customer-E:1028") in new stack [Aug 14 14:14:38] DEBUG[24671]: pbx.c:1843 pbx_extension_helper: Launching 'Queue' -- Executing [s@default-application-acd-customer-new-english:13] Queue("SIP/siptrunk-b58dc998", "default-customer-new-english|tH|||60|") in new stack [Aug 14 14:14:38] DEBUG[24671]: app_queue.c:3966 queue_exec: SIP/siptrunk-b58dc998: Got priority 0 from ${QUEUE_PRIO}. [Aug 14 14:14:38] DEBUG[24671]: app_queue.c:3998 queue_exec: queue: default-customer-new-english, options: tH, url: , announce: , expires: 1250273738, priority: 0 [Aug 14 14:14:38] DEBUG[24671]: app_queue.c:1352 update_realtime_members: Queue default-customer-new-english has no realtime members defined. No need for update [Aug 14 14:14:38] DEBUG[24671]: app_queue.c:1497 join_queue: Queue 'default-customer-new-english' Join, Channel 'SIP/siptrunk-b58dc998', Position '3' -- Started music on hold, class 'default', on SIP/siptrunk-b58dc998 [Aug 14 14:14:38] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:38] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[24671]: channel.c:3090 set_format: Set channel SIP/siptrunk-b58dc998 to write format slin [Aug 14 14:14:38] DEBUG[24671]: res_musiconhold.c:261 ast_moh_files_next: SIP/siptrunk-b58dc998 Opened file 1 '/var/lib/asterisk/moh//fpm-world-mix' [Aug 14 14:14:38] DEBUG[24671]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:38] DEBUG[24671]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:38] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[24634]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[24635]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:38] DEBUG[24637]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[24638]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] DEBUG[2499]: manager.c:2230 process_message: Manager received command 'Ping' [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:38] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] DEBUG[23836]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] DEBUG[23837]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:39] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] DEBUG[24660]: rtp.c:2712 ast_rtp_raw_write: Difference is 720, ms is 110 [Aug 14 14:14:39] DEBUG[24659]: rtp.c:2712 ast_rtp_raw_write: Difference is 728, ms is 111 [Aug 14 14:14:39] DEBUG[24647]: rtp.c:2712 ast_rtp_raw_write: Difference is 744, ms is 113 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:39] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:39] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] DEBUG[24639]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:39] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] DEBUG[2483]: chan_iax2.c:9040 iax2_do_register: Allocate call number [Aug 14 14:14:40] DEBUG[2483]: chan_iax2.c:9046 iax2_do_register: Registration created on call 9069 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:40] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:40] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:40] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] DEBUG[24660]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:40] DEBUG[24659]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] DEBUG[24663]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:40] DEBUG[23948]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] DEBUG[24487]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:40] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] DEBUG[24665]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:41] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:41] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:41] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:41] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] DEBUG[24128]: rtp.c:923 ast_rtcp_read: Got RTCP report of 44 bytes [Aug 14 14:14:42] DEBUG[24672]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] DEBUG[24486]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:42] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:42] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:42] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] DEBUG[23891]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] DEBUG[24161]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:42] DEBUG[24160]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:42] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[24337]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:43] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 asterisk*CLI> <--- SIP read from 192.168.30.146:5060 ---> REGISTER sip:agent_1614 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK0c0f3c0a;rport From: ;tag=as285d9239 To: Call-ID: 22e436f577ce15d40722fbb250d54fc0@127.0.0.1 CSeq: 50628 REGISTER User-Agent: Test Framework 1.0 Max-Forwards: 70 Authorization: Digest username="1614", realm="asterisk", algorithm=MD5, uri="sip:agent_1614", nonce="0e71ec92", response="1095f1c7ebe98f269dde2a89b4e18c3b" Expires: 120 Contact: Event: registration Content-Length: 0 <-------------> --- (13 headers 0 lines) --- [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 22e436f577ce15d40722fbb250d54fc0@127.0.0.1 - REGISTER (No RTP) Using latest REGISTER request as basis request Sending to 192.168.30.146 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK0c0f3c0a;received=192.168.30.146;rport=5060 From: ;tag=as285d9239 To: Call-ID: 22e436f577ce15d40722fbb250d54fc0@127.0.0.1 CSeq: 50628 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK0c0f3c0a;received=192.168.30.146;rport=5060 From: ;tag=as285d9239 To: ;tag=as07553d8d Call-ID: 22e436f577ce15d40722fbb250d54fc0@127.0.0.1 CSeq: 50628 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2541c271" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '22e436f577ce15d40722fbb250d54fc0@127.0.0.1' in 32000 ms (Method: REGISTER) asterisk*CLI> <--- SIP read from 192.168.30.146:5060 ---> REGISTER sip:agent_1614 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK2fec327d;rport From: ;tag=as3fc6f4d1 To: Call-ID: 22e436f577ce15d40722fbb250d54fc0@127.0.0.1 CSeq: 50629 REGISTER User-Agent: Test Framework 1.0 Max-Forwards: 70 Authorization: Digest username="1614", realm="asterisk", algorithm=MD5, uri="sip:agent_1614", nonce="2541c271", response="f30eee739d2a37e48f56977723b64e3c" Expires: 120 Contact: Event: registration Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.30.146 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK2fec327d;received=192.168.30.146;rport=5060 From: ;tag=as3fc6f4d1 To: Call-ID: 22e436f577ce15d40722fbb250d54fc0@127.0.0.1 CSeq: 50629 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:43] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.146:5060: OPTIONS sip:1614@192.168.30.146 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK31caea64;rport From: "asterisk" ;tag=as016f4213 To: Contact: Call-ID: 340090676c97e13d4e057cf33aa01be0@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK2fec327d;received=192.168.30.146;rport=5060 From: ;tag=as3fc6f4d1 To: ;tag=as07553d8d Call-ID: 22e436f577ce15d40722fbb250d54fc0@127.0.0.1 CSeq: 50629 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 120 Contact: ;expires=120 Date: Fri, 14 Aug 2009 18:14:43 GMT Content-Length: 0 <------------> [Aug 14 14:14:43] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1614 Scheduling destruction of SIP dialog '22e436f577ce15d40722fbb250d54fc0@127.0.0.1' in 32000 ms (Method: REGISTER) [Aug 14 14:14:43] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1614 [Aug 14 14:14:43] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/1614 - state 1 (Not in use) [Aug 14 14:14:43] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1614 <--- SIP read from 192.168.30.146:5060 ---> REGISTER sip:agent_1616 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK61707b18;rport From: ;tag=as2dac9fa7 To: Call-ID: 2edbb4d121feeb3d6be3c6c72ad8bfbf@127.0.0.1 CSeq: 50661 REGISTER User-Agent: Test Framework 1.0 Max-Forwards: 70 Authorization: Digest username="1616", realm="asterisk", algorithm=MD5, uri="sip:agent_1616", nonce="2ff06d79", response="34734fcf19e84caa7868771400d1eaf7" Expires: 120 Contact: Event: registration Content-Length: 0 <-------------> --- (13 headers 0 lines) --- [Aug 14 14:14:43] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/1614' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 2edbb4d121feeb3d6be3c6c72ad8bfbf@127.0.0.1 - REGISTER (No RTP) Using latest REGISTER request as basis request Sending to 192.168.30.146 : 5060 (no NAT) asterisk*CLI> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK61707b18;received=192.168.30.146;rport=5060 From: ;tag=as2dac9fa7 To: Call-ID: 2edbb4d121feeb3d6be3c6c72ad8bfbf@127.0.0.1 CSeq: 50661 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> asterisk*CLI> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK61707b18;received=192.168.30.146;rport=5060 From: ;tag=as2dac9fa7 To: ;tag=as44f7850b Call-ID: 2edbb4d121feeb3d6be3c6c72ad8bfbf@127.0.0.1 CSeq: 50661 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08a63471" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2edbb4d121feeb3d6be3c6c72ad8bfbf@127.0.0.1' in 32000 ms (Method: REGISTER) asterisk*CLI> <--- SIP read from 192.168.30.146:5060 ---> REGISTER sip:agent_1649 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK27c38f9d;rport From: ;tag=as46538833 To: Call-ID: 63d5603a462c22ee3e2daeb2401f20d5@127.0.0.1 CSeq: 50655 REGISTER User-Agent: Test Framework 1.0 Max-Forwards: 70 Authorization: Digest username="1649", realm="asterisk", algorithm=MD5, uri="sip:agent_1649", nonce="18e9ed00", response="25e82013e2e6af3e94769aac6cdd5105" Expires: 120 Contact: Event: registration Content-Length: 0 <-------------> --- (13 headers 0 lines) --- [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 63d5603a462c22ee3e2daeb2401f20d5@127.0.0.1 - REGISTER (No RTP) Using latest REGISTER request as basis request Sending to 192.168.30.146 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK27c38f9d;received=192.168.30.146;rport=5060 From: ;tag=as46538833 To: Call-ID: 63d5603a462c22ee3e2daeb2401f20d5@127.0.0.1 CSeq: 50655 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK27c38f9d;received=192.168.30.146;rport=5060 From: ;tag=as46538833 To: ;tag=as3d80588d Call-ID: 63d5603a462c22ee3e2daeb2401f20d5@127.0.0.1 CSeq: 50655 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6b294598" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '63d5603a462c22ee3e2daeb2401f20d5@127.0.0.1' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.30.146:5060 ---> REGISTER sip:agent_1645 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK2543877f;rport From: ;tag=as544793c6 To: Call-ID: 0944fc6663de06897be333e50965caba@127.0.0.1 CSeq: 50670 REGISTER User-Agent: Test Framework 1.0 Max-Forwards: 70 Authorization: Digest username="1645", realm="asterisk", algorithm=MD5, uri="sip:agent_1645", nonce="7bee1630", response="aa0c381fbf7c682a3b1c25d51c3d74a6" Expires: 120 Contact: Event: registration Content-Length: 0 <-------------> --- (13 headers 0 lines) --- [Aug 14 14:14:43] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:43] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 0944fc6663de06897be333e50965caba@127.0.0.1 - REGISTER (No RTP) Using latest REGISTER request as basis request Sending to 192.168.30.146 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK2543877f;received=192.168.30.146;rport=5060 From: ;tag=as544793c6 To: Call-ID: 0944fc6663de06897be333e50965caba@127.0.0.1 CSeq: 50670 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> asterisk*CLI> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK2543877f;received=192.168.30.146;rport=5060 From: ;tag=as544793c6 To: ;tag=as4c267950 Call-ID: 0944fc6663de06897be333e50965caba@127.0.0.1 CSeq: 50670 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76497acb" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0944fc6663de06897be333e50965caba@127.0.0.1' in 32000 ms (Method: REGISTER) asterisk*CLI> <--- SIP read from 192.168.30.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK31caea64;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as016f4213 To: ;tag=as75b97878 Call-ID: 340090676c97e13d4e057cf33aa01be0@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '340090676c97e13d4e057cf33aa01be0@192.168.30.165' of Request 102: Match Found <--- SIP read from 192.168.30.146:5060 ---> REGISTER sip:agent_1616 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK29962e3f;rport From: ;tag=as62215d6d To: Call-ID: 2edbb4d121feeb3d6be3c6c72ad8bfbf@127.0.0.1 CSeq: 50662 REGISTER User-Agent: Test Framework 1.0 Max-Forwards: 70 Authorization: Digest username="1616", realm="asterisk", algorithm=MD5, uri="sip:agent_1616", nonce="08a63471", response="ba81a50f7385bd7307430eac741e4b01" Expires: 120 Contact: Event: registration Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.30.146 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK29962e3f;received=192.168.30.146;rport=5060 From: ;tag=as62215d6d To: Call-ID: 2edbb4d121feeb3d6be3c6c72ad8bfbf@127.0.0.1 CSeq: 50662 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.146:5060: OPTIONS sip:1616@192.168.30.146 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK7d0c7e46;rport From: "asterisk" ;tag=as23561bcc To: Contact: Call-ID: 5da7aaab38d71adf127d384816a0eb2e@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- asterisk*CLI> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK29962e3f;received=192.168.30.146;rport=5060 From: ;tag=as62215d6d To: ;tag=as44f7850b Call-ID: 2edbb4d121feeb3d6be3c6c72ad8bfbf@127.0.0.1 CSeq: 50662 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 120 Contact: ;expires=120 Date: Fri, 14 Aug 2009 18:14:43 GMT Content-Length: 0 <------------> [Aug 14 14:14:43] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1616 [Aug 14 14:14:43] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1616 [Aug 14 14:14:43] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/1616 - state 1 (Not in use) [Aug 14 14:14:43] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1616 [Aug 14 14:14:43] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/1616' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Scheduling destruction of SIP dialog '2edbb4d121feeb3d6be3c6c72ad8bfbf@127.0.0.1' in 32000 ms (Method: REGISTER) asterisk*CLI> <--- SIP read from 192.168.30.146:5060 ---> REGISTER sip:agent_1649 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK2422d713;rport From: ;tag=as21798368 To: Call-ID: 63d5603a462c22ee3e2daeb2401f20d5@127.0.0.1 CSeq: 50656 REGISTER User-Agent: Test Framework 1.0 Max-Forwards: 70 Authorization: Digest username="1649", realm="asterisk", algorithm=MD5, uri="sip:agent_1649", nonce="6b294598", response="9c68fe79f696d2820aae12a8a1fb3e40" Expires: 120 Contact: Event: registration Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.30.146 : 5060 (no NAT) asterisk*CLI> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK2422d713;received=192.168.30.146;rport=5060 From: ;tag=as21798368 To: Call-ID: 63d5603a462c22ee3e2daeb2401f20d5@127.0.0.1 CSeq: 50656 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.146:5060: OPTIONS sip:1649@192.168.30.146 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK7bf45bf7;rport From: "asterisk" ;tag=as73318001 To: Contact: Call-ID: 2fe63cf55dff5b403ef28b341d02547b@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- asterisk*CLI> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK2422d713;received=192.168.30.146;rport=5060 From: ;tag=as21798368 To: ;tag=as3d80588d Call-ID: 63d5603a462c22ee3e2daeb2401f20d5@127.0.0.1 CSeq: 50656 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 120 Contact: ;expires=120 Date: Fri, 14 Aug 2009 18:14:43 GMT Content-Length: 0 <------------> [Aug 14 14:14:43] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1649 [Aug 14 14:14:43] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1649 [Aug 14 14:14:43] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/1649 - state 2 (In use) [Aug 14 14:14:43] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1649 [Aug 14 14:14:43] DEBUG[2541]: app_queue.c:680 handle_statechange: Scheduling destruction of SIP dialog '63d5603a462c22ee3e2daeb2401f20d5@127.0.0.1' in 32000 ms (Method: REGISTER) Device 'SIP/1649' changed to state '2' (In use) asterisk*CLI> <--- SIP read from 192.168.30.146:5060 ---> REGISTER sip:agent_1645 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK6e817fd7;rport From: ;tag=as53ce26d3 To: Call-ID: 0944fc6663de06897be333e50965caba@127.0.0.1 CSeq: 50671 REGISTER User-Agent: Test Framework 1.0 Max-Forwards: 70 Authorization: Digest username="1645", realm="asterisk", algorithm=MD5, uri="sip:agent_1645", nonce="76497acb", response="d317ea593edf1693f429237ff01edf1a" Expires: 120 Contact: Event: registration Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.30.146 : 5060 (no NAT) asterisk*CLI> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK6e817fd7;received=192.168.30.146;rport=5060 From: ;tag=as53ce26d3 To: Call-ID: 0944fc6663de06897be333e50965caba@127.0.0.1 CSeq: 50671 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.146:5060: OPTIONS sip:1645@192.168.30.146 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK41afdec8;rport From: "asterisk" ;tag=as07b88288 To: Contact: Call-ID: 16429a2449e689954f72d2fa0b538090@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- asterisk*CLI> <--- Transmitting (no NAT) to 192.168.30.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.146:5060;branch=z9hG4bK6e817fd7;received=192.168.30.146;rport=5060 From: ;tag=as53ce26d3 To: ;tag=as4c267950 Call-ID: 0944fc6663de06897be333e50965caba@127.0.0.1 CSeq: 50671 REGISTER User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 120 Contact: ;expires=120 Date: Fri, 14 Aug 2009 18:14:43 GMT Content-Length: 0 <------------> [Aug 14 14:14:43] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1645 [Aug 14 14:14:43] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1645 [Aug 14 14:14:43] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/1645 - state 1 (Not in use) [Aug 14 14:14:43] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1645 [Aug 14 14:14:43] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/1645' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Scheduling destruction of SIP dialog '0944fc6663de06897be333e50965caba@127.0.0.1' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.30.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK7d0c7e46;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as23561bcc To: ;tag=as0c59a26e Call-ID: 5da7aaab38d71adf127d384816a0eb2e@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '5da7aaab38d71adf127d384816a0eb2e@192.168.30.165' of Request 102: Match Found <--- SIP read from 192.168.30.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK7bf45bf7;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as73318001 To: ;tag=as61240120 Call-ID: 2fe63cf55dff5b403ef28b341d02547b@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '2fe63cf55dff5b403ef28b341d02547b@192.168.30.165' of Request 102: Match Found <--- SIP read from 192.168.30.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK41afdec8;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as07b88288 To: ;tag=as5a6127e6 Call-ID: 16429a2449e689954f72d2fa0b538090@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:43] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '16429a2449e689954f72d2fa0b538090@192.168.30.165' of Request 102: Match Found [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '16429a2449e689954f72d2fa0b538090@192.168.30.165' Method: OPTIONS Really destroying SIP dialog '2fe63cf55dff5b403ef28b341d02547b@192.168.30.165' Method: OPTIONS Really destroying SIP dialog '5da7aaab38d71adf127d384816a0eb2e@192.168.30.165' Method: OPTIONS [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '340090676c97e13d4e057cf33aa01be0@192.168.30.165' Method: OPTIONS [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[2499]: manager.c:2230 process_message: Manager received command 'Ping' [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[2499]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[2499]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:43] DEBUG[2499]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] DEBUG[2499]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:43] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:44] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:44] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:44] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24672]: rtp.c:2712 ast_rtp_raw_write: Difference is 15696, ms is 1982 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24671]: rtp.c:2712 ast_rtp_raw_write: Difference is 15704, ms is 1983 [Aug 14 14:14:44] DEBUG[24663]: rtp.c:2712 ast_rtp_raw_write: Difference is 15696, ms is 1982 [Aug 14 14:14:44] DEBUG[24638]: rtp.c:2712 ast_rtp_raw_write: Difference is 15712, ms is 1984 [Aug 14 14:14:44] DEBUG[24660]: rtp.c:2712 ast_rtp_raw_write: Difference is 15832, ms is 1999 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24647]: rtp.c:2712 ast_rtp_raw_write: Difference is 15944, ms is 2013 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24659]: rtp.c:2712 ast_rtp_raw_write: Difference is 16064, ms is 2028 [Aug 14 14:14:44] DEBUG[24486]: rtp.c:2712 ast_rtp_raw_write: Difference is 15984, ms is 2018 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24637]: rtp.c:2712 ast_rtp_raw_write: Difference is 16160, ms is 2040 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24634]: rtp.c:2712 ast_rtp_raw_write: Difference is 16320, ms is 2060 [Aug 14 14:14:44] DEBUG[24635]: rtp.c:2712 ast_rtp_raw_write: Difference is 16304, ms is 2058 [Aug 14 14:14:44] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1658@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK56a82079;rport From: "asterisk" ;tag=as2ee29c79 To: Contact: Call-ID: 2ce4dcaf0822064b0f7743d834b9064a@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- asterisk*CLI> <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK56a82079;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as2ee29c79 To: ;tag=as4a7e49bc Call-ID: 2ce4dcaf0822064b0f7743d834b9064a@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:44] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '2ce4dcaf0822064b0f7743d834b9064a@192.168.30.165' of Request 102: Match Found [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24639]: rtp.c:2712 ast_rtp_raw_write: Difference is 16400, ms is 2070 [Aug 14 14:14:44] DEBUG[24487]: rtp.c:2712 ast_rtp_raw_write: Difference is 16432, ms is 2074 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '2ce4dcaf0822064b0f7743d834b9064a@192.168.30.165' Method: OPTIONS [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] DEBUG[24671]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] DEBUG[24647]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] DEBUG[23779]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] DEBUG[24634]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] DEBUG[24635]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] DEBUG[24637]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] DEBUG[24638]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] DEBUG[23836]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] DEBUG[23837]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] DEBUG[24639]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:44] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:45] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:45] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:45] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] DEBUG[23948]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:45] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[24660]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:46] DEBUG[24659]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:46] DEBUG[24663]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:46] DEBUG[24487]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:46] DEBUG[24665]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:46] DEBUG[24665]: rtp.c:2712 ast_rtp_raw_write: Difference is 2200, ms is 295 [Aug 14 14:14:46] DEBUG[24660]: rtp.c:2712 ast_rtp_raw_write: Difference is 4472, ms is 579 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[24487]: rtp.c:2712 ast_rtp_raw_write: Difference is 3280, ms is 430 [Aug 14 14:14:46] DEBUG[24659]: rtp.c:2712 ast_rtp_raw_write: Difference is 4520, ms is 585 [Aug 14 14:14:46] DEBUG[24663]: rtp.c:2712 ast_rtp_raw_write: Difference is 4280, ms is 555 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 asterisk*CLI> <--- SIP read from 192.168.30.254:5060 ---> BYE sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4b0ea389;rport From: "1039" ;tag=as3e653cff To: ;tag=as7dd36c66 Call-ID: 23736d4664e4b432166cc90933a774c6@192.168.30.254 CSeq: 104 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165", nonce="2de5fe04", response="88e6f8649ab2b5c5bb169a243f0e82f1" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.30.254 : 5060 (NAT) [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:1701 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 23736d4664e4b432166cc90933a774c6@192.168.30.254 [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:15771 handle_request_bye: Received bye, issuing owner hangup asterisk*CLI> <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4b0ea389;received=192.168.30.254;rport=5060 From: "1039" ;tag=as3e653cff To: ;tag=as7dd36c66 Call-ID: 23736d4664e4b432166cc90933a774c6@192.168.30.254 CSeq: 104 BYE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Aug 14 14:14:46] DEBUG[24128]: channel.c:4229 ast_generic_bridge: Didn't get a frame from channel: SIP/siptrunk-b46907c8 [Aug 14 14:14:46] DEBUG[24128]: channel.c:4596 ast_channel_bridge: Bridge stops bridging channels SIP/siptrunk-b46907c8 and Local/1643@default-agent-9ec7,1 [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [h@default-application-acd-customer-new-english:1] Goto("SIP/siptrunk-b46907c8", "all-hangup|s|1") in new stack -- Goto (all-hangup,s,1) [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '2' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [s@all-hangup:1] GotoIf("SIP/siptrunk-b46907c8", "0?all-faxnotify|s|1:2") in new stack -- Goto (all-hangup,s,2) [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1843 pbx_extension_helper: Launching 'ResetCDR' -- Executing [s@all-hangup:2] ResetCDR("SIP/siptrunk-b46907c8", "w") in new stack [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '"customer-E:1039" <1039>' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1039' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 's' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'default-application-acd-customer-new-english' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'SIP/siptrunk-b46907c8' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'Local/1643@default-agent-9ec7,1' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'Queue' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'default-customer-new-english|tH|||60|' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:13:21' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:13:21' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:46' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '85' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '85' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'siptrunk' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1250273601.6241' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '5000' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1843 pbx_extension_helper: Launching 'NoCDR' -- Executing [s@all-hangup:3] NoCDR("SIP/siptrunk-b46907c8", "") in new stack [Aug 14 14:14:46] DEBUG[24128]: pbx.c:1843 pbx_extension_helper: Launching 'System' -- Executing [s@all-hangup:4] System("SIP/siptrunk-b46907c8", "/var/www/scopserv/telephony/scripts/billing/cdr.sh 1250273601.6241") in new stack [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:46] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1689@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK384b36f3;rport From: "asterisk" ;tag=as2104e592 To: Contact: Call-ID: 1758c3d5655e40c91ffe5ca93a1b2434@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- asterisk*CLI> <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK384b36f3;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as2104e592 To: ;tag=as05cf271b Call-ID: 1758c3d5655e40c91ffe5ca93a1b2434@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '1758c3d5655e40c91ffe5ca93a1b2434@192.168.30.165' of Request 102: Match Found [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1666@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK2b69719f;rport From: "asterisk" ;tag=as2752fa3c To: Contact: Call-ID: 0fbd1d06060697d1135bd47236b1f03e@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1687@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK49f946d8;rport From: "asterisk" ;tag=as0016a841 To: Contact: Call-ID: 047dfc7f74d73f665f5c5927528dd929@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK2b69719f;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as2752fa3c To: ;tag=as367ff00c Call-ID: 0fbd1d06060697d1135bd47236b1f03e@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '0fbd1d06060697d1135bd47236b1f03e@192.168.30.165' of Request 102: Match Found <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK49f946d8;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as0016a841 To: ;tag=as3e3fea44 Call-ID: 047dfc7f74d73f665f5c5927528dd929@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '047dfc7f74d73f665f5c5927528dd929@192.168.30.165' of Request 102: Match Found [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:46] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[24635]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:46] DEBUG[24635]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '047dfc7f74d73f665f5c5927528dd929@192.168.30.165' Method: OPTIONS Really destroying SIP dialog '0fbd1d06060697d1135bd47236b1f03e@192.168.30.165' Method: OPTIONS Really destroying SIP dialog '1758c3d5655e40c91ffe5ca93a1b2434@192.168.30.165' Method: OPTIONS [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[24634]: rtp.c:2712 ast_rtp_raw_write: Difference is 728, ms is 111 [Aug 14 14:14:46] DEBUG[24639]: rtp.c:2712 ast_rtp_raw_write: Difference is 744, ms is 113 [Aug 14 14:14:46] DEBUG[24486]: rtp.c:2712 ast_rtp_raw_write: Difference is 728, ms is 111 [Aug 14 14:14:46] DEBUG[24663]: rtp.c:2712 ast_rtp_raw_write: Difference is 704, ms is 108 [Aug 14 14:14:46] DEBUG[24647]: rtp.c:2712 ast_rtp_raw_write: Difference is 736, ms is 112 [Aug 14 14:14:46] DEBUG[24671]: rtp.c:2712 ast_rtp_raw_write: Difference is 720, ms is 110 [Aug 14 14:14:46] DEBUG[24659]: rtp.c:2712 ast_rtp_raw_write: Difference is 736, ms is 112 [Aug 14 14:14:46] DEBUG[24665]: rtp.c:2712 ast_rtp_raw_write: Difference is 728, ms is 111 [Aug 14 14:14:46] DEBUG[24128]: channel.c:1546 ast_hangup: Hanging up channel 'Local/1643@default-agent-9ec7,1' [Aug 14 14:14:46] DEBUG[24337]: channel.c:4229 ast_generic_bridge: Didn't get a frame from channel: Local/1643@default-agent-9ec7,2 [Aug 14 14:14:46] DEBUG[24128]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel Local/1643@default-agent [Aug 14 14:14:46] DEBUG[24337]: channel.c:4596 ast_channel_bridge: Bridge stops bridging channels Local/1643@default-agent-9ec7,2 and SIP/1643-0a0597b0 [Aug 14 14:14:46] DEBUG[24337]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/1643-0a0597b0' [Aug 14 14:14:46] DEBUG[24337]: chan_sip.c:3636 sip_hangup: Hangup call SIP/1643-0a0597b0, SIP callid 0bf733720c6b5729317a71f60d1b1c5a@192.168.30.165) [Aug 14 14:14:46] DEBUG[24337]: chan_sip.c:3645 sip_hangup: update_call_counter(1643) - decrement call limit counter on hangup [Aug 14 14:14:46] DEBUG[24337]: chan_sip.c:3319 update_call_counter: Updating call counter for outgoing call [Aug 14 14:14:46] DEBUG[24337]: chan_sip.c:3370 update_call_counter: Call to peer '1643' removed from call limit 16 [Aug 14 14:14:46] DEBUG[24337]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1643 Scheduling destruction of SIP dialog '0bf733720c6b5729317a71f60d1b1c5a@192.168.30.165' in 6400 ms (Method: INVITE) [Aug 14 14:14:46] DEBUG[24337]: chan_sip.c:6283 reqprep: Strict routing enforced for session 0bf733720c6b5729317a71f60d1b1c5a@192.168.30.165 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.146, port 5060 Reliably Transmitting (no NAT) to 192.168.30.146:5060: BYE sip:1643@192.168.30.146 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK0cf42e26;rport From: "customer-E:1039" ;tag=as590b9f81 To: ;tag=as7eb5fecd Call-ID: 0bf733720c6b5729317a71f60d1b1c5a@192.168.30.165 CSeq: 103 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Remote-Party-ID: "customer-E:1039" ;privacy=off;screen=no X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 14 14:14:46] DEBUG[24337]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1643 [Aug 14 14:14:46] DEBUG[24337]: rtp.c:1559 ast_rtp_early_bridge: Channel 'Local/1643@default-agent-9ec7,2' has no RTP, not doing anything [Aug 14 14:14:46] DEBUG[24337]: app_dial.c:1824 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Aug 14 14:14:46] DEBUG[24337]: pbx.c:2392 __ast_pbx_run: Spawn extension (default-local-devices,1643,3) exited non-zero on 'Local/1643@default-agent-9ec7,2' == Spawn extension (default-local-devices, 1643, 3) exited non-zero on 'Local/1643@default-agent-9ec7,2' [Aug 14 14:14:46] DEBUG[24337]: channel.c:1453 ast_softhangup_nolock: Soft-Hanging up channel 'Local/1643@default-agent-9ec7,2' [Aug 14 14:14:46] DEBUG[24337]: channel.c:1546 ast_hangup: Hanging up channel 'Local/1643@default-agent-9ec7,2' [Aug 14 14:14:46] DEBUG[24128]: pbx.c:2392 __ast_pbx_run: Spawn extension (all-hangup,s,13) exited non-zero on 'SIP/siptrunk-b46907c8' == Spawn extension (all-hangup, s, 13) exited non-zero on 'SIP/siptrunk-b46907c8' [Aug 14 14:14:46] DEBUG[24337]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel Local/1643@default-agent [Aug 14 14:14:46] DEBUG[2406]: chan_local.c:145 local_devicestate: Checking if extension 1643@default-agent exists (devicestate) [Aug 14 14:14:46] DEBUG[24128]: channel.c:1453 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/siptrunk-b46907c8' [Aug 14 14:14:46] DEBUG[24128]: channel.c:1546 ast_hangup: Hanging up channel 'SIP/siptrunk-b46907c8' <--- SIP read from 192.168.30.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK0cf42e26;received=192.168.30.165;rport=5060 From: "customer-E:1039" ;tag=as590b9f81 To: ;tag=as7eb5fecd Call-ID: 0bf733720c6b5729317a71f60d1b1c5a@192.168.30.165 CSeq: 103 BYE User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:46] DEBUG[24128]: chan_sip.c:3636 sip_hangup: Hangup call SIP/siptrunk-b46907c8, SIP callid 23736d4664e4b432166cc90933a774c6@192.168.30.254) [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[24128]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:46] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for Local/1643@default-agent - state 1 (Not in use) [Aug 14 14:14:46] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'Local/1643@default-agent' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:46] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1643 [Aug 14 14:14:46] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/1643 - state 1 (Not in use) [Aug 14 14:14:46] DEBUG[2541]: app_queue.c:680 handle_statechange: Device 'SIP/1643' changed to state '1' (Not in use) [Aug 14 14:14:46] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1643 [Aug 14 14:14:46] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1643 [Aug 14 14:14:46] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/1643 - state 1 (Not in use) [Aug 14 14:14:46] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1643 [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '0bf733720c6b5729317a71f60d1b1c5a@192.168.30.165' of Request 103: Match Found [Aug 14 14:14:46] DEBUG[2541]: app_queue.c:680 handle_statechange: Device 'SIP/1643' changed to state '1' (Not in use) [Aug 14 14:14:46] DEBUG[2406]: chan_local.c:145 local_devicestate: Checking if extension 1643@default-agent exists (devicestate) [Aug 14 14:14:46] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for Local/1643@default-agent - state 1 (Not in use) [Aug 14 14:14:46] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:46] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:46] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'Local/1643@default-agent' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:46] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '0bf733720c6b5729317a71f60d1b1c5a@192.168.30.165' Method: INVITE Really destroying SIP dialog '23736d4664e4b432166cc90933a774c6@192.168.30.254' Method: BYE [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[24638]: rtp.c:2712 ast_rtp_raw_write: Difference is 1968, ms is 266 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[24637]: rtp.c:2712 ast_rtp_raw_write: Difference is 1416, ms is 197 [Aug 14 14:14:46] DEBUG[24634]: rtp.c:2712 ast_rtp_raw_write: Difference is 1440, ms is 200 [Aug 14 14:14:46] DEBUG[24635]: rtp.c:2712 ast_rtp_raw_write: Difference is 1392, ms is 194 [Aug 14 14:14:46] DEBUG[24486]: rtp.c:2712 ast_rtp_raw_write: Difference is 1440, ms is 200 [Aug 14 14:14:46] DEBUG[24672]: rtp.c:2712 ast_rtp_raw_write: Difference is 1424, ms is 198 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1651@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK46c5f089;rport From: "asterisk" ;tag=as747171a8 To: Contact: Call-ID: 6fbd2bc339c42813417caa9c44855046@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- asterisk*CLI> <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK46c5f089;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as747171a8 To: ;tag=as61766556 Call-ID: 6fbd2bc339c42813417caa9c44855046@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '6fbd2bc339c42813417caa9c44855046@192.168.30.165' of Request 102: Match Found [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1686@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK7a7f90c5;rport From: "asterisk" ;tag=as5a322cee To: Contact: Call-ID: 20cac6ca56f1db0b05d4778433af1058@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- asterisk*CLI> <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK7a7f90c5;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as5a322cee To: ;tag=as66869a19 Call-ID: 20cac6ca56f1db0b05d4778433af1058@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '20cac6ca56f1db0b05d4778433af1058@192.168.30.165' of Request 102: Match Found [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1668@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK42e026f7;rport From: "asterisk" ;tag=as58909434 To: Contact: Call-ID: 3256f58a14556bd6355a516922b99151@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1671@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK09a52a26;rport From: "asterisk" ;tag=as4f99e816 To: Contact: Call-ID: 001fae511ecc741b461f6e575b0da291@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1696@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK427772c9;rport From: "asterisk" ;tag=as2eaaa56d To: Contact: Call-ID: 4441d34b41e2f99541f9a1d606dab555@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK42e026f7;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as58909434 To: ;tag=as442011c1 Call-ID: 3256f58a14556bd6355a516922b99151@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '3256f58a14556bd6355a516922b99151@192.168.30.165' of Request 102: Match Found <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK09a52a26;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as4f99e816 To: ;tag=as74ce7f5b Call-ID: 001fae511ecc741b461f6e575b0da291@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '001fae511ecc741b461f6e575b0da291@192.168.30.165' of Request 102: Match Found <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK427772c9;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as2eaaa56d To: ;tag=as1feabbc9 Call-ID: 4441d34b41e2f99541f9a1d606dab555@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:46] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '4441d34b41e2f99541f9a1d606dab555@192.168.30.165' of Request 102: Match Found [Aug 14 14:14:46] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '4441d34b41e2f99541f9a1d606dab555@192.168.30.165' Method: OPTIONS Really destroying SIP dialog '001fae511ecc741b461f6e575b0da291@192.168.30.165' Method: OPTIONS Really destroying SIP dialog '3256f58a14556bd6355a516922b99151@192.168.30.165' Method: OPTIONS [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '20cac6ca56f1db0b05d4778433af1058@192.168.30.165' Method: OPTIONS Really destroying SIP dialog '6fbd2bc339c42813417caa9c44855046@192.168.30.165' Method: OPTIONS [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24635]: rtp.c:2712 ast_rtp_raw_write: Difference is 1264, ms is 178 [Aug 14 14:14:47] DEBUG[24639]: rtp.c:2712 ast_rtp_raw_write: Difference is 1264, ms is 178 [Aug 14 14:14:47] DEBUG[24663]: rtp.c:2712 ast_rtp_raw_write: Difference is 1256, ms is 177 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24671]: rtp.c:2712 ast_rtp_raw_write: Difference is 1304, ms is 183 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24487]: rtp.c:2712 ast_rtp_raw_write: Difference is 2048, ms is 276 [Aug 14 14:14:47] DEBUG[24672]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:47] DEBUG[24486]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:47] DEBUG[24486]: rtp.c:2712 ast_rtp_raw_write: Difference is 2112, ms is 284 [Aug 14 14:14:47] DEBUG[24672]: rtp.c:2712 ast_rtp_raw_write: Difference is 2080, ms is 280 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24647]: rtp.c:2712 ast_rtp_raw_write: Difference is 2072, ms is 279 [Aug 14 14:14:47] DEBUG[24659]: rtp.c:2712 ast_rtp_raw_write: Difference is 2096, ms is 282 [Aug 14 14:14:47] DEBUG[24638]: rtp.c:2712 ast_rtp_raw_write: Difference is 2176, ms is 292 [Aug 14 14:14:47] DEBUG[24660]: rtp.c:2712 ast_rtp_raw_write: Difference is 2152, ms is 289 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24671]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 asterisk*CLI> <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3ad171a4;rport From: "1047" ;tag=as3d36fb90 To: Contact: Call-ID: 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 12454 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off asterisk*CLI> <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3ad171a4;received=192.168.30.254;rport=5060 From: "1047" ;tag=as3d36fb90 To: ;tag=as72a05dd8 Call-ID: 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74b0f218" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5cb22217076bbec46541829c5dbc2fb1@192.168.30.254' in 32000 ms (Method: INVITE) asterisk*CLI> <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3ad171a4;rport From: "1047" ;tag=as3d36fb90 To: ;tag=as72a05dd8 Contact: Call-ID: 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '5cb22217076bbec46541829c5dbc2fb1@192.168.30.254' of Response 102: Match Found asterisk*CLI> <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK29e88eb9;rport From: "1047" ;tag=as3d36fb90 To: Contact: Call-ID: 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="74b0f218", response="71fe670a27e384d8a00bf3f48c2f8b61" Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 12454 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:12454 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:12454 [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:47] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Ping' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:47] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[2453]: channel.c:1130 channel_find_locked: Avoiding initial deadlock for channel '0xb5800930' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b4851840: New call is still down.... Trying... asterisk*CLI> <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK29e88eb9;received=192.168.30.254;rport=5060 From: "1047" ;tag=as3d36fb90 To: Call-ID: 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:47] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:47] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:47] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:47] DEBUG[2453]: channel.c:1130 channel_find_locked: Avoiding initial deadlock for channel '0xb5800930' [Aug 14 14:14:47] DEBUG[24647]: app_queue.c:2448 is_our_turn: There is 1 available member. [Aug 14 14:14:47] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' asterisk*CLI> <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0ed70124;rport From: "1015" ;tag=as17a16708 To: Contact: Call-ID: 0b05caec1722026c32e056b47802442f@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 15768 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 0b05caec1722026c32e056b47802442f@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 0b05caec1722026c32e056b47802442f@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off asterisk*CLI> <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0ed70124;received=192.168.30.254;rport=5060 From: "1015" ;tag=as17a16708 To: ;tag=as58e7eeb2 Call-ID: 0b05caec1722026c32e056b47802442f@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="483436c3" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0b05caec1722026c32e056b47802442f@192.168.30.254' in 32000 ms (Method: INVITE) -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b4851840", "") in new stack [Aug 14 14:14:47] DEBUG[24695]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:47] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:47] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:47] DEBUG[24695]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b4851840 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24695]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:47] DEBUG[24695]: chan_sip.c:6761 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Aug 14 14:14:47] DEBUG[24695]: chan_sip.c:6762 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.30.165 port 11796 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Aug 14 14:14:47] DEBUG[24695]: chan_sip.c:6879 add_sdp: -- Done with adding codecs to SDP [Aug 14 14:14:47] DEBUG[24695]: chan_sip.c:6988 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) asterisk*CLI> <--- Reliably Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK29e88eb9;received=192.168.30.254;rport=5060 From: "1047" ;tag=as3d36fb90 To: ;tag=as2f669adb Call-ID: 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 11796 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b4851840", "CHANNEL(musicclass)=default") in new stack asterisk*CLI> <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0ed70124;rport From: "1015" ;tag=as17a16708 To: ;tag=as58e7eeb2 Contact: Call-ID: 0b05caec1722026c32e056b47802442f@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '0b05caec1722026c32e056b47802442f@192.168.30.254' of Response 102: Match Found asterisk*CLI> <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7e718dff;rport From: "1015" ;tag=as17a16708 To: Contact: Call-ID: 0b05caec1722026c32e056b47802442f@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="483436c3", response="cc58790ab5d0efcc8df34e155902bd99" Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 15768 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 0b05caec1722026c32e056b47802442f@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:15768 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:15768 [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b50fe470: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7e718dff;received=192.168.30.254;rport=5060 From: "1015" ;tag=as17a16708 To: Call-ID: 0b05caec1722026c32e056b47802442f@192.168.30.254 CSeq: 103 INVITE ser-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:47] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:47] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:47] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3286e4ac;rport From: "1047" ;tag=as3d36fb90 To: ;tag=as2f669adb Contact: Call-ID: 5cb22217076bbec46541829c5dbc2fb1@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '5cb22217076bbec46541829c5dbc2fb1@192.168.30.254' of Response 103: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK47b1c6fa;rport From: "1034" ;tag=as09dfed79 To: Contact: Call-ID: 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 12782 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK47b1c6fa;received=192.168.30.254;rport=5060 From: "1034" ;tag=as09dfed79 To: ;tag=as019c1d78 Call-ID: 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7b5fa15b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2d1573c349d1308731d3db8f2e529afa@192.168.30.254' in 32000 ms (Method: INVITE) [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1047' [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1047' <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK47b1c6fa;rport From: "1034" ;tag=as09dfed79 To: ;tag=as019c1d78 Contact: Call-ID: 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '2d1573c349d1308731d3db8f2e529afa@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0ed93fce;rport From: "1034" ;tag=as09dfed79 To: Contact: Call-ID: 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="7b5fa15b", response="97431a0989be0e4115c50ec194f8219e" Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 12782 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 [Aug 14 14:14:47] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:12782 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:12782 [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b580f580: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0ed93fce;received=192.168.30.254;rport=5060 From: "1034" ;tag=as09dfed79 To: Call-ID: 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:47] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:47] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:47] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b4851840", ""INCOMING CALL FROM CALLER ID: 1047 (1047)"") in new stack [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'Set' [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Executing [5000@private-siptrunk-incoming:4] Set("SIP/siptrunk-b4851840", "__INCOMING_DNIS=5000") in new stack [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b4851840", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b4851840", "CDR(userfield)=5000") in new stack [Aug 14 14:14:47] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b4851840", "10") in new stack -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:47] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b4851840", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b50fe470", "") in new stack [Aug 14 14:14:47] DEBUG[24696]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:47] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:47] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:47] DEBUG[24696]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b50fe470 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24696]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:47] DEBUG[24696]: chan_sip.c:6761 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Aug 14 14:14:47] DEBUG[24696]: chan_sip.c:6762 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.30.165 port 14928 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Aug 14 14:14:47] DEBUG[24696]: chan_sip.c:6879 add_sdp: -- Done with adding codecs to SDP [Aug 14 14:14:47] DEBUG[24696]: chan_sip.c:6988 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) <--- Reliably Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7e718dff;received=192.168.30.254;rport=5060 From: "1015" ;tag=as17a16708 To: ;tag=as64b517f1 Call-ID: 0b05caec1722026c32e056b47802442f@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 14928 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b50fe470", "CHANNEL(musicclass)=default") in new stack [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1015' [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1015' [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b50fe470", ""INCOMING CALL FROM CALLER ID: 1015 (1015)"") in new stack [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:4] Set("SIP/siptrunk-b50fe470", "__INCOMING_DNIS=5000") in new stack [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b50fe470", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b50fe470", "CDR(userfield)=5000") in new stack [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b50fe470", "10") in new stack -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:47] DEBUG[24696]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b50fe470", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK57c918ff;rport From: "1015" ;tag=as17a16708 To: ;tag=as64b517f1 Contact: Call-ID: 0b05caec1722026c32e056b47802442f@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '0b05caec1722026c32e056b47802442f@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b580f580", "") in new stack [Aug 14 14:14:47] DEBUG[24697]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:47] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:47] DEBUG[24697]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b580f580 [Aug 14 14:14:47] DEBUG[24697]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:47] DEBUG[24697]: chan_sip.c:6761 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Aug 14 14:14:47] DEBUG[24697]: chan_sip.c:6762 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.30.165 port 18136 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Aug 14 14:14:47] DEBUG[24697]: chan_sip.c:6879 add_sdp: -- Done with adding codecs to SDP [Aug 14 14:14:47] DEBUG[24697]: chan_sip.c:6988 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) <--- Reliably Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0ed93fce;received=192.168.30.254;rport=5060 From: "1034" ;tag=as09dfed79 To: ;tag=as3abaa8b8 Call-ID: 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 18136 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b580f580", "CHANNEL(musicclass)=default") in new stack [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1034' [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1034' [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b580f580", ""INCOMING CALL FROM CALLER ID: 1034 (1034)"") in new stack [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:4] Set("SIP/siptrunk-b580f580", "__INCOMING_DNIS=5000") in new stack [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b580f580", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b580f580", "CDR(userfield)=5000") in new stack [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b580f580", "10") in new stack [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:47] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b580f580", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1dc7c6a3;rport From: "1034" ;tag=as09dfed79 To: ;tag=as3abaa8b8 Contact: Call-ID: 2d1573c349d1308731d3db8f2e529afa@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '2d1573c349d1308731d3db8f2e529afa@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24671]: app_queue.c:2448 is_our_turn: There is 1 available member. [Aug 14 14:14:47] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0c55ddd2;rport From: "1002" ;tag=as74a23113 To: Contact: Call-ID: 33c541550a6e6ee471ffd80362a664de@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 14196 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 33c541550a6e6ee471ffd80362a664de@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces ending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 33c541550a6e6ee471ffd80362a664de@192.168.30.254 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0c55ddd2;received=192.168.30.254;rport=5060 From: "1002" ;tag=as74a23113 To: ;tag=as71473cbb Call-ID: 33c541550a6e6ee471ffd80362a664de@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="106b13b5" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '33c541550a6e6ee471ffd80362a664de@192.168.30.254' in 32000 ms (Method: INVITE) [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2448 is_our_turn: There is 1 available member. [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2462 is_our_turn: It's our turn (SIP/siptrunk-b584f380). [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2845 try_calling: SIP/siptrunk-b584f380 is trying to call a queue member. [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1627@default-agent/n' with metric 0 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1627@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1607@default-agent/n' with metric 1 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1607@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1610@default-agent/n' with metric 2 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1610@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1604@default-agent/n' with metric 3 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1604@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1608@default-agent/n' with metric 4 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1608@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1649@default-agent/n' with metric 5 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1649@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1605@default-agent/n' with metric 6 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1605@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1622@default-agent/n' with metric 7 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1622@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1636@default-agent/n' with metric 8 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1636@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1602@default-agent/n' with metric 9 [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:1889 ring_entry: Local/1602@default-agent/n in use, can't receive call [Aug 14 14:14:47] DEBUG[24635]: app_queue.c:2069 ring_one: Trying 'Local/1643@default-agent/n' with metric 10 <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0c55ddd2;rport From: "1002" ;tag=as74a23113 To: ;tag=as71473cbb Contact: Call-ID: 33c541550a6e6ee471ffd80362a664de@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '33c541550a6e6ee471ffd80362a664de@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK2d1f18d1;rport From: "1002" ;tag=as74a23113 To: Contact: Call-ID: 33c541550a6e6ee471ffd80362a664de@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="106b13b5", response="38a6ff07712f3ca2063324e493465a36" Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 14196 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) [Aug 14 14:14:47] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:47] DEBUG[24635]: channel.c:3688 ast_channel_inherit_variables: Copying hard-transferable variable ALLOW_TRANSFER. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3688 ast_channel_inherit_variables: Copying hard-transferable variable ACD_TO_OUTGOING. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3693 ast_channel_inherit_variables: Not copying variable MONITOR_FILENAME. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3688 ast_channel_inherit_variables: Copying hard-transferable variable QUEUENAME. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3693 ast_channel_inherit_variables: Not copying variable ~GOSUB~STACK~. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3693 ast_channel_inherit_variables: Not copying variable QUEUE_PRIO. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3688 ast_channel_inherit_variables: Copying hard-transferable variable INCOMINGLINE. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3693 ast_channel_inherit_variables: Not copying variable FAXNUMBER. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3693 ast_channel_inherit_variables: Not copying variable AGISTATUS. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3688 ast_channel_inherit_variables: Copying hard-transferable variable INCOMING_DNIS. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3693 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3693 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3693 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Aug 14 14:14:47] DEBUG[24635]: channel.c:3693 ast_channel_inherit_variables: Not copying variable SIPURI. -- Called Local/1643@default-agent/n [Aug 14 14:14:47] DEBUG[24635]: chan_sip.c:17010 sip_devicestate: Checking device state for peer 1643 Using INVITE request as basis request - 33c541550a6e6ee471ffd80362a664de@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:14196 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:14196 [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b58bee08: New call is still down.... Trying... [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK2d1f18d1;received=192.168.30.254;rport=5060 From: "1002" ;tag=as74a23113 To: Call-ID: 33c541550a6e6ee471ffd80362a664de@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:47] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:47] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:47] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'NoCDR' -- Executing [1643@default-agent:1] NoCDR("Local/1643@default-agent-430e,2", "") in new stack [Aug 14 14:14:47] DEBUG[24701]: db.c:196 ast_db_get: Unable to find key '1643/DoNotDisturb' in family 'default' [Aug 14 14:14:47] DEBUG[24701]: func_db.c:70 function_db_read: DB: default/1643/DoNotDisturb not found in database. [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [1643@default-agent:2] Set("Local/1643@default-agent-430e,2", "DND=") in new stack [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [1643@default-agent:3] GotoIf("Local/1643@default-agent-430e,2", "0?30") in new stack [Aug 14 14:14:47] DEBUG[24701]: pbx.c:6065 pbx_builtin_gotoif: Not taking any branch [Aug 14 14:14:47] DEBUG[24701]: db.c:196 ast_db_get: Unable to find key '1643/OutOffice' in family 'default' [Aug 14 14:14:47] DEBUG[24701]: func_db.c:70 function_db_read: DB: default/1643/OutOffice not found in database. [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [1643@default-agent:4] Set("Local/1643@default-agent-430e,2", "OutOffice=") in new stack [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [1643@default-agent:5] GotoIf("Local/1643@default-agent-430e,2", "0?30") in new stack [Aug 14 14:14:47] DEBUG[24701]: pbx.c:6065 pbx_builtin_gotoif: Not taking any branch [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '0' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '0' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [1643@default-agent:6] Set("Local/1643@default-agent-430e,2", "GROUPCOUNT=0") in new stack [Aug 14 14:14:47] DEBUG[24701]: db.c:196 ast_db_get: Unable to find key 'agent//maxcalls' in family 'default' [Aug 14 14:14:47] DEBUG[24701]: func_db.c:70 function_db_read: DB: default/agent//maxcalls not found in database. [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [1643@default-agent:7] Set("Local/1643@default-agent-430e,2", "AGENTMAXCALL=") in new stack [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '9' [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '12' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [1643@default-agent:8] GotoIf("Local/1643@default-agent-430e,2", "1?9:12") in new stack -- Goto (default-agent,1643,9) [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [1643@default-agent:9] Set("Local/1643@default-agent-430e,2", "AGENTMAXCALL=") in new stack [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '11' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '12' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [1643@default-agent:10] GotoIf("Local/1643@default-agent-430e,2", "1?11:12") in new stack -- Goto (default-agent,1643,11) [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [1643@default-agent:11] Set("Local/1643@default-agent-430e,2", "AGENTMAXCALL=1") in new stack [Aug 14 14:14:47] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [1643@default-agent:12] GotoIf("Local/1643@default-agent-430e,2", "0?30") in new stack [Aug 14 14:14:47] DEBUG[24701]: pbx.c:6065 pbx_builtin_gotoif: Not taking any branch [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '16' [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [1643@default-agent:13] GotoIf("Local/1643@default-agent-430e,2", "1?16") in new stack -- Goto (default-agent,1643,16) [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [1643@default-agent:16] Set("Local/1643@default-agent-430e,2", "OUTBOUND_GROUP_ONCE=1643@INCOMING") in new stack [Aug 14 14:14:47] DEBUG[24701]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [1643@default-agent:17] Set("Local/1643@default-agent-430e,2", "DB(default/wrapup/1643/lastcall)=1250273687.6313") in new stack <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3e7a3f02;rport From: "1044" ;tag=as3636bc48 To: Contact: Call-ID: 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 14956 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3e7a3f02;received=192.168.30.254;rport=5060 From: "1044" ;tag=as3636bc48 To: ;tag=as050427e4 Call-ID: 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27a0f0da" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254' in 32000 ms (Method: INVITE) [Aug 14 14:14:47] DEBUG[24702]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b58bee08", "") in new stack [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3e7a3f02;rport From: "1044" ;tag=as3636bc48 To: ;tag=as050427e4 Contact: Call-ID: 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK362f017b;rport From: "1044" ;tag=as3636bc48 To: Contact: Call-ID: 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="27a0f0da", response="260eca8c7255b661a8731e471cd373b3" Date: Fri, 14 Aug 2009 18:14:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 14956 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:14956 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:14956 [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:47] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b487b4e0: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK362f017b;received=192.168.30.254;rport=5060 From: "1044" ;tag=as3636bc48 To: Call-ID: 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:47] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:47] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:47] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:47] DEBUG[24702]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[24703]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b487b4e0", "") in new stack [Aug 14 14:14:47] DEBUG[24703]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:47] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:47] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:47] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:47] DEBUG[24487]: pbx.c:2392 __ast_pbx_run: Spawn extension (private-siptrunk-incoming,5000,501) exited non-zero on 'SIP/siptrunk-b581b0a0' == Spawn extension (private-siptrunk-incoming, 5000, 501) exited non-zero on 'SIP/siptrunk-b581b0a0' [Aug 14 14:14:47] DEBUG[24487]: channel.c:1453 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/siptrunk-b581b0a0' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [h@private-siptrunk-incoming:1] Goto("SIP/siptrunk-b581b0a0", "all-hangup|s|1") in new stack -- Goto (all-hangup,s,1) [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '2' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [s@all-hangup:1] GotoIf("SIP/siptrunk-b581b0a0", "0?all-faxnotify|s|1:2") in new stack -- Goto (all-hangup,s,2) [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1843 pbx_extension_helper: Launching 'ResetCDR' -- Executing [s@all-hangup:2] ResetCDR("SIP/siptrunk-b581b0a0", "w") in new stack [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '"1037" <1037>' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1037' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '5000' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'private-siptrunk-incoming' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'SIP/siptrunk-b581b0a0' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '(null)' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'Congestion' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '30' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:15' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:15' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:47' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '32' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '32' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'siptrunk' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1250273655.6296' [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '5000' [Aug 14 14:14:47] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1843 pbx_extension_helper: Launching 'NoCDR' -- Executing [s@all-hangup:3] NoCDR("SIP/siptrunk-b581b0a0", "") in new stack [Aug 14 14:14:47] DEBUG[24487]: pbx.c:1843 pbx_extension_helper: Launching 'System' -- Executing [s@all-hangup:4] System("SIP/siptrunk-b581b0a0", "/var/www/scopserv/telephony/scripts/billing/cdr.sh 1250273655.6296") in new stack [Aug 14 14:14:47] DEBUG[24703]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b487b4e0 [Aug 14 14:14:47] DEBUG[24703]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:47] DEBUG[24703]: chan_sip.c:6761 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Aug 14 14:14:47] DEBUG[24703]: chan_sip.c:6762 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.30.165 port 14356 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Aug 14 14:14:47] DEBUG[24703]: chan_sip.c:6879 add_sdp: -- Done with adding codecs to SDP [Aug 14 14:14:47] DEBUG[24703]: chan_sip.c:6988 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) <--- Reliably Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK362f017b;received=192.168.30.254;rport=5060 From: "1044" ;tag=as3636bc48 To: ;tag=as2d771630 Call-ID: 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 14356 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Aug 14 14:14:47] DEBUG[24703]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b487b4e0", "CHANNEL(musicclass)=default") in new stack <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK51dec8a8;rport From: "1044" ;tag=as3636bc48 To: ;tag=as2d771630 Contact: Call-ID: 2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:47] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '2d3893ad7bf7ed57290d611a72de91a2@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:47] DEBUG[24703]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1044' [Aug 14 14:14:47] DEBUG[24703]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1044' [Aug 14 14:14:47] DEBUG[24703]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b487b4e0", ""INCOMING CALL FROM CALLER ID: 1044 (1044)"") in new stack [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK776c5650;rport From: "1036" ;tag=as745704ea To: Contact: Call-ID: 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 19092 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK776c5650;received=192.168.30.254;rport=5060 From: "1036" ;tag=as745704ea To: ;tag=as48e19038 Call-ID: 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="527e368b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK776c5650;rport From: "1036" ;tag=as745704ea To: ;tag=as48e19038 Contact: Call-ID: 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7e2ab3f0;rport From: "1036" ;tag=as745704ea To: Contact: Call-ID: 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="527e368b", response="1dd4f52ada970bcf3cf8f14ac0fff2db" Date: Fri, 14 Aug 2009 18:14:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 19092 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:19092 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:19092 [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b58c3240: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7e2ab3f0;received=192.168.30.254;rport=5060 From: "1036" ;tag=as745704ea To: Call-ID: 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:48] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:48] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:48] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:48] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b58c3240", "") in new stack [Aug 14 14:14:48] DEBUG[24708]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:48] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:48] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:48] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:48] DEBUG[24708]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b58c3240 [Aug 14 14:14:48] DEBUG[24708]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:48] DEBUG[24708]: chan_sip.c:6761 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[24708]: chan_sip.c:6762 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.30.165 port 12462 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Aug 14 14:14:48] DEBUG[24708]: chan_sip.c:6879 add_sdp: -- Done with adding codecs to SDP [Aug 14 14:14:48] DEBUG[24708]: chan_sip.c:6988 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) <--- Reliably Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7e2ab3f0;received=192.168.30.254;rport=5060 From: "1036" ;tag=as745704ea To: ;tag=as6bee1110 Call-ID: 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 12462 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0d3f7563;rport From: "1036" ;tag=as745704ea To: ;tag=as6bee1110 Contact: Call-ID: 0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '0588b8dc4bc65a7a022d29527f8c04db@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b58c3240", "CHANNEL(musicclass)=default") in new stack [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1036' [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1036' [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b58c3240", ""INCOMING CALL FROM CALLER ID: 1036 (1036)"") in new stack [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:4] Set("SIP/siptrunk-b58c3240", "__INCOMING_DNIS=5000") in new stack [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b58c3240", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[23779]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b58c3240", "CDR(userfield)=5000") in new stack [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b58c3240", "10") in new stack -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:48] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b58c3240", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:48] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:48] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:48] DEBUG[24647]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:48] DEBUG[24647]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b4830b88). [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Reliably Transmitting (no NAT) to 192.168.30.148:5060: OPTIONS sip:1665@192.168.30.148 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK4ba3034e;rport From: "asterisk" ;tag=as048a4d05 To: Contact: Call-ID: 674583c8462e014542f64ad111f6c0fe@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.30.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.165:5060;branch=z9hG4bK4ba3034e;received=192.168.30.165;rport=5060 From: "asterisk" ;tag=as048a4d05 To: ;tag=as0ee84284 Call-ID: 674583c8462e014542f64ad111f6c0fe@192.168.30.165 CSeq: 102 OPTIONS User-Agent: Test Framework 1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '674583c8462e014542f64ad111f6c0fe@192.168.30.165' of Request 102: Match Found [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[24486]: pbx.c:2392 __ast_pbx_run: Spawn extension (private-siptrunk-incoming,5000,501) exited non-zero on 'SIP/siptrunk-b4d1ef50' == Spawn extension (private-siptrunk-incoming, 5000, 501) exited non-zero on 'SIP/siptrunk-b4d1ef50' [Aug 14 14:14:48] DEBUG[24486]: channel.c:1453 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/siptrunk-b4d1ef50' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [h@private-siptrunk-incoming:1] Goto("SIP/siptrunk-b4d1ef50", "all-hangup|s|1") in new stack -- Goto (all-hangup,s,1) [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '2' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [s@all-hangup:1] GotoIf("SIP/siptrunk-b4d1ef50", "0?all-faxnotify|s|1:2") in new stack -- Goto (all-hangup,s,2) [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1843 pbx_extension_helper: Launching 'ResetCDR' -- Executing [s@all-hangup:2] ResetCDR("SIP/siptrunk-b4d1ef50", "w") in new stack [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '"1044" <1044>' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1044' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '5000' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'private-siptrunk-incoming' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'SIP/siptrunk-b4d1ef50' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '(null)' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'Congestion' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '30' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:15' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:15' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:48' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '33' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '33' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'siptrunk' [Aug 14 14:14:48] DEBUG[24486]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1250273655.6295' [Aug 14 14:14:48] DEBUG[24486]: :2258 __sip_ack: Stopping retransmission on '193c44b83a5ee0d775e6164410573547@192.168.30.165' of Request 102: Match Found [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:48] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '221b05e239af1e7632e545244c11eae3@127.0.0.1' [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 221b05e239af1e7632e545244c11eae3@127.0.0.1 Really destroying SIP dialog '221b05e239af1e7632e545244c11eae3@127.0.0.1' Method: REGISTER Really destroying SIP dialog '193c44b83a5ee0d775e6164410573547@192.168.30.165' Method: OPTIONS [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[24634]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '58c86fbb7ffbf003721f2c0c0729cade@127.0.0.1' [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 58c86fbb7ffbf003721f2c0c0729cade@127.0.0.1 Really destroying SIP dialog '58c86fbb7ffbf003721f2c0c0729cade@127.0.0.1' Method: REGISTER [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[24635]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:48] DEBUG[24637]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[24638]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:2163 __sip_autodestruct: Auto destroying SIP dialog '721b96992076c0ec235493d3180f25d3@127.0.0.1' [Aug 14 14:14:48] DEBUG[2654]: chan_sip.c:3428 sip_destroy: Destroying SIP dialog 721b96992076c0ec235493d3180f25d3@127.0.0.1 Really destroying SIP dialog '721b96992076c0ec235493d3180f25d3@127.0.0.1' Method: REGISTER [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[24696]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:48] DEBUG[24696]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b50fe470). [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:48] DEBUG[2499]: manager.c:2230 process_message: Manager received command 'Ping' [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:48] DEBUG[24702]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b58bee08", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:48] DEBUG[24702]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '51' [Aug 14 14:14:48] DEBUG[24702]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:48] DEBUG[24702]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b58bee08", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:48] DEBUG[24702]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b58bee08", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:48] DEBUG[24702]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b58bee08", "30") in new stack [Aug 14 14:14:48] DEBUG[24702]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b58bee08' does not support indication 8, emulating it [Aug 14 14:14:48] DEBUG[24702]: channel.c:3090 set_format: Set channel SIP/siptrunk-b58bee08 to write format slin [Aug 14 14:14:48] DEBUG[24702]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:48] DEBUG[24702]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:48] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[23836]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[23837]: rtp.c:923 ast_rtcp_read: Got RTCP report of 44 bytes [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally lon -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:49] DEBUG[24706]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b58c1960", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:49] DEBUG[24706]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b58c1960", "30") in new stack [Aug 14 14:14:49] DEBUG[24706]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b58c1960' does not support indication 8, emulating it [Aug 14 14:14:49] DEBUG[24706]: channel.c:3090 set_format: Set channel SIP/siptrunk-b58c1960 to write format slin [Aug 14 14:14:49] DEBUG[24706]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:49] DEBUG[24706]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b58c3240", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:49] DEBUG[24708]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '54' [Aug 14 14:14:49] DEBUG[24708]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:49] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b58c3240", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:49] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b58c3240", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:49] DEBUG[24708]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b58c3240", "30") in new stack [Aug 14 14:14:49] DEBUG[24708]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b58c3240' does not support indication 8, emulating it [Aug 14 14:14:49] DEBUG[24708]: channel.c:3090 set_format: Set channel SIP/siptrunk-b58c3240 to write format slin [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24647]: rtp.c:2712 ast_rtp_raw_write: Difference is 696, ms is 107 [Aug 14 14:14:49] DEBUG[24702]: rtp.c:2712 ast_rtp_raw_write: Difference is 720, ms is 110 [Aug 14 14:14:49] DEBUG[24671]: rtp.c:2712 ast_rtp_raw_write: Difference is 696, ms is 107 [Aug 14 14:14:49] DEBUG[24706]: rtp.c:2712 ast_rtp_raw_write: Difference is 656, ms is 102 [Aug 14 14:14:49] DEBUG[24665]: rtp.c:2712 ast_rtp_raw_write: Difference is 696, ms is 107 [Aug 14 14:14:49] DEBUG[24637]: rtp.c:2712 ast_rtp_raw_write: Difference is 672, ms is 104 [Aug 14 14:14:49] DEBUG[24634]: rtp.c:2712 ast_rtp_raw_write: Difference is 664, ms is 103 [Aug 14 14:14:49] DEBUG[24638]: rtp.c:2712 ast_rtp_raw_write: Difference is 768, ms is 116 [Aug 14 14:14:49] DEBUG[24672]: rtp.c:2712 ast_rtp_raw_write: Difference is 760, ms is 115 [Aug 14 14:14:49] DEBUG[24696]: rtp.c:2712 ast_rtp_raw_write: Difference is 744, ms is 113 [Aug 14 14:14:49] DEBUG[24708]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:49] DEBUG[24708]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:49] DEBUG[24659]: rtp.c:2712 ast_rtp_raw_write: Difference is 744, ms is 113 [Aug 14 14:14:49] DEBUG[24660]: rtp.c:2712 ast_rtp_raw_write: Difference is 744, ms is 113 [Aug 14 14:14:49] DEBUG[24703]: rtp.c:2712 ast_rtp_raw_write: Difference is 712, ms is 109 [Aug 14 14:14:49] DEBUG[24663]: rtp.c:2712 ast_rtp_raw_write: Difference is 712, ms is 109 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b580f580", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:49] DEBUG[24697]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '55' [Aug 14 14:14:49] DEBUG[24697]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:49] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b580f580", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:49] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b580f580", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:49] DEBUG[24697]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b580f580", "30") in new stack [Aug 14 14:14:49] DEBUG[24697]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b580f580' does not support indication 8, emulating it [Aug 14 14:14:49] DEBUG[24697]: channel.c:3090 set_format: Set channel SIP/siptrunk-b580f580 to write format slin [Aug 14 14:14:49] DEBUG[24697]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:49] DEBUG[24697]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b4851840", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:49] DEBUG[24695]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '56' [Aug 14 14:14:49] DEBUG[24695]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:49] DEBUG[24695]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@ctact: Call-ID: 6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 18808 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1ed4f797;received=192.168.30.254;rport=5060 From: "1041" ;tag=as423e0f94 To: ;tag=as132f3a23 Call-ID: 6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42ca5b2f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1ed4f797;rport From: "1041" ;tag=as423e0f94 To: ;tag=as132f3a23 Contact: Call-ID: 6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4616a7bd;rport From: "1041" ;tag=as423e0f94 To: Contact: Call-ID: 6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="42ca5b2f", response="cf58ffaffd04bb210a37fe562916ed70" Date: Fri, 14 Aug 2009 18:14:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 18808 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:18808 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:18808 [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: list_route: hop: [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:14975 handle_request_invite: SIP/siptrunk-b51f77b8: New call is still down.... Trying... <--- Transmitting (NAT) to 192.168.30.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4616a7bd;received=192.168.30.254;rport=5060 From: "1041" ;tag=as423e0f94 To: Call-ID: 6f9b22ff310bc02d43ce58722f4eaa92@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Aug 14 14:14:49] DEBUG[2654]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Ping' [Aug 14 14:14:49] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:49] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:49] DEBUG[2453]: manager.c:2230 process_message: Manager received command 'Command' [Aug 14 14:14:49] DEBUG[24715]: pbx.c:1843 pbx_extension_helper: Launching 'Answer' -- Executing [5000@private-siptrunk-incoming:1] Answer("SIP/siptrunk-b51f77b8", "") in new stack [Aug 14 14:14:49] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:49] DEBUG[24715]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/siptrunk [Aug 14 14:14:49] DEBUG[2406]: chan_sip.c:17010 sip_devicestate: Checking device state for peer siptrunk [Aug 14 14:14:49] DEBUG[2406]: devicestate.c:287 do_state_change: Changing state for SIP/siptrunk - state 1 (Not in use) [Aug 14 14:14:49] DEBUG[2541]: app_queue.c:675 handle_statechange: Device 'SIP/siptrunk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24715]: chan_sip.c:3809 sip_answer: SIP answering channel: SIP/siptrunk-b51f77b8 [Aug 14 14:14:49] DEBUG[24715]: chan_sip.c:7046 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 14 14:14:49] DEBUG[24715]: chan_sip.c: Set("SIP/siptrunk-b51f77b8", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24715]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b51f77b8", "CDR(userfield)=5000") in new stack [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24715]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b51f77b8", "10") in new stack -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:49] DEBUG[24715]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b51f77b8", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24671]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:49] DEBUG[24671]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b58dc998). [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3a87a120;rport From: "1016" ;tag=as70db19bb To: Contact: Call-ID: 77e74c186c3e510252952def1b926fba@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Date: Fri, 14 Aug 2009 18:14:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28162 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 16846 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4711 sip_alloc: Allocating new SIP dialog for 77e74c186c3e510252952def1b926fba@192.168.30.254 - INVITE (With RTP) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:1748 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:1756 parse_sip_options: Found SIP option: -replaces- [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:1762 parse_sip_options: Matched SIP option: replaces Sending to 192.168.30.254 : 5060 (no NAT) Using INVITE request as basis request - 77e74c186c3e510252952def1b926fba@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off <--- Reliably Transmitting (no NAT) to 192.168.30.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3a87a120;received=192.168.30.254;rport=5060 From: "1016" ;tag=as70db19bb To: ;tag=as75bf0577 Call-ID: 77e74c186c3e510252952def1b926fba@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01ee4d61" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '77e74c186c3e510252952def1b926fba@192.168.30.254' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3a87a120;rport From: "1016" ;tag=as70db19bb To: ;tag=as75bf0577 Contact: Call-ID: 77e74c186c3e510252952def1b926fba@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '77e74c186c3e510252952def1b926fba@192.168.30.254' of Response 102: Match Found <--- SIP read from 192.168.30.254:5060 ---> INVITE sip:5000@192.168.30.165;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK184eddde;rport From: "1016" ;tag=as70db19bb To: Contact: Call-ID: 77e74c186c3e510252952def1b926fba@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165;user=phone", nonce="01ee4d61", response="4ae623fe0363bcf3c98b54a1dfcd8a11" Date: Fri, 14 Aug 2009 18:14:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 28162 28163 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 16846 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 192.168.30.254 : 5060 (NAT) Using INVITE request as basis request - 77e74c186c3e510252952def1b926fba@192.168.30.254 Found peer 'siptrunk' [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2835 do_setnat: Setting NAT on RTP to Off [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2840 do_setnat: Setting NAT on VRTP to Off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.254:16846 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:5726 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.30.254:16846 [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:5806 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:14876 handle_request_invite: Checking SIP call limits for device siptrunk [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:3319 update_call_counter: Updating call counter for incoming call Looking for 5000 in private-siptrunk-incoming (domain 192.168.30.165) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4179 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4180 sip_new: *** Joint capabilities are 0x4 (ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4181 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4182 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:4205 sip_new: This channel will not be able to handle video. [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:8719 build_route: build_route: Contact hop: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK184eddde;received=192.168.30.254;rport=5060 From: "1016" ;tag=as70db19bb To: ;tag=as30a8eb81 Call-ID: 77e74c186c3e510252952def1b926fba@192.168.30.254 CSeq: 103 INVITE User-Agent: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2398 2398 IN IP4 192.168.30.165 s=session c=IN IP4 192.168.30.165 t=0 0 m=audio 14876 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Aug 14 14:14:49] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:2] Set("SIP/siptrunk-b58197c0", "CHANNEL(musicclass)=default") in new stack [Aug 14 14:14:49] DEBUG[24717]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1016' [Aug 14 14:14:49] DEBUG[24717]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1016' [Aug 14 14:14:49] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:3] NoOp("SIP/siptrunk-b58197c0", ""INCOMING CALL FROM CALLER ID: 1016 (1016)"") in new stack [Aug 14 14:14:49] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:4] Set("SIP/siptrunk-b58197c0", "__INCOMING_DNIS=5000") in new stack [Aug 14 14:14:49] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:5] Set("SIP/siptrunk-b58197c0", "CALLERID(dnid)=5000") in new stack [Aug 14 14:14:49] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:6] Set("SIP/siptrunk-b58197c0", "CDR(userfield)=5000") in new stack [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [5000@private-siptrunk-incoming:7] Goto("SIP/siptrunk-b58197c0", "10") in new stack -- Goto (private-siptrunk-incoming,5000,10) [Aug 14 14:14:49] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'AGI' -- Executing [5000@private-siptrunk-incoming:10] AGI("SIP/siptrunk-b58197c0", "/var/www/scopserv/telephony/scripts/agi/fixcidname.php") in new stack Really destroying SIP dialog '653026167d1385ee072537a162b302b4@192.168.30.254' Method: ACK [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Really destroying SIP dialog '2628be613135fa725e34faad249ac9f2@192.168.30.254' Method: ACK -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php <--- SIP read from 192.168.30.254:5060 ---> ACK sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK10a55ec7;rport From: "1016" ;tag=as70db19bb To: ;tag=as30a8eb81 Contact: Call-ID: 77e74c186c3e510252952def1b926fba@192.168.30.254 CSeq: 103 ACK User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Aug 14 14:14:49] DEBUG[2654]: chan_sip.c:2258 __sip_ack: Stopping retransmission on '77e74c186c3e510252952def1b926fba@192.168.30.254' of Response 103: Match Found [Aug 14 14:14:49] DEBUG[24696]: rtp.c:2712 ast_rtp_raw_write: Difference is 912, ms is 134 [Aug 14 14:14:49] DEBUG[24702]: rtp.c:2712 ast_rtp_raw_write: Difference is 920, ms is 135 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24696]: app_queue.c:2448 is_our_turn: There are 0 available members. [Aug 14 14:14:49] DEBUG[24696]: app_queue.c:2466 is_our_turn: It's not our turn (SIP/siptrunk-b50fe470). [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:49] DEBUG[24639]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 <--- SIP read from 192.168.30.254:5060 ---> BYE sip:5000@192.168.30.165 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK44cc29e8;rport From: "1026" ;tag=as732e53e0 To: ;tag=as3abfab47 Call-ID: 3064269139d64ce57fb86c2c39bf5450@192.168.30.254 CSeq: 104 BYE User-Agent: Asterisk PBX (asterisk) Max-Forwards: 70 Proxy-Authorization: Digest username="siptrunk", realm="asterisk", algorithm=MD5, uri="sip:5000@192.168.30.165", nonce="3cfa0644", response="05136b9efbdf81a0d82e0beadbc93b48" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 Sending to 192.168.30.254 : 5060 (NAT) [Aug 14 14:14:49] WARNING[23948]: channel.c:951 __ast_queue_frame: E[0;37;40m:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b51f77b8", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1843 pbx_extension_helper: Launching 'Goto' -- Executing [h@default-application-acd-customer-new-english:1] Goto("SIP/siptrunk-b5b758a8", "all-hangup|s|1") in new stack -- Goto (all-hangup,s,1) [Aug 14 14:14:50] DEBUG[24715]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b51f77b8", "30") in new stack [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '0' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '2' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [s@all-hangup:1] GotoIf("SIP/siptrunk-b5b758a8", "0?all-faxnotify|s|1:2") in new stack -- Goto (all-hangup,s,2) [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1843 pbx_extension_helper: Launching 'ResetCDR' -- Executing [s@all-hangup:2] ResetCDR("SIP/siptrunk-b5b758a8", "w") in new stack [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '"customer-E:1026" <1026>' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1026' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 's' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'default-application-acd-customer-new-english' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'SIP/siptrunk-b5b758a8' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'Local/1602@default-agent-92ce,1' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'Queue' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'default-customer-new-english|tH|||60|' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:12:28' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:12:28' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '2009-08-14 14:14:50' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '142' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '142' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is 'siptrunk' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '1250273548.6206' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '5000' [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1843 pbx_extension_helper: Launching 'NoCDR' -- Executing [s@all-hangup:3] NoCDR("SIP/siptrunk-b5b758a8", "") in new stack [Aug 14 14:14:50] DEBUG[23837]: pbx.c:1843 pbx_extension_helper: Launching 'System' -- Executing [s@all-hangup:4] System("SIP/siptrunk-b5b758a8", "/var/www/scopserv/telephony/scripts/billing/cdr.sh 1250273548.6206") in new stack [Aug 14 14:14:50] DEBUG[24715]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b51f77b8' does not support indication 8, emulating it [Aug 14 14:14:50] DEBUG[24715]: channel.c:3090 set_format: Set channel SIP/siptrunk-b51f77b8 to write format slin [Aug 14 14:14:50] DEBUG[24715]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:50] DEBUG[24715]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 -- AGI Script /var/www/scopserv/telephony/scripts/agi/fixcidname.php completed, returning 0 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:50] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'Set' -- Executing [5000@private-siptrunk-incoming:11] Set("SIP/siptrunk-b58197c0", "GROUP(siptrunk)=INCOMING") in new stack [Aug 14 14:14:50] DEBUG[24717]: pbx.c:1691 pbx_substitute_variables_helper_full: Function result is '56' [Aug 14 14:14:50] DEBUG[24717]: pbx.c:1759 pbx_substitute_variables_helper_full: Expression result is '1' [Aug 14 14:14:50] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'GotoIf' -- Executing [5000@private-siptrunk-incoming:12] GotoIf("SIP/siptrunk-b58197c0", "1?500") in new stack -- Goto (private-siptrunk-incoming,5000,500) [Aug 14 14:14:50] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'NoOp' -- Executing [5000@private-siptrunk-incoming:500] NoOp("SIP/siptrunk-b58197c0", ""INCOMING CALL LIMIT REACH"") in new stack [Aug 14 14:14:50] DEBUG[24717]: pbx.c:1843 pbx_extension_helper: Launching 'Congestion' -- Executing [5000@private-siptrunk-incoming:501] Congestion("SIP/siptrunk-b58197c0", "30") in new stack [Aug 14 14:14:50] DEBUG[24717]: channel.c:2650 ast_indicate_data: Driver for channel 'SIP/siptrunk-b58197c0' does not support indication 8, emulating it [Aug 14 14:14:50] DEBUG[24717]: channel.c:3090 set_format: Set channel SIP/siptrunk-b58197c0 to write format slin [Aug 14 14:14:50] DEBUG[24717]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 14 14:14:50] DEBUG[24717]: rtp.c:2902 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1649@default-agent-0f2c,1 [Aug 14 14:14:50] WARNING[23948]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queu