[Dec 17 11:38:04] Asterisk 11.15.0-1 built by root @ pbx-dev32m on a i686 running Linux on 2014-12-16 13:01:42 UTC [Dec 17 11:38:04] DEBUG[3299] config.c: Parsing /etc/asterisk/logger.conf [Dec 17 11:38:08] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:08] DEBUG[3296] db.c: Unable to find key '100' in family 'CFIM' [Dec 17 11:38:08] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:08] DEBUG[3296] db.c: Unable to find key '100' in family 'CFUN' [Dec 17 11:38:08] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:08] DEBUG[3296] db.c: Unable to find key '100' in family 'CFBS' [Dec 17 11:38:08] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:08] DEBUG[3296] db.c: Unable to find key '100' in family 'CFNOCID' [Dec 17 11:38:08] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:08] DEBUG[3296] db.c: Unable to find key '100' in family 'CFFB' [Dec 17 11:38:08] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:08] DEBUG[3296] db.c: Unable to find key '100' in family 'CFFBTO' [Dec 17 11:38:09] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:09] DEBUG[3296] db.c: Unable to find key '100' in family 'DND' [Dec 17 11:38:09] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:09] DEBUG[3296] db.c: Unable to find key '100' in family 'DND_ALLOW' [Dec 17 11:38:09] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:09] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:09] DEBUG[3296] db.c: Unable to find key '100' in family 'PARALLEL' [Dec 17 11:38:09] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:09] DEBUG[3296] db.c: Unable to find key '100' in family 'PAR_DELAY' [Dec 17 11:38:09] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:09] DEBUG[3296] db.c: Unable to find key '100' in family 'PAR_AMD' [Dec 17 11:38:18] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SUBSCRIBE sip:hans@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bKebde33f842c914d86 Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKebde33f842c914d86 Max-Forwards: 69 From: "Hans Mustermann" ;tag=536ac1974e To: Call-ID: 8df0d8f279dfac42 CSeq: 27766 SUBSCRIBE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="hans",realm="asterisk",nonce="7b49ee3f",uri="sip:hans@192.168.10.75:5060",response="2913c69ce8d397ea00995b388f184756",algorithm=MD5 Contact: "Hans Mustermann" ;+sip.instance="" Event: message-summary Expires: 3600 Supported: path User-Agent: Aastra 53i/3.3.1.2217 Content-Length: 0 <-------------> [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 0 [ 46]: SUBSCRIBE sip:hans@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bKebde33f842c914d86 [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 2 [ 68]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKebde33f842c914d86 [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 3 [ 16]: Max-Forwards: 69 [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 4 [ 68]: From: "Hans Mustermann" ;tag=536ac1974e [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 5 [ 33]: To: [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 6 [ 25]: Call-ID: 8df0d8f279dfac42 [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 7 [ 21]: CSeq: 27766 SUBSCRIBE [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 8 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 9 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 10 [163]: Authorization: Digest username="hans",realm="asterisk",nonce="7b49ee3f",uri="sip:hans@192.168.10.75:5060",response="2913c69ce8d397ea00995b388f184756",algorithm=MD5 [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 11 [135]: Contact: "Hans Mustermann" ;+sip.instance="" [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 12 [ 22]: Event: message-summary [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 14 [ 15]: Supported: path [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 15 [ 33]: User-Agent: Aastra 53i/3.3.1.2217 [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Header 16 [ 17]: Content-Length: 0 [Dec 17 11:38:18] VERBOSE[3289] chan_sip.c: --- (17 headers 0 lines) --- [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: = Looking for Call ID: 8df0d8f279dfac42 (Checking From) --From tag 536ac1974e --To-tag [Dec 17 11:38:18] DEBUG[3289] acl.c: For destination '192.168.10.75', our source address is '192.168.10.75'. [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:25060 [Dec 17 11:38:18] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:18] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:18] VERBOSE[3289] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Allocating new SIP dialog for 8df0d8f279dfac42 - SUBSCRIBE (No RTP) [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Dec 17 11:38:18] VERBOSE[3289] chan_sip.c: Creating new subscription [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 8df0d8f279dfac42 [Dec 17 11:38:18] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:18] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:18] VERBOSE[3289] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: build_route: Contact hop: "Hans Mustermann" ;+sip.instance="" [Dec 17 11:38:18] VERBOSE[3289] chan_sip.c: list_route: hop: [Dec 17 11:38:18] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:18] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:18] VERBOSE[3289] chan_sip.c: No matching peer for 'hans' from '192.168.10.75:5060' [Dec 17 11:38:18] VERBOSE[3289] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bKebde33f842c914d86;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKebde33f842c914d86 From: "Hans Mustermann" ;tag=536ac1974e To: ;tag=as2694a72d Call-ID: 8df0d8f279dfac42 CSeq: 27766 SUBSCRIBE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="396f332f" Content-Length: 0 <------------> [Dec 17 11:38:18] DEBUG[3289] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:18] VERBOSE[3289] chan_sip.c: Scheduling destruction of SIP dialog '8df0d8f279dfac42' in 32000 ms (Method: SUBSCRIBE) [Dec 17 11:38:24] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> INVITE sip:100@192.168.10.75:25060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b Max-Forwards: 69 From: "PhoneA" ;tag=80931a0487 To: Call-ID: 0dcce801bbc3a2c1 CSeq: 8089 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "PhoneA" ;+sip.instance="" Supported: path, 100rel, replaces User-Agent: Aastra 55i/3.3.1.2217 Content-Type: application/sdp Content-Length: 620 v=0 o=MxSIP 0 1 IN IP4 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 0 [ 53]: INVITE sip:100@192.168.10.75:25060;user=phone SIP/2.0 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 1 [ 85]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 2 [ 86]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 3 [ 16]: Max-Forwards: 69 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 4 [ 62]: From: "PhoneA" ;tag=80931a0487 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 5 [ 43]: To: [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 6 [ 25]: Call-ID: 0dcce801bbc3a2c1 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 7 [ 17]: CSeq: 8089 INVITE [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 8 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 9 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 10 [129]: Contact: "PhoneA" ;+sip.instance="" [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 11 [ 33]: Supported: path, 100rel, replaces [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 12 [ 33]: User-Agent: Aastra 55i/3.3.1.2217 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 14 [ 19]: Content-Length: 620 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 15 [ 0]: [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 0 [ 3]: v=0 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.201 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 2 [ 10]: s=SIP Call [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 5 [ 70]: m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 21 [ 25]: a=silenceSupp:off - - - - [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 22 [ 19]: a=fmtp:18 annexb=no [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Body 25 [ 10]: a=sendrecv [Dec 17 11:38:24] VERBOSE[3289] chan_sip.c: --- (15 headers 26 lines) --- [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: = Looking for Call ID: 0dcce801bbc3a2c1 (Checking From) --From tag 80931a0487 --To-tag [Dec 17 11:38:24] DEBUG[3289] acl.c: For destination '192.168.10.75', our source address is '192.168.10.75'. [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:25060 [Dec 17 11:38:24] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:24] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:24] VERBOSE[3289] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Allocating new SIP dialog for 0dcce801bbc3a2c1 - INVITE (No RTP) [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Dec 17 11:38:24] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Begin: parsing SIP "Supported: path, 100rel, replaces" [Dec 17 11:38:24] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Found SIP option: -path- [Dec 17 11:38:24] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Matched SIP option: path [Dec 17 11:38:24] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Found SIP option: -100rel- [Dec 17 11:38:24] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Matched SIP option: 100rel [Dec 17 11:38:24] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Found SIP option: -replaces- [Dec 17 11:38:24] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Matched SIP option: replaces [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 0dcce801bbc3a2c1 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Using INVITE request as basis request - 0dcce801bbc3a2c1 [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found peer 'phone-a' for 'phone-a' from 192.168.10.75:5060 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9759f6c' [Dec 17 11:38:24] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Allocated port 19658 for RTP instance '0x9759f6c' [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: RTP instance '0x9759f6c' is setup and ready to go [Dec 17 11:38:24] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9759f6c' [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.201... OK. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.201' into... [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.201' and port ''. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 0 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 18 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 18 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 106 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 106 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 107 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 107 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 113 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 113 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 110 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 110 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 111 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 111 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 112 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 112 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 98 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 98 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 97 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 97 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 115 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 115 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 96 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 9 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 9 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 8 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 101 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 101 based on m type on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format G729 for ID 18 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Unsetting payload 106 on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found unknown media description format BV16 for ID 106 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Unsetting payload 107 on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found unknown media description format BV32 for ID 107 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format L16 for ID 113 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Unsetting payload 110 on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found unknown media description format PCMU for ID 110 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Unsetting payload 111 on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found unknown media description format PCMA for ID 111 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format L16 for ID 112 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Unsetting payload 98 on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found unknown media description format G726-16 for ID 98 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Unsetting payload 97 on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found unknown media description format G726-24 for ID 97 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format G726-32 for ID 115 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Unsetting payload 96 on 0xb5ba9240 [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found unknown media description format G726-40 for ID 96 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format G722 for ID 9 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g726|slin|g729|g722|slin16)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 17 11:38:24] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9759f6c' [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 0 from 0xb5ba9240 to 0x975a118 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 8 from 0xb5ba9240 to 0x975a118 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 9 from 0xb5ba9240 to 0x975a118 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 18 from 0xb5ba9240 to 0x975a118 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 101 from 0xb5ba9240 to 0x975a118 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 112 from 0xb5ba9240 to 0x975a118 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 113 from 0xb5ba9240 to 0x975a118 [Dec 17 11:38:24] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 115 from 0xb5ba9240 to 0x975a118 [Dec 17 11:38:24] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x9759f6c' [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: We're settling with these formats: (ulaw|alaw) [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Checking SIP call limits for device phone-a [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Updating call counter for incoming call [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Call from peer 'phone-a' is 1 out of 2147483647 [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.75:25060' into... [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:24] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: Looking for 100 in Standard (domain 192.168.10.75) [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: *** Our native formats are (alaw) [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: *** Joint capabilities are (ulaw|alaw) [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: *** Our capabilities are (ulaw|alaw) [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: This channel will not be able to handle video. [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: build_route: Contact hop: "PhoneA" ;+sip.instance="" [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: list_route: hop: [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: SIP/phone-a-00000000: New call is still down.... Trying... [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b From: "PhoneA" ;tag=80931a0487 To: Call-ID: 0dcce801bbc3a2c1 CSeq: 8089 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:24] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-a [Dec 17 11:38:24] DEBUG[3251] chan_sip.c: Checking device state for peer phone-a [Dec 17 11:38:24] DEBUG[3251] devicestate.c: Changing state for SIP/phone-a - state 2 (In use) [Dec 17 11:38:24] DEBUG[3251] devicestate.c: device 'SIP/phone-a' state '2' [Dec 17 11:38:24] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-a [Dec 17 11:38:24] DEBUG[3251] chan_sip.c: Checking device state for peer phone-a [Dec 17 11:38:24] DEBUG[3251] devicestate.c: Changing state for SIP/phone-a - state 2 (In use) [Dec 17 11:38:24] DEBUG[3251] devicestate.c: device 'SIP/phone-a' state '2' [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone-a-00000000 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 180 CallerIDName: PhoneA AccountCode: Exten: 100 Context: Standard Uniqueid: 1418812704.0 [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone-a-00000000 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 180 CallerIDName: PhoneA ConnectedLineNum: ConnectedLineName: Uniqueid: 1418812704.0 [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: 100 Context: Standard Channel: SIP/phone-a-00000000 Application: AppData: EventTime: 2014-12-17 11:38:24 AMAFlags: DOCUMENTATION UniqueID: 1418812704.0 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'Gosub' [Dec 17 11:38:24] DEBUG[3345][C-00000000] app_stack.c: Channel SIP/phone-a-00000000 has no datastore, so we're allocating one. [Dec 17 11:38:24] DEBUG[3345][C-00000000] app_stack.c: Setting 'ARG1' to '100' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Function CDR(userfield) result is '(null)' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'ARG1' is '100' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'Set' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Function SIP_HEADER(User-Agent) result is 'Aastra 55i/3.3.1.2217' [Dec 17 11:38:24] DEBUG[3345][C-00000000] func_strings.c: FUNCTION REGEX (^BSIP )(Aastra 55i/3.3.1.2217) [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Function REGEX("^BSIP " Aastra 55i/3.3.1.2217) result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'Set' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'Originated' is NULL [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '1' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'GotoIf' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'Set' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'AGI' [Dec 17 11:38:24] DEBUG[3253] app_queue.c: Extension '180@_extensions' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:24] DEBUG[3253] chan_sip.c: Strict routing enforced for session CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 [Dec 17 11:38:24] VERBOSE[3253] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:24] DEBUG[3253] netsock2.c: Splitting '192.168.10.124:5060' into... [Dec 17 11:38:24] DEBUG[3253] netsock2.c: ...host '192.168.10.124' and port '5060'. [Dec 17 11:38:24] VERBOSE[3253] chan_sip.c: set_destination: set destination to 192.168.10.124:5060 [Dec 17 11:38:24] VERBOSE[3253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.124:5060: NOTIFY sip:phone-b@192.168.10.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK0c22846c Max-Forwards: 70 From: ;tag=as6db3dcd7 To: ;tag=7C2F8008F5C1_T2062061876;user=phone Contact: Call-ID: CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 CSeq: 103 NOTIFY User-Agent: IPTAM PBX (Version 20141216/6814) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 203 confirmed --- [Dec 17 11:38:24] DEBUG[3253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #31 [Dec 17 11:38:24] DEBUG[3253] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.124:5060 [Dec 17 11:38:24] DEBUG[3295] app_queue.c: Device 'SIP/phone-a' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:24] DEBUG[3295] app_queue.c: Device 'SIP/phone-a' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: _extensions Hint: SIP/phone-a Status: 1 [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'ARG1' is '100' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'OptionCD' is '1' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'OptionCFNOCIDfirst' is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'TrunkSelection' is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'SIPTrunkSelection' is '' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'HopCountMax' is '7' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'CAUSE' is '' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'XMPPactive' is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DOOR_AGENT' is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'PubVM' is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'CFFB_ACTIVE' is NULL [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'FALLBACK' is NULL [Dec 17 11:38:24] DEBUG[3345][C-00000000] db.c: Unable to find key '100' in family 'LANGUAGE' [Dec 17 11:38:24] DEBUG[3345][C-00000000] db.c: Unable to find key '100' in family 'CFFB' [Dec 17 11:38:24] DEBUG[3345][C-00000000] db.c: Unable to find key '100' in family 'CFIM' [Dec 17 11:38:24] DEBUG[3345][C-00000000] db.c: Unable to find key '100' in family 'DND' [Dec 17 11:38:24] DEBUG[3345][C-00000000] db.c: Unable to find key '100' in family 'PARALLEL' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'SIPREFERREDBYHDR' is NULL [Dec 17 11:38:24] DEBUG[3345][C-00000000] db.c: Unable to find key '100' in family 'CFUN' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'SourceIsExt' is NULL [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'USE_CALLBACK_NR' is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDEXT' is '100' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Function DB(LIMIT/100) result is '1' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'Set' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'limit' is '1' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '1' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'limit' is '1' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'GotoIf' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Not taking any branch [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Function CALLERID(num) result is '180' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDEXT' is '100' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'GotoIf' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Not taking any branch [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-b' [Dec 17 11:38:24] DEBUG[3345][C-00000000] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Function DEVICE_STATE(SIP/phone-b) result is 'NOT_INUSE' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-b' [Dec 17 11:38:24] DEBUG[3345][C-00000000] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Function DEVICE_STATE(SIP/phone-b) result is 'NOT_INUSE' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-b' [Dec 17 11:38:24] DEBUG[3345][C-00000000] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Function DEVICE_STATE(SIP/phone-b) result is 'NOT_INUSE' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-b' [Dec 17 11:38:24] DEBUG[3345][C-00000000] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Function DEVICE_STATE(SIP/phone-b) result is 'NOT_INUSE' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'GotoIf' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Not taking any branch [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDEXT' is '100' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'CHANNEL' is 'SIP/phone-a-00000000' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'UserEvent' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDEXT' is '100' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'Set' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDEXT' is '100' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'Goto' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'ActiveAmt' is NULL [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '1' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'GotoIf' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-b' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'AGI' [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: UserEvent Privilege: user,all UserEvent: CalledExt Uniqueid: 1418812704.0 Dst: 100 Channel: SIP/phone-a-00000000 [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DISTINCTIVE_RINGING' is NULL [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'ICOM_ACTIVE' is '0' [Dec 17 11:38:24] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.124:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK0c22846c From: ;tag=as6db3dcd7 To: ;tag=7C2F8008F5C1_T2062061876;user=phone Call-ID: CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 CSeq: 103 NOTIFY User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Content-Length: 0 <-------------> [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK0c22846c [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 2 [ 44]: From: ;tag=as6db3dcd7 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 3 [ 71]: To: ;tag=7C2F8008F5C1_T2062061876;user=phone [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 4 [ 57]: Call-ID: CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 103 NOTIFY [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 6 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 17 11:38:24] VERBOSE[3289] chan_sip.c: --- (8 headers 0 lines) --- [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 (Checking To) --From tag as6db3dcd7 --To-tag 7C2F8008F5C1_T2062061876 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Acked pending invite 103 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #31 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Stopping retransmission on 'CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124' of Request 103: Match Found [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'Set' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'UseT2M' is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Expression result is '0' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'ExecIf' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Result of 'DIAL_STRING' is 'SIP/phone-b,' [Dec 17 11:38:24] DEBUG[3345][C-00000000] pbx.c: Launching 'Dial' [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Allocating new SIP dialog for 4dd918936296624743905c163429e5cc@192.168.10.75 - INVITE (No RTP) [Dec 17 11:38:24] DEBUG[3345][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x97662e4' [Dec 17 11:38:24] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Allocated port 18368 for RTP instance '0x97662e4' [Dec 17 11:38:24] DEBUG[3345][C-00000000] rtp_engine.c: RTP instance '0x97662e4' is setup and ready to go [Dec 17 11:38:24] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x97662e4' [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 17 11:38:24] DEBUG[3345][C-00000000] acl.c: For destination '192.168.10.75', our source address is '192.168.10.75'. [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:25060 [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: SIP call-id changed from '4dd918936296624743905c163429e5cc@192.168.10.75' to '21bcfc8b3931cfce66d332554203c767@192.168.10.75' [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: *** Our native formats are (alaw) [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: *** Joint capabilities are (alaw) [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: *** Our capabilities are (ulaw|alaw) [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: This channel will not be able to handle video. [Dec 17 11:38:24] DEBUG[3345][C-00000000] channel_internal_api.c: Channel Call ID changing from [C-00000000] to [C-00000000] [Dec 17 11:38:24] DEBUG[3345][C-00000000] channel.c: Inheriting variable PICKUPMARK from SIP/phone-a-00000000 to SIP/phone-b-00000001. [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Outgoing Call for phone-b [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Updating call counter for outgoing call [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Call to peer 'phone-b' is 1 out of 2147483647 [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: ** Our capability: (ulaw|alaw) Video flag: False Text flag: False [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: ** Our prefcodec: (alaw) [Dec 17 11:38:24] VERBOSE[3345][C-00000000] chan_sip.c: Audio is at 18368 [Dec 17 11:38:24] VERBOSE[3345][C-00000000] chan_sip.c: Adding codec 100004 (alaw) to SDP [Dec 17 11:38:24] VERBOSE[3345][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Dec 17 11:38:24] VERBOSE[3345][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw) [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 0 [ 40]: INVITE sip:phone-b@192.168.10.75 SIP/2.0 [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK38ab9f12 [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 3 [ 59]: From: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 4 [ 31]: To: [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 5 [ 38]: Contact: [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 6 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 8 [ 45]: User-Agent: IPTAM PBX (Version 20141216/6814) [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 9 [ 35]: Date: Wed, 17 Dec 2014 10:38:24 GMT [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 12 [ 53]: P-Asserted-Identity: "PhoneA" [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Dec 17 11:38:24] VERBOSE[3345][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.75:5060: INVITE sip:phone-b@192.168.10.75 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK38ab9f12 Max-Forwards: 70 From: "PhoneA" ;tag=as7a4ce21e To: Contact: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 102 INVITE User-Agent: IPTAM PBX (Version 20141216/6814) Date: Wed, 17 Dec 2014 10:38:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: "PhoneA" Content-Type: application/sdp Content-Length: 262 v=0 o=root 387003927 387003927 IN IP4 192.168.10.75 s=Asterisk PBX 11.15.0-1 c=IN IP4 192.168.10.75 t=0 0 m=audio 18368 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #32 [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone-b-00000001 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: PhoneB AccountCode: Exten: Context: Standard Uniqueid: 1418812704.1 [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone-a-00000000 Destination: SIP/phone-b-00000001 CallerIDNum: 180 CallerIDName: PhoneA ConnectedLineNum: ConnectedLineName: UniqueID: 1418812704.0 DestUniqueID: 1418812704.1 Dialstring: phone-b [Dec 17 11:38:24] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:24] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:24] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 6 (Ringing) [Dec 17 11:38:24] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '6' [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: APP_START AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: 180 CallerIDrdnis: CallerIDdnid: 100 Exten: 100 Context: sub-dial-intern Channel: SIP/phone-a-00000000 Application: Dial AppData: SIP/phone-b, EventTime: 2014-12-17 11:38:24 AMAFlags: DOCUMENTATION UniqueID: 1418812704.0 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: s Context: Standard Channel: SIP/phone-b-00000001 Application: AppData: EventTime: 2014-12-17 11:38:24 AMAFlags: DOCUMENTATION UniqueID: 1418812704.1 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:24] DEBUG[3253] app_queue.c: Extension '100@_extensions' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Dec 17 11:38:24] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: _extensions Hint: SIP/phone-b Status: 8 [Dec 17 11:38:24] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK38ab9f12 From: "PhoneA" ;tag=as7a4ce21e To: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 102 INVITE Server: OpenSIPS (1.11.2-notls (i386/linux)) Content-Length: 0 <-------------> [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK38ab9f12 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 2 [ 59]: From: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 3 [ 31]: To: [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 4 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 6 [ 44]: Server: OpenSIPS (1.11.2-notls (i386/linux)) [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 17 11:38:24] VERBOSE[3289] chan_sip.c: --- (8 headers 0 lines) --- [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: = Looking for Call ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 (Checking To) --From tag as7a4ce21e --To-tag [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: *** SIP TIMER: Cancelling retransmission #32 - INVITE (got response) [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '21bcfc8b3931cfce66d332554203c767@192.168.10.75' Request 102: Found [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: SIP response 100 to standard invite [Dec 17 11:38:24] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK38ab9f12 From: "PhoneA" ;tag=as7a4ce21e To: ;tag=7C2F8008F5C1_T1926987239 Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 Contact: CSeq: 102 INVITE User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO Allow-Events: talk, hold Content-Length: 0 <-------------> [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK38ab9f12 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 2 [ 59]: From: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 3 [ 60]: To: ;tag=7C2F8008F5C1_T1926987239 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 4 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 5 [ 42]: Contact: [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 8 [ 85]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 9 [ 24]: Allow-Events: talk, hold [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Dec 17 11:38:24] VERBOSE[3289] chan_sip.c: --- (11 headers 0 lines) --- [Dec 17 11:38:24] DEBUG[3289] chan_sip.c: = Looking for Call ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 (Checking To) --From tag as7a4ce21e --To-tag 7C2F8008F5C1_T1926987239 [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '21bcfc8b3931cfce66d332554203c767@192.168.10.75' Request 102: Found [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: SIP response 180 to standard invite [Dec 17 11:38:24] DEBUG[3289][C-00000000] chan_sip.c: build_route: Contact hop: [Dec 17 11:38:24] VERBOSE[3289][C-00000000] chan_sip.c: list_route: hop: [Dec 17 11:38:24] DEBUG[3345][C-00000000] rtp_engine.c: Setting early bridge SDP of 'SIP/phone-a-00000000' with that of 'SIP/phone-b-00000001' [Dec 17 11:38:24] VERBOSE[3345][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b From: "PhoneA" ;tag=80931a0487 To: ;tag=as07820be1 Call-ID: 0dcce801bbc3a2c1 CSeq: 8089 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: P-Asserted-Identity: "PhoneB" Content-Length: 0 <------------> [Dec 17 11:38:24] DEBUG[3345][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:24] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:24] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:24] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 6 (Ringing) [Dec 17 11:38:24] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '6' [Dec 17 11:38:24] DEBUG[3296] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone-b-00000001 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 100 CallerIDName: PhoneB ConnectedLineNum: 180 ConnectedLineName: PhoneA Uniqueid: 1418812704.1 [Dec 17 11:38:24] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Dec 17 11:38:29] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK38ab9f12 From: "PhoneA" ;tag=as7a4ce21e To: ;tag=7C2F8008F5C1_T1926987239 Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 Contact: CSeq: 102 INVITE User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO Supported: 100rel,timer,replaces Content-Type: application/sdp Content-Length: 241 v=0 o=- 123226729 123226729 IN IP4 192.168.10.124 s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1 c=IN IP4 192.168.10.124 t=0 0 m=audio 5004 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK38ab9f12 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 2 [ 59]: From: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 3 [ 60]: To: ;tag=7C2F8008F5C1_T1926987239 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 4 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 5 [ 42]: Contact: [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 8 [ 85]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 9 [ 32]: Supported: 100rel,timer,replaces [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 11 [ 19]: Content-Length: 241 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 12 [ 0]: [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 0 [ 3]: v=0 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 1 [ 45]: o=- 123226729 123226729 IN IP4 192.168.10.124 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 2 [ 39]: s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.124 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 5 [ 26]: m=audio 5004 RTP/AVP 8 101 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 6 [ 22]: a=rtpmap:8 PCMA/8000/1 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Body 9 [ 10]: a=sendrecv [Dec 17 11:38:29] VERBOSE[3289] chan_sip.c: --- (12 headers 10 lines) --- [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: = Looking for Call ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 (Checking To) --From tag as7a4ce21e --To-tag 7C2F8008F5C1_T1926987239 [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Acked pending invite 102 [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Stopping retransmission on '21bcfc8b3931cfce66d332554203c767@192.168.10.75' of Request 102: Match Found [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: SIP response 200 to standard invite [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP o=- 123226729 123226729 IN IP4 192.168.10.124... OK. [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1... UNSUPPORTED OR FAILED. [Dec 17 11:38:29] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.124' into... [Dec 17 11:38:29] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.124' and port ''. [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.124... OK. [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 8 [Dec 17 11:38:29] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0xb5ba82f0 [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 101 [Dec 17 11:38:29] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 101 based on m type on 0xb5ba82f0 [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK. [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 17 11:38:29] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97662e4' [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: Peer audio RTP is at port 192.168.10.124:5004 [Dec 17 11:38:29] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 8 from 0xb5ba82f0 to 0x9766490 [Dec 17 11:38:29] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 101 from 0xb5ba82f0 to 0x9766490 [Dec 17 11:38:29] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x97662e4' [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Updating call counter for outgoing call [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: build_route: Contact hop: [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: list_route: hop: [Dec 17 11:38:29] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.124:5060' into... [Dec 17 11:38:29] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.124' and port '5060'. [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Strict routing enforced for session 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:29] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.124:5060' into... [Dec 17 11:38:29] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.124' and port '5060'. [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: set_destination: set destination to 192.168.10.124:5060 [Dec 17 11:38:29] VERBOSE[3289][C-00000000] chan_sip.c: Transmitting (no NAT) to 192.168.10.124:5060: ACK sip:phone-b@192.168.10.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK476bbde6 Max-Forwards: 70 From: "PhoneA" ;tag=as7a4ce21e To: ;tag=7C2F8008F5C1_T1926987239 Contact: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 102 ACK User-Agent: IPTAM PBX (Version 20141216/6814) Content-Length: 0 --- [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.124:5060 [Dec 17 11:38:29] DEBUG[3345][C-00000000] rtp_engine.c: Setting early bridge SDP of 'SIP/phone-a-00000000' with that of 'SIP/phone-b-00000001' [Dec 17 11:38:29] DEBUG[3345][C-00000000] chan_sip.c: SIP answering channel: SIP/phone-a-00000000 [Dec 17 11:38:29] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 17 11:38:29] DEBUG[3345][C-00000000] chan_sip.c: Setting framing from config on incoming call [Dec 17 11:38:29] DEBUG[3345][C-00000000] chan_sip.c: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True [Dec 17 11:38:29] DEBUG[3345][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [Dec 17 11:38:29] VERBOSE[3345][C-00000000] chan_sip.c: Audio is at 19658 [Dec 17 11:38:29] VERBOSE[3345][C-00000000] chan_sip.c: Adding codec 100004 (alaw) to SDP [Dec 17 11:38:29] VERBOSE[3345][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Dec 17 11:38:29] VERBOSE[3345][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 17 11:38:29] DEBUG[3345][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Dec 17 11:38:29] DEBUG[3345][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw) [Dec 17 11:38:29] VERBOSE[3345][C-00000000] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK261de477b28ac618b.0f67a77cb65f9d87b From: "PhoneA" ;tag=80931a0487 To: ;tag=as07820be1 Call-ID: 0dcce801bbc3a2c1 CSeq: 8089 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: P-Asserted-Identity: "PhoneB" Content-Type: application/sdp Content-Length: 264 v=0 o=root 2046840620 2046840620 IN IP4 192.168.10.75 s=Asterisk PBX 11.15.0-1 c=IN IP4 192.168.10.75 t=0 0 m=audio 19658 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Dec 17 11:38:29] DEBUG[3345][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #35 [Dec 17 11:38:29] DEBUG[3345][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:29] DEBUG[3345][C-00000000] features.c: bridge answer set, chan answer set [Dec 17 11:38:29] DEBUG[3345][C-00000000] features.c: Removing dialed interfaces datastore on SIP/phone-b-00000001 since we're bridging [Dec 17 11:38:29] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 17 11:38:29] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 17 11:38:29] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:29] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:29] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 2 (In use) [Dec 17 11:38:29] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '2' [Dec 17 11:38:29] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:29] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:29] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 2 (In use) [Dec 17 11:38:29] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '2' [Dec 17 11:38:29] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-a [Dec 17 11:38:29] DEBUG[3251] chan_sip.c: Checking device state for peer phone-a [Dec 17 11:38:29] DEBUG[3251] devicestate.c: Changing state for SIP/phone-a - state 2 (In use) [Dec 17 11:38:29] DEBUG[3251] devicestate.c: device 'SIP/phone-a' state '2' [Dec 17 11:38:29] DEBUG[3296] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone-b-00000001 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 100 CallerIDName: PhoneB ConnectedLineNum: 180 ConnectedLineName: PhoneA Uniqueid: 1418812704.1 [Dec 17 11:38:29] DEBUG[3296] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone-a-00000000 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 180 CallerIDName: PhoneA ConnectedLineNum: 100 ConnectedLineName: PhoneB Uniqueid: 1418812704.0 [Dec 17 11:38:29] DEBUG[3296] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone-b-00000001 Uniqueid: 1418812704.1 AccountCode: OldAccountCode: [Dec 17 11:38:29] DEBUG[3296] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone-a-00000000 Channel2: SIP/phone-b-00000001 Uniqueid1: 1418812704.0 Uniqueid2: 1418812704.1 CallerID1: 180 CallerID2: 100 [Dec 17 11:38:29] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: ANSWER AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: Exten: 100 Context: Standard Channel: SIP/phone-b-00000001 Application: AppDial AppData: (Outgoing Line) EventTime: 2014-12-17 11:38:29 AMAFlags: DOCUMENTATION UniqueID: 1418812704.1 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:29] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: ANSWER AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: 180 CallerIDrdnis: CallerIDdnid: 100 Exten: 100 Context: sub-dial-intern Channel: SIP/phone-a-00000000 Application: Dial AppData: SIP/phone-b, EventTime: 2014-12-17 11:38:29 AMAFlags: DOCUMENTATION UniqueID: 1418812704.0 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:29] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: BRIDGE_START AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: 180 CallerIDrdnis: CallerIDdnid: 100 Exten: 100 Context: sub-dial-intern Channel: SIP/phone-a-00000000 Application: Dial AppData: SIP/phone-b, EventTime: 2014-12-17 11:38:29 AMAFlags: DOCUMENTATION UniqueID: 1418812704.0 LinkedID: 1418812704.0 Userfield: Peer: SIP/phone-b-00000001 PeerAccount: Extra: [Dec 17 11:38:29] DEBUG[3253] app_queue.c: Extension '100@_extensions' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:29] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:29] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:29] DEBUG[3295] app_queue.c: Device 'SIP/phone-a' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:29] DEBUG[3296] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: _extensions Hint: SIP/phone-b Status: 1 [Dec 17 11:38:29] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> ACK sip:100@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK4fe99baef64770cec Max-Forwards: 70 From: "PhoneA" ;tag=80931a0487 To: ;tag=as07820be1 Call-ID: 0dcce801bbc3a2c1 CSeq: 8089 ACK Authorization: Digest username="phone-a",realm="pbx5",nonce="54915d3e000000362d585188f75662b74d8994e612eddae8",uri="sip:100@192.168.10.75:5060;user=phone",response="be6fffee63a192be2cee242749a7c2dd",qop=auth,cnonce="9256a1ce",nc=00000001 User-Agent: Aastra 55i/3.3.1.2217 Content-Length: 0 <-------------> [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 0 [ 39]: ACK sip:100@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK4fe99baef64770cec [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 3 [ 62]: From: "PhoneA" ;tag=80931a0487 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 4 [ 58]: To: ;tag=as07820be1 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 5 [ 25]: Call-ID: 0dcce801bbc3a2c1 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 6 [ 14]: CSeq: 8089 ACK [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 7 [237]: Authorization: Digest username="phone-a",realm="pbx5",nonce="54915d3e000000362d585188f75662b74d8994e612eddae8",uri="sip:100@192.168.10.75:5060;user=phone",response="be6fffee63a192be2cee242749a7c2dd",qop=auth,cnonce="9256a1ce",nc=00000001 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 8 [ 33]: User-Agent: Aastra 55i/3.3.1.2217 [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 17 11:38:29] VERBOSE[3289] chan_sip.c: --- (10 headers 0 lines) --- [Dec 17 11:38:29] DEBUG[3289] chan_sip.c: = Looking for Call ID: 0dcce801bbc3a2c1 (Checking From) --From tag 80931a0487 --To-tag as07820be1 [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #35 [Dec 17 11:38:29] DEBUG[3289][C-00000000] chan_sip.c: Stopping retransmission on '0dcce801bbc3a2c1' of Response 8089: Match Found [Dec 17 11:38:29] DEBUG[3345][C-00000000] res_rtp_asterisk.c: 0x975eec0 -- Probation learning mode pass with source address 192.168.10.201:3000 [Dec 17 11:38:29] DEBUG[3345][C-00000000] res_rtp_asterisk.c: 0x9749e80 -- Probation learning mode pass with source address 192.168.10.124:5004 [Dec 17 11:38:30] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Got RTCP report of 23 bytes [Dec 17 11:38:30] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> INVITE sip:180@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T198ACAC4 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T198ACAC4;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 1 INVITE Contact: Max-Forwards: 69 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO Supported: 100rel,timer,replaces User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Content-Type: application/sdp Content-Length: 234 v=0 o=- 123226729 123226730 IN IP4 192.168.10.124 s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1 c=IN IP4 0.0.0.0 t=0 0 m=audio 5004 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendonly <-------------> [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 0 [ 42]: INVITE sip:180@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T198ACAC4 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T198ACAC4;rport=5060 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 3 [ 62]: From: ;tag=7C2F8008F5C1_T1926987239 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 4 [ 57]: To: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 5 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 6 [ 14]: CSeq: 1 INVITE [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 7 [ 42]: Contact: [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 8 [ 16]: Max-Forwards: 69 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 9 [ 85]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 10 [ 32]: Supported: 100rel,timer,replaces [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 11 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 13 [ 19]: Content-Length: 234 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Header 14 [ 0]: [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 0 [ 3]: v=0 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 1 [ 45]: o=- 123226729 123226730 IN IP4 192.168.10.124 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 2 [ 39]: s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 3 [ 16]: c=IN IP4 0.0.0.0 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 5 [ 26]: m=audio 5004 RTP/AVP 8 101 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 6 [ 22]: a=rtpmap:8 PCMA/8000/1 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: Body 9 [ 10]: a=sendonly [Dec 17 11:38:30] VERBOSE[3289] chan_sip.c: --- (14 headers 10 lines) --- [Dec 17 11:38:30] DEBUG[3289] chan_sip.c: = Looking for Call ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 (Checking From) --From tag 7C2F8008F5C1_T1926987239 --To-tag as7a4ce21e [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Dec 17 11:38:30] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,timer,replaces" [Dec 17 11:38:30] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Found SIP option: -100rel- [Dec 17 11:38:30] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Matched SIP option: 100rel [Dec 17 11:38:30] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Found SIP option: -timer- [Dec 17 11:38:30] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Matched SIP option: timer [Dec 17 11:38:30] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Found SIP option: -replaces- [Dec 17 11:38:30] DEBUG[3289][C-00000000] sip/reqresp_parser.c: Matched SIP option: replaces [Dec 17 11:38:30] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:30] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Sending to 192.168.10.124:5060 (no NAT) [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP o=- 123226729 123226730 IN IP4 192.168.10.124... OK. [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1... UNSUPPORTED OR FAILED. [Dec 17 11:38:30] DEBUG[3289][C-00000000] netsock2.c: Splitting '0.0.0.0' into... [Dec 17 11:38:30] DEBUG[3289][C-00000000] netsock2.c: ...host '0.0.0.0' and port ''. [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 0.0.0.0... OK. [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 8 [Dec 17 11:38:30] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0xb5ba9240 [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 101 [Dec 17 11:38:30] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 101 based on m type on 0xb5ba9240 [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK. [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 17 11:38:30] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97662e4' [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Peer audio RTP is at port 0.0.0.0:5004 [Dec 17 11:38:30] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 8 from 0xb5ba9240 to 0x9766490 [Dec 17 11:38:30] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 101 from 0xb5ba9240 to 0x9766490 [Dec 17 11:38:30] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x97662e4' [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: We're settling with these formats: (alaw) [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Dec 17 11:38:30] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97662e4' [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Got a SIP re-invite for call 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Incoming INVITE with 'timer' option supported [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: SIP/phone-b-00000001: This call is UP.... [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T198ACAC4;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T198ACAC4;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 1 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Setting framing from config on incoming call [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: ** Our prefcodec: (alaw) [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Audio is at 18368 [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Adding codec 100004 (alaw) to SDP [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Dec 17 11:38:30] VERBOSE[3289][C-00000000] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T198ACAC4;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T198ACAC4;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 1 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 387003927 387003928 IN IP4 192.168.10.75 s=Asterisk PBX 11.15.0-1 c=IN IP4 192.168.10.75 t=0 0 m=audio 18368 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly <------------> [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #36 [Dec 17 11:38:30] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:30] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 17 11:38:30] DEBUG[3345][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Dec 17 11:38:30] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 17 11:38:30] DEBUG[3296] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/phone-a-00000000 UniqueID: 1418812704.0 Class: default [Dec 17 11:38:30] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:30] DEBUG[3345][C-00000000] channel.c: Set channel SIP/phone-a-00000000 to write format slin [Dec 17 11:38:30] DEBUG[3345][C-00000000] res_musiconhold.c: SIP/phone-a-00000000 Opened file 0 '/var/lib/asterisk/moh/iptam/iptam_moh' [Dec 17 11:38:30] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Dec 17 11:38:30] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Dec 17 11:38:30] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:30] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> ACK sip:180@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T253C6AB0 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T253C6AB0;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 1 ACK User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Contact: Max-Forwards: 69 Content-Length: 0 <-------------> [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 0 [ 39]: ACK sip:180@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T253C6AB0 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T253C6AB0;rport=5060 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 3 [ 62]: From: ;tag=7C2F8008F5C1_T1926987239 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 4 [ 57]: To: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 5 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 6 [ 11]: CSeq: 1 ACK [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 8 [ 42]: Contact: [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 9 [ 16]: Max-Forwards: 69 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Dec 17 11:38:31] VERBOSE[3289] chan_sip.c: --- (11 headers 0 lines) --- [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: = Looking for Call ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 (Checking From) --From tag 7C2F8008F5C1_T1926987239 --To-tag as7a4ce21e [Dec 17 11:38:31] DEBUG[3289][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 17 11:38:31] DEBUG[3289][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #36 [Dec 17 11:38:31] DEBUG[3289][C-00000000] chan_sip.c: Stopping retransmission on '21bcfc8b3931cfce66d332554203c767@192.168.10.75' of Response 1: Match Found [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> INVITE sip:200@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T30D180B0 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;rport=5060 Session-Expires: 1800 From: " PhoneB" ;tag=7C2F8008F5C1_T1957041132 To: Call-ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 CSeq: 2 INVITE Contact: Max-Forwards: 69 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO Supported: 100rel,timer,replaces User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Content-Type: application/sdp Content-Length: 241 v=0 o=- 123302710 123302710 IN IP4 192.168.10.124 s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1 c=IN IP4 192.168.10.124 t=0 0 m=audio 5004 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 0 [ 42]: INVITE sip:200@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T30D180B0 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;rport=5060 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 3 [ 21]: Session-Expires: 1800 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 4 [ 72]: From: " PhoneB" ;tag=7C2F8008F5C1_T1957041132 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 5 [ 27]: To: [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 6 [ 57]: Call-ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 7 [ 14]: CSeq: 2 INVITE [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 8 [ 42]: Contact: [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 9 [ 16]: Max-Forwards: 69 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 10 [ 85]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 11 [ 32]: Supported: 100rel,timer,replaces [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 12 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 14 [ 19]: Content-Length: 241 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Header 15 [ 0]: [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 0 [ 3]: v=0 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 1 [ 45]: o=- 123302710 123302710 IN IP4 192.168.10.124 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 2 [ 39]: s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.124 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 5 [ 26]: m=audio 5004 RTP/AVP 8 101 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 6 [ 22]: a=rtpmap:8 PCMA/8000/1 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Body 9 [ 10]: a=sendrecv [Dec 17 11:38:31] VERBOSE[3289] chan_sip.c: --- (15 headers 10 lines) --- [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 (Checking From) --From tag 7C2F8008F5C1_T1957041132 --To-tag [Dec 17 11:38:31] DEBUG[3289] acl.c: For destination '192.168.10.75', our source address is '192.168.10.75'. [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:25060 [Dec 17 11:38:31] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:31] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:31] VERBOSE[3289] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:31] DEBUG[3289] chan_sip.c: Allocating new SIP dialog for CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 - INVITE (No RTP) [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Dec 17 11:38:31] DEBUG[3289][C-00000001] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,timer,replaces" [Dec 17 11:38:31] DEBUG[3289][C-00000001] sip/reqresp_parser.c: Found SIP option: -100rel- [Dec 17 11:38:31] DEBUG[3289][C-00000001] sip/reqresp_parser.c: Matched SIP option: 100rel [Dec 17 11:38:31] DEBUG[3289][C-00000001] sip/reqresp_parser.c: Found SIP option: -timer- [Dec 17 11:38:31] DEBUG[3289][C-00000001] sip/reqresp_parser.c: Matched SIP option: timer [Dec 17 11:38:31] DEBUG[3289][C-00000001] sip/reqresp_parser.c: Found SIP option: -replaces- [Dec 17 11:38:31] DEBUG[3289][C-00000001] sip/reqresp_parser.c: Matched SIP option: replaces [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Using INVITE request as basis request - CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: Splitting '192.168.10.75' into... [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Found peer 'phone-b' for 'phone-b' from 192.168.10.75:5060 [Dec 17 11:38:31] DEBUG[3289][C-00000001] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x97932dc' [Dec 17 11:38:31] DEBUG[3289][C-00000001] res_rtp_asterisk.c: Allocated port 14520 for RTP instance '0x97932dc' [Dec 17 11:38:31] DEBUG[3289][C-00000001] rtp_engine.c: RTP instance '0x97932dc' is setup and ready to go [Dec 17 11:38:31] DEBUG[3289][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x97932dc' [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Processing session-level SDP o=- 123302710 123302710 IN IP4 192.168.10.124... OK. [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Processing session-level SDP s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1... UNSUPPORTED OR FAILED. [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: Splitting '192.168.10.124' into... [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: ...host '192.168.10.124' and port ''. [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.124... OK. [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Found RTP audio format 8 [Dec 17 11:38:31] DEBUG[3289][C-00000001] rtp_engine.c: Setting payload 8 based on m type on 0xb5ba9240 [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Found RTP audio format 101 [Dec 17 11:38:31] DEBUG[3289][C-00000001] rtp_engine.c: Setting payload 101 based on m type on 0xb5ba9240 [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8 [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK. [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 17 11:38:31] DEBUG[3289][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97932dc' [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Peer audio RTP is at port 192.168.10.124:5004 [Dec 17 11:38:31] DEBUG[3289][C-00000001] rtp_engine.c: Copying payload 8 from 0xb5ba9240 to 0x9793488 [Dec 17 11:38:31] DEBUG[3289][C-00000001] rtp_engine.c: Copying payload 101 from 0xb5ba9240 to 0x9793488 [Dec 17 11:38:31] DEBUG[3289][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x97932dc' [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: We're settling with these formats: (alaw) [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Checking SIP call limits for device phone-b [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Updating call counter for incoming call [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Call from peer 'phone-b' is 2 out of 2147483647 [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: Splitting '192.168.10.75:25060' into... [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: Splitting '192.168.10.75' into... [Dec 17 11:38:31] DEBUG[3289][C-00000001] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: Looking for 200 in Standard (domain 192.168.10.75) [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Incoming INVITE with 'timer' option supported [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: INVITE also has "Session-Expires" header. [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Session-Expires: 1800 [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: *** Our native formats are (alaw) [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: *** Joint capabilities are (alaw) [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: *** Our capabilities are (ulaw|alaw) [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: This channel will not be able to handle video. [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: build_route: Contact hop: [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: list_route: hop: [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: SIP/phone-b-00000002: New call is still down.... Trying... [Dec 17 11:38:31] VERBOSE[3289][C-00000001] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T1957041132 To: Call-ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 CSeq: 2 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Dec 17 11:38:31] DEBUG[3289][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:31] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:31] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:31] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 2 (In use) [Dec 17 11:38:31] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '2' [Dec 17 11:38:31] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:31] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:31] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 2 (In use) [Dec 17 11:38:31] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '2' [Dec 17 11:38:31] DEBUG[3296] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone-b-00000002 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: PhoneB AccountCode: Exten: 200 Context: Standard Uniqueid: 1418812711.2 [Dec 17 11:38:31] DEBUG[3296] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone-b-00000002 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: PhoneB ConnectedLineNum: ConnectedLineName: Uniqueid: 1418812711.2 [Dec 17 11:38:31] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: 200 Context: Standard Channel: SIP/phone-b-00000002 Application: AppData: EventTime: 2014-12-17 11:38:31 AMAFlags: DOCUMENTATION UniqueID: 1418812711.2 LinkedID: 1418812711.2 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Launching 'Gosub' [Dec 17 11:38:31] DEBUG[3348][C-00000001] app_stack.c: Channel SIP/phone-b-00000002 has no datastore, so we're allocating one. [Dec 17 11:38:31] DEBUG[3348][C-00000001] app_stack.c: Setting 'ARG1' to '200' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Function CDR(userfield) result is '(null)' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'ARG1' is '200' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Launching 'Set' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Function SIP_HEADER(User-Agent) result is 'DE700 IP PRO/61.02.00.15;7C2F8008F5C1' [Dec 17 11:38:31] DEBUG[3348][C-00000001] func_strings.c: FUNCTION REGEX (^BSIP )(DE700 IP PRO/61.02.00.15;7C2F8008F5C1) [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Function REGEX("^BSIP " DE700 IP PRO/61.02.00.15;7C2F8008F5C1) result is '0' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Launching 'Set' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'Originated' is NULL [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Expression result is '1' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Launching 'GotoIf' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Launching 'Set' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Launching 'AGI' [Dec 17 11:38:31] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:31] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'ARG1' is '200' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'OptionCD' is '1' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'OptionCFNOCIDfirst' is '0' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'TrunkSelection' is '0' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'SIPTrunkSelection' is '' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'HopCountMax' is '7' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'CAUSE' is '' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'XMPPactive' is '0' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'DOOR_AGENT' is '0' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'PubVM' is '0' [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'CFFB_ACTIVE' is NULL [Dec 17 11:38:31] DEBUG[3348][C-00000001] pbx.c: Result of 'FALLBACK' is NULL [Dec 17 11:38:31] DEBUG[3348][C-00000001] db.c: Unable to find key '200' in family 'LANGUAGE' [Dec 17 11:38:31] DEBUG[3348][C-00000001] db.c: Unable to find key '200' in family 'CFFB' [Dec 17 11:38:31] DEBUG[3348][C-00000001] db.c: Unable to find key '200' in family 'CFIM' [Dec 17 11:38:31] DEBUG[3348][C-00000001] db.c: Unable to find key '200' in family 'DND' [Dec 17 11:38:31] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3348][C-00000001] db.c: Unable to find key '200' in family 'PARALLEL' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'SIPREFERREDBYHDR' is NULL [Dec 17 11:38:32] DEBUG[3348][C-00000001] db.c: Unable to find key '200' in family 'CFUN' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'SourceIsExt' is NULL [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'USE_CALLBACK_NR' is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDEXT' is '200' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Function DB(LIMIT/200) result is '1' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'Set' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'limit' is '1' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '1' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'limit' is '1' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'GotoIf' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Not taking any branch [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Function CALLERID(num) result is '100' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDEXT' is '200' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'GotoIf' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Not taking any branch [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-c' [Dec 17 11:38:32] DEBUG[3348][C-00000001] devicestate.c: No provider found, checking channel drivers for SIP - phone-c [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Checking device state for peer phone-c [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Function DEVICE_STATE(SIP/phone-c) result is 'NOT_INUSE' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-c' [Dec 17 11:38:32] DEBUG[3348][C-00000001] devicestate.c: No provider found, checking channel drivers for SIP - phone-c [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Checking device state for peer phone-c [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Function DEVICE_STATE(SIP/phone-c) result is 'NOT_INUSE' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-c' [Dec 17 11:38:32] DEBUG[3348][C-00000001] devicestate.c: No provider found, checking channel drivers for SIP - phone-c [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Checking device state for peer phone-c [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Function DEVICE_STATE(SIP/phone-c) result is 'NOT_INUSE' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-c' [Dec 17 11:38:32] DEBUG[3348][C-00000001] devicestate.c: No provider found, checking channel drivers for SIP - phone-c [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Checking device state for peer phone-c [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Function DEVICE_STATE(SIP/phone-c) result is 'NOT_INUSE' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'GotoIf' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Not taking any branch [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDEXT' is '200' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'CHANNEL' is 'SIP/phone-b-00000002' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'UserEvent' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDEXT' is '200' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'Set' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDEXT' is '200' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'Goto' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'ActiveAmt' is NULL [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '1' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'GotoIf' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIALEDACC' is 'SIP/phone-c' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'AGI' [Dec 17 11:38:32] DEBUG[3296] manager.c: Examining event: Event: UserEvent Privilege: user,all UserEvent: CalledExt Uniqueid: 1418812711.2 Dst: 200 Channel: SIP/phone-b-00000002 [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DISTINCTIVE_RINGING' is NULL [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'ICOM_ACTIVE' is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'Set' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'UseT2M' is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Expression result is '0' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'ExecIf' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Result of 'DIAL_STRING' is 'SIP/phone-c,' [Dec 17 11:38:32] DEBUG[3348][C-00000001] pbx.c: Launching 'Dial' [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Allocating new SIP dialog for 1a67ee856377df3a3c84f83b7e331381@192.168.10.75 - INVITE (No RTP) [Dec 17 11:38:32] DEBUG[3348][C-00000001] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x97aa8dc' [Dec 17 11:38:32] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Allocated port 10922 for RTP instance '0x97aa8dc' [Dec 17 11:38:32] DEBUG[3348][C-00000001] rtp_engine.c: RTP instance '0x97aa8dc' is setup and ready to go [Dec 17 11:38:32] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x97aa8dc' [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 17 11:38:32] DEBUG[3348][C-00000001] acl.c: For destination '192.168.10.75', our source address is '192.168.10.75'. [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:25060 [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: SIP call-id changed from '1a67ee856377df3a3c84f83b7e331381@192.168.10.75' to '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: *** Our native formats are (alaw) [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: *** Joint capabilities are (alaw) [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: *** Our capabilities are (ulaw|alaw) [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: This channel will not be able to handle video. [Dec 17 11:38:32] DEBUG[3348][C-00000001] channel_internal_api.c: Channel Call ID changing from [C-00000001] to [C-00000001] [Dec 17 11:38:32] DEBUG[3348][C-00000001] channel.c: Inheriting variable PICKUPMARK from SIP/phone-b-00000002 to SIP/phone-c-00000003. [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Outgoing Call for phone-c [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Updating call counter for outgoing call [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Call to peer 'phone-c' is 1 out of 2147483647 [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: ** Our capability: (ulaw|alaw) Video flag: False Text flag: False [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: ** Our prefcodec: (alaw) [Dec 17 11:38:32] VERBOSE[3348][C-00000001] chan_sip.c: Audio is at 10922 [Dec 17 11:38:32] VERBOSE[3348][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP [Dec 17 11:38:32] VERBOSE[3348][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Dec 17 11:38:32] VERBOSE[3348][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw) [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 0 [ 40]: INVITE sip:phone-c@192.168.10.75 SIP/2.0 [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 3 [ 59]: From: "PhoneB" ;tag=as2623879c [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 4 [ 31]: To: [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 5 [ 38]: Contact: [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 6 [ 55]: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 8 [ 45]: User-Agent: IPTAM PBX (Version 20141216/6814) [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 9 [ 35]: Date: Wed, 17 Dec 2014 10:38:32 GMT [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 12 [ 53]: P-Asserted-Identity: "PhoneB" [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Dec 17 11:38:32] VERBOSE[3348][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.75:5060: INVITE sip:phone-c@192.168.10.75 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 Max-Forwards: 70 From: "PhoneB" ;tag=as2623879c To: Contact: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 CSeq: 102 INVITE User-Agent: IPTAM PBX (Version 20141216/6814) Date: Wed, 17 Dec 2014 10:38:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: "PhoneB" Content-Type: application/sdp Content-Length: 264 v=0 o=root 1125161233 1125161233 IN IP4 192.168.10.75 s=Asterisk PBX 11.15.0-1 c=IN IP4 192.168.10.75 t=0 0 m=audio 10922 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #39 [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:32] DEBUG[3296] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone-c-00000003 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 200 CallerIDName: PhoneC AccountCode: Exten: Context: Standard Uniqueid: 1418812712.3 [Dec 17 11:38:32] DEBUG[3296] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone-b-00000002 Destination: SIP/phone-c-00000003 CallerIDNum: 100 CallerIDName: PhoneB ConnectedLineNum: ConnectedLineName: UniqueID: 1418812711.2 DestUniqueID: 1418812712.3 Dialstring: phone-c [Dec 17 11:38:32] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-c [Dec 17 11:38:32] DEBUG[3251] chan_sip.c: Checking device state for peer phone-c [Dec 17 11:38:32] DEBUG[3251] devicestate.c: Changing state for SIP/phone-c - state 6 (Ringing) [Dec 17 11:38:32] DEBUG[3251] devicestate.c: device 'SIP/phone-c' state '6' [Dec 17 11:38:32] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: APP_START AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: 200 Exten: 200 Context: sub-dial-intern Channel: SIP/phone-b-00000002 Application: Dial AppData: SIP/phone-c, EventTime: 2014-12-17 11:38:32 AMAFlags: DOCUMENTATION UniqueID: 1418812711.2 LinkedID: 1418812711.2 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:32] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 200 CallerIDname: PhoneC CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: s Context: Standard Channel: SIP/phone-c-00000003 Application: AppData: EventTime: 2014-12-17 11:38:32 AMAFlags: DOCUMENTATION UniqueID: 1418812712.3 LinkedID: 1418812711.2 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:32] DEBUG[3253] app_queue.c: Extension '200@_extensions' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Dec 17 11:38:32] DEBUG[3253] chan_sip.c: Strict routing enforced for session CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 [Dec 17 11:38:32] VERBOSE[3253] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:32] DEBUG[3253] netsock2.c: Splitting '192.168.10.124:5060' into... [Dec 17 11:38:32] DEBUG[3253] netsock2.c: ...host '192.168.10.124' and port '5060'. [Dec 17 11:38:32] VERBOSE[3253] chan_sip.c: set_destination: set destination to 192.168.10.124:5060 [Dec 17 11:38:32] VERBOSE[3253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.124:5060: NOTIFY sip:phone-b@192.168.10.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK167f66d8 Max-Forwards: 70 From: ;tag=as08125298 To: ;tag=7C2F8008F5C1_T461496925;user=phone Contact: Call-ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 CSeq: 103 NOTIFY User-Agent: IPTAM PBX (Version 20141216/6814) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 573 sip:200@192.168.10.75 sip:200@192.168.10.75 early --- [Dec 17 11:38:32] DEBUG[3253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #41 [Dec 17 11:38:32] DEBUG[3253] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.124:5060 [Dec 17 11:38:32] DEBUG[3295] app_queue.c: Device 'SIP/phone-c' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Dec 17 11:38:32] DEBUG[3296] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 200 Context: _extensions Hint: SIP/phone-c Status: 8 [Dec 17 11:38:32] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 From: "PhoneB" ;tag=as2623879c To: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 CSeq: 102 INVITE Server: OpenSIPS (1.11.2-notls (i386/linux)) Content-Length: 0 <-------------> [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 2 [ 59]: From: "PhoneB" ;tag=as2623879c [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 3 [ 31]: To: [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 4 [ 55]: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 6 [ 44]: Server: OpenSIPS (1.11.2-notls (i386/linux)) [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 17 11:38:32] VERBOSE[3289] chan_sip.c: --- (8 headers 0 lines) --- [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: = Looking for Call ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 (Checking To) --From tag as2623879c --To-tag [Dec 17 11:38:32] DEBUG[3289][C-00000001] chan_sip.c: *** SIP TIMER: Cancelling retransmission #39 - INVITE (got response) [Dec 17 11:38:32] DEBUG[3289][C-00000001] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' Request 102: Found [Dec 17 11:38:32] DEBUG[3289][C-00000001] chan_sip.c: SIP response 100 to standard invite [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.124:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK167f66d8 From: ;tag=as08125298 To: ;tag=7C2F8008F5C1_T461496925;user=phone Call-ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 CSeq: 103 NOTIFY User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Content-Length: 0 <-------------> [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK167f66d8 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 2 [ 44]: From: ;tag=as08125298 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 3 [ 70]: To: ;tag=7C2F8008F5C1_T461496925;user=phone [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 4 [ 57]: Call-ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 103 NOTIFY [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 6 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 17 11:38:32] VERBOSE[3289] chan_sip.c: --- (8 headers 0 lines) --- [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 (Checking To) --From tag as08125298 --To-tag 7C2F8008F5C1_T461496925 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Acked pending invite 103 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #41 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Stopping retransmission on 'CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124' of Request 103: Match Found [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 From: "PhoneB" ;tag=as2623879c To: ;tag=7C2F8020F6BA_T1173291550 Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 Contact: CSeq: 102 INVITE User-Agent: DE900 IP PRO/60.02.00.15;7C2F8020F6BA Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO Allow-Events: talk, hold Content-Length: 0 <-------------> [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 2 [ 59]: From: "PhoneB" ;tag=as2623879c [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 3 [ 60]: To: ;tag=7C2F8020F6BA_T1173291550 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 4 [ 55]: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 5 [ 42]: Contact: [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE900 IP PRO/60.02.00.15;7C2F8020F6BA [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 8 [ 85]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 9 [ 24]: Allow-Events: talk, hold [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Dec 17 11:38:32] VERBOSE[3289] chan_sip.c: --- (11 headers 0 lines) --- [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: = Looking for Call ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 (Checking To) --From tag as2623879c --To-tag 7C2F8020F6BA_T1173291550 [Dec 17 11:38:32] DEBUG[3289][C-00000001] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' Request 102: Found [Dec 17 11:38:32] DEBUG[3289][C-00000001] chan_sip.c: SIP response 180 to standard invite [Dec 17 11:38:32] DEBUG[3289][C-00000001] chan_sip.c: build_route: Contact hop: [Dec 17 11:38:32] VERBOSE[3289][C-00000001] chan_sip.c: list_route: hop: [Dec 17 11:38:32] DEBUG[3348][C-00000001] rtp_engine.c: Setting early bridge SDP of 'SIP/phone-b-00000002' with that of 'SIP/phone-c-00000003' [Dec 17 11:38:32] VERBOSE[3348][C-00000001] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T1957041132 To: ;tag=as7ae09457 Call-ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 CSeq: 2 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: P-Asserted-Identity: "PhoneC" Content-Length: 0 <------------> [Dec 17 11:38:32] DEBUG[3348][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:32] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-c [Dec 17 11:38:32] DEBUG[3251] chan_sip.c: Checking device state for peer phone-c [Dec 17 11:38:32] DEBUG[3251] devicestate.c: Changing state for SIP/phone-c - state 6 (Ringing) [Dec 17 11:38:32] DEBUG[3251] devicestate.c: device 'SIP/phone-c' state '6' [Dec 17 11:38:32] DEBUG[3296] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone-c-00000003 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 200 CallerIDName: PhoneC ConnectedLineNum: 100 ConnectedLineName: PhoneB Uniqueid: 1418812712.3 [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3253] chan_sip.c: Strict routing enforced for session CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 [Dec 17 11:38:32] VERBOSE[3253] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:32] DEBUG[3253] netsock2.c: Splitting '192.168.10.124:5060' into... [Dec 17 11:38:32] DEBUG[3253] netsock2.c: ...host '192.168.10.124' and port '5060'. [Dec 17 11:38:32] VERBOSE[3253] chan_sip.c: set_destination: set destination to 192.168.10.124:5060 [Dec 17 11:38:32] VERBOSE[3253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.124:5060: NOTIFY sip:phone-b@192.168.10.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK4f139d6d Max-Forwards: 70 From: ;tag=as08125298 To: ;tag=7C2F8008F5C1_T461496925;user=phone Contact: Call-ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 CSeq: 104 NOTIFY User-Agent: IPTAM PBX (Version 20141216/6814) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 579 sip:100@192.168.10.75 sip:200@192.168.10.75 early --- [Dec 17 11:38:32] DEBUG[3253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #43 [Dec 17 11:38:32] DEBUG[3253] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.124:5060 [Dec 17 11:38:32] DEBUG[3295] app_queue.c: Device 'SIP/phone-c' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.124:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK4f139d6d From: ;tag=as08125298 To: ;tag=7C2F8008F5C1_T461496925;user=phone Call-ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 CSeq: 104 NOTIFY User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Content-Length: 0 <-------------> [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK4f139d6d [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 2 [ 44]: From: ;tag=as08125298 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 3 [ 70]: To: ;tag=7C2F8008F5C1_T461496925;user=phone [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 4 [ 57]: Call-ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 104 NOTIFY [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 6 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 17 11:38:32] VERBOSE[3289] chan_sip.c: --- (8 headers 0 lines) --- [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 (Checking To) --From tag as08125298 --To-tag 7C2F8008F5C1_T461496925 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Acked pending invite 104 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #43 [Dec 17 11:38:32] DEBUG[3289] chan_sip.c: Stopping retransmission on 'CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124' of Request 104: Match Found [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:32] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:33] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Got RTCP report of 88 bytes [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.10.201:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 20697 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x97662e4' so dropping frame [Dec 17 11:38:34] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> REFER sip:180@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T7A157459 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T7A157459;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 2 REFER Refer-To: Referred-by: Contact: Max-Forwards: 69 Content-Length: 0 <-------------> [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 0 [ 41]: REFER sip:180@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T7A157459 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T7A157459;rport=5060 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 3 [ 62]: From: ;tag=7C2F8008F5C1_T1926987239 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 4 [ 57]: To: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 5 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 6 [ 13]: CSeq: 2 REFER [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 7 [153]: Refer-To: [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 8 [ 40]: Referred-by: [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 9 [ 42]: Contact: [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 10 [ 16]: Max-Forwards: 69 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Dec 17 11:38:34] VERBOSE[3289] chan_sip.c: --- (12 headers 0 lines) --- [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: = Looking for Call ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 (Checking From) --From tag 7C2F8008F5C1_T1926987239 --To-tag as7a4ce21e [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Dec 17 11:38:34] VERBOSE[3289][C-00000000] chan_sip.c: Call 21bcfc8b3931cfce66d332554203c767@192.168.10.75 got a SIP call transfer from caller: (REFER)! [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: Attended transfer: Will use Replace-Call-ID : CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 F-tag: 7C2F8008F5C1_T1957041132 T-tag: as7ae09457 [Dec 17 11:38:34] VERBOSE[3289][C-00000000] chan_sip.c: SIP transfer to extension 200@Standard by phone-b@192.168.10.75 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: SIP attended transfer: Transferer channel SIP/phone-b-00000001, transferee channel SIP/phone-a-00000000 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/phone-a-00000000' [Dec 17 11:38:34] VERBOSE[3289][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T7A157459;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T7A157459;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 2 REFER Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: Looking for callid CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 (fromtag 7C2F8008F5C1_T1957041132 totag as7ae09457) [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: Matched INCOMING call - their tag is 7C2F8008F5C1_T1957041132 Our tag is as7ae09457 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: SIP attended transfer: Attempting transfer in ringing state [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: SIP attended transfer: trying to bridge SIP/phone-b-00000002 and SIP/phone-a-00000000 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: Sip transfer:-------------------- [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: -- Transferer to PBX channel: SIP/phone-b-00000001 State Up [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: -- Transferer to PBX second channel (target): SIP/phone-b-00000002 State Ring [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: -- Bridged call to transferee: SIP/phone-a-00000000 State Up [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Attended Channel: SIP/phone-b-00000001 Uniqueid: 1418812704.1 SIP-Callid: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 TargetChannel: SIP/phone-b-00000002 TargetUniqueid: 1418812711.2 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: -- No target second channel --- [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: -- END Sip transfer:-------------------- [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: SIP transfer: Four channels to handle [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Set channel SIP/phone-a-00000000 to write format alaw [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: SIP transfer: trying to masquerade SIP/phone-a-00000000 into SIP/phone-b-00000002 [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Planning to masquerade channel SIP/phone-a-00000000 into the structure of SIP/phone-b-00000002 [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Done planning to masquerade channel SIP/phone-a-00000000 into the structure of SIP/phone-b-00000002 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: Strict routing enforced for session 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:34] VERBOSE[3289][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:34] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.124:5060' into... [Dec 17 11:38:34] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.124' and port '5060'. [Dec 17 11:38:34] VERBOSE[3289][C-00000000] chan_sip.c: set_destination: set destination to 192.168.10.124:5060 [Dec 17 11:38:34] VERBOSE[3289][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.124:5060: NOTIFY sip:phone-b@192.168.10.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK172f5afe Max-Forwards: 70 From: "PhoneA" ;tag=as7a4ce21e To: ;tag=7C2F8008F5C1_T1926987239 Contact: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 103 NOTIFY User-Agent: IPTAM PBX (Version 20141216/6814) Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 16 SIP/2.0 200 OK --- [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #44 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.124:5060 [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Actually Masquerading SIP/phone-a-00000000(6) into the structure of SIP/phone-b-00000002(4) [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Putting channel SIP/phone-a-00000000 in alaw/alaw formats [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: SIP Fixup: New owner for dialogue 0dcce801bbc3a2c1: SIP/phone-a-00000000 (Old parent: SIP/phone-b-00000002) [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: SIP Fixup: New owner for dialogue CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124: SIP/phone-b-00000002 (Old parent: SIP/phone-a-00000000) [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Driver for channel 'SIP/phone-a-00000000' does not support indication 3, emulating it [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Set channel SIP/phone-a-00000000 to write format slin [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Done Masquerading SIP/phone-a-00000000 (6) [Dec 17 11:38:34] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Changing ssrc from 752796134 to 795196775 due to a source change [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Driver for channel 'SIP/phone-a-00000000' does not support indication 3, emulating it [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Set channel SIP/phone-a-00000000 to write format alaw [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Set channel SIP/phone-a-00000000 to write format slin [Dec 17 11:38:34] DEBUG[3289][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/phone-a-00000000 UniqueID: 1418812704.0 [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/phone-a-00000000 CloneState: Up Original: SIP/phone-b-00000002 OriginalState: Ring [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone-a-00000000 Newname: SIP/phone-a-00000000 Uniqueid: 1418812704.0 [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone-b-00000002 Newname: SIP/phone-a-00000000 Uniqueid: 1418812711.2 [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone-a-00000000 Newname: SIP/phone-b-00000002 Uniqueid: 1418812704.0 [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone-a-00000000 CallerIDNum: 180 CallerIDName: PhoneA Uniqueid: 1418812711.2 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Strict routing enforced for session 0dcce801bbc3a2c1 [Dec 17 11:38:34] VERBOSE[3348][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:34] DEBUG[3348][C-00000001] netsock2.c: Splitting '192.168.10.201:5060' into... [Dec 17 11:38:34] DEBUG[3348][C-00000001] netsock2.c: ...host '192.168.10.201' and port '5060'. [Dec 17 11:38:34] VERBOSE[3348][C-00000001] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [Dec 17 11:38:34] VERBOSE[3348][C-00000001] chan_sip.c: Audio is at 19658 [Dec 17 11:38:34] VERBOSE[3348][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP [Dec 17 11:38:34] VERBOSE[3348][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Dec 17 11:38:34] VERBOSE[3348][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw) [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Initializing already initialized SIP dialog 0dcce801bbc3a2c1 (presumably reinvite) [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 0 [ 60]: INVITE sip:phone-a@192.168.10.201:5060;transport=udp SIP/2.0 [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK6a08126e [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 3 [ 60]: From: ;tag=as07820be1 [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 4 [ 60]: To: "PhoneA" ;tag=80931a0487 [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 5 [ 38]: Contact: [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 6 [ 25]: Call-ID: 0dcce801bbc3a2c1 [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 8 [ 45]: User-Agent: IPTAM PBX (Version 20141216/6814) [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 9 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 11 [ 53]: P-Asserted-Identity: "PhoneC" [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Dec 17 11:38:34] VERBOSE[3348][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: INVITE sip:phone-a@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK6a08126e Max-Forwards: 70 From: ;tag=as07820be1 To: "PhoneA" ;tag=80931a0487 Contact: Call-ID: 0dcce801bbc3a2c1 CSeq: 102 INVITE User-Agent: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: "PhoneC" Content-Type: application/sdp Content-Length: 264 v=0 o=root 2046840620 2046840621 IN IP4 192.168.10.75 s=Asterisk PBX 11.15.0-1 c=IN IP4 192.168.10.75 t=0 0 m=audio 19658 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #45 [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.201:5060 [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Strict routing enforced for session 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:34] VERBOSE[3348][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.75:5060: UPDATE sip:phone-c@192.168.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK701fe496 Max-Forwards: 70 From: "PhoneB" ;tag=as2623879c To: ;tag=7C2F8020F6BA_T1173291550 Contact: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 CSeq: 103 UPDATE User-Agent: IPTAM PBX (Version 20141216/6814) P-Asserted-Identity: "PhoneA" X-Asterisk-rpid-update: Yes Content-Length: 0 --- [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #46 [Dec 17 11:38:34] DEBUG[3348][C-00000001] chan_sip.c: Trying to put 'UPDATE sip:' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:34] DEBUG[3345][C-00000000] rtp_engine.c: rtp-engine-local-bridge: Oooh, something is weird, backing out [Dec 17 11:38:34] DEBUG[3345][C-00000000] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/phone-b-00000002, c1=SIP/phone-b-00000001, flags: Yes,Yes,No,No [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 17 11:38:34] DEBUG[3345][C-00000000] channel.c: Bridge stops bridging channels SIP/phone-b-00000002 and SIP/phone-b-00000001 [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: ATTENDEDTRANSFER AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: 200 Exten: 200 Context: sub-dial-intern Channel: SIP/phone-b-00000002 Application: Dial AppData: SIP/phone-c, EventTime: 2014-12-17 11:38:34 AMAFlags: DOCUMENTATION UniqueID: 1418812711.2 LinkedID: 1418812711.2 Userfield: Peer: PeerAccount: Extra: 1418812704.0 [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Unlink Bridgetype: core Channel1: SIP/phone-b-00000002 Channel2: SIP/phone-b-00000001 Uniqueid1: 1418812704.0 Uniqueid2: 1418812704.1 CallerID1: 100 CallerID2: 100 [Dec 17 11:38:34] DEBUG[3345][C-00000000] channel.c: Hanging up channel 'SIP/phone-b-00000001' [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: update_call_counter(phone-b) - decrement call limit counter on hangup [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: Updating call counter for outgoing call [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: Call to peer 'phone-b' removed from call limit 2147483647 [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 21bcfc8b3931cfce66d332554203c767@192.168.10.75. [Dec 17 11:38:34] VERBOSE[3345][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '21bcfc8b3931cfce66d332554203c767@192.168.10.75' in 32000 ms (Method: REFER) [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 180 Destination: 100 DestinationContext: sub-dial-intern CallerID: "PhoneA" <180> Channel: SIP/phone-a-00000000 DestinationChannel: SIP/phone-b-00000001 LastApplication: Dial LastData: SIP/phone-b, StartTime: 2014-12-17 11:38:24 AnswerTime: 2014-12-17 11:38:29 EndTime: 2014-12-17 11:38:34 Duration: 10 BillableSeconds: 5 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1418812704.0 UserField: D:200# [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone-b-00000001 Uniqueid: 1418812704.1 CallerIDNum: 100 CallerIDName: PhoneB ConnectedLineNum: 180 ConnectedLineName: PhoneA AccountCode: Cause: 16 Cause-txt: Normal Clearing [Dec 17 11:38:34] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:34] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:34] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 2 (In use) [Dec 17 11:38:34] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '2' [Dec 17 11:38:34] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:34] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:34] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 2 (In use) [Dec 17 11:38:34] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '2' [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: Exten: Context: sub-dial-intern Channel: SIP/phone-b-00000001 Application: AppDial AppData: (Outgoing Line) EventTime: 2014-12-17 11:38:34 AMAFlags: DOCUMENTATION UniqueID: 1418812704.1 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: 16,, [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_END AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: Exten: Context: sub-dial-intern Channel: SIP/phone-b-00000001 Application: AppDial AppData: (Outgoing Line) EventTime: 2014-12-17 11:38:34 AMAFlags: DOCUMENTATION UniqueID: 1418812704.1 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:34] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:34] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:34] DEBUG[3296] manager.c: Running action 'Setvar' [Dec 17 11:38:34] DEBUG[3345][C-00000000] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Dec 17 11:38:34] DEBUG[3345][C-00000000] pbx.c: Spawn extension (sub-dial-intern,100,6) exited non-zero on 'SIP/phone-b-00000002' [Dec 17 11:38:34] DEBUG[3345][C-00000000] channel.c: Soft-Hanging up channel 'SIP/phone-b-00000002' [Dec 17 11:38:34] DEBUG[3345][C-00000000] channel.c: Hanging up channel 'SIP/phone-b-00000002' [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: Hangup call SIP/phone-b-00000002, SIP callid CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: update_call_counter(phone-b) - decrement call limit counter on hangup [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: Updating call counter for incoming call [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: Call from peer 'phone-b' removed from call limit 2147483647 [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: Hanging up channel in state Ring (not UP) [Dec 17 11:38:34] DEBUG[3345][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97932dc' [Dec 17 11:38:34] VERBOSE[3345][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124' in 32000 ms (Method: INVITE) [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: AST hangup cause 16 (no match found in SIP) [Dec 17 11:38:34] VERBOSE[3345][C-00000000] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T1957041132 To: ;tag=as7ae09457 Call-ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 CSeq: 2 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 <------------> [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #49 [Dec 17 11:38:34] DEBUG[3345][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 603' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone-b-00000002 UniqueID: 1418812704.0 DialStatus: ANSWER [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: SoftHangupRequest Privilege: call,all Channel: SIP/phone-b-00000002 Uniqueid: 1418812704.0 Cause: 16 [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone-b-00000002 Uniqueid: 1418812704.0 CallerIDNum: 100 CallerIDName: PhoneB ConnectedLineNum: 200 ConnectedLineName: PhoneC AccountCode: Cause: 16 Cause-txt: Normal Clearing [Dec 17 11:38:34] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:34] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:34] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:34] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 1 (Not in use) [Dec 17 11:38:34] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '1' [Dec 17 11:38:34] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:34] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:34] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 1 (Not in use) [Dec 17 11:38:34] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '1' [Dec 17 11:38:34] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.124:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK172f5afe From: "PhoneA" ;tag=as7a4ce21e To: ;tag=7C2F8008F5C1_T1926987239 Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 103 NOTIFY User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Content-Length: 0 <-------------> [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK172f5afe [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 2 [ 59]: From: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 3 [ 60]: To: ;tag=7C2F8008F5C1_T1926987239 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 4 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 103 NOTIFY [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 6 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 17 11:38:34] VERBOSE[3289] chan_sip.c: --- (8 headers 0 lines) --- [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: = Looking for Call ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 (Checking To) --From tag as7a4ce21e --To-tag 7C2F8008F5C1_T1926987239 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #44 [Dec 17 11:38:34] DEBUG[3289][C-00000000] chan_sip.c: Stopping retransmission on '21bcfc8b3931cfce66d332554203c767@192.168.10.75' of Request 103: Match Found [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: APP_END AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: 200 Exten: 100 Context: sub-dial-intern Channel: SIP/phone-b-00000002 Application: Dial AppData: SIP/phone-b, EventTime: 2014-12-17 11:38:34 AMAFlags: DOCUMENTATION UniqueID: 1418812704.0 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: 200 Exten: 100 Context: sub-dial-intern Channel: SIP/phone-b-00000002 Application: AppData: EventTime: 2014-12-17 11:38:34 AMAFlags: DOCUMENTATION UniqueID: 1418812704.0 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: 16,,ANSWER [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_END AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: 200 Exten: 100 Context: sub-dial-intern Channel: SIP/phone-b-00000002 Application: AppData: EventTime: 2014-12-17 11:38:34 AMAFlags: DOCUMENTATION UniqueID: 1418812704.0 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:34] DEBUG[3253] app_queue.c: Extension '100@_extensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:34] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:34] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:34] DEBUG[3296] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: _extensions Hint: SIP/phone-b Status: 0 [Dec 17 11:38:34] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK701fe496 From: "PhoneB" ;tag=as2623879c To: ;tag=7C2F8020F6BA_T1173291550 Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 CSeq: 103 UPDATE Content-Length: 0 <-------------> [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK701fe496 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 2 [ 59]: From: "PhoneB" ;tag=as2623879c [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 3 [ 60]: To: ;tag=7C2F8020F6BA_T1173291550 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 4 [ 55]: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 103 UPDATE [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Dec 17 11:38:34] VERBOSE[3289] chan_sip.c: --- (7 headers 0 lines) --- [Dec 17 11:38:34] DEBUG[3289] chan_sip.c: = Looking for Call ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 (Checking To) --From tag as2623879c --To-tag 7C2F8020F6BA_T1173291550 [Dec 17 11:38:34] DEBUG[3289][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #46 [Dec 17 11:38:34] DEBUG[3289][C-00000001] chan_sip.c: Stopping retransmission on '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' of Request 103: Match Found [Dec 17 11:38:34] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:34] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:34] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:34] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:34] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK6a08126e From: ;tag=as07820be1 To: "PhoneA" ;tag=80931a0487 Call-ID: 0dcce801bbc3a2c1 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "PhoneA" ;+sip.instance="" Server: Aastra 55i/3.3.1.2217 Supported: path, replaces Content-Type: application/sdp Content-Length: 261 v=0 o=MxSIP 0 2 IN IP4 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK6a08126e [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 2 [ 60]: From: ;tag=as07820be1 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 3 [ 60]: To: "PhoneA" ;tag=80931a0487 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 4 [ 25]: Call-ID: 0dcce801bbc3a2c1 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 8 [129]: Contact: "PhoneA" ;+sip.instance="" [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 9 [ 29]: Server: Aastra 55i/3.3.1.2217 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 10 [ 25]: Supported: path, replaces [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 12 [ 19]: Content-Length: 261 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 13 [ 0]: [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 0 [ 3]: v=0 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 2 IN IP4 192.168.10.201 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 2 [ 10]: s=SIP Call [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Body 12 [ 10]: a=sendrecv [Dec 17 11:38:35] VERBOSE[3289] chan_sip.c: --- (13 headers 13 lines) --- [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: = Looking for Call ID: 0dcce801bbc3a2c1 (Checking To) --From tag as07820be1 --To-tag 80931a0487 [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Acked pending invite 102 [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #45 [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Stopping retransmission on '0dcce801bbc3a2c1' of Request 102: Match Found [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call 0dcce801bbc3a2c1 [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.10.201... OK. [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. [Dec 17 11:38:35] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.201' into... [Dec 17 11:38:35] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.201' and port ''. [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 8 [Dec 17 11:38:35] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0xb5ba82f0 [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 0 [Dec 17 11:38:35] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0xb5ba82f0 [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 101 [Dec 17 11:38:35] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 101 based on m type on 0xb5ba82f0 [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED. [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 17 11:38:35] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9759f6c' [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Dec 17 11:38:35] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 0 from 0xb5ba82f0 to 0x975a118 [Dec 17 11:38:35] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 8 from 0xb5ba82f0 to 0x975a118 [Dec 17 11:38:35] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 101 from 0xb5ba82f0 to 0x975a118 [Dec 17 11:38:35] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x9759f6c' [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: We're settling with these formats: (ulaw|alaw) [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw) [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Updating call counter for incoming call [Dec 17 11:38:35] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.201:5060' into... [Dec 17 11:38:35] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.201' and port '5060'. [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Strict routing enforced for session 0dcce801bbc3a2c1 [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:35] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.201:5060' into... [Dec 17 11:38:35] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.201' and port '5060'. [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Transmitting (no NAT) to 192.168.10.201:5060: ACK sip:phone-a@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK105a2d44 Max-Forwards: 70 From: ;tag=as07820be1 To: "PhoneA" ;tag=80931a0487 Contact: Call-ID: 0dcce801bbc3a2c1 CSeq: 102 ACK User-Agent: IPTAM PBX (Version 20141216/6814) Content-Length: 0 --- [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Dec 17 11:38:35] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-a [Dec 17 11:38:35] DEBUG[3251] chan_sip.c: Checking device state for peer phone-a [Dec 17 11:38:35] DEBUG[3251] devicestate.c: Changing state for SIP/phone-a - state 2 (In use) [Dec 17 11:38:35] DEBUG[3251] devicestate.c: device 'SIP/phone-a' state '2' [Dec 17 11:38:35] DEBUG[3295] app_queue.c: Device 'SIP/phone-a' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: 0x975eec0 -- Probation learning mode pass with source address 192.168.10.201:3000 [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: SIP TIMER: Rescheduling retransmission #49 (1) SIP/2.0 - 1 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #49)) [Dec 17 11:38:35] VERBOSE[3289] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.10.75:5060: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T1957041132 To: ;tag=as7ae09457 Call-ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 CSeq: 2 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0 --- [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Trying to put 'SIP/2.0 603' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> BYE sip:180@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T078DB36E Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T078DB36E;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 3 BYE User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Contact: Max-Forwards: 69 Content-Length: 0 <-------------> [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 0 [ 39]: BYE sip:180@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T078DB36E [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T078DB36E;rport=5060 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 3 [ 62]: From: ;tag=7C2F8008F5C1_T1926987239 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 4 [ 57]: To: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 5 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 6 [ 11]: CSeq: 3 BYE [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 8 [ 42]: Contact: [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 9 [ 16]: Max-Forwards: 69 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Dec 17 11:38:35] VERBOSE[3289] chan_sip.c: --- (11 headers 0 lines) --- [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: = Looking for Call ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 (Checking From) --From tag 7C2F8008F5C1_T1926987239 --To-tag as7a4ce21e [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Initializing initreq for method BYE - callid 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:35] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:35] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Sending to 192.168.10.124:5060 (no NAT) [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Setting SIP_ALREADYGONE on dialog 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:35] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97662e4' [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '21bcfc8b3931cfce66d332554203c767@192.168.10.75' in 32000 ms (Method: BYE) [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Received bye, no owner, selfdestruct soon. [Dec 17 11:38:35] VERBOSE[3289][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T078DB36E;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T078DB36E;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 3 BYE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Dec 17 11:38:35] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> ACK sip:200@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T30D180B0 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T1957041132 To: ;tag=as7ae09457 Call-ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 CSeq: 2 ACK User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Route: Contact: Max-Forwards: 69 Content-Length: 0 <-------------> [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 0 [ 39]: ACK sip:200@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T30D180B0 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T30D180B0;rport=5060 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 3 [ 72]: From: " PhoneB" ;tag=7C2F8008F5C1_T1957041132 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 4 [ 42]: To: ;tag=as7ae09457 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 5 [ 57]: Call-ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 6 [ 11]: CSeq: 2 ACK [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 8 [ 30]: Route: [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 9 [ 42]: Contact: [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 10 [ 16]: Max-Forwards: 69 [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Dec 17 11:38:35] VERBOSE[3289] chan_sip.c: --- (12 headers 0 lines) --- [Dec 17 11:38:35] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 (Checking From) --From tag 7C2F8008F5C1_T1957041132 --To-tag as7ae09457 [Dec 17 11:38:35] DEBUG[3289][C-00000001] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 17 11:38:35] DEBUG[3289][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #49 [Dec 17 11:38:35] DEBUG[3289][C-00000001] chan_sip.c: Stopping retransmission on 'CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124' of Response 2: Match Found [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3296] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 192.168.10.201:3001 OurSSRC: 795196775 SentNTP: 1418812715.3896553472 SentRTP: 39840 SentPackets: 249 SentOctets: 39840 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0006 TheirLastSR: 8003256 DLSR: 1.9370 (sec) [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:35] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> BYE sip:180@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T078DB36E Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T078DB36E;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 3 BYE User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Contact: Max-Forwards: 69 Content-Length: 0 <-------------> [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 0 [ 39]: BYE sip:180@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T078DB36E [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T078DB36E;rport=5060 [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 3 [ 62]: From: ;tag=7C2F8008F5C1_T1926987239 [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 4 [ 57]: To: "PhoneA" ;tag=as7a4ce21e [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 5 [ 55]: Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 6 [ 11]: CSeq: 3 BYE [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 8 [ 42]: Contact: [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 9 [ 16]: Max-Forwards: 69 [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Dec 17 11:38:36] VERBOSE[3289] chan_sip.c: --- (11 headers 0 lines) --- [Dec 17 11:38:36] DEBUG[3289] chan_sip.c: = Looking for Call ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 (Checking From) --From tag 7C2F8008F5C1_T1926987239 --To-tag as7a4ce21e [Dec 17 11:38:36] DEBUG[3289][C-00000000] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Dec 17 11:38:36] DEBUG[3289][C-00000000] chan_sip.c: Ignoring SIP message because of retransmit (BYE Seqno 3, ours 3) [Dec 17 11:38:36] DEBUG[3289][C-00000000] chan_sip.c: Initializing initreq for method BYE - callid 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:36] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:36] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:36] VERBOSE[3289][C-00000000] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:36] DEBUG[3289][C-00000000] chan_sip.c: Setting SIP_ALREADYGONE on dialog 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:38:36] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97662e4' [Dec 17 11:38:36] DEBUG[3289][C-00000000] chan_sip.c: Received bye, no owner, selfdestruct soon. [Dec 17 11:38:36] VERBOSE[3289][C-00000000] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T078DB36E;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T078DB36E;rport=5060 From: ;tag=7C2F8008F5C1_T1926987239 To: "PhoneA" ;tag=as7a4ce21e Call-ID: 21bcfc8b3931cfce66d332554203c767@192.168.10.75 CSeq: 3 BYE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Dec 17 11:38:36] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:36] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:37] DEBUG[3296] db.c: Unable to find key '200' in family 'CFIM' [Dec 17 11:38:37] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:37] DEBUG[3296] db.c: Unable to find key '200' in family 'CFUN' [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:37] DEBUG[3296] db.c: Unable to find key '200' in family 'CFBS' [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:37] DEBUG[3296] db.c: Unable to find key '200' in family 'CFNOCID' [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:37] DEBUG[3296] db.c: Unable to find key '200' in family 'CFFB' [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:37] DEBUG[3296] db.c: Unable to find key '200' in family 'CFFBTO' [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:37] DEBUG[3296] db.c: Unable to find key '200' in family 'DND' [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:37] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:37] DEBUG[3296] db.c: Unable to find key '200' in family 'DND_ALLOW' [Dec 17 11:38:37] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:38] DEBUG[3296] db.c: Unable to find key '200' in family 'PARALLEL' [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:38] DEBUG[3296] db.c: Unable to find key '200' in family 'PAR_DELAY' [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3296] manager.c: Running action 'DBGet' [Dec 17 11:38:38] DEBUG[3296] db.c: Unable to find key '200' in family 'PAR_AMD' [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:38] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Got RTCP report of 88 bytes [Dec 17 11:38:39] DEBUG[3296] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.10.201:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 20951 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:39] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3296] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 192.168.10.201:3001 OurSSRC: 795196775 SentNTP: 1418812720.3901775872 SentRTP: 79840 SentPackets: 499 SentOctets: 79840 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0012 TheirLastSR: 8336179 DLSR: 1.8640 (sec) [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:40] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> INVITE sip:*8200@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T14E68038 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T14E68038;rport=5060 Session-Expires: 1800 From: " PhoneB" ;tag=7C2F8008F5C1_T343468775 To: Call-ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 CSeq: 2 INVITE Contact: Max-Forwards: 69 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO Supported: 100rel,timer,replaces User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Content-Type: application/sdp Content-Length: 241 v=0 o=- 123243601 123243601 IN IP4 192.168.10.124 s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1 c=IN IP4 192.168.10.124 t=0 0 m=audio 5004 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 0 [ 44]: INVITE sip:*8200@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T14E68038 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T14E68038;rport=5060 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 3 [ 21]: Session-Expires: 1800 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 4 [ 71]: From: " PhoneB" ;tag=7C2F8008F5C1_T343468775 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 5 [ 29]: To: [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 6 [ 57]: Call-ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 7 [ 14]: CSeq: 2 INVITE [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 8 [ 42]: Contact: [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 9 [ 16]: Max-Forwards: 69 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 10 [ 85]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 11 [ 32]: Supported: 100rel,timer,replaces [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 12 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 14 [ 19]: Content-Length: 241 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 15 [ 0]: [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 0 [ 3]: v=0 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 1 [ 45]: o=- 123243601 123243601 IN IP4 192.168.10.124 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 2 [ 39]: s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.124 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 5 [ 26]: m=audio 5004 RTP/AVP 8 101 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 6 [ 22]: a=rtpmap:8 PCMA/8000/1 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 9 [ 10]: a=sendrecv [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: --- (15 headers 10 lines) --- [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 (Checking From) --From tag 7C2F8008F5C1_T343468775 --To-tag [Dec 17 11:38:41] DEBUG[3289] acl.c: For destination '192.168.10.75', our source address is '192.168.10.75'. [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:25060 [Dec 17 11:38:41] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:41] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Allocating new SIP dialog for CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 - INVITE (No RTP) [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Dec 17 11:38:41] DEBUG[3289][C-00000002] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,timer,replaces" [Dec 17 11:38:41] DEBUG[3289][C-00000002] sip/reqresp_parser.c: Found SIP option: -100rel- [Dec 17 11:38:41] DEBUG[3289][C-00000002] sip/reqresp_parser.c: Matched SIP option: 100rel [Dec 17 11:38:41] DEBUG[3289][C-00000002] sip/reqresp_parser.c: Found SIP option: -timer- [Dec 17 11:38:41] DEBUG[3289][C-00000002] sip/reqresp_parser.c: Matched SIP option: timer [Dec 17 11:38:41] DEBUG[3289][C-00000002] sip/reqresp_parser.c: Found SIP option: -replaces- [Dec 17 11:38:41] DEBUG[3289][C-00000002] sip/reqresp_parser.c: Matched SIP option: replaces [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Using INVITE request as basis request - CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: Splitting '192.168.10.75' into... [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Found peer 'phone-b' for 'phone-b' from 192.168.10.75:5060 [Dec 17 11:38:41] DEBUG[3289][C-00000002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x97c09a4' [Dec 17 11:38:41] DEBUG[3289][C-00000002] res_rtp_asterisk.c: Allocated port 12810 for RTP instance '0x97c09a4' [Dec 17 11:38:41] DEBUG[3289][C-00000002] rtp_engine.c: RTP instance '0x97c09a4' is setup and ready to go [Dec 17 11:38:41] DEBUG[3289][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x97c09a4' [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Setting NAT on RTP to Off [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Processing session-level SDP o=- 123243601 123243601 IN IP4 192.168.10.124... OK. [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Processing session-level SDP s=DE700 IP PRO/61.02.00.15;7C2F8008F5C1... UNSUPPORTED OR FAILED. [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: Splitting '192.168.10.124' into... [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: ...host '192.168.10.124' and port ''. [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.124... OK. [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Found RTP audio format 8 [Dec 17 11:38:41] DEBUG[3289][C-00000002] rtp_engine.c: Setting payload 8 based on m type on 0xb5ba9240 [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Found RTP audio format 101 [Dec 17 11:38:41] DEBUG[3289][C-00000002] rtp_engine.c: Setting payload 101 based on m type on 0xb5ba9240 [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8 [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK. [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 17 11:38:41] DEBUG[3289][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97c09a4' [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Peer audio RTP is at port 192.168.10.124:5004 [Dec 17 11:38:41] DEBUG[3289][C-00000002] rtp_engine.c: Copying payload 8 from 0xb5ba9240 to 0x97c0b50 [Dec 17 11:38:41] DEBUG[3289][C-00000002] rtp_engine.c: Copying payload 101 from 0xb5ba9240 to 0x97c0b50 [Dec 17 11:38:41] DEBUG[3289][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x97c09a4' [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: We're settling with these formats: (alaw) [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Checking SIP call limits for device phone-b [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Updating call counter for incoming call [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Call from peer 'phone-b' is 1 out of 2147483647 [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: Splitting '192.168.10.75:25060' into... [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: Splitting '192.168.10.75' into... [Dec 17 11:38:41] DEBUG[3289][C-00000002] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: Looking for *8200 in Standard (domain 192.168.10.75) [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Incoming INVITE with 'timer' option supported [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: INVITE also has "Session-Expires" header. [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Session-Expires: 1800 [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: *** Our native formats are (alaw) [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: *** Joint capabilities are (alaw) [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: *** Our capabilities are (ulaw|alaw) [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: This channel will not be able to handle video. [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: build_route: Contact hop: [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: list_route: hop: [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: SIP/phone-b-00000004: New call is still down.... Trying... [Dec 17 11:38:41] VERBOSE[3289][C-00000002] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T14E68038;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T14E68038;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T343468775 To: Call-ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 CSeq: 2 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:41] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:41] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:41] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 2 (In use) [Dec 17 11:38:41] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '2' [Dec 17 11:38:41] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:41] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:41] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 2 (In use) [Dec 17 11:38:41] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '2' [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone-b-00000004 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: PhoneB AccountCode: Exten: *8200 Context: Standard Uniqueid: 1418812721.4 [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone-b-00000004 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: PhoneB ConnectedLineNum: ConnectedLineName: Uniqueid: 1418812721.4 [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: *8200 Context: Standard Channel: SIP/phone-b-00000004 Application: AppData: EventTime: 2014-12-17 11:38:41 AMAFlags: DOCUMENTATION UniqueID: 1418812721.4 LinkedID: 1418812721.4 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:41] DEBUG[3357][C-00000002] pbx.c: Result of 'EXTEN' is '*8200' [Dec 17 11:38:41] DEBUG[3357][C-00000002] pbx.c: Launching 'Pickup' [Dec 17 11:38:41] DEBUG[3357][C-00000002] features.c: Call pickup on 'SIP/phone-c-00000003' by 'SIP/phone-b-00000004' [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: SIP answering channel: SIP/phone-b-00000004 [Dec 17 11:38:41] DEBUG[3357][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Setting framing from config on incoming call [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [Dec 17 11:38:41] VERBOSE[3357][C-00000002] chan_sip.c: Audio is at 12810 [Dec 17 11:38:41] VERBOSE[3357][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP [Dec 17 11:38:41] VERBOSE[3357][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Dec 17 11:38:41] VERBOSE[3357][C-00000002] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T14E68038;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T14E68038;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T343468775 To: ;tag=as5454b2c8 Call-ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 CSeq: 2 INVITE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: P-Asserted-Identity: "PhoneA" Content-Type: application/sdp Require: timer Content-Length: 236 v=0 o=root 76852669 76852669 IN IP4 192.168.10.75 s=Asterisk PBX 11.15.0-1 c=IN IP4 192.168.10.75 t=0 0 m=audio 12810 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #52 [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Session timer started: 53 - CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 900000ms [Dec 17 11:38:41] DEBUG[3253] app_queue.c: Extension '100@_extensions' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:41] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:41] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone-b-00000004 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 100 CallerIDName: PhoneB ConnectedLineNum: 180 ConnectedLineName: PhoneA Uniqueid: 1418812721.4 [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: _extensions Hint: SIP/phone-b Status: 1 [Dec 17 11:38:41] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:41] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:41] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 2 (In use) [Dec 17 11:38:41] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '2' [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: PICKUP AccountCode: CallerIDnum: 200 CallerIDname: PhoneC CallerIDani: 200 CallerIDrdnis: CallerIDdnid: Exten: 200 Context: Standard Channel: SIP/phone-c-00000003 Application: AppDial AppData: (Outgoing Line) EventTime: 2014-12-17 11:38:41 AMAFlags: DOCUMENTATION UniqueID: 1418812712.3 LinkedID: 1418812711.2 Userfield: Peer: SIP/phone-b-00000004 PeerAccount: Extra: [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: ANSWER AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: *8200 Exten: *8200 Context: Standard Channel: SIP/phone-b-00000004 Application: Pickup AppData: 200@PICKUPMARK&888@PICKUPMARK&889@PICKUPMARK&789@PICKUPMARK EventTime: 2014-12-17 11:38:41 AMAFlags: DOCUMENTATION UniqueID: 1418812721.4 LinkedID: 1418812721.4 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:41] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> ACK sip:*8200@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T1D959F24 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T1D959F24;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T343468775 To: ;tag=as5454b2c8 Call-ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 CSeq: 2 ACK User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Contact: Max-Forwards: 69 Content-Length: 0 <-------------> [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 0 [ 41]: ACK sip:*8200@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T1D959F24 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T1D959F24;rport=5060 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 3 [ 71]: From: " PhoneB" ;tag=7C2F8008F5C1_T343468775 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 4 [ 44]: To: ;tag=as5454b2c8 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 5 [ 57]: Call-ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 6 [ 11]: CSeq: 2 ACK [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 8 [ 42]: Contact: [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 9 [ 16]: Max-Forwards: 69 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: --- (11 headers 0 lines) --- [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 (Checking From) --From tag 7C2F8008F5C1_T343468775 --To-tag as5454b2c8 [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #52 [Dec 17 11:38:41] DEBUG[3289][C-00000002] chan_sip.c: Stopping retransmission on 'CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124' of Response 2: Match Found [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x97aa8dc' so dropping frame [Dec 17 11:38:41] DEBUG[3357][C-00000002] res_rtp_asterisk.c: 0x97b79a0 -- Probation learning mode pass with source address 192.168.10.124:5004 [Dec 17 11:38:41] DEBUG[3357][C-00000002] channel.c: Planning to masquerade channel SIP/phone-b-00000004 into the structure of SIP/phone-c-00000003 [Dec 17 11:38:41] DEBUG[3357][C-00000002] channel.c: Done planning to masquerade channel SIP/phone-b-00000004 into the structure of SIP/phone-c-00000003 [Dec 17 11:38:41] DEBUG[3357][C-00000002] channel.c: Actually Masquerading SIP/phone-b-00000004(6) into the structure of SIP/phone-c-00000003(5) [Dec 17 11:38:41] DEBUG[3357][C-00000002] channel.c: Putting channel SIP/phone-b-00000004 in alaw/alaw formats [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: SIP Fixup: New owner for dialogue CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124: SIP/phone-b-00000004 (Old parent: SIP/phone-c-00000003) [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: SIP Fixup: New owner for dialogue 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75: SIP/phone-c-00000003 (Old parent: SIP/phone-b-00000004) [Dec 17 11:38:41] DEBUG[3357][C-00000002] channel.c: Done Masquerading SIP/phone-b-00000004 (6) [Dec 17 11:38:41] DEBUG[3357][C-00000002] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Dec 17 11:38:41] DEBUG[3357][C-00000002] pbx.c: Spawn extension (Standard,*8200,1) exited non-zero on 'SIP/phone-c-00000003' [Dec 17 11:38:41] DEBUG[3357][C-00000002] channel.c: Soft-Hanging up channel 'SIP/phone-c-00000003' [Dec 17 11:38:41] DEBUG[3357][C-00000002] channel.c: Soft-Hanging up channel 'SIP/phone-c-00000003' [Dec 17 11:38:41] DEBUG[3357][C-00000002] pbx.c: Launching 'NoOp' [Dec 17 11:38:41] DEBUG[3357][C-00000002] channel.c: Hanging up channel 'SIP/phone-c-00000003' [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: This call was answered elsewhere [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Hangup call SIP/phone-c-00000003, SIP callid 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: update_call_counter(phone-c) - decrement call limit counter on hangup [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Updating call counter for outgoing call [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Call to peer 'phone-c' removed from call limit 2147483647 [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Hanging up channel in state Ringing (not UP) [Dec 17 11:38:41] DEBUG[3357][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97aa8dc' [Dec 17 11:38:41] VERBOSE[3357][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' in 32000 ms (Method: INVITE) [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' Request 102: Found [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Strict routing enforced for session 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:41] VERBOSE[3357][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.75:5060: CANCEL sip:phone-c@192.168.10.75 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 Max-Forwards: 70 From: "PhoneB" ;tag=as2623879c To: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 CSeq: 102 CANCEL User-Agent: IPTAM PBX (Version 20141216/6814) Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 --- [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #56 [Dec 17 11:38:41] DEBUG[3357][C-00000002] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:41] VERBOSE[3357][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' in 32000 ms (Method: INVITE) [Dec 17 11:38:41] DEBUG[3348][C-00000001] channel.c: Set channel SIP/phone-a-00000000 to write format alaw [Dec 17 11:38:41] DEBUG[3348][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Dec 17 11:38:41] DEBUG[3348][C-00000001] features.c: bridge answer set, chan answer set [Dec 17 11:38:41] DEBUG[3348][C-00000001] features.c: Removing dialed interfaces datastore on SIP/phone-b-00000004 since we're bridging [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [Dec 17 11:38:41] DEBUG[3348][C-00000001] channel.c: Dropping duplicate answer! [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Strict routing enforced for session 0dcce801bbc3a2c1 [Dec 17 11:38:41] VERBOSE[3348][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:41] DEBUG[3348][C-00000001] netsock2.c: Splitting '192.168.10.201:5060' into... [Dec 17 11:38:41] DEBUG[3348][C-00000001] netsock2.c: ...host '192.168.10.201' and port '5060'. [Dec 17 11:38:41] VERBOSE[3348][C-00000001] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [Dec 17 11:38:41] VERBOSE[3348][C-00000001] chan_sip.c: Audio is at 19658 [Dec 17 11:38:41] VERBOSE[3348][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP [Dec 17 11:38:41] VERBOSE[3348][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Dec 17 11:38:41] VERBOSE[3348][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw) [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Initializing already initialized SIP dialog 0dcce801bbc3a2c1 (presumably reinvite) [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 0 [ 60]: INVITE sip:phone-a@192.168.10.201:5060;transport=udp SIP/2.0 [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK58dda877 [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 3 [ 60]: From: ;tag=as07820be1 [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 4 [ 60]: To: "PhoneA" ;tag=80931a0487 [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 5 [ 38]: Contact: [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 6 [ 25]: Call-ID: 0dcce801bbc3a2c1 [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 8 [ 45]: User-Agent: IPTAM PBX (Version 20141216/6814) [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 9 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 11 [ 53]: P-Asserted-Identity: "PhoneB" [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Dec 17 11:38:41] VERBOSE[3348][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: INVITE sip:phone-a@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK58dda877 Max-Forwards: 70 From: ;tag=as07820be1 To: "PhoneA" ;tag=80931a0487 Contact: Call-ID: 0dcce801bbc3a2c1 CSeq: 103 INVITE User-Agent: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: "PhoneB" Content-Type: application/sdp Content-Length: 264 v=0 o=root 2046840620 2046840622 IN IP4 192.168.10.75 s=Asterisk PBX 11.15.0-1 c=IN IP4 192.168.10.75 t=0 0 m=audio 19658 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #58 [Dec 17 11:38:41] DEBUG[3348][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.201:5060 [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Pickup Privilege: call,all Channel: SIP/phone-b-00000004 TargetChannel: SIP/phone-c-00000003 [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/phone-b-00000004 CloneState: Up Original: SIP/phone-c-00000003 OriginalState: Ringing [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone-b-00000004 Newname: SIP/phone-b-00000004 Uniqueid: 1418812721.4 [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone-c-00000003 Newname: SIP/phone-b-00000004 Uniqueid: 1418812712.3 [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone-b-00000004 Newname: SIP/phone-c-00000003 Uniqueid: 1418812721.4 [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone-b-00000004 CallerIDNum: 100 CallerIDName: PhoneB Uniqueid: 1418812712.3 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: SoftHangupRequest Privilege: call,all Channel: SIP/phone-c-00000003 Uniqueid: 1418812721.4 Cause: 16 [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone-c-00000003 Uniqueid: 1418812721.4 CallerIDNum: 200 CallerIDName: PhoneC ConnectedLineNum: 180 ConnectedLineName: PhoneA AccountCode: Cause: 26 Cause-txt: Answered elsewhere [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone-b-00000004 Uniqueid: 1418812712.3 AccountCode: OldAccountCode: [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone-a-00000000 Channel2: SIP/phone-b-00000004 Uniqueid1: 1418812711.2 Uniqueid2: 1418812712.3 CallerID1: 180 CallerID2: 100 [Dec 17 11:38:41] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-c [Dec 17 11:38:41] DEBUG[3251] chan_sip.c: Checking device state for peer phone-c [Dec 17 11:38:41] DEBUG[3251] devicestate.c: Changing state for SIP/phone-c - state 1 (Not in use) [Dec 17 11:38:41] DEBUG[3251] devicestate.c: device 'SIP/phone-c' state '1' [Dec 17 11:38:41] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-c [Dec 17 11:38:41] DEBUG[3251] chan_sip.c: Checking device state for peer phone-c [Dec 17 11:38:41] DEBUG[3251] devicestate.c: Changing state for SIP/phone-c - state 1 (Not in use) [Dec 17 11:38:41] DEBUG[3251] devicestate.c: device 'SIP/phone-c' state '1' [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: LINKEDID_END AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: *8200 Exten: *8200 Context: Standard Channel: SIP/phone-b-00000004 Application: Pickup AppData: 200@PICKUPMARK&888@PICKUPMARK&889@PICKUPMARK&789@PICKUPMARK EventTime: 2014-12-17 11:38:41 AMAFlags: DOCUMENTATION UniqueID: 1418812721.4 LinkedID: 1418812721.4 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 200 CallerIDname: PhoneC CallerIDani: 200 CallerIDrdnis: CallerIDdnid: Exten: h Context: Standard Channel: SIP/phone-c-00000003 Application: AppData: EventTime: 2014-12-17 11:38:41 AMAFlags: DOCUMENTATION UniqueID: 1418812721.4 LinkedID: 1418812711.2 Userfield: Peer: PeerAccount: Extra: 26,, [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_END AccountCode: CallerIDnum: 200 CallerIDname: PhoneC CallerIDani: 200 CallerIDrdnis: CallerIDdnid: Exten: h Context: Standard Channel: SIP/phone-c-00000003 Application: AppData: EventTime: 2014-12-17 11:38:41 AMAFlags: DOCUMENTATION UniqueID: 1418812721.4 LinkedID: 1418812711.2 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: LINKEDID_END AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: *8200 Exten: Context: Standard Channel: SIP/phone-b-00000004 Application: AppDial AppData: (Outgoing Line) EventTime: 2014-12-17 11:38:41 AMAFlags: DOCUMENTATION UniqueID: 1418812712.3 LinkedID: 1418812711.2 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: BRIDGE_START AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: 180 CallerIDrdnis: CallerIDdnid: 100 Exten: 200 Context: sub-dial-intern Channel: SIP/phone-a-00000000 Application: Dial AppData: SIP/phone-c, EventTime: 2014-12-17 11:38:41 AMAFlags: DOCUMENTATION UniqueID: 1418812711.2 LinkedID: 1418812704.0 Userfield: Peer: SIP/phone-b-00000004 PeerAccount: Extra: [Dec 17 11:38:41] DEBUG[3253] app_queue.c: Extension '200@_extensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:41] DEBUG[3253] chan_sip.c: Strict routing enforced for session CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 [Dec 17 11:38:41] VERBOSE[3253] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:41] DEBUG[3253] netsock2.c: Splitting '192.168.10.124:5060' into... [Dec 17 11:38:41] DEBUG[3253] netsock2.c: ...host '192.168.10.124' and port '5060'. [Dec 17 11:38:41] VERBOSE[3253] chan_sip.c: set_destination: set destination to 192.168.10.124:5060 [Dec 17 11:38:41] VERBOSE[3253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.124:5060: NOTIFY sip:phone-b@192.168.10.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK531770fe Max-Forwards: 70 From: ;tag=as08125298 To: ;tag=7C2F8008F5C1_T461496925;user=phone Contact: Call-ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 CSeq: 105 NOTIFY User-Agent: IPTAM PBX (Version 20141216/6814) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 204 terminated --- [Dec 17 11:38:41] DEBUG[3253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #59 [Dec 17 11:38:41] DEBUG[3253] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.124:5060 [Dec 17 11:38:41] DEBUG[3295] app_queue.c: Device 'SIP/phone-c' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:41] DEBUG[3295] app_queue.c: Device 'SIP/phone-c' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:41] DEBUG[3296] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 200 Context: _extensions Hint: SIP/phone-c Status: 0 [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SIP/2.0 200 canceling Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 From: "PhoneB" ;tag=as2623879c To: ;tag=fc7bec6af74f3dd97fe73542f53baeeb-1fc8 Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 CSeq: 102 CANCEL Server: OpenSIPS (1.11.2-notls (i386/linux)) Content-Length: 0 <-------------> [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 0 [ 21]: SIP/2.0 200 canceling [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 2 [ 59]: From: "PhoneB" ;tag=as2623879c [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 3 [ 73]: To: ;tag=fc7bec6af74f3dd97fe73542f53baeeb-1fc8 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 4 [ 55]: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 102 CANCEL [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 6 [ 44]: Server: OpenSIPS (1.11.2-notls (i386/linux)) [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: --- (8 headers 0 lines) --- [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: = Looking for Call ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 (Checking To) --From tag as2623879c --To-tag fc7bec6af74f3dd97fe73542f53baeeb-1fc8 [Dec 17 11:38:41] DEBUG[3289][C-00000001] chan_sip.c: Acked pending invite 102 [Dec 17 11:38:41] DEBUG[3289][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #56 [Dec 17 11:38:41] DEBUG[3289][C-00000001] chan_sip.c: Stopping retransmission on '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' of Request 102: Match Found [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SIP/2.0 487 Terminated Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 From: "PhoneB" ;tag=as2623879c To: ;tag=7C2F8020F6BA_T1173291550 Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 Contact: CSeq: 102 INVITE User-Agent: DE900 IP PRO/60.02.00.15;7C2F8020F6BA Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO Content-Length: 0 <-------------> [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 0 [ 22]: SIP/2.0 487 Terminated [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 2 [ 59]: From: "PhoneB" ;tag=as2623879c [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 3 [ 60]: To: ;tag=7C2F8020F6BA_T1173291550 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 4 [ 55]: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 5 [ 42]: Contact: [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE900 IP PRO/60.02.00.15;7C2F8020F6BA [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 8 [ 85]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,MESSAGE,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: --- (10 headers 0 lines) --- [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: = Looking for Call ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 (Checking To) --From tag as2623879c --To-tag 7C2F8020F6BA_T1173291550 [Dec 17 11:38:41] DEBUG[3289][C-00000001] chan_sip.c: Stopping retransmission on '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' of Request 102: Match Found [Dec 17 11:38:41] DEBUG[3289][C-00000001] chan_sip.c: SIP response 487 to standard invite [Dec 17 11:38:41] DEBUG[3289][C-00000001] chan_sip.c: Strict routing enforced for session 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:38:41] VERBOSE[3289][C-00000001] chan_sip.c: Transmitting (no NAT) to 192.168.10.75:5060: ACK sip:phone-c@192.168.10.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK77148e48 Max-Forwards: 70 From: "PhoneB" ;tag=as2623879c To: ;tag=7C2F8020F6BA_T1173291550 Contact: Call-ID: 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 CSeq: 102 ACK User-Agent: IPTAM PBX (Version 20141216/6814) Content-Length: 0 --- [Dec 17 11:38:41] DEBUG[3289][C-00000001] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:41] DEBUG[3289][C-00000001] chan_sip.c: Updating call counter for outgoing call [Dec 17 11:38:41] DEBUG[3289][C-00000001] chan_sip.c: Call to peer 'phone-c' removed from call limit 2147483647 [Dec 17 11:38:41] VERBOSE[3289][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' in 32000 ms (Method: INVITE) [Dec 17 11:38:41] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-c [Dec 17 11:38:41] DEBUG[3251] chan_sip.c: Checking device state for peer phone-c [Dec 17 11:38:41] DEBUG[3251] devicestate.c: Changing state for SIP/phone-c - state 1 (Not in use) [Dec 17 11:38:41] DEBUG[3251] devicestate.c: device 'SIP/phone-c' state '1' [Dec 17 11:38:41] DEBUG[3295] app_queue.c: Device 'SIP/phone-c' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK58dda877 From: ;tag=as07820be1 To: "PhoneA" ;tag=80931a0487 Call-ID: 0dcce801bbc3a2c1 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "PhoneA" ;+sip.instance="" Server: Aastra 55i/3.3.1.2217 Supported: path, replaces Content-Type: application/sdp Content-Length: 261 v=0 o=MxSIP 0 3 IN IP4 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK58dda877 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 2 [ 60]: From: ;tag=as07820be1 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 3 [ 60]: To: "PhoneA" ;tag=80931a0487 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 4 [ 25]: Call-ID: 0dcce801bbc3a2c1 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 8 [129]: Contact: "PhoneA" ;+sip.instance="" [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 9 [ 29]: Server: Aastra 55i/3.3.1.2217 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 10 [ 25]: Supported: path, replaces [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 12 [ 19]: Content-Length: 261 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 13 [ 0]: [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 0 [ 3]: v=0 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.201 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 2 [ 10]: s=SIP Call [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Body 12 [ 10]: a=sendrecv [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: --- (13 headers 13 lines) --- [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: = Looking for Call ID: 0dcce801bbc3a2c1 (Checking To) --From tag as07820be1 --To-tag 80931a0487 [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Acked pending invite 103 [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #58 [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Stopping retransmission on '0dcce801bbc3a2c1' of Request 103: Match Found [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call 0dcce801bbc3a2c1 [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.10.201... OK. [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED OR FAILED. [Dec 17 11:38:41] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.201' into... [Dec 17 11:38:41] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.201' and port ''. [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 8 [Dec 17 11:38:41] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0xb5ba82f0 [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 0 [Dec 17 11:38:41] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0xb5ba82f0 [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Found RTP audio format 101 [Dec 17 11:38:41] DEBUG[3289][C-00000000] rtp_engine.c: Setting payload 101 based on m type on 0xb5ba82f0 [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED. [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Dec 17 11:38:41] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9759f6c' [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Dec 17 11:38:41] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 0 from 0xb5ba82f0 to 0x975a118 [Dec 17 11:38:41] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 8 from 0xb5ba82f0 to 0x975a118 [Dec 17 11:38:41] DEBUG[3289][C-00000000] rtp_engine.c: Copying payload 101 from 0xb5ba82f0 to 0x975a118 [Dec 17 11:38:41] DEBUG[3289][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x9759f6c' [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: We're settling with these formats: (ulaw|alaw) [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw) [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Updating call counter for incoming call [Dec 17 11:38:41] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.201:5060' into... [Dec 17 11:38:41] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.201' and port '5060'. [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Strict routing enforced for session 0dcce801bbc3a2c1 [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:41] DEBUG[3289][C-00000000] netsock2.c: Splitting '192.168.10.201:5060' into... [Dec 17 11:38:41] DEBUG[3289][C-00000000] netsock2.c: ...host '192.168.10.201' and port '5060'. [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Dec 17 11:38:41] VERBOSE[3289][C-00000000] chan_sip.c: Transmitting (no NAT) to 192.168.10.201:5060: ACK sip:phone-a@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK22354d90 Max-Forwards: 70 From: ;tag=as07820be1 To: "PhoneA" ;tag=80931a0487 Contact: Call-ID: 0dcce801bbc3a2c1 CSeq: 103 ACK User-Agent: IPTAM PBX (Version 20141216/6814) Content-Length: 0 --- [Dec 17 11:38:41] DEBUG[3289][C-00000000] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Dec 17 11:38:41] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-a [Dec 17 11:38:41] DEBUG[3251] chan_sip.c: Checking device state for peer phone-a [Dec 17 11:38:41] DEBUG[3251] devicestate.c: Changing state for SIP/phone-a - state 2 (In use) [Dec 17 11:38:41] DEBUG[3251] devicestate.c: device 'SIP/phone-a' state '2' [Dec 17 11:38:41] DEBUG[3295] app_queue.c: Device 'SIP/phone-a' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.124:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK531770fe From: ;tag=as08125298 To: ;tag=7C2F8008F5C1_T461496925;user=phone Call-ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 CSeq: 105 NOTIFY User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Content-Length: 0 <-------------> [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK531770fe [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 2 [ 44]: From: ;tag=as08125298 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 3 [ 70]: To: ;tag=7C2F8008F5C1_T461496925;user=phone [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 4 [ 57]: Call-ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 105 NOTIFY [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 6 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 17 11:38:41] VERBOSE[3289] chan_sip.c: --- (8 headers 0 lines) --- [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124 (Checking To) --From tag as08125298 --To-tag 7C2F8008F5C1_T461496925 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Acked pending invite 105 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #59 [Dec 17 11:38:41] DEBUG[3289] chan_sip.c: Stopping retransmission on 'CALL_ID2_7C2F8008F5C1_T1933425886@192_168_10_124' of Request 105: Match Found [Dec 17 11:38:41] DEBUG[3348][C-00000001] res_rtp_asterisk.c: 0x975eec0 -- Probation learning mode pass with source address 192.168.10.201:3000 [Dec 17 11:38:44] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Got RTCP report of 88 bytes [Dec 17 11:38:44] DEBUG[3296] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.10.201:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 201326615 FractionLost: 23 PacketsLost: 12 HighestSequence: 131 SequenceNumberCycles: 0 IAJitter: 13114 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Dec 17 11:38:45] DEBUG[3296] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 192.168.10.201:3001 OurSSRC: 795196775 SentNTP: 1418812725.3897102336 SentRTP: 160 SentPackets: 528 SentOctets: 84480 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0005 TheirLastSR: 8667791 DLSR: 1.7960 (sec) [Dec 17 11:38:45] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Dec 17 11:38:45] DEBUG[3296] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.10.124:5005 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 1929576460 FractionLost: 12 PacketsLost: 883 HighestSequence: 11529 SequenceNumberCycles: 0 IAJitter: 37 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Dec 17 11:38:48] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SUBSCRIBE sip:hans@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK72e36415dfb4a36fa Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK72e36415dfb4a36fa Max-Forwards: 69 From: "Hans Mustermann" ;tag=536ac1974e To: Call-ID: 8df0d8f279dfac42 CSeq: 27767 SUBSCRIBE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="hans",realm="asterisk",nonce="396f332f",uri="sip:hans@192.168.10.75:5060",response="a48c22ec20fe2a724321b259636adfd6",algorithm=MD5 Contact: "Hans Mustermann" ;+sip.instance="" Event: message-summary Expires: 3600 Supported: path User-Agent: Aastra 53i/3.3.1.2217 Content-Length: 0 <-------------> [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 0 [ 46]: SUBSCRIBE sip:hans@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK72e36415dfb4a36fa [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 2 [ 68]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK72e36415dfb4a36fa [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 3 [ 16]: Max-Forwards: 69 [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 4 [ 68]: From: "Hans Mustermann" ;tag=536ac1974e [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 5 [ 33]: To: [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 6 [ 25]: Call-ID: 8df0d8f279dfac42 [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 7 [ 21]: CSeq: 27767 SUBSCRIBE [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 8 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 9 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 10 [163]: Authorization: Digest username="hans",realm="asterisk",nonce="396f332f",uri="sip:hans@192.168.10.75:5060",response="a48c22ec20fe2a724321b259636adfd6",algorithm=MD5 [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 11 [135]: Contact: "Hans Mustermann" ;+sip.instance="" [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 12 [ 22]: Event: message-summary [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 14 [ 15]: Supported: path [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 15 [ 33]: User-Agent: Aastra 53i/3.3.1.2217 [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Header 16 [ 17]: Content-Length: 0 [Dec 17 11:38:48] VERBOSE[3289] chan_sip.c: --- (17 headers 0 lines) --- [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: = Looking for Call ID: 8df0d8f279dfac42 (Checking From) --From tag 536ac1974e --To-tag [Dec 17 11:38:48] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:25060' into... [Dec 17 11:38:48] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '25060'. [Dec 17 11:38:48] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:25060' into... [Dec 17 11:38:48] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '25060'. [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Got a new subscription 8df0d8f279dfac42 (possibly with auth) or retransmission [Dec 17 11:38:48] VERBOSE[3289] chan_sip.c: Creating new subscription [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 8df0d8f279dfac42 [Dec 17 11:38:48] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:48] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:48] VERBOSE[3289] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: build_route: Retaining previous route: [Dec 17 11:38:48] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:48] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:38:48] VERBOSE[3289] chan_sip.c: No matching peer for 'hans' from '192.168.10.75:5060' [Dec 17 11:38:48] NOTICE[3289] chan_sip.c: Failed to authenticate device "Hans Mustermann" ;tag=536ac1974e for SUBSCRIBE [Dec 17 11:38:48] VERBOSE[3289] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK72e36415dfb4a36fa;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK72e36415dfb4a36fa From: "Hans Mustermann" ;tag=536ac1974e To: ;tag=as2694a72d Call-ID: 8df0d8f279dfac42 CSeq: 27767 SUBSCRIBE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Trying to put 'SIP/2.0 403' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:48] DEBUG[3289] chan_sip.c: Destroying SIP dialog 8df0d8f279dfac42 [Dec 17 11:38:48] VERBOSE[3289] chan_sip.c: Really destroying SIP dialog '8df0d8f279dfac42' Method: SUBSCRIBE [Dec 17 11:38:49] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Got RTCP report of 88 bytes [Dec 17 11:38:49] DEBUG[3296] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.10.201:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 201326592 FractionLost: 0 PacketsLost: 12 HighestSequence: 381 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Dec 17 11:38:50] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Dec 17 11:38:50] DEBUG[3296] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.10.124:5005 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 1929576448 FractionLost: 0 PacketsLost: 883 HighestSequence: 11764 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Dec 17 11:38:53] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Got RTCP report of 23 bytes [Dec 17 11:38:53] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> BYE sip:*8200@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T706193FD Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T706193FD;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T343468775 To: ;tag=as5454b2c8 Call-ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 CSeq: 3 BYE User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Contact: Max-Forwards: 69 Content-Length: 0 <-------------> [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 0 [ 41]: BYE sip:*8200@192.168.10.75:25060 SIP/2.0 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T706193FD [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 2 [109]: Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T706193FD;rport=5060 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 3 [ 71]: From: " PhoneB" ;tag=7C2F8008F5C1_T343468775 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 4 [ 44]: To: ;tag=as5454b2c8 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 5 [ 57]: Call-ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 6 [ 11]: CSeq: 3 BYE [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 7 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 8 [ 42]: Contact: [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 9 [ 16]: Max-Forwards: 69 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Dec 17 11:38:53] VERBOSE[3289] chan_sip.c: --- (11 headers 0 lines) --- [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 (Checking From) --From tag 7C2F8008F5C1_T343468775 --To-tag as5454b2c8 [Dec 17 11:38:53] DEBUG[3289][C-00000002] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Dec 17 11:38:53] DEBUG[3289][C-00000002] chan_sip.c: Initializing initreq for method BYE - callid CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:38:53] DEBUG[3289][C-00000002] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:38:53] DEBUG[3289][C-00000002] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:38:53] VERBOSE[3289][C-00000002] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:38:53] DEBUG[3289][C-00000002] chan_sip.c: Setting SIP_ALREADYGONE on dialog CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:38:53] DEBUG[3289][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97c09a4' [Dec 17 11:38:53] DEBUG[3289][C-00000002] chan_sip.c: Session timer stopped: 53 - CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:38:53] VERBOSE[3289][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog 'CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124' in 32000 ms (Method: BYE) [Dec 17 11:38:53] DEBUG[3289][C-00000002] chan_sip.c: Received bye, issuing owner hangup [Dec 17 11:38:53] VERBOSE[3289][C-00000002] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK_7C2F8008F5C1_T706193FD;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.124:5060;received=192.168.10.124;branch=z9hG4bK_7C2F8008F5C1_T706193FD;rport=5060 From: " PhoneB" ;tag=7C2F8008F5C1_T343468775 To: ;tag=as5454b2c8 Call-ID: CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 CSeq: 3 BYE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Dec 17 11:38:53] DEBUG[3289][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: HangupRequest Privilege: call,all Channel: SIP/phone-b-00000004 Uniqueid: 1418812712.3 [Dec 17 11:38:53] DEBUG[3348][C-00000001] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a hangup [Dec 17 11:38:53] DEBUG[3348][C-00000001] channel.c: Returning from native bridge, channels: SIP/phone-a-00000000, SIP/phone-b-00000004 [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Unlink Bridgetype: core Channel1: SIP/phone-a-00000000 Channel2: SIP/phone-b-00000004 Uniqueid1: 1418812711.2 Uniqueid2: 1418812712.3 CallerID1: 180 CallerID2: 100 [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: BRIDGE_END AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: 180 CallerIDrdnis: CallerIDdnid: 100 Exten: 200 Context: sub-dial-intern Channel: SIP/phone-a-00000000 Application: Dial AppData: SIP/phone-c, EventTime: 2014-12-17 11:38:53 AMAFlags: DOCUMENTATION UniqueID: 1418812711.2 LinkedID: 1418812704.0 Userfield: Peer: SIP/phone-b-00000004 PeerAccount: Extra: [Dec 17 11:38:53] DEBUG[3348][C-00000001] channel.c: Hanging up channel 'SIP/phone-b-00000004' [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: Hangup call SIP/phone-b-00000004, SIP callid CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: update_call_counter(phone-b) - decrement call limit counter on hangup [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: Updating call counter for incoming call [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: Call from peer 'phone-b' removed from call limit 2147483647 [Dec 17 11:38:53] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x97c09a4' [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 180 Destination: 200 DestinationContext: sub-dial-intern CallerID: "PhoneA" <180> Channel: SIP/phone-a-00000000 DestinationChannel: SIP/phone-b-00000004 LastApplication: Dial LastData: SIP/phone-c, StartTime: 2014-12-17 11:38:34 AnswerTime: 2014-12-17 11:38:41 EndTime: 2014-12-17 11:38:53 Duration: 19 BillableSeconds: 12 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1418812704.0 UserField: D:100# [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone-b-00000004 Uniqueid: 1418812712.3 CallerIDNum: 100 CallerIDName: PhoneB ConnectedLineNum: 180 ConnectedLineName: PhoneA AccountCode: Cause: 16 Cause-txt: Normal Clearing [Dec 17 11:38:53] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:53] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:53] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 1 (Not in use) [Dec 17 11:38:53] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '1' [Dec 17 11:38:53] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:38:53] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:38:53] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 1 (Not in use) [Dec 17 11:38:53] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '1' [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: *8200 Exten: Context: sub-dial-intern Channel: SIP/phone-b-00000004 Application: AppDial AppData: (Outgoing Line) EventTime: 2014-12-17 11:38:53 AMAFlags: DOCUMENTATION UniqueID: 1418812712.3 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: 16,SIP/phone-b-00000004, [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_END AccountCode: CallerIDnum: 100 CallerIDname: PhoneB CallerIDani: 100 CallerIDrdnis: CallerIDdnid: *8200 Exten: Context: sub-dial-intern Channel: SIP/phone-b-00000004 Application: AppDial AppData: (Outgoing Line) EventTime: 2014-12-17 11:38:53 AMAFlags: DOCUMENTATION UniqueID: 1418812712.3 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:53] DEBUG[3253] app_queue.c: Extension '100@_extensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:53] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:53] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: _extensions Hint: SIP/phone-b Status: 0 [Dec 17 11:38:53] DEBUG[3348][C-00000001] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Dec 17 11:38:53] DEBUG[3348][C-00000001] pbx.c: Spawn extension (sub-dial-intern,200,6) exited non-zero on 'SIP/phone-a-00000000' [Dec 17 11:38:53] DEBUG[3348][C-00000001] channel.c: Soft-Hanging up channel 'SIP/phone-a-00000000' [Dec 17 11:38:53] DEBUG[3348][C-00000001] channel.c: Hanging up channel 'SIP/phone-a-00000000' [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: Hangup call SIP/phone-a-00000000, SIP callid 0dcce801bbc3a2c1 [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: update_call_counter(phone-a) - decrement call limit counter on hangup [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: Updating call counter for incoming call [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: Call from peer 'phone-a' removed from call limit 2147483647 [Dec 17 11:38:53] DEBUG[3348][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9759f6c' [Dec 17 11:38:53] VERBOSE[3348][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '0dcce801bbc3a2c1' in 32000 ms (Method: ACK) [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: Strict routing enforced for session 0dcce801bbc3a2c1 [Dec 17 11:38:53] VERBOSE[3348][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:53] DEBUG[3348][C-00000001] netsock2.c: Splitting '192.168.10.201:5060' into... [Dec 17 11:38:53] DEBUG[3348][C-00000001] netsock2.c: ...host '192.168.10.201' and port '5060'. [Dec 17 11:38:53] VERBOSE[3348][C-00000001] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Dec 17 11:38:53] VERBOSE[3348][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: BYE sip:phone-a@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK727051d6 Max-Forwards: 70 From: ;tag=as07820be1 To: "PhoneA" ;tag=80931a0487 Call-ID: 0dcce801bbc3a2c1 CSeq: 104 BYE User-Agent: IPTAM PBX (Version 20141216/6814) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #63 [Dec 17 11:38:53] DEBUG[3348][C-00000001] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone-a-00000000 UniqueID: 1418812711.2 DialStatus: ANSWER [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: SoftHangupRequest Privilege: call,all Channel: SIP/phone-a-00000000 Uniqueid: 1418812711.2 Cause: 16 [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone-a-00000000 Uniqueid: 1418812711.2 CallerIDNum: 180 CallerIDName: PhoneA ConnectedLineNum: 100 ConnectedLineName: PhoneB AccountCode: Cause: 16 Cause-txt: Normal Clearing [Dec 17 11:38:53] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-a [Dec 17 11:38:53] DEBUG[3251] chan_sip.c: Checking device state for peer phone-a [Dec 17 11:38:53] DEBUG[3251] devicestate.c: Changing state for SIP/phone-a - state 1 (Not in use) [Dec 17 11:38:53] DEBUG[3251] devicestate.c: device 'SIP/phone-a' state '1' [Dec 17 11:38:53] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-a [Dec 17 11:38:53] DEBUG[3251] chan_sip.c: Checking device state for peer phone-a [Dec 17 11:38:53] DEBUG[3251] devicestate.c: Changing state for SIP/phone-a - state 1 (Not in use) [Dec 17 11:38:53] DEBUG[3251] devicestate.c: device 'SIP/phone-a' state '1' [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: APP_END AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: 180 CallerIDrdnis: CallerIDdnid: 100 Exten: 200 Context: sub-dial-intern Channel: SIP/phone-a-00000000 Application: Dial AppData: SIP/phone-c, EventTime: 2014-12-17 11:38:53 AMAFlags: DOCUMENTATION UniqueID: 1418812711.2 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: 180 CallerIDrdnis: CallerIDdnid: 100 Exten: 200 Context: sub-dial-intern Channel: SIP/phone-a-00000000 Application: AppData: EventTime: 2014-12-17 11:38:53 AMAFlags: DOCUMENTATION UniqueID: 1418812711.2 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: 16,SIP/phone-b-00000004,ANSWER [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_END AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: 180 CallerIDrdnis: CallerIDdnid: 100 Exten: 200 Context: sub-dial-intern Channel: SIP/phone-a-00000000 Application: AppData: EventTime: 2014-12-17 11:38:53 AMAFlags: DOCUMENTATION UniqueID: 1418812711.2 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: CEL Privilege: call,all EventName: LINKEDID_END AccountCode: CallerIDnum: 180 CallerIDname: PhoneA CallerIDani: 180 CallerIDrdnis: CallerIDdnid: 100 Exten: 200 Context: sub-dial-intern Channel: SIP/phone-a-00000000 Application: AppData: EventTime: 2014-12-17 11:38:53 AMAFlags: DOCUMENTATION UniqueID: 1418812711.2 LinkedID: 1418812704.0 Userfield: Peer: PeerAccount: Extra: [Dec 17 11:38:53] DEBUG[3253] app_queue.c: Extension '180@_extensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:53] DEBUG[3253] chan_sip.c: Strict routing enforced for session CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 [Dec 17 11:38:53] VERBOSE[3253] chan_sip.c: set_destination: Parsing for address/port to send to [Dec 17 11:38:53] DEBUG[3253] netsock2.c: Splitting '192.168.10.124:5060' into... [Dec 17 11:38:53] DEBUG[3253] netsock2.c: ...host '192.168.10.124' and port '5060'. [Dec 17 11:38:53] VERBOSE[3253] chan_sip.c: set_destination: set destination to 192.168.10.124:5060 [Dec 17 11:38:53] VERBOSE[3253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.124:5060: NOTIFY sip:phone-b@192.168.10.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK5a56755f Max-Forwards: 70 From: ;tag=as6db3dcd7 To: ;tag=7C2F8008F5C1_T2062061876;user=phone Contact: Call-ID: CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 CSeq: 104 NOTIFY User-Agent: IPTAM PBX (Version 20141216/6814) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 204 terminated --- [Dec 17 11:38:53] DEBUG[3253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #64 [Dec 17 11:38:53] DEBUG[3253] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.124:5060 [Dec 17 11:38:53] DEBUG[3295] app_queue.c: Device 'SIP/phone-a' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:53] DEBUG[3295] app_queue.c: Device 'SIP/phone-a' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:38:53] DEBUG[3296] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: _extensions Hint: SIP/phone-a Status: 0 [Dec 17 11:38:53] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.124:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK5a56755f From: ;tag=as6db3dcd7 To: ;tag=7C2F8008F5C1_T2062061876;user=phone Call-ID: CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 CSeq: 104 NOTIFY User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 Content-Length: 0 <-------------> [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK5a56755f [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 2 [ 44]: From: ;tag=as6db3dcd7 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 3 [ 71]: To: ;tag=7C2F8008F5C1_T2062061876;user=phone [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 4 [ 57]: Call-ID: CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 5 [ 16]: CSeq: 104 NOTIFY [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 6 [ 49]: User-Agent: DE700 IP PRO/61.02.00.15;7C2F8008F5C1 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 17 11:38:53] VERBOSE[3289] chan_sip.c: --- (8 headers 0 lines) --- [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: = Looking for Call ID: CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124 (Checking To) --From tag as6db3dcd7 --To-tag 7C2F8008F5C1_T2062061876 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Acked pending invite 104 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #64 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Stopping retransmission on 'CALL_ID7_7C2F8008F5C1_T2027213775@192_168_10_124' of Request 104: Match Found [Dec 17 11:38:53] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK727051d6 From: ;tag=as07820be1 To: "PhoneA" ;tag=80931a0487 Call-ID: 0dcce801bbc3a2c1 CSeq: 104 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 55i/3.3.1.2217 Supported: path Content-Length: 0 <-------------> [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK727051d6 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 2 [ 60]: From: ;tag=as07820be1 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 3 [ 60]: To: "PhoneA" ;tag=80931a0487 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 4 [ 25]: Call-ID: 0dcce801bbc3a2c1 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 8 [ 29]: Server: Aastra 55i/3.3.1.2217 [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 9 [ 15]: Supported: path [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Dec 17 11:38:53] VERBOSE[3289] chan_sip.c: --- (11 headers 0 lines) --- [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: = Looking for Call ID: 0dcce801bbc3a2c1 (Checking To) --From tag as07820be1 --To-tag 80931a0487 [Dec 17 11:38:53] DEBUG[3289][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #63 [Dec 17 11:38:53] DEBUG[3289][C-00000000] chan_sip.c: Stopping retransmission on '0dcce801bbc3a2c1' of Request 104: Match Found [Dec 17 11:38:53] DEBUG[3289] chan_sip.c: Destroying SIP dialog 0dcce801bbc3a2c1 [Dec 17 11:38:53] VERBOSE[3289] chan_sip.c: Really destroying SIP dialog '0dcce801bbc3a2c1' Method: ACK [Dec 17 11:38:53] DEBUG[3289] rtp_engine.c: Destroyed RTP instance '0x9759f6c' [Dec 17 11:39:06] DEBUG[3289] chan_sip.c: Auto destroying SIP dialog 'CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124' [Dec 17 11:39:06] DEBUG[3289] chan_sip.c: Session timer stopped: -1 - CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 [Dec 17 11:39:06] DEBUG[3289] chan_sip.c: Destroying SIP dialog CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124 [Dec 17 11:39:06] VERBOSE[3289] chan_sip.c: Really destroying SIP dialog 'CALL_ID12_7C2F8008F5C1_T163313201@192_168_10_124' Method: ACK [Dec 17 11:39:06] DEBUG[3289] rtp_engine.c: Destroyed RTP instance '0x97932dc' [Dec 17 11:39:07] DEBUG[3289] chan_sip.c: Auto destroying SIP dialog '21bcfc8b3931cfce66d332554203c767@192.168.10.75' [Dec 17 11:39:07] DEBUG[3289] chan_sip.c: Destroying SIP dialog 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:39:07] VERBOSE[3289] chan_sip.c: Really destroying SIP dialog '21bcfc8b3931cfce66d332554203c767@192.168.10.75' Method: BYE [Dec 17 11:39:07] DEBUG[3289] chan_sip.c: Updating call counter for outgoing call [Dec 17 11:39:07] DEBUG[3289] chan_sip.c: Call to peer 'phone-b' removed from call limit 2147483647 [Dec 17 11:39:07] DEBUG[3289] chan_sip.c: This call did not properly clean up call limits. Call ID 21bcfc8b3931cfce66d332554203c767@192.168.10.75 [Dec 17 11:39:07] DEBUG[3289] rtp_engine.c: Destroyed RTP instance '0x97662e4' [Dec 17 11:39:07] DEBUG[3251] devicestate.c: No provider found, checking channel drivers for SIP - phone-b [Dec 17 11:39:07] DEBUG[3251] chan_sip.c: Checking device state for peer phone-b [Dec 17 11:39:07] DEBUG[3251] devicestate.c: Changing state for SIP/phone-b - state 1 (Not in use) [Dec 17 11:39:07] DEBUG[3251] devicestate.c: device 'SIP/phone-b' state '1' [Dec 17 11:39:07] DEBUG[3295] app_queue.c: Device 'SIP/phone-b' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 17 11:39:13] DEBUG[3289] chan_sip.c: Auto destroying SIP dialog '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' [Dec 17 11:39:13] DEBUG[3289] chan_sip.c: Destroying SIP dialog 3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75 [Dec 17 11:39:13] VERBOSE[3289] chan_sip.c: Really destroying SIP dialog '3d7c3a6b5bc0cd711bfed09262ea73e2@192.168.10.75' Method: INVITE [Dec 17 11:39:13] DEBUG[3289] rtp_engine.c: Destroyed RTP instance '0x97aa8dc' [Dec 17 11:39:18] VERBOSE[3289] chan_sip.c: <--- SIP read from UDP:192.168.10.75:5060 ---> SUBSCRIBE sip:hans@192.168.10.75:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK72a68d9e0bc35fb27 Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK72a68d9e0bc35fb27 Max-Forwards: 69 From: "Hans Mustermann" ;tag=536ac1974e To: Call-ID: 8df0d8f279dfac42 CSeq: 27768 SUBSCRIBE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="hans",realm="asterisk",nonce="396f332f",uri="sip:hans@192.168.10.75:5060",response="a48c22ec20fe2a724321b259636adfd6",algorithm=MD5 Contact: "Hans Mustermann" ;+sip.instance="" Event: message-summary Expires: 3600 Supported: path User-Agent: Aastra 53i/3.3.1.2217 Content-Length: 0 <-------------> [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 0 [ 46]: SUBSCRIBE sip:hans@192.168.10.75:25060 SIP/2.0 [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK72a68d9e0bc35fb27 [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 2 [ 68]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK72a68d9e0bc35fb27 [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 3 [ 16]: Max-Forwards: 69 [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 4 [ 68]: From: "Hans Mustermann" ;tag=536ac1974e [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 5 [ 33]: To: [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 6 [ 25]: Call-ID: 8df0d8f279dfac42 [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 7 [ 21]: CSeq: 27768 SUBSCRIBE [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 8 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 9 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 10 [163]: Authorization: Digest username="hans",realm="asterisk",nonce="396f332f",uri="sip:hans@192.168.10.75:5060",response="a48c22ec20fe2a724321b259636adfd6",algorithm=MD5 [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 11 [135]: Contact: "Hans Mustermann" ;+sip.instance="" [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 12 [ 22]: Event: message-summary [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 14 [ 15]: Supported: path [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 15 [ 33]: User-Agent: Aastra 53i/3.3.1.2217 [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Header 16 [ 17]: Content-Length: 0 [Dec 17 11:39:18] VERBOSE[3289] chan_sip.c: --- (17 headers 0 lines) --- [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: = Looking for Call ID: 8df0d8f279dfac42 (Checking From) --From tag 536ac1974e --To-tag [Dec 17 11:39:18] DEBUG[3289] acl.c: For destination '192.168.10.75', our source address is '192.168.10.75'. [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:25060 [Dec 17 11:39:18] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:39:18] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:39:18] VERBOSE[3289] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Allocating new SIP dialog for 8df0d8f279dfac42 - SUBSCRIBE (No RTP) [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Dec 17 11:39:18] VERBOSE[3289] chan_sip.c: Creating new subscription [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 8df0d8f279dfac42 [Dec 17 11:39:18] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:39:18] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port '5060'. [Dec 17 11:39:18] VERBOSE[3289] chan_sip.c: Sending to 192.168.10.75:5060 (no NAT) [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: build_route: Contact hop: "Hans Mustermann" ;+sip.instance="" [Dec 17 11:39:18] VERBOSE[3289] chan_sip.c: list_route: hop: [Dec 17 11:39:18] DEBUG[3289] netsock2.c: Splitting '192.168.10.75:5060' into... [Dec 17 11:39:18] DEBUG[3289] netsock2.c: ...host '192.168.10.75' and port ''. [Dec 17 11:39:18] VERBOSE[3289] chan_sip.c: No matching peer for 'hans' from '192.168.10.75:5060' [Dec 17 11:39:18] VERBOSE[3289] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.75:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK72a68d9e0bc35fb27;received=192.168.10.75 Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK72a68d9e0bc35fb27 From: "Hans Mustermann" ;tag=536ac1974e To: ;tag=as7a19aaec Call-ID: 8df0d8f279dfac42 CSeq: 27768 SUBSCRIBE Server: IPTAM PBX (Version 20141216/6814) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7dbe23f6" Content-Length: 0 <------------> [Dec 17 11:39:18] DEBUG[3289] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.75:5060 [Dec 17 11:39:18] VERBOSE[3289] chan_sip.c: Scheduling destruction of SIP dialog '8df0d8f279dfac42' in 32000 ms (Method: SUBSCRIBE) [Dec 17 11:39:25] DEBUG[3289] chan_sip.c: Auto destroying SIP dialog 'CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124' [Dec 17 11:39:25] DEBUG[3289] chan_sip.c: Destroying SIP dialog CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124 [Dec 17 11:39:25] VERBOSE[3289] chan_sip.c: Really destroying SIP dialog 'CALL_ID13_7C2F8008F5C1_T751186558@192_168_10_124' Method: BYE [Dec 17 11:39:25] DEBUG[3289] rtp_engine.c: Destroyed RTP instance '0x97c09a4'