[Nov 11 10:49:25] Asterisk 11.0.1 built by root @ localhost on a ppc64 running Linux on 2012-11-10 10:45:58 UTC [Nov 11 10:49:25] DEBUG[5689] config.c: Parsing /etc/asterisk/logger.conf [Nov 11 10:49:25] VERBOSE[5689] config.c: == Parsing '/etc/asterisk/logger.conf': Found [Nov 11 10:49:25] VERBOSE[5689] logger.c: Asterisk Queue Logger restarted [Nov 11 10:49:29] VERBOSE[28473] asterisk.c: -- Remote UNIX connection [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 0 [ 41]: INVITE sip:808@192.168.24.17:5060 SIP/2.0 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 1 [ 34]: Via: SIP/2.0/UDP 192.168.24.3:5060 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 2 [ 52]: From: "950" ;tag=31B26EA8-19FB [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 3 [ 27]: To: [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 4 [ 35]: Date: Sun, 11 Nov 2012 09:49:34 GMT [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 5 [ 57]: Call-ID: EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 6 [ 23]: Supported: timer,100rel [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 7 [ 12]: Min-SE: 1800 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 8 [ 53]: Cisco-Guid: 3985621941-723194338-2205135534-552245383 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 9 [ 37]: User-Agent: Cisco-SIPGateway/IOS-12.x [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 10 [ 86]: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 11 [ 16]: CSeq: 101 INVITE [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 12 [ 15]: Max-Forwards: 6 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 13 [ 75]: Remote-Party-ID: ;party=calling;screen=no;privacy=off [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 14 [ 21]: Timestamp: 1352627374 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 15 [ 36]: Contact: [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 16 [ 12]: Expires: 180 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 17 [ 29]: Allow-Events: telephone-event [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 18 [ 29]: Content-Type: application/sdp [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 19 [ 19]: Content-Length: 214 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Header 20 [ 0]: [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 1 [ 59]: o=CiscoSystemsSIP-GW-UserAgent 7802 154 IN IP4 192.168.24.3 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.24.3 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 5 [ 26]: m=audio 17464 RTP/AVP 8 19 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 6 [ 21]: c=IN IP4 192.168.24.3 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 8 [ 19]: a=rtpmap:19 CN/8000 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: = Looking for Call ID: EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 (Checking From) --From tag 31B26EA8-19FB --To-tag [Nov 11 10:49:34] DEBUG[28487] logger.c: CALL_ID [C-0000003b] created by thread. [Nov 11 10:49:34] DEBUG[28487] acl.c: For destination '192.168.24.3', our source address is '192.168.24.17'. [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.24.17:5060 [Nov 11 10:49:34] DEBUG[28487] chan_sip.c: Allocating new SIP dialog for EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 - INVITE (No RTP) [Nov 11 10:49:34] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 11 10:49:34] DEBUG[28487][C-0000003b] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer,100rel" [Nov 11 10:49:34] DEBUG[28487][C-0000003b] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 11 10:49:34] DEBUG[28487][C-0000003b] sip/reqresp_parser.c: Matched SIP option: timer [Nov 11 10:49:34] DEBUG[28487][C-0000003b] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 11 10:49:34] DEBUG[28487][C-0000003b] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.24.3:5060' into... [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.24.3' and port '5060'. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: NAT detected for 192.168.24.3:5060 / 192.168.24.3:50146 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Initializing initreq for method INVITE - callid EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.24.3' into... [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.24.3' and port ''. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x10b1e4a4' [Nov 11 10:49:34] DEBUG[28487][C-0000003b] res_rtp_asterisk.c: Allocated port 15312 for RTP instance '0x10b1e4a4' [Nov 11 10:49:34] DEBUG[28487][C-0000003b] rtp_engine.c: RTP instance '0x10b1e4a4' is setup and ready to go [Nov 11 10:49:34] DEBUG[28487][C-0000003b] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x10b1e4a4' [Nov 11 10:49:34] VERBOSE[28487][C-0000003b] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Setting NAT on RTP to On [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP o=CiscoSystemsSIP-GW-UserAgent 7802 154 IN IP4 192.168.24.3... UNSUPPORTED OR FAILED. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.24.3' into... [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.24.3' and port ''. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.24.3... OK. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] rtp_engine.c: Setting payload 8 based on m type on 0xf73c4128 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] rtp_engine.c: Setting payload 19 based on m type on 0xf73c4128 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.24.3' into... [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.24.3' and port ''. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 192.168.24.3... OK. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:19 CN/8000... OK. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x10b1e4a4' [Nov 11 10:49:34] DEBUG[28487][C-0000003b] rtp_engine.c: Copying payload 8 from 0xf73c4128 to 0x10b1e650 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] rtp_engine.c: Copying payload 19 from 0xf73c4128 to 0x10b1e650 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x10b1e4a4' [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: We're settling with these formats: (alaw) [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Checking SIP call limits for device 9950 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Updating call counter for incoming call [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.24.17:5060' into... [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.24.17' and port ''. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.24.3' into... [Nov 11 10:49:34] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.24.3' and port ''. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] format_pref.c: Could not find preferred codec - Going for the best codec [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: *** Our native formats are (alaw) [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: *** Joint capabilities are (alaw) [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw) [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: This channel will not be able to handle video. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: build_route: Contact hop: [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: SIP/950-00000051: New call is still down.... Trying... [Nov 11 10:49:34] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.24.3:5060 [Nov 11 10:49:34] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 950 [Nov 11 10:49:34] DEBUG[28478] chan_sip.c: Checking device state for peer 950 [Nov 11 10:49:34] DEBUG[28478] devicestate.c: Changing state for SIP/950 - state 1 (Not in use) [Nov 11 10:49:34] DEBUG[28478] devicestate.c: device 'SIP/950' state '1' [Nov 11 10:49:34] DEBUG[28514] app_queue.c: Device 'SIP/950' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:49:34] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:34] DEBUG[5711][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Launching 'Macro' [Nov 11 10:49:34] VERBOSE[5711][C-0000003b] pbx.c: -- Executing [808@localnumbers:1] Macro("SIP/950-00000051", "growl-remote") in new stack [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is '950' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is 'Local User' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Expression result is '0' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is '950' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is '950' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is 'Local User' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Expression result is '0' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is '950' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is 'Local User' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is '950' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is '"Local User <950>' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is '"Local User <950>' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Launching 'Set' [Nov 11 10:49:34] VERBOSE[5711][C-0000003b] pbx.c: -- Executing [s@macro-growl-remote:1] Set("SIP/950-00000051", "CALLER="Local User <950>"") in new stack [Nov 11 10:49:34] DEBUG[5711][C-0000003b] app_macro.c: Executed application: Set [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Result of 'EPOCH' is '1352627374' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is '10:49:34' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Result of 'CALLER' is '"Local User <950>"' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=91, consumed=1, args.allowed=[0-9][A-Z][a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=48, c2=57 [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=93, consumed=1, args.allowed=][A-Z][a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=91, consumed=1, args.allowed=[A-Z][a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=65, c2=90 [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=93, consumed=1, args.allowed=][a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=91, consumed=1, args.allowed=[a-z][åäöÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=97, c2=122 [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=93, consumed=1, args.allowed=][åäöÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=91, consumed=1, args.allowed=[åäöÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=195, consumed=1, args.allowed=åäöÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=165, consumed=1, args.allowed=¥Ã¤Ã¶Ã…ÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=195, consumed=1, args.allowed=äöÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=164, consumed=1, args.allowed=¤Ã¶Ã…ÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=195, consumed=1, args.allowed=öÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=182, consumed=1, args.allowed=¶Ã…ÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=195, consumed=1, args.allowed=ÅÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=133, consumed=1, args.allowed=…ÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=195, consumed=1, args.allowed=ÄÖ][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=132, consumed=1, args.allowed=„Ö][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=195, consumed=1, args.allowed=Ö][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=150, consumed=1, args.allowed=–][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=93, consumed=1, args.allowed=][+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=91, consumed=1, args.allowed=[+-><()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=43, c2=62 [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=60, consumed=1, args.allowed=<()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=40, consumed=1, args.allowed=()][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=41, consumed=1, args.allowed=)][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=93, consumed=1, args.allowed=][ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=91, consumed=1, args.allowed=[ ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=32, consumed=1, args.allowed= ] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: c1=93, consumed=1, args.allowed=] [Nov 11 10:49:34] DEBUG[5711][C-0000003b] func_strings.c: Allowed: ()+,-./0123456789:;<=>ABCDEFGHIJKLMNOPQRSTUVWXYZ[]abcdefghijklmnopqrstuvwxyz„…–¤¥¶Ã [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Function result is 'Local User <950>' [Nov 11 10:49:34] DEBUG[5711][C-0000003b] pbx.c: Launching 'System' [Nov 11 10:49:34] VERBOSE[5711][C-0000003b] pbx.c: -- Executing [s@macro-growl-remote:2] System("SIP/950-00000051", "growl 83.251.x.x ******** "Inkommande samtal" "[10:49:34] Local User <950>"") in new stack [Nov 11 10:49:35] DEBUG[5711][C-0000003b] app_macro.c: Executed application: System [Nov 11 10:49:35] DEBUG[5711][C-0000003b] pbx.c: Launching 'Dial' [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] pbx.c: -- Executing [808@localnumbers:2] Dial("SIP/950-00000051", "SIP/808&SIP/809,240,gtwWr") in new stack [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Allocating new SIP dialog for 39df53762d85e2cf4cb99ba9578c84bc@sip.server.domain - INVITE (No RTP) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x10c45ad4' [Nov 11 10:49:35] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Allocated port 13126 for RTP instance '0x10c45ad4' [Nov 11 10:49:35] DEBUG[5711][C-0000003b] rtp_engine.c: RTP instance '0x10c45ad4' is setup and ready to go [Nov 11 10:49:35] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x10c45ad4' [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Setting NAT on RTP to On [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 11 10:49:35] DEBUG[5711][C-0000003b] acl.c: For destination '83.251.x.x', our source address is '192.168.24.17'. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Target address 83.251.x.x:5060 is not local, substituting externaddr [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 85.224.x.x:5060 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] format_pref.c: Could not find preferred codec - Going for the best codec [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** Our native formats are (alaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** Joint capabilities are (alaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: This channel will not be able to handle video. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel_internal_api.c: Channel Call ID changing from [C-0000003b] to [C-0000003b] [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable DIALEDTIME. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable ANSWEREDTIME. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable DIALEDPEERNAME. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable DIALEDPEERNUMBER. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable DIALSTATUS. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable MACRO_DEPTH. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable SYSTEMSTATUS. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable CALLER. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable SIPCALLID. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable SIPDOMAIN. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable SIPURI. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Allocating new SIP dialog for 11ff4efe51f01ee746a333e94c17c368@sip.server.domain - INVITE (No RTP) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x10c1c1b4' [Nov 11 10:49:35] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Allocated port 16642 for RTP instance '0x10c1c1b4' [Nov 11 10:49:35] DEBUG[5711][C-0000003b] rtp_engine.c: RTP instance '0x10c1c1b4' is setup and ready to go [Nov 11 10:49:35] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x10c1c1b4' [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Setting NAT on RTP to On [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 11 10:49:35] DEBUG[5711][C-0000003b] acl.c: For destination '83.251.x.x', our source address is '192.168.24.17'. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Target address 83.251.x.x:5060 is not local, substituting externaddr [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 85.224.x.x:5060 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] format_pref.c: Could not find preferred codec - Going for the best codec [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** Our native formats are (alaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** Joint capabilities are (alaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: This channel will not be able to handle video. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel_internal_api.c: Channel Call ID changing from [C-0000003b] to [C-0000003b] [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable DIALEDTIME. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable ANSWEREDTIME. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable DIALEDPEERNAME. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable DIALEDPEERNUMBER. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable DIALSTATUS. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable MACRO_DEPTH. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable SYSTEMSTATUS. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable CALLER. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable SIPCALLID. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable SIPDOMAIN. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] channel.c: Not copying variable SIPURI. [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Outgoing Call for 808 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Updating call counter for outgoing call [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: This call needs video offers, but there's no video support enabled! [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: ** Our capability: (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: ** Our prefcodec: (alaw) [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Audio is at 13126 [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding codec 100004 (alaw) to SDP [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding codec 100002 (gsm) to SDP [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding codec 100017 (testlaw) to SDP [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: -- Done with adding codecs to SDP [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|h263|testlaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Initializing initreq for method INVITE - callid 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 0 [ 40]: INVITE sip:808@192.168.0.11:5060 SIP/2.0 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 3 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 4 [ 31]: To: [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 5 [ 37]: Contact: [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 6 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 8 [ 30]: User-Agent: SIP Server [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 9 [ 35]: Date: Sun, 11 Nov 2012 09:49:35 GMT [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Reliably Transmitting (no NAT) to 83.251.x.x:5060: INVITE sip:808@192.168.0.11:5060 SIP/2.0 Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Max-Forwards: 70 From: "Local User" ;tag=as0b062a95 To: Contact: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE User-Agent: SIP Server Date: Sun, 11 Nov 2012 09:49:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 311 v=0 o=root 2121496268 2121496268 IN IP4 85.224.x.x s=Asterisk PBX 11.0.1 c=IN IP4 85.224.x.x t=0 0 m=audio 13126 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39593 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] app_dial.c: -- Called SIP/808 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Outgoing Call for 809 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Updating call counter for outgoing call [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: This call needs video offers, but there's no video support enabled! [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: ** Our capability: (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: ** Our prefcodec: (alaw) [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Audio is at 16642 [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding codec 100004 (alaw) to SDP [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding codec 100002 (gsm) to SDP [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding codec 100017 (testlaw) to SDP [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: -- Done with adding codecs to SDP [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|h263|testlaw) [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Initializing initreq for method INVITE - callid 0fc118831c9aeda968fba28f12ce888e@sip.server.domain [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 0 [ 40]: INVITE sip:809@192.168.0.11:5060 SIP/2.0 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 3 [ 49]: From: "Local User" ;tag=as6cc38166 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 4 [ 31]: To: [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 5 [ 37]: Contact: [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 6 [ 52]: Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 8 [ 30]: User-Agent: SIP Server [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 9 [ 35]: Date: Sun, 11 Nov 2012 09:49:35 GMT [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] chan_sip.c: Reliably Transmitting (no NAT) to 83.251.x.x:5060: INVITE sip:809@192.168.0.11:5060 SIP/2.0 Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 Max-Forwards: 70 From: "Local User" ;tag=as6cc38166 To: Contact: Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain CSeq: 102 INVITE User-Agent: SIP Server Date: Sun, 11 Nov 2012 09:49:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 311 v=0 o=root 2020512287 2020512287 IN IP4 85.224.x.x s=Asterisk PBX 11.0.1 c=IN IP4 85.224.x.x t=0 0 m=audio 16642 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39595 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] app_dial.c: -- Called SIP/809 [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.24.3:5060 [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] app_dial.c: -- SIP/809-00000053 connected line has changed. Saving it until answer for SIP/950-00000051 [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] app_dial.c: -- SIP/808-00000052 connected line has changed. Saving it until answer for SIP/950-00000051 [Nov 11 10:49:35] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 100 Trying To: From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 1 [ 31]: To: [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 6 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 11 10:49:35] VERBOSE[28487] chan_sip.c: --- (8 headers 0 lines) --- [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag [Nov 11 10:49:35] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: *** SIP TIMER: Cancelling retransmission #39593 - INVITE (got response) [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' Request 102: Found [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: SIP response 100 to standard invite [Nov 11 10:49:35] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:35] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 100 Trying To: From: "Local User" ;tag=as6cc38166 Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 1 [ 31]: To: [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as6cc38166 [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 6 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 11 10:49:35] VERBOSE[28487] chan_sip.c: --- (8 headers 0 lines) --- [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: = Looking for Call ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain (Checking To) --From tag as6cc38166 --To-tag [Nov 11 10:49:35] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: *** SIP TIMER: Cancelling retransmission #39595 - INVITE (got response) [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0fc118831c9aeda968fba28f12ce888e@sip.server.domain' Request 102: Found [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: SIP response 100 to standard invite [Nov 11 10:49:35] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:35] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 180 Ringing To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 6 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 11 10:49:35] VERBOSE[28487] chan_sip.c: --- (8 headers 0 lines) --- [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:49:35] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' Request 102: Found [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: SIP response 180 to standard invite [Nov 11 10:49:35] VERBOSE[28487][C-0000003b] chan_sip.c: list_route: no route [Nov 11 10:49:35] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 808 [Nov 11 10:49:35] DEBUG[28478] chan_sip.c: Checking device state for peer 808 [Nov 11 10:49:35] DEBUG[28478] devicestate.c: Changing state for SIP/808 - state 1 (Not in use) [Nov 11 10:49:35] DEBUG[28478] devicestate.c: device 'SIP/808' state '1' [Nov 11 10:49:35] DEBUG[28514] app_queue.c: Device 'SIP/808' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:49:35] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] app_dial.c: -- SIP/808-00000052 is ringing [Nov 11 10:49:35] DEBUG[5711][C-0000003b] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.24.3:5060 [Nov 11 10:49:35] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 180 Ringing To: ;tag=4984674f679bd045i1 From: "Local User" ;tag=as6cc38166 Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 1 [ 54]: To: ;tag=4984674f679bd045i1 [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as6cc38166 [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 6 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 11 10:49:35] VERBOSE[28487] chan_sip.c: --- (8 headers 0 lines) --- [Nov 11 10:49:35] DEBUG[28487] chan_sip.c: = Looking for Call ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain (Checking To) --From tag as6cc38166 --To-tag 4984674f679bd045i1 [Nov 11 10:49:35] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0fc118831c9aeda968fba28f12ce888e@sip.server.domain' Request 102: Found [Nov 11 10:49:35] DEBUG[28487][C-0000003b] chan_sip.c: SIP response 180 to standard invite [Nov 11 10:49:35] VERBOSE[28487][C-0000003b] chan_sip.c: list_route: no route [Nov 11 10:49:35] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:35] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 809 [Nov 11 10:49:35] DEBUG[28478] chan_sip.c: Checking device state for peer 809 [Nov 11 10:49:35] DEBUG[28478] devicestate.c: Changing state for SIP/809 - state 1 (Not in use) [Nov 11 10:49:35] DEBUG[28478] devicestate.c: device 'SIP/809' state '1' [Nov 11 10:49:35] DEBUG[28514] app_queue.c: Device 'SIP/809' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:49:35] VERBOSE[5711][C-0000003b] app_dial.c: -- SIP/809-00000053 is ringing [Nov 11 10:49:38] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:49:38] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Acked pending invite 102 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Found [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: SIP response 200 to standard invite [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP o=- 62125256 62125256 IN IP4 192.168.0.11... UNSUPPORTED OR FAILED. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11' into... [Nov 11 10:49:38] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port ''. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.11... OK. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Found RTP audio format 8 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] rtp_engine.c: Setting payload 8 based on m type on 0xf73c3dd8 [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Found RTP audio format 100 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] rtp_engine.c: Setting payload 100 based on m type on 0xf73c3dd8 [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Found RTP audio format 101 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] rtp_engine.c: Setting payload 101 based on m type on 0xf73c3dd8 [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] rtp_engine.c: Unsetting payload 100 on 0xf73c3dd8 [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Found unknown media description format NSE for ID 100 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:100 NSE/8000... UNSUPPORTED OR FAILED. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=fmtp:100 192-193... UNSUPPORTED OR FAILED. [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 11 10:49:38] DEBUG[28487][C-0000003b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x10c45ad4' [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Peer audio RTP is at port 192.168.0.11:16414 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] rtp_engine.c: Copying payload 8 from 0xf73c3dd8 to 0x10c45c80 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] rtp_engine.c: Copying payload 101 from 0xf73c3dd8 to 0x10c45c80 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x10c45ad4' [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: We're settling with these formats: (alaw) [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: We have an owner, now see if we need to change this call [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Nov 11 10:49:38] DEBUG[28487][C-0000003b] format_pref.c: Could not find preferred codec - Going for the best codec [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Updating call counter for outgoing call [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: build_route: Contact hop: 808 [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: list_route: hop: [Nov 11 10:49:38] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:38] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:38] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:38] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:38] VERBOSE[5711][C-0000003b] app_dial.c: -- SIP/808-00000052 connected line has changed. Saving it until answer for SIP/950-00000051 [Nov 11 10:49:38] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 808 [Nov 11 10:49:38] DEBUG[28478] chan_sip.c: Checking device state for peer 808 [Nov 11 10:49:38] DEBUG[28478] devicestate.c: Changing state for SIP/808 - state 1 (Not in use) [Nov 11 10:49:38] DEBUG[28478] devicestate.c: device 'SIP/808' state '1' [Nov 11 10:49:38] DEBUG[28514] app_queue.c: Device 'SIP/808' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:49:38] VERBOSE[5711][C-0000003b] app_dial.c: -- SIP/808-00000052 answered SIP/950-00000051 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] channel.c: Hanging up channel 'SIP/809-00000053' [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: This call was answered elsewhere [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: Hangup call SIP/809-00000053, SIP callid 0fc118831c9aeda968fba28f12ce888e@sip.server.domain [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: Hanging up channel in state Ringing (not UP) [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x10c1c1b4' [Nov 11 10:49:38] VERBOSE[5711][C-0000003b] chan_sip.c: Scheduling destruction of SIP dialog '0fc118831c9aeda968fba28f12ce888e@sip.server.domain' in 32000 ms (Method: INVITE) [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0fc118831c9aeda968fba28f12ce888e@sip.server.domain' Request 102: Found [Nov 11 10:49:38] VERBOSE[5711][C-0000003b] chan_sip.c: Reliably Transmitting (no NAT) to 83.251.x.x:5060: CANCEL sip:809@192.168.0.11:5060 SIP/2.0 Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 Max-Forwards: 70 From: "Local User" ;tag=as6cc38166 To: Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain CSeq: 102 CANCEL User-Agent: SIP Server Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 --- [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39600 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:49:38] VERBOSE[5711][C-0000003b] chan_sip.c: Scheduling destruction of SIP dialog '0fc118831c9aeda968fba28f12ce888e@sip.server.domain' in 32000 ms (Method: INVITE) [Nov 11 10:49:38] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 809 [Nov 11 10:49:38] DEBUG[28478] chan_sip.c: Checking device state for peer 809 [Nov 11 10:49:38] DEBUG[28478] devicestate.c: Changing state for SIP/809 - state 1 (Not in use) [Nov 11 10:49:38] DEBUG[28478] devicestate.c: device 'SIP/809' state '1' [Nov 11 10:49:38] DEBUG[28514] app_queue.c: Device 'SIP/809' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:49:38] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 950 [Nov 11 10:49:38] DEBUG[28478] chan_sip.c: Checking device state for peer 950 [Nov 11 10:49:38] DEBUG[28478] devicestate.c: Changing state for SIP/950 - state 1 (Not in use) [Nov 11 10:49:38] DEBUG[28478] devicestate.c: device 'SIP/950' state '1' [Nov 11 10:49:38] DEBUG[28514] app_queue.c: Device 'SIP/950' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: SIP answering channel: SIP/950-00000051 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: Setting framing from config on incoming call [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: ** Our prefcodec: (nothing) [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: -- Done with adding codecs to SDP [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39602 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.24.3:5060 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] features.c: bridge answer set, chan answer set [Nov 11 10:49:38] DEBUG[5711][C-0000003b] features.c: Removing dialed interfaces datastore on SIP/808-00000052 since we're bridging [Nov 11 10:49:38] DEBUG[5711][C-0000003b] channel.c: setting peeraccount to 0920 for SIP/808-00000052 from data on channel SIP/950-00000051 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] channel.c: setting peeraccount to 0920 for SIP/950-00000051 from data on channel SIP/808-00000052 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 0 [ 38]: ACK sip:808@192.168.24.17:5060 SIP/2.0 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 1 [ 34]: Via: SIP/2.0/UDP 192.168.24.3:5060 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 2 [ 52]: From: "950" ;tag=31B26EA8-19FB [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 3 [ 42]: To: ;tag=as323de783 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 4 [ 35]: Date: Sun, 11 Nov 2012 09:49:34 GMT [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 5 [ 57]: Call-ID: EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 6 [ 15]: Max-Forwards: 6 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 8 [ 13]: CSeq: 101 ACK [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: = Looking for Call ID: EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 (Checking From) --From tag 31B26EA8-19FB --To-tag as323de783 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #39602 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on 'EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3' of Response 101: Match Found [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:38] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 487 Request Terminated To: ;tag=4984674f679bd045i1 From: "Local User" ;tag=as6cc38166 Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 0 [ 30]: SIP/2.0 487 Request Terminated [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 1 [ 54]: To: ;tag=4984674f679bd045i1 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as6cc38166 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 6 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 11 10:49:38] VERBOSE[28487] chan_sip.c: --- (8 headers 0 lines) --- [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: = Looking for Call ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain (Checking To) --From tag as6cc38166 --To-tag 4984674f679bd045i1 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Acked pending invite 102 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '0fc118831c9aeda968fba28f12ce888e@sip.server.domain' of Request 102: Match Found [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: SIP response 487 to standard invite [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Transmitting (no NAT) to 83.251.x.x:5060: ACK sip:809@192.168.0.11:5060 SIP/2.0 Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 Max-Forwards: 70 From: "Local User" ;tag=as6cc38166 To: ;tag=4984674f679bd045i1 Contact: Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain CSeq: 102 ACK User-Agent: SIP Server Content-Length: 0 --- [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:809' onto UDP socket destined for 83.251.x.x:5060 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Updating call counter for outgoing call [Nov 11 10:49:38] VERBOSE[28487][C-0000003b] chan_sip.c: Scheduling destruction of SIP dialog '0fc118831c9aeda968fba28f12ce888e@sip.server.domain' in 32000 ms (Method: INVITE) [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:38] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=4984674f679bd045i1 From: "Local User" ;tag=as6cc38166 Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain CSeq: 102 CANCEL Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 0 <-------------> [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 1 [ 54]: To: ;tag=4984674f679bd045i1 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as6cc38166 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 CANCEL [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK06e66c70 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 6 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 11 10:49:38] VERBOSE[28487] chan_sip.c: --- (8 headers 0 lines) --- [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: = Looking for Call ID: 0fc118831c9aeda968fba28f12ce888e@sip.server.domain (Checking To) --From tag as6cc38166 --To-tag 4984674f679bd045i1 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #39600 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '0fc118831c9aeda968fba28f12ce888e@sip.server.domain' of Request 102: Match Found [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- start learning mode pass with addr = 192.168.24.3:17464 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- probation = 4, seq = 8424 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- start learning mode pass with addr = 192.168.24.3:17464 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- probation = 3, seq = 8425 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- start learning mode pass with addr = 192.168.24.3:17464 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- probation = 2, seq = 8426 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- Condition for learning hasn't exited, so reject the frame. [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- start learning mode pass with addr = 192.168.24.3:17464 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- probation = 1, seq = 8427 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: 0x10a56880 -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 192.168.24.3:17464 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Nov 11 10:49:38] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x10c45ad4' [Nov 11 10:49:38] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:49:38] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:49:38] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:38] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:38] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:38] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:38] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:39] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:49:39] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:49:39] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:49:39] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:39] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:49:39] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:39] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:39] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:39] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:39] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:40] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Got RTCP report of 108 bytes [Nov 11 10:49:41] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:49:41] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:49:41] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:49:41] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:41] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:49:41] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:41] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:41] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:41] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:41] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:43] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Got RTCP report of 108 bytes [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '280a063d31e1dea339afbfca618d7d1f@sip.server.domain' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 280a063d31e1dea339afbfca618d7d1f@sip.server.domain [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '2cd74b3d5f283731430f3e2e7bfe3fcf@sip.server.domain' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 2cd74b3d5f283731430f3e2e7bfe3fcf@sip.server.domain [Nov 11 10:49:45] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:49:45] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:49:45] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:45] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:49:45] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:45] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:45] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:45] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:45] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '74f93b2a1fbffe6a19fcae5c2337d98b@80.244.65.70:5060' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 74f93b2a1fbffe6a19fcae5c2337d98b@80.244.65.70:5060 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '0afb91392339ea344f1c851a4c455598@sip.server.domain' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 0afb91392339ea344f1c851a4c455598@sip.server.domain [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '71b100c94fae865578bd45573f3158f4@80.244.65.70:5060' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 71b100c94fae865578bd45573f3158f4@80.244.65.70:5060 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '698f779d53a6501911b11d2c76b7d67d@sip.server.domain' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 698f779d53a6501911b11d2c76b7d67d@sip.server.domain [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '699d54015b54cf30095aebbf40f2398b@80.244.65.70:5060' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 699d54015b54cf30095aebbf40f2398b@80.244.65.70:5060 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '6b20d3533178dbae633badfb75f43488@sip.server.domain' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 6b20d3533178dbae633badfb75f43488@sip.server.domain [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '4661a30b434bdac823e2accb2124ac51@80.244.65.70:5060' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 4661a30b434bdac823e2accb2124ac51@80.244.65.70:5060 [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Auto destroying SIP dialog '027cdcd5122801181152bfe62c9a8bf3@sip.server.domain' [Nov 11 10:49:45] DEBUG[28487] chan_sip.c: Destroying SIP dialog 027cdcd5122801181152bfe62c9a8bf3@sip.server.domain [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 0 [ 38]: BYE sip:808@192.168.24.17:5060 SIP/2.0 [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 1 [ 34]: Via: SIP/2.0/UDP 192.168.24.3:5060 [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 2 [ 52]: From: "950" ;tag=31B26EA8-19FB [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 3 [ 42]: To: ;tag=as323de783 [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 4 [ 35]: Date: Sun, 11 Nov 2012 09:49:34 GMT [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 5 [ 57]: Call-ID: EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 6 [ 37]: User-Agent: Cisco-SIPGateway/IOS-12.x [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 7 [ 15]: Max-Forwards: 6 [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 8 [ 21]: Timestamp: 1352627386 [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 9 [ 13]: CSeq: 102 BYE [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 11 10:49:46] DEBUG[28487] chan_sip.c: = Looking for Call ID: EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 (Checking From) --From tag 31B26EA8-19FB --To-tag as323de783 [Nov 11 10:49:46] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:46] DEBUG[28487][C-0000003b] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 11 10:49:46] DEBUG[28487][C-0000003b] chan_sip.c: Initializing initreq for method BYE - callid EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 [Nov 11 10:49:46] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.24.3:5060' into... [Nov 11 10:49:46] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.24.3' and port '5060'. [Nov 11 10:49:46] DEBUG[28487][C-0000003b] chan_sip.c: NAT detected for 192.168.24.3:5060 / 192.168.24.3:51529 [Nov 11 10:49:46] DEBUG[28487][C-0000003b] chan_sip.c: Setting SIP_ALREADYGONE on dialog EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 [Nov 11 10:49:46] DEBUG[28487][C-0000003b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x10b1e4a4' [Nov 11 10:49:46] DEBUG[28487][C-0000003b] chan_sip.c: Received bye, issuing owner hangup [Nov 11 10:49:46] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.24.3:51529 [Nov 11 10:49:46] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:46] DEBUG[5711][C-0000003b] channel.c: Didn't get a frame from channel: SIP/950-00000051 [Nov 11 10:49:46] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 11 10:49:46] DEBUG[5711][C-0000003b] channel.c: Bridge stops bridging channels SIP/950-00000051 and SIP/808-00000052 [Nov 11 10:49:46] DEBUG[5711][C-0000003b] cdr_mysql.c: Inserting a CDR record. [Nov 11 10:49:46] DEBUG[5711][C-0000003b] cdr_mysql.c: SQL command as follows: INSERT INTO calls (`calldate`,`clid`,`src`,`dst`,`dcontext`,`channel`,`dstchannel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`accountcode`) VALUES ('2012-11-11 10:49:34','\"Local User\" <950>','950','808','localnumbers','SIP/950-00000051','SIP/808-00000052','Dial','SIP/808&SIP/809,240,gtwWr','12','8','ANSWERED','3','0920') [Nov 11 10:49:46] DEBUG[5711][C-0000003b] channel.c: Hanging up channel 'SIP/808-00000052' [Nov 11 10:49:46] DEBUG[5711][C-0000003b] chan_sip.c: Hangup call SIP/808-00000052, SIP callid 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:46] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x10c45ad4' [Nov 11 10:49:46] DEBUG[5711][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:46] DEBUG[5711][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:46] DEBUG[5711][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:46] DEBUG[5711][C-0000003b] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39608 [Nov 11 10:49:46] DEBUG[5711][C-0000003b] chan_sip.c: Trying to put 'BYE sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:46] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 808 [Nov 11 10:49:46] DEBUG[28478] chan_sip.c: Checking device state for peer 808 [Nov 11 10:49:46] DEBUG[28478] devicestate.c: Changing state for SIP/808 - state 1 (Not in use) [Nov 11 10:49:46] DEBUG[28478] devicestate.c: device 'SIP/808' state '1' [Nov 11 10:49:46] DEBUG[28514] app_queue.c: Device 'SIP/808' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:49:46] DEBUG[5711][C-0000003b] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 11 10:49:46] DEBUG[5711][C-0000003b] pbx.c: Spawn extension (localnumbers,808,2) exited non-zero on 'SIP/950-00000051' [Nov 11 10:49:46] VERBOSE[5711][C-0000003b] pbx.c: == Spawn extension (localnumbers, 808, 2) exited non-zero on 'SIP/950-00000051' [Nov 11 10:49:46] DEBUG[5711][C-0000003b] channel.c: Soft-Hanging up channel 'SIP/950-00000051' [Nov 11 10:49:46] DEBUG[5711][C-0000003b] channel.c: Hanging up channel 'SIP/950-00000051' [Nov 11 10:49:46] DEBUG[5711][C-0000003b] chan_sip.c: Hangup call SIP/950-00000051, SIP callid EF12BBEE-2B1B11E2-8372B2AE-20EA9887@192.168.24.3 [Nov 11 10:49:46] DEBUG[5711][C-0000003b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x10b1e4a4' [Nov 11 10:49:46] DEBUG[28478] devicestate.c: No provider found, checking channel drivers for SIP - 950 [Nov 11 10:49:46] DEBUG[28478] chan_sip.c: Checking device state for peer 950 [Nov 11 10:49:46] DEBUG[28478] devicestate.c: Changing state for SIP/950 - state 1 (Not in use) [Nov 11 10:49:46] DEBUG[28478] devicestate.c: device 'SIP/950' state '1' [Nov 11 10:49:46] DEBUG[28514] app_queue.c: Device 'SIP/950' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 11 10:49:47] DEBUG[28487] chan_sip.c: SIP TIMER: Rescheduling retransmission #39608 (1) BYE - 8 [Nov 11 10:49:47] DEBUG[28487] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #39608)) [Nov 11 10:49:47] DEBUG[28487] chan_sip.c: Trying to put 'BYE sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:48] DEBUG[28487] chan_sip.c: SIP TIMER: Rescheduling retransmission #39608 (2) BYE - 8 [Nov 11 10:49:48] DEBUG[28487] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #39608)) [Nov 11 10:49:48] DEBUG[28487] chan_sip.c: Trying to put 'BYE sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:49] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:49:49] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:49:49] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:49:49] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:49] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:49:49] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:49] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:49] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:49] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:49] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:50] DEBUG[28487] chan_sip.c: SIP TIMER: Rescheduling retransmission #39608 (3) BYE - 8 [Nov 11 10:49:50] DEBUG[28487] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #39608)) [Nov 11 10:49:50] DEBUG[28487] chan_sip.c: Trying to put 'BYE sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:53] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:49:53] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:49:53] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:49:53] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:53] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:49:53] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:53] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:53] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:53] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:53] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:54] DEBUG[28487] chan_sip.c: SIP TIMER: Rescheduling retransmission #39608 (4) BYE - 8 [Nov 11 10:49:54] DEBUG[28487] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #39608)) [Nov 11 10:49:54] DEBUG[28487] chan_sip.c: Trying to put 'BYE sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:55] VERBOSE[5704] asterisk.c: -- Remote UNIX connection disconnected [Nov 11 10:49:57] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:49:57] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:49:57] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:49:57] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:49:57] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:49:57] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:49:57] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:49:57] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:49:57] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:49:57] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:49:58] DEBUG[28487] chan_sip.c: SIP TIMER: Rescheduling retransmission #39608 (5) BYE - 8 [Nov 11 10:49:58] DEBUG[28487] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #39608)) [Nov 11 10:49:58] DEBUG[28487] chan_sip.c: Trying to put 'BYE sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:50:01] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:50:01] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:50:01] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:50:01] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:50:01] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:50:01] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:50:01] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:50:01] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:50:01] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:50:01] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:50:02] DEBUG[28487] chan_sip.c: SIP TIMER: Rescheduling retransmission #39608 (6) BYE - 8 [Nov 11 10:50:02] DEBUG[28487] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #39608)) [Nov 11 10:50:02] DEBUG[28487] chan_sip.c: Trying to put 'BYE sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:50:05] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:50:05] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:50:05] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:50:05] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:50:05] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:50:05] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:50:05] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:50:05] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:50:05] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:50:05] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread. [Nov 11 10:50:06] DEBUG[28487] chan_sip.c: SIP TIMER: Rescheduling retransmission #39608 (7) BYE - 8 [Nov 11 10:50:06] DEBUG[28487] chan_sip.c: ** SIP timers: Rescheduling retransmission 8 to 4000 ms (t1 500 ms (Retrans id #39608)) [Nov 11 10:50:06] DEBUG[28487] chan_sip.c: Trying to put 'BYE sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:50:06] VERBOSE[28473] asterisk.c: -- Remote UNIX connection [Nov 11 10:50:09] VERBOSE[28487] chan_sip.c: <--- SIP read from UDP:83.251.x.x:5060 ---> SIP/2.0 200 OK To: ;tag=40a38d0575fc4b6i0 From: "Local User" ;tag=as0b062a95 Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain CSeq: 102 INVITE Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f Contact: 808 Server: Linksys/PAP2T-5.1.3(LS) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 62125256 62125256 IN IP4 192.168.0.11 s=- c=IN IP4 192.168.0.11 t=0 0 m=audio 16414 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 1 [ 53]: To: ;tag=40a38d0575fc4b6i0 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 2 [ 49]: From: "Local User" ;tag=as0b062a95 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 3 [ 52]: Call-ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 5 [ 58]: Via: SIP/2.0/UDP 85.224.x.x:5060;branch=z9hG4bK1e6f592f [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 6 [ 40]: Contact: 808 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 7 [ 31]: Server: Linksys/PAP2T-5.1.3(LS) [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 8 [ 19]: Content-Length: 257 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 10 [ 29]: Supported: x-sipura, replaces [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Header 12 [ 0]: [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 0 [ 3]: v=0 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 1 [ 41]: o=- 62125256 62125256 IN IP4 192.168.0.11 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 2 [ 3]: s=- [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.0.11 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 5 [ 31]: m=audio 16414 RTP/AVP 8 100 101 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 7 [ 21]: a=rtpmap:100 NSE/8000 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 8 [ 18]: a=fmtp:100 192-193 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 11 [ 10]: a=ptime:30 [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: Body 12 [ 10]: a=sendrecv [Nov 11 10:50:09] VERBOSE[28487] chan_sip.c: --- (12 headers 13 lines) --- [Nov 11 10:50:09] DEBUG[28487] chan_sip.c: = Looking for Call ID: 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain (Checking To) --From tag as0b062a95 --To-tag 40a38d0575fc4b6i0 [Nov 11 10:50:09] DEBUG[28487][C-0000003b] logger.c: CALL_ID [C-0000003b] bound to thread. [Nov 11 10:50:09] DEBUG[28487][C-0000003b] chan_sip.c: Stopping retransmission on '5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain' of Request 102: Match Not Found [Nov 11 10:50:09] DEBUG[28487][C-0000003b] chan_sip.c: Strict routing enforced for session 5b3c2a3d7513bdb8109fdf2d61680d28@sip.server.domain [Nov 11 10:50:09] DEBUG[28487][C-0000003b] netsock2.c: Splitting '192.168.0.11:5060' into... [Nov 11 10:50:09] DEBUG[28487][C-0000003b] netsock2.c: ...host '192.168.0.11' and port '5060'. [Nov 11 10:50:09] DEBUG[28487][C-0000003b] chan_sip.c: Trying to put 'ACK sip:808' onto UDP socket destined for 192.168.0.11:5060 [Nov 11 10:50:09] DEBUG[28487][C-0000003b] logger.c: Call_ID [C-0000003b] being removed from thread.