[Aug 15 18:38:12] VERBOSE[5271] config.c: == Parsing '/etc/asterisk/logger.conf': [Aug 15 18:38:12] DEBUG[5271] config.c: Parsing /etc/asterisk/logger.conf [Aug 15 18:38:12] VERBOSE[5271] config.c: == Found [Aug 15 18:38:12] VERBOSE[5271] logger.c: Asterisk Queue Logger restarted [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.209:5060 ---> INVITE sip:106@192.168.10.75:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK08a0c3a448db20d56.c9c6ff2d4112bce6b Max-Forwards: 70 From: "user 3" ;tag=f1865c1211 To: Call-ID: 848bff54a41b66ee CSeq: 24500 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "user 3" ;+sip.instance="" Supported: path, 100rel, replaces User-Agent: Aastra 53i/3.2.1.43 Content-Type: application/sdp Content-Length: 634 v=0 o=MxSIP 0 1 IN IP4 192.168.10.209 s=SIP Call c=IN IP4 192.168.10.209 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:101 0-15 a=ptime:30 a=rtcp:3001 IN IP4 192.168.10.209 a=sendrecv <-------------> [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 0 [ 52]: INVITE sip:106@192.168.10.75:5060;user=phone SIP/2.0 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 1 [ 81]: Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK08a0c3a448db20d56.c9c6ff2d4112bce6b [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 3 [ 61]: From: "user 3" ;tag=f1865c1211 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 4 [ 43]: To: [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 5 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 6 [ 18]: CSeq: 24500 INVITE [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 9 [128]: Contact: "user 3" ;+sip.instance="" [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 10 [ 33]: Supported: path, 100rel, replaces [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 53i/3.2.1.43 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 13 [ 19]: Content-Length: 634 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 14 [ 0]: [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.209 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 2 [ 10]: s=SIP Call [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.209 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:27] DEBUG[5261] 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 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 21 [ 25]: a=silenceSupp:off - - - - [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 22 [ 15]: a=fmtp:101 0-15 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 23 [ 10]: a=ptime:30 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 24 [ 33]: a=rtcp:3001 IN IP4 192.168.10.209 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Body 25 [ 10]: a=sendrecv [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: --- (14 headers 26 lines) --- [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: = Looking for Call ID: 848bff54a41b66ee (Checking From) --From tag f1865c1211 --To-tag [Aug 15 18:38:27] DEBUG[5261] acl.c: For destination '192.168.10.209', our source address is '192.168.10.75'. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Allocating new SIP dialog for 848bff54a41b66ee - INVITE (No RTP) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 15 18:38:27] DEBUG[5261] sip/reqresp_parser.c: Begin: parsing SIP "Supported: path, 100rel, replaces" [Aug 15 18:38:27] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -path- [Aug 15 18:38:27] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: path [Aug 15 18:38:27] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -100rel- [Aug 15 18:38:27] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: 100rel [Aug 15 18:38:27] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -replaces- [Aug 15 18:38:27] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: replaces [Aug 15 18:38:27] DEBUG[5261] netsock2.c: Splitting '192.168.10.209' into... [Aug 15 18:38:27] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port ''. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.209:5060 (no NAT) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Initializing initreq for method INVITE - callid 848bff54a41b66ee [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Using INVITE request as basis request - 848bff54a41b66ee [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found peer 'phone3' for 'phone3' from 192.168.10.209:5060 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9062008' [Aug 15 18:38:27] DEBUG[5261] res_rtp_asterisk.c: Allocated port 15448 for RTP instance '0x9062008' [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: RTP instance '0x9062008' is setup and ready to go [Aug 15 18:38:27] DEBUG[5261] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9062008' [Aug 15 18:38:27] VERBOSE[5261] netsock2.c: == Using SIP RTP TOS bits 184 [Aug 15 18:38:27] VERBOSE[5261] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Setting NAT on RTP to Off [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.209... UNSUPPORTED. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Aug 15 18:38:27] DEBUG[5261] netsock2.c: Splitting '192.168.10.209' into... [Aug 15 18:38:27] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port ''. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.209... OK. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 0 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 0 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 18 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 18 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 106 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 106 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 107 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 113 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 110 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 110 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 111 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 111 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 112 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 112 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 98 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 98 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 97 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 97 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 115 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 115 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 96 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 9 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 9 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 8 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 8 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found RTP audio format 101 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Setting payload 101 based on m type on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found audio description format G729 for ID 18 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Unsetting payload 106 on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found unknown media description format BV16 for ID 106 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... UNSUPPORTED. [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Unsetting payload 107 on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found unknown media description format BV32 for ID 107 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... UNSUPPORTED. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found audio description format L16 for ID 113 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Unsetting payload 110 on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found unknown media description format PCMU for ID 110 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... UNSUPPORTED. [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Unsetting payload 111 on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found unknown media description format PCMA for ID 111 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... UNSUPPORTED. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found audio description format L16 for ID 112 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Unsetting payload 98 on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found unknown media description format G726-16 for ID 98 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... UNSUPPORTED. [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Unsetting payload 97 on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found unknown media description format G726-24 for ID 97 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... UNSUPPORTED. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found audio description format G726-32 for ID 115 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Unsetting payload 96 on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found unknown media description format G726-40 for ID 96 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... UNSUPPORTED. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found audio description format G722 for ID 9 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtcp:3001 IN IP4 192.168.10.209... UNSUPPORTED. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Incorporating payload 0 on 0xb616bf88 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Incorporating payload 8 on 0xb616bf88 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Incorporating payload 9 on 0xb616bf88 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Incorporating payload 18 on 0xb616bf88 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Incorporating payload 101 on 0xb616bf88 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Incorporating payload 112 on 0xb616bf88 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Incorporating payload 113 on 0xb616bf88 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Incorporating payload 115 on 0xb616bf88 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x994c (ulaw|alaw|g726|slin|g729|g722|slin16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 15 18:38:27] DEBUG[5261] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9062008' [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Peer audio RTP is at port 192.168.10.209:3000 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Copying payload 0 from 0xb616bf88 to 0x90621b4 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Copying payload 8 from 0xb616bf88 to 0x90621b4 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Copying payload 9 from 0xb616bf88 to 0x90621b4 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Copying payload 18 from 0xb616bf88 to 0x90621b4 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Copying payload 101 from 0xb616bf88 to 0x90621b4 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Copying payload 112 from 0xb616bf88 to 0x90621b4 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Copying payload 113 from 0xb616bf88 to 0x90621b4 [Aug 15 18:38:27] DEBUG[5261] rtp_engine.c: Copying payload 115 from 0xb616bf88 to 0x90621b4 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Checking SIP call limits for device phone3 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Updating call counter for incoming call [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Call from peer 'phone3' is 1 out of 2147483647 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: Looking for 106 in Standard (domain 192.168.10.75:5060) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: *** Our native formats are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: This channel will not be able to handle video. [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: build_route: Contact hop: "user 3" ;+sip.instance="" [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: list_route: hop: [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: SIP/phone3-00000003: New call is still down.... Trying... [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.209:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK08a0c3a448db20d56.c9c6ff2d4112bce6b;received=192.168.10.209 From: "user 3" ;tag=f1865c1211 To: Call-ID: 848bff54a41b66ee CSeq: 24500 INVITE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 1 [105]: Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK08a0c3a448db20d56.c9c6ff2d4112bce6b;received=192.168.10.209 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 2 [ 61]: From: "user 3" ;tag=f1865c1211 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 3 [ 43]: To: [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 4 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 5 [ 18]: CSeq: 24500 INVITE [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 9 [ 37]: Contact: [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 11 [ 0]: [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:38:27] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Aug 15 18:38:27] DEBUG[5240] chan_sip.c: Checking device state for peer phone3 [Aug 15 18:38:27] DEBUG[5240] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Aug 15 18:38:27] DEBUG[5240] devicestate.c: device 'SIP/phone3' state '2' [Aug 15 18:38:27] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Aug 15 18:38:27] DEBUG[5240] chan_sip.c: Checking device state for peer phone3 [Aug 15 18:38:27] DEBUG[5240] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Aug 15 18:38:27] DEBUG[5240] devicestate.c: device 'SIP/phone3' state '2' [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone3-00000003 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 103 CallerIDName: User 3 AccountCode: Exten: 106 Context: Standard Uniqueid: 1313426307.3 [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000003 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 103 CallerIDName: User 3 ConnectedLineNum: ConnectedLineName: Uniqueid: 1313426307.3 [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 103 CallerIDname: User 3 CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: 106 Context: Standard Channel: SIP/phone3-00000003 Application: AppData: EventTime: 2011-08-15 18:38:27 AMAFlags: DOCUMENTATION UniqueID: 1313426307.3 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:27] DEBUG[5291] pbx.c: Launching 'Dial' [Aug 15 18:38:27] VERBOSE[5291] pbx.c: -- Executing [106@Standard:1] Dial("SIP/phone3-00000003", "SIP/phone1&SIP/phone2") in new stack [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Allocating new SIP dialog for 283bc7562720ff8a20b07a0352ee930d@192.168.10.75 - INVITE (No RTP) [Aug 15 18:38:27] DEBUG[5291] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9052e60' [Aug 15 18:38:27] DEBUG[5291] res_rtp_asterisk.c: Allocated port 18176 for RTP instance '0x9052e60' [Aug 15 18:38:27] DEBUG[5291] rtp_engine.c: RTP instance '0x9052e60' is setup and ready to go [Aug 15 18:38:27] DEBUG[5291] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9052e60' [Aug 15 18:38:27] VERBOSE[5291] netsock2.c: == Using SIP RTP TOS bits 184 [Aug 15 18:38:27] VERBOSE[5291] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Setting NAT on RTP to Off [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 15 18:38:27] DEBUG[5291] acl.c: For destination '192.168.10.207', our source address is '192.168.10.75'. [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** Our native formats are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: This channel will not be able to handle video. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable DIALEDTIME. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable ANSWEREDTIME. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable DIALEDPEERNAME. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable DIALEDPEERNUMBER. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable DIALSTATUS. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable SIPCALLID. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable SIPDOMAIN. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable SIPURI. [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Outgoing Call for phone1 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Updating call counter for outgoing call [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Call to peer 'phone1' is 1 out of 2147483647 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Audio is at 5060 [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: -- Done with adding codecs to SDP [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Initializing initreq for method INVITE - callid 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.207:2048;line=4dsger5q SIP/2.0 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 3 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 4 [ 50]: To: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 6 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 9 [ 35]: Date: Mon, 15 Aug 2011 16:38:27 GMT [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 12 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.207:2048: INVITE sip:phone1@192.168.10.207:2048;line=4dsger5q SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 Max-Forwards: 70 From: "User 3" ;tag=as1426c059 To: Contact: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 INVITE User-Agent: IPTAM PBX Date: Mon, 15 Aug 2011 16:38:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "User 3" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1925556413 1925556413 IN IP4 192.168.10.75 s=Asterisk PBX 1.8.5-2 c=IN IP4 192.168.10.75 t=0 0 m=audio 18176 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 --- [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.207:2048;line=4dsger5q SIP/2.0 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 3 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 4 [ 50]: To: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 6 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 9 [ 35]: Date: Mon, 15 Aug 2011 16:38:27 GMT [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 12 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 14 [ 19]: Content-Length: 262 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 15 [ 0]: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 1 [ 49]: o=root 1925556413 1925556413 IN IP4 192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.5-2 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 5 [ 29]: m=audio 18176 RTP/AVP 8 0 101 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #155 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.207:2048 [Aug 15 18:38:27] VERBOSE[5291] app_dial.c: -- Called SIP/phone1 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Allocating new SIP dialog for 02d3fd0a1984f33911cd691b70487b00@192.168.10.75 - INVITE (No RTP) [Aug 15 18:38:27] DEBUG[5291] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9078ed8' [Aug 15 18:38:27] DEBUG[5291] res_rtp_asterisk.c: Allocated port 16524 for RTP instance '0x9078ed8' [Aug 15 18:38:27] DEBUG[5291] rtp_engine.c: RTP instance '0x9078ed8' is setup and ready to go [Aug 15 18:38:27] DEBUG[5291] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9078ed8' [Aug 15 18:38:27] VERBOSE[5291] netsock2.c: == Using SIP RTP TOS bits 184 [Aug 15 18:38:27] VERBOSE[5291] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Setting NAT on RTP to Off [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Aug 15 18:38:27] DEBUG[5291] acl.c: For destination '192.168.10.208', our source address is '192.168.10.75'. [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** Our native formats are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: This channel will not be able to handle video. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable DIALEDTIME. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable ANSWEREDTIME. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable DIALEDPEERNAME. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable DIALEDPEERNUMBER. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable DIALSTATUS. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable SIPCALLID. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable SIPDOMAIN. [Aug 15 18:38:27] DEBUG[5291] channel.c: Not copying variable SIPURI. [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Outgoing Call for phone2 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Updating call counter for outgoing call [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Call to peer 'phone2' is 1 out of 2147483647 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Audio is at 5060 [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: -- Done with adding codecs to SDP [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Initializing initreq for method INVITE - callid 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.208:5060;transport=udp SIP/2.0 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 3 [ 53]: From: "User 3" ;tag=as59eca913 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 4 [ 50]: To: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 6 [ 55]: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 9 [ 35]: Date: Mon, 15 Aug 2011 16:38:27 GMT [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 12 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.208:5060: INVITE sip:phone2@192.168.10.208:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 Max-Forwards: 70 From: "User 3" ;tag=as59eca913 To: Contact: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 CSeq: 102 INVITE User-Agent: IPTAM PBX Date: Mon, 15 Aug 2011 16:38:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "User 3" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1508918561 1508918561 IN IP4 192.168.10.75 s=Asterisk PBX 1.8.5-2 c=IN IP4 192.168.10.75 t=0 0 m=audio 16524 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 --- [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.208:5060;transport=udp SIP/2.0 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 3 [ 53]: From: "User 3" ;tag=as59eca913 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 4 [ 50]: To: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 6 [ 55]: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 9 [ 35]: Date: Mon, 15 Aug 2011 16:38:27 GMT [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 12 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 14 [ 19]: Content-Length: 262 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 15 [ 0]: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 1 [ 49]: o=root 1508918561 1508918561 IN IP4 192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.5-2 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.75 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 5 [ 29]: m=audio 16524 RTP/AVP 8 0 101 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #157 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.208:5060 [Aug 15 18:38:27] VERBOSE[5291] app_dial.c: -- Called SIP/phone2 [Aug 15 18:38:27] VERBOSE[5291] app_dial.c: -- SIP/phone1-00000004 connected line has changed. Saving it until answer for SIP/phone3-00000003 [Aug 15 18:38:27] VERBOSE[5291] app_dial.c: -- SIP/phone2-00000005 connected line has changed. Saving it until answer for SIP/phone3-00000003 [Aug 15 18:38:27] DEBUG[5241] app_queue.c: Extension '103@_extensions' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:27] DEBUG[5267] app_queue.c: Device 'SIP/phone3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:27] DEBUG[5267] app_queue.c: Device 'SIP/phone3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone1-00000004 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 101 CallerIDName: User 1 AccountCode: Exten: Context: Standard Uniqueid: 1313426307.4 [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone3-00000003 Destination: SIP/phone1-00000004 CallerIDNum: 103 CallerIDName: User 3 ConnectedLineNum: ConnectedLineName: UniqueID: 1313426307.3 DestUniqueID: 1313426307.4 Dialstring: phone1 [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-00000005 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 102 CallerIDName: User 2 AccountCode: Exten: Context: Standard Uniqueid: 1313426307.5 [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone3-00000003 Destination: SIP/phone2-00000005 CallerIDNum: 103 CallerIDName: User 3 ConnectedLineNum: ConnectedLineName: UniqueID: 1313426307.3 DestUniqueID: 1313426307.5 Dialstring: phone2 [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 103 Context: _extensions Hint: SIP/phone3 Status: 1 [Aug 15 18:38:27] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Aug 15 18:38:27] DEBUG[5240] chan_sip.c: Checking device state for peer phone1 [Aug 15 18:38:27] DEBUG[5240] devicestate.c: Changing state for SIP/phone1 - state 6 (Ringing) [Aug 15 18:38:27] DEBUG[5240] devicestate.c: device 'SIP/phone1' state '6' [Aug 15 18:38:27] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Aug 15 18:38:27] DEBUG[5240] chan_sip.c: Checking device state for peer phone2 [Aug 15 18:38:27] DEBUG[5240] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Aug 15 18:38:27] DEBUG[5240] devicestate.c: device 'SIP/phone2' state '6' [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: APP_START AccountCode: CallerIDnum: 103 CallerIDname: User 3 CallerIDani: 103 CallerIDrdnis: CallerIDdnid: 106 Exten: 106 Context: Standard Channel: SIP/phone3-00000003 Application: Dial AppData: SIP/phone1&SIP/phone2 EventTime: 2011-08-15 18:38:27 AMAFlags: DOCUMENTATION UniqueID: 1313426307.3 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 101 CallerIDname: User 1 CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: s Context: Standard Channel: SIP/phone1-00000004 Application: AppData: EventTime: 2011-08-15 18:38:27 AMAFlags: DOCUMENTATION UniqueID: 1313426307.4 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 102 CallerIDname: User 2 CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: s Context: Standard Channel: SIP/phone2-00000005 Application: AppData: EventTime: 2011-08-15 18:38:27 AMAFlags: DOCUMENTATION UniqueID: 1313426307.5 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:27] DEBUG[5241] app_queue.c: Extension '101@_extensions' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Aug 15 18:38:27] DEBUG[5241] chan_sip.c: Strict routing enforced for session 3c2670e67e76-t62560t3kg7w [Aug 15 18:38:27] VERBOSE[5241] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:27] DEBUG[5241] netsock2.c: Splitting '192.168.10.203:2053' into... [Aug 15 18:38:27] DEBUG[5241] netsock2.c: ...host '192.168.10.203' and port '2053'. [Aug 15 18:38:27] VERBOSE[5241] chan_sip.c: set_destination: set destination to 192.168.10.203:2053 [Aug 15 18:38:27] VERBOSE[5241] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:2053: NOTIFY sip:phone5@192.168.10.203:2053;line=ixa0bhmu SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK632028ec;rport Max-Forwards: 70 From: ;tag=as08d89766 To: ;tag=gij4ob7fro Contact: Call-ID: 3c2670e67e76-t62560t3kg7w CSeq: 103 NOTIFY User-Agent: IPTAM PBX Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 521 sip:101@192.168.10.75 sip:101@192.168.10.75 early --- [Aug 15 18:38:27] DEBUG[5241] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #159 [Aug 15 18:38:27] DEBUG[5241] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.203:2053 [Aug 15 18:38:27] VERBOSE[5241] chan_sip.c: == Extension Changed 101[_extensions] new state Ringing for Notify User phone5 [Aug 15 18:38:27] DEBUG[5241] app_queue.c: Extension '102@_extensions' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Aug 15 18:38:27] DEBUG[5241] chan_sip.c: Strict routing enforced for session 3c2670e6785d-xzn3ch211ngt [Aug 15 18:38:27] VERBOSE[5241] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:27] DEBUG[5241] netsock2.c: Splitting '192.168.10.203:2053' into... [Aug 15 18:38:27] DEBUG[5241] netsock2.c: ...host '192.168.10.203' and port '2053'. [Aug 15 18:38:27] VERBOSE[5241] chan_sip.c: set_destination: set destination to 192.168.10.203:2053 [Aug 15 18:38:27] VERBOSE[5241] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:2053: NOTIFY sip:phone5@192.168.10.203:2053;line=ixa0bhmu SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK1f28f11c;rport Max-Forwards: 70 From: ;tag=as193adead To: ;tag=86yhofjdv1 Contact: Call-ID: 3c2670e6785d-xzn3ch211ngt CSeq: 103 NOTIFY User-Agent: IPTAM PBX Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 521 sip:102@192.168.10.75 sip:102@192.168.10.75 early --- [Aug 15 18:38:27] DEBUG[5241] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #160 [Aug 15 18:38:27] DEBUG[5241] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.203:2053 [Aug 15 18:38:27] VERBOSE[5241] chan_sip.c: == Extension Changed 102[_extensions] new state Ringing for Notify User phone5 [Aug 15 18:38:27] DEBUG[5267] app_queue.c: Device 'SIP/phone1' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Aug 15 18:38:27] DEBUG[5267] app_queue.c: Device 'SIP/phone2' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 101 Context: _extensions Hint: SIP/phone1 Status: 8 [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 102 Context: _extensions Hint: SIP/phone2 Status: 8 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.207:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 From: "User 3" ;tag=as1426c059 To: ;tag=2el7dj9iuz Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=2el7dj9iuz [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: --- (10 headers 0 lines) --- [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: = Looking for Call ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 (Checking To) --From tag as1426c059 --To-tag 2el7dj9iuz [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: *** SIP TIMER: Cancelling retransmission #155 - INVITE (got response) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' Request 102: Found [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: SIP response 180 to standard invite [Aug 15 18:38:27] VERBOSE[5291] app_dial.c: -- SIP/phone1-00000004 is ringing [Aug 15 18:38:27] VERBOSE[5291] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.209:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK08a0c3a448db20d56.c9c6ff2d4112bce6b;received=192.168.10.209 From: "user 3" ;tag=f1865c1211 To: ;tag=as5b35b701 Call-ID: 848bff54a41b66ee CSeq: 24500 INVITE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 1 [105]: Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK08a0c3a448db20d56.c9c6ff2d4112bce6b;received=192.168.10.209 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 2 [ 61]: From: "user 3" ;tag=f1865c1211 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 3 [ 58]: To: ;tag=as5b35b701 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 4 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 5 [ 18]: CSeq: 24500 INVITE [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 9 [ 37]: Contact: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Header 11 [ 0]: [Aug 15 18:38:27] DEBUG[5291] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:38:27] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Aug 15 18:38:27] DEBUG[5240] chan_sip.c: Checking device state for peer phone1 [Aug 15 18:38:27] DEBUG[5240] devicestate.c: Changing state for SIP/phone1 - state 6 (Ringing) [Aug 15 18:38:27] DEBUG[5240] devicestate.c: device 'SIP/phone1' state '6' [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-00000004 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 101 CallerIDName: User 1 ConnectedLineNum: 103 ConnectedLineName: User 3 Uniqueid: 1313426307.4 [Aug 15 18:38:27] DEBUG[5267] app_queue.c: Device 'SIP/phone1' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2053 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK632028ec;rport=5060 From: ;tag=as08d89766 To: ;tag=gij4ob7fro Call-ID: 3c2670e67e76-t62560t3kg7w CSeq: 103 NOTIFY Content-Length: 0 <-------------> [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK632028ec;rport=5060 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 2 [ 55]: From: ;tag=as08d89766 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 3 [ 45]: To: ;tag=gij4ob7fro [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2670e67e76-t62560t3kg7w [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 103 NOTIFY [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: --- (7 headers 0 lines) --- [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2670e67e76-t62560t3kg7w (Checking To) --From tag as08d89766 --To-tag gij4ob7fro [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Acked pending invite 103 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #159 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Stopping retransmission on '3c2670e67e76-t62560t3kg7w' of Request 103: Match Found [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.208:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 From: "User 3" ;tag=as59eca913 To: ;tag=1190011780 Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "User 2" ;+sip.instance="" Server: Aastra 51i/2.6.0.2019 Supported: gruu, path Content-Length: 0 <-------------> [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as59eca913 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=1190011780 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 8 [128]: Contact: "User 2" ;+sip.instance="" [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.2019 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: --- (12 headers 0 lines) --- [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: = Looking for Call ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 (Checking To) --From tag as59eca913 --To-tag 1190011780 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: *** SIP TIMER: Cancelling retransmission #157 - INVITE (got response) [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '38a269487dde38005bbd0a773e00a253@192.168.10.75' Request 102: Found [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: SIP response 180 to standard invite [Aug 15 18:38:27] VERBOSE[5291] app_dial.c: -- SIP/phone2-00000005 is ringing [Aug 15 18:38:27] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Aug 15 18:38:27] DEBUG[5240] chan_sip.c: Checking device state for peer phone2 [Aug 15 18:38:27] DEBUG[5240] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Aug 15 18:38:27] DEBUG[5240] devicestate.c: device 'SIP/phone2' state '6' [Aug 15 18:38:27] DEBUG[5268] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000005 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 102 CallerIDName: User 2 ConnectedLineNum: 103 ConnectedLineName: User 3 Uniqueid: 1313426307.5 [Aug 15 18:38:27] DEBUG[5267] app_queue.c: Device 'SIP/phone2' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2053 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK1f28f11c;rport=5060 From: ;tag=as193adead To: ;tag=86yhofjdv1 Call-ID: 3c2670e6785d-xzn3ch211ngt CSeq: 103 NOTIFY Content-Length: 0 <-------------> [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK1f28f11c;rport=5060 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 2 [ 55]: From: ;tag=as193adead [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 3 [ 45]: To: ;tag=86yhofjdv1 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2670e6785d-xzn3ch211ngt [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 103 NOTIFY [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: --- (7 headers 0 lines) --- [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2670e6785d-xzn3ch211ngt (Checking To) --From tag as193adead --To-tag 86yhofjdv1 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Acked pending invite 103 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #160 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Stopping retransmission on '3c2670e6785d-xzn3ch211ngt' of Request 103: Match Found [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.207:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 From: "User 3" ;tag=as1426c059 To: ;tag=2el7dj9iuz Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=2el7dj9iuz [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:27] VERBOSE[5261] chan_sip.c: --- (10 headers 0 lines) --- [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: = Looking for Call ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 (Checking To) --From tag as1426c059 --To-tag 2el7dj9iuz [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' Request 102: Found [Aug 15 18:38:27] DEBUG[5261] chan_sip.c: SIP response 180 to standard invite [Aug 15 18:38:27] VERBOSE[5291] app_dial.c: -- SIP/phone1-00000004 is ringing [Aug 15 18:38:28] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.207:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 From: "User 3" ;tag=as1426c059 To: ;tag=2el7dj9iuz Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=2el7dj9iuz [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:28] VERBOSE[5261] chan_sip.c: --- (10 headers 0 lines) --- [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: = Looking for Call ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 (Checking To) --From tag as1426c059 --To-tag 2el7dj9iuz [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' Request 102: Found [Aug 15 18:38:28] DEBUG[5261] chan_sip.c: SIP response 180 to standard invite [Aug 15 18:38:28] VERBOSE[5291] app_dial.c: -- SIP/phone1-00000004 is ringing [Aug 15 18:38:30] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.207:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 From: "User 3" ;tag=as1426c059 To: ;tag=2el7dj9iuz Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=2el7dj9iuz [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:30] VERBOSE[5261] chan_sip.c: --- (10 headers 0 lines) --- [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: = Looking for Call ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 (Checking To) --From tag as1426c059 --To-tag 2el7dj9iuz [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' Request 102: Found [Aug 15 18:38:30] DEBUG[5261] chan_sip.c: SIP response 180 to standard invite [Aug 15 18:38:30] VERBOSE[5291] app_dial.c: -- SIP/phone1-00000004 is ringing [Aug 15 18:38:33] DEBUG[5261] chan_sip.c: Auto destroying SIP dialog '6b06460c523fb0c1' [Aug 15 18:38:33] DEBUG[5261] chan_sip.c: Destroying SIP dialog 6b06460c523fb0c1 [Aug 15 18:38:33] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '6b06460c523fb0c1' Method: REGISTER [Aug 15 18:38:35] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.207:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 From: "User 3" ;tag=as1426c059 To: ;tag=2el7dj9iuz Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=2el7dj9iuz [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 8 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:35] VERBOSE[5261] chan_sip.c: --- (10 headers 0 lines) --- [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: = Looking for Call ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 (Checking To) --From tag as1426c059 --To-tag 2el7dj9iuz [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' Request 102: Found [Aug 15 18:38:35] DEBUG[5261] chan_sip.c: SIP response 180 to standard invite [Aug 15 18:38:35] VERBOSE[5291] app_dial.c: -- SIP/phone1-00000004 is ringing [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.201:2051 ---> INVITE sip:*8@192.168.10.75;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-k2set6x6jcec;rport From: "User 4" ;tag=vo1tbz2k84 To: Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.3.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 248 v=0 o=root 2087804736 2087804736 IN IP4 192.168.10.201 s=call c=IN IP4 192.168.10.201 t=0 0 m=audio 52952 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 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 46]: INVITE sip:*8@192.168.10.75;user=phone SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-k2set6x6jcec;rport [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 56]: From: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 37]: To: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 41]: P-Key-Flags: resolution="31x13", keys="4" [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 26]: User-Agent: snom360/7.3.30 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 23]: Accept: application/sdp [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 11 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 12 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 13 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 14 [ 35]: Session-Expires: 3600;refresher=uas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 15 [ 10]: Min-SE: 90 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 16 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 17 [ 19]: Content-Length: 248 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 18 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 1 [ 50]: o=root 2087804736 2087804736 IN IP4 192.168.10.201 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 2 [ 6]: s=call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 5 [ 29]: m=audio 52952 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 pcma/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (18 headers 12 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671b15914-fot3bvnvp7v5 (Checking From) --From tag vo1tbz2k84 --To-tag [Aug 15 18:38:41] DEBUG[5261] acl.c: For destination '192.168.10.201', our source address is '192.168.10.75'. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Allocating new SIP dialog for 3c2671b15914-fot3bvnvp7v5 - INVITE (No RTP) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -timer- [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: timer [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -100rel- [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: 100rel [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -replaces- [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: replaces [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -from-change- [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: from-change [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.201:2051 (no NAT) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Initializing initreq for method INVITE - callid 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Using INVITE request as basis request - 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found peer 'phone4' for 'phone4' from 192.168.10.201:2051 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9081ba0' [Aug 15 18:38:41] DEBUG[5261] res_rtp_asterisk.c: Allocated port 11808 for RTP instance '0x9081ba0' [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: RTP instance '0x9081ba0' is setup and ready to go [Aug 15 18:38:41] DEBUG[5261] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9081ba0' [Aug 15 18:38:41] VERBOSE[5261] netsock2.c: == Using SIP RTP TOS bits 184 [Aug 15 18:38:41] VERBOSE[5261] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Setting NAT on RTP to Off [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP o=root 2087804736 2087804736 IN IP4 192.168.10.201... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.201' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port ''. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 8 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 8 based on m type on 0xb616bf88 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 0 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 0 based on m type on 0xb616bf88 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 101 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 101 based on m type on 0xb616bf88 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format pcma for ID 8 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format pcmu for ID 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 0 on 0xb616bf88 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 8 on 0xb616bf88 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 101 on 0xb616bf88 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 15 18:38:41] DEBUG[5261] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9081ba0' [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Peer audio RTP is at port 192.168.10.201:52952 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 0 from 0xb616bf88 to 0x9081d4c [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 8 from 0xb616bf88 to 0x9081d4c [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 101 from 0xb616bf88 to 0x9081d4c [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Checking SIP call limits for device phone4 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Updating call counter for incoming call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Call from peer 'phone4' is 1 out of 2147483647 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Looking for *8 in Standard (domain 192.168.10.75) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: *** Our native formats are 0x8 (alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: This channel will not be able to handle video. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: list_route: hop: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Session-Expires: 3600 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Refresher: UAS [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Received Min-SE: 90 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Session timer started: 162 - 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: SIP/phone4-00000006: New call is still down.... Trying... [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:2051 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-k2set6x6jcec;received=192.168.10.201;rport=2051 From: "User 4" ;tag=vo1tbz2k84 To: Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 1 INVITE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-k2set6x6jcec;received=192.168.10.201;rport=2051 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 56]: From: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 37]: To: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 12 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Started Group pickup thread on channel SIP/phone4-00000006 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone4 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone4 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone4 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone4' state '2' [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone4 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone4 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone4 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone4' state '2' [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone4-00000006 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 104 CallerIDName: User 4 AccountCode: Exten: *8 Context: Standard Uniqueid: 1313426321.6 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone4-00000006 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 104 CallerIDName: User 4 ConnectedLineNum: ConnectedLineName: Uniqueid: 1313426321.6 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 104 CallerIDname: User 4 CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: *8 Context: Standard Channel: SIP/phone4-00000006 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.6 LinkedID: 1313426321.6 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5292] features.c: pickup attempt by SIP/phone4-00000006 [Aug 15 18:38:41] NOTICE[5292] features.c: pickup SIP/phone1-00000004 attempt by SIP/phone4-00000006 [Aug 15 18:38:41] DEBUG[5292] features.c: Call pickup on 'SIP/phone1-00000004' by 'SIP/phone4-00000006' [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: SIP answering channel: SIP/phone4-00000006 [Aug 15 18:38:41] DEBUG[5292] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Setting framing from config on incoming call [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 15 18:38:41] VERBOSE[5292] chan_sip.c: Audio is at 5060 [Aug 15 18:38:41] VERBOSE[5292] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 15 18:38:41] VERBOSE[5292] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 15 18:38:41] VERBOSE[5292] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: -- Done with adding codecs to SDP [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5292] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.201:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-k2set6x6jcec;received=192.168.10.201;rport=2051 From: "User 4" ;tag=vo1tbz2k84 To: ;tag=as5c196cbb Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 1 INVITE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: P-Asserted-Identity: "User 3" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1166121089 1166121089 IN IP4 192.168.10.75 s=Asterisk PBX 1.8.5-2 c=IN IP4 192.168.10.75 t=0 0 m=audio 11808 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 <------------> [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-k2set6x6jcec;received=192.168.10.201;rport=2051 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 2 [ 56]: From: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 3 [ 52]: To: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 10 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 11 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 14 [ 0]: [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 1 [ 49]: o=root 1166121089 1166121089 IN IP4 192.168.10.75 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.5-2 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.75 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 5 [ 29]: m=audio 11808 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #164 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:41] DEBUG[5241] app_queue.c: Extension '104@_extensions' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone4' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone4' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone4 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone4 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone4 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone4' state '2' [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone4-00000006 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 104 CallerIDName: User 4 ConnectedLineNum: 103 ConnectedLineName: User 3 Uniqueid: 1313426321.6 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 104 Context: _extensions Hint: SIP/phone4 Status: 1 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: PICKUP AccountCode: CallerIDnum: 101 CallerIDname: User 1 CallerIDani: 101 CallerIDrdnis: CallerIDdnid: Exten: 106 Context: Standard Channel: SIP/phone1-00000004 Application: AppDial AppData: (Outgoing Line) EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426307.4 LinkedID: 1313426307.3 Userfield: Peer: SIP/phone4-00000006 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: ANSWER AccountCode: CallerIDnum: 104 CallerIDname: User 4 CallerIDani: 104 CallerIDrdnis: CallerIDdnid: *8 Exten: *8 Context: Standard Channel: SIP/phone4-00000006 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.6 LinkedID: 1313426321.6 Userfield: Peer: [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2053 ---> INVITE sip:*8@192.168.10.75;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:2053;branch=z9hG4bK-ubdsoo7405rh;rport From: "User 5" ;tag=rth2lmmswn To: Call-ID: 3c2671a0557b-e3xeyovy6nas CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 00041324C0A7 P-Key-Flags: keys="3" User-Agent: snom320/8.4.31 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 246 v=0 o=root 488521303 488521303 IN IP4 192.168.10.203 s=call c=IN IP4 192.168.10.203 t=0 0 m=audio 58238 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 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 46]: INVITE sip:*8@192.168.10.75;user=phone SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.203:2053;branch=z9hG4bK-ubdsoo7405rh;rport [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 56]: From: "User 5" ;tag=rth2lmmswn [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 37]: To: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671a0557b-e3xeyovy6nas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 28]: X-Serialnumber: 00041324C0A7 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 21]: P-Key-Flags: keys="3" [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 26]: User-Agent: snom320/8.4.31 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 11 [ 23]: Accept: application/sdp [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 12 [ 96]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 13 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 14 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 15 [ 35]: Session-Expires: 3600;refresher=uas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 16 [ 10]: Min-SE: 90 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 17 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 18 [ 19]: Content-Length: 246 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 19 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 1 [ 48]: o=root 488521303 488521303 IN IP4 192.168.10.203 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 2 [ 6]: s=call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 5 [ 29]: m=audio 58238 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (19 headers 12 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671a0557b-e3xeyovy6nas (Checking From) --From tag rth2lmmswn --To-tag [Aug 15 18:38:41] DEBUG[5261] acl.c: For destination '192.168.10.203', our source address is '192.168.10.75'. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Allocating new SIP dialog for 3c2671a0557b-e3xeyovy6nas - INVITE (No RTP) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -timer- [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: timer [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -100rel- [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: 100rel [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -replaces- [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: replaces [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Found SIP option: -from-change- [Aug 15 18:38:41] DEBUG[5261] sip/reqresp_parser.c: Matched SIP option: from-change [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.203:2053' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.203' and port '2053'. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.203:2053 (no NAT) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Initializing initreq for method INVITE - callid 3c2671a0557b-e3xeyovy6nas [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Using INVITE request as basis request - 3c2671a0557b-e3xeyovy6nas [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found peer 'phone5' for 'phone5' from 192.168.10.203:2053 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9057c30' [Aug 15 18:38:41] DEBUG[5261] res_rtp_asterisk.c: Allocated port 11706 for RTP instance '0x9057c30' [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: RTP instance '0x9057c30' is setup and ready to go [Aug 15 18:38:41] DEBUG[5261] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9057c30' [Aug 15 18:38:41] VERBOSE[5261] netsock2.c: == Using SIP RTP TOS bits 184 [Aug 15 18:38:41] VERBOSE[5261] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Setting NAT on RTP to Off [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP o=root 488521303 488521303 IN IP4 192.168.10.203... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.203' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.203' and port ''. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 8 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 8 based on m type on 0xb616bf88 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 0 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 0 based on m type on 0xb616bf88 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 101 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 101 based on m type on 0xb616bf88 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 0 on 0xb616bf88 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 8 on 0xb616bf88 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 101 on 0xb616bf88 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 15 18:38:41] DEBUG[5261] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9057c30' [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Peer audio RTP is at port 192.168.10.203:58238 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 0 from 0xb616bf88 to 0x9057ddc [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 8 from 0xb616bf88 to 0x9057ddc [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 101 from 0xb616bf88 to 0x9057ddc [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Checking SIP call limits for device phone5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Updating call counter for incoming call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Call from peer 'phone5' is 1 out of 2147483647 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Looking for *8 in Standard (domain 192.168.10.75) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: *** Our native formats are 0x8 (alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: This channel will not be able to handle video. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: list_route: hop: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Session-Expires: 3600 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Refresher: UAS [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Received Min-SE: 90 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Session timer started: 165 - 3c2671a0557b-e3xeyovy6nas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: SIP/phone5-00000007: New call is still down.... Trying... [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:2053 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.203:2053;branch=z9hG4bK-ubdsoo7405rh;received=192.168.10.203;rport=2053 From: "User 5" ;tag=rth2lmmswn To: Call-ID: 3c2671a0557b-e3xeyovy6nas CSeq: 1 INVITE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.203:2053;branch=z9hG4bK-ubdsoo7405rh;received=192.168.10.203;rport=2053 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 56]: From: "User 5" ;tag=rth2lmmswn [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 37]: To: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671a0557b-e3xeyovy6nas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 12 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:2053 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Started Group pickup thread on channel SIP/phone5-00000007 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone5 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone5 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone5 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone5' state '2' [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone5 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone5 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone5 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone5' state '2' [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone5-00000007 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 105 CallerIDName: User 5 AccountCode: Exten: *8 Context: Standard Uniqueid: 1313426321.7 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone5-00000007 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 105 CallerIDName: User 5 ConnectedLineNum: ConnectedLineName: Uniqueid: 1313426321.7 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_START AccountCode: CallerIDnum: 105 CallerIDname: User 5 CallerIDani: CallerIDrdnis: CallerIDdnid: Exten: *8 Context: Standard Channel: SIP/phone5-00000007 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.7 LinkedID: 1313426321.7 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5293] features.c: pickup attempt by SIP/phone5-00000007 [Aug 15 18:38:41] NOTICE[5293] features.c: pickup SIP/phone2-00000005 attempt by SIP/phone5-00000007 [Aug 15 18:38:41] DEBUG[5293] features.c: Call pickup on 'SIP/phone2-00000005' by 'SIP/phone5-00000007' [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: SIP answering channel: SIP/phone5-00000007 [Aug 15 18:38:41] DEBUG[5293] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Setting framing from config on incoming call [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 15 18:38:41] VERBOSE[5293] chan_sip.c: Audio is at 5060 [Aug 15 18:38:41] VERBOSE[5293] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 15 18:38:41] VERBOSE[5293] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 15 18:38:41] VERBOSE[5293] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: -- Done with adding codecs to SDP [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5293] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.203:2053 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.203:2053;branch=z9hG4bK-ubdsoo7405rh;received=192.168.10.203;rport=2053 From: "User 5" ;tag=rth2lmmswn To: ;tag=as2385dc10 Call-ID: 3c2671a0557b-e3xeyovy6nas CSeq: 1 INVITE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: P-Asserted-Identity: "User 3" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1032337777 1032337777 IN IP4 192.168.10.75 s=Asterisk PBX 1.8.5-2 c=IN IP4 192.168.10.75 t=0 0 m=audio 11706 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 <------------> [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.203:2053;branch=z9hG4bK-ubdsoo7405rh;received=192.168.10.203;rport=2053 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 2 [ 56]: From: "User 5" ;tag=rth2lmmswn [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 3 [ 52]: To: ;tag=as2385dc10 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671a0557b-e3xeyovy6nas [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 10 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 11 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Header 14 [ 0]: [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 1 [ 49]: o=root 1032337777 1032337777 IN IP4 192.168.10.75 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.5-2 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.75 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 5 [ 29]: m=audio 11706 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #167 [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2053 [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone4' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone5 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone5 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone5 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone5' state '2' [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone5-00000007 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 105 CallerIDName: User 5 ConnectedLineNum: 103 ConnectedLineName: User 3 Uniqueid: 1313426321.7 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: PICKUP AccountCode: CallerIDnum: 102 CallerIDname: User 2 CallerIDani: 102 CallerIDrdnis: CallerIDdnid: Exten: 106 Context: Standard Channel: SIP/phone2-00000005 Application: AppDial AppData: (Outgoing Line) EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426307.5 LinkedID: 1313426307.3 Userfield: Peer: SIP/phone5-00000007 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: ANSWER AccountCode: CallerIDnum: 105 CallerIDname: User 5 CallerIDani: 105 CallerIDrdnis: CallerIDdnid: *8 Exten: *8 Context: Standard Channel: SIP/phone5-00000007 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.7 LinkedID: 1313426321.7 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5241] app_queue.c: Extension '105@_extensions' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone5' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone5' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 105 Context: _extensions Hint: SIP/phone5 Status: 1 [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone5' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2053 ---> ACK sip:*8@192.168.10.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:2053;branch=z9hG4bK-jqpy8vm271fs;rport From: "User 5" ;tag=rth2lmmswn To: ;tag=as2385dc10 Call-ID: 3c2671a0557b-e3xeyovy6nas CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 37]: ACK sip:*8@192.168.10.75:5060 SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.203:2053;branch=z9hG4bK-jqpy8vm271fs;rport [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 56]: From: "User 5" ;tag=rth2lmmswn [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 52]: To: ;tag=as2385dc10 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671a0557b-e3xeyovy6nas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (9 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671a0557b-e3xeyovy6nas (Checking From) --From tag rth2lmmswn --To-tag as2385dc10 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #167 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '3c2671a0557b-e3xeyovy6nas' of Response 1: Match Found [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.201:2051 ---> ACK sip:*8@192.168.10.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-zqrm5gt05xkq;rport From: "User 4" ;tag=vo1tbz2k84 To: ;tag=as5c196cbb Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 37]: ACK sip:*8@192.168.10.75:5060 SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-zqrm5gt05xkq;rport [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 56]: From: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 52]: To: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (9 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671b15914-fot3bvnvp7v5 (Checking From) --From tag vo1tbz2k84 --To-tag as5c196cbb [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #164 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '3c2671b15914-fot3bvnvp7v5' of Response 1: Match Found [Aug 15 18:38:41] DEBUG[5292] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.10.201:52953 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 16777216 FractionLost: 0 PacketsLost: 1 HighestSequence: 0 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Aug 15 18:38:41] DEBUG[5293] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From: 192.168.10.203:58239 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 16777216 FractionLost: 0 PacketsLost: 1 HighestSequence: 0 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Aug 15 18:38:41] DEBUG[5292] channel.c: Planning to masquerade channel SIP/phone4-00000006 into the structure of SIP/phone1-00000004 [Aug 15 18:38:41] DEBUG[5292] channel.c: Done planning to masquerade channel SIP/phone4-00000006 into the structure of SIP/phone1-00000004 [Aug 15 18:38:41] DEBUG[5292] channel.c: Actually Masquerading SIP/phone4-00000006(6) into the structure of SIP/phone1-00000004(5) [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: SIP Fixup: New owner for dialogue 453246b6506d5ebe1d40824369b95d3f@192.168.10.75: SIP/phone4-00000006 (Old parent: SIP/phone4-00000006) [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Hangup call SIP/phone4-00000006, SIP callid 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: update_call_counter(phone1) - decrement call limit counter on hangup [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Updating call counter for outgoing call [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Call to peer 'phone1' removed from call limit 2147483647 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Hanging up channel in state Ringing (not UP) [Aug 15 18:38:41] DEBUG[5292] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9052e60' [Aug 15 18:38:41] VERBOSE[5292] chan_sip.c: Scheduling destruction of SIP dialog '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' in 32000 ms (Method: INVITE) [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' Request 102: Found [Aug 15 18:38:41] VERBOSE[5292] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.207:2048: CANCEL sip:phone1@192.168.10.207:2048;line=4dsger5q SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 Max-Forwards: 70 From: "User 3" ;tag=as1426c059 To: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 CANCEL User-Agent: IPTAM PBX Content-Length: 0 --- [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 0 [ 59]: CANCEL sip:phone1@192.168.10.207:2048;line=4dsger5q SIP/2.0 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 3 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 4 [ 50]: To: [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 5 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 6 [ 16]: CSeq: 102 CANCEL [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 7 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Header 9 [ 0]: [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #170 [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 192.168.10.207:2048 [Aug 15 18:38:41] VERBOSE[5292] chan_sip.c: Scheduling destruction of SIP dialog '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' in 32000 ms (Method: INVITE) [Aug 15 18:38:41] DEBUG[5292] channel.c: Putting channel SIP/phone4-00000006 in alaw/alaw formats [Aug 15 18:38:41] DEBUG[5292] chan_sip.c: SIP Fixup: New owner for dialogue 3c2671b15914-fot3bvnvp7v5: SIP/phone4-00000006 (Old parent: SIP/phone1-00000004) [Aug 15 18:38:41] DEBUG[5292] channel.c: Released clone lock on 'SIP/phone1-00000004' [Aug 15 18:38:41] DEBUG[5292] channel.c: Done Masquerading SIP/phone4-00000006 (6) [Aug 15 18:38:41] DEBUG[5292] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Aug 15 18:38:41] DEBUG[5292] channel.c: Hanging up zombie 'SIP/phone1-00000004' [Aug 15 18:38:41] VERBOSE[5291] app_dial.c: -- SIP/phone4-00000006 answered SIP/phone3-00000003 [Aug 15 18:38:41] DEBUG[5291] channel.c: Hanging up channel 'SIP/phone2-00000005' [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: This call was answered elsewhere[Aug 15 18:38:41] DEBUG[5291] chan_sip.c: ####### It's the cause code, buddy. The cause code!!! [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Hangup call SIP/phone2-00000005, SIP callid 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: update_call_counter(phone2) - decrement call limit counter on hangup [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Updating call counter for outgoing call [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Call to peer 'phone2' removed from call limit 2147483647 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Hanging up channel in state Ringing (not UP) [Aug 15 18:38:41] DEBUG[5291] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9078ed8' [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Scheduling destruction of SIP dialog '38a269487dde38005bbd0a773e00a253@192.168.10.75' in 32000 ms (Method: INVITE) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '38a269487dde38005bbd0a773e00a253@192.168.10.75' Request 102: Found [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.208:5060: CANCEL sip:phone2@192.168.10.208:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 Max-Forwards: 70 From: "User 3" ;tag=as59eca913 To: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 CSeq: 102 CANCEL User-Agent: IPTAM PBX Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 --- [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 0 [ 59]: CANCEL sip:phone2@192.168.10.208:5060;transport=udp SIP/2.0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 3 [ 53]: From: "User 3" ;tag=as59eca913 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 4 [ 50]: To: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 5 [ 55]: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 6 [ 16]: CSeq: 102 CANCEL [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 7 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 8 [ 53]: Reason: SIP;cause=200;text="Call completed elsewhere" [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 10 [ 0]: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #173 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 192.168.10.208:5060 [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Scheduling destruction of SIP dialog '38a269487dde38005bbd0a773e00a253@192.168.10.75' in 32000 ms (Method: INVITE) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: SIP answering channel: SIP/phone3-00000003 [Aug 15 18:38:41] DEBUG[5291] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Setting framing from config on incoming call [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Audio is at 5060 [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: -- Done with adding codecs to SDP [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.209:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK08a0c3a448db20d56.c9c6ff2d4112bce6b;received=192.168.10.209 From: "user 3" ;tag=f1865c1211 To: ;tag=as5b35b701 Call-ID: 848bff54a41b66ee CSeq: 24500 INVITE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "User 1" Content-Type: application/sdp Content-Length: 260 v=0 o=root 217296799 217296799 IN IP4 192.168.10.75 s=Asterisk PBX 1.8.5-2 c=IN IP4 192.168.10.75 t=0 0 m=audio 15448 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 <------------> [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 1 [105]: Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK08a0c3a448db20d56.c9c6ff2d4112bce6b;received=192.168.10.209 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 2 [ 61]: From: "user 3" ;tag=f1865c1211 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 3 [ 58]: To: ;tag=as5b35b701 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 4 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 5 [ 18]: CSeq: 24500 INVITE [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 9 [ 37]: Contact: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 10 [ 58]: P-Asserted-Identity: "User 1" [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 12 [ 19]: Content-Length: 260 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 13 [ 0]: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 1 [ 47]: o=root 217296799 217296799 IN IP4 192.168.10.75 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.5-2 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.75 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 5 [ 29]: m=audio 15448 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #175 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:38:41] DEBUG[5291] features.c: bridge answer set, chan answer set [Aug 15 18:38:41] DEBUG[5291] features.c: Removing dialed interfaces datastore on SIP/phone4-00000006 since we're bridging [Aug 15 18:38:41] DEBUG[5291] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 15 18:38:41] DEBUG[5291] res_rtp_asterisk.c: Setting the marker bit due to a source update [Aug 15 18:38:41] VERBOSE[5291] rtp_engine.c: -- Remotely bridging SIP/phone3-00000003 and SIP/phone4-00000006 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Deferring reinvite on SIP '848bff54a41b66ee' - It's audio will be redirected to IP 192.168.10.201:52952 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Sending reinvite on SIP '3c2671b15914-fot3bvnvp7v5' - It's audio soon redirected to IP 192.168.10.209:3000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Strict routing enforced for session 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:41] DEBUG[5291] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:41] DEBUG[5291] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: set_destination: set destination to 192.168.10.201:2051 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Audio is at 5060 [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: -- Done with adding codecs to SDP [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Initializing already initialized SIP dialog 3c2671b15914-fot3bvnvp7v5 (presumably reinvite) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK11c13298;rport [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 3 [ 54]: From: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 4 [ 54]: To: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 5 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 6 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 14 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:2051: INVITE sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK11c13298;rport Max-Forwards: 70 From: ;tag=as5c196cbb To: "User 4" ;tag=vo1tbz2k84 Contact: Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 102 INVITE User-Agent: IPTAM PBX Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "User 3" Content-Type: application/sdp Content-Length: 263 v=0 o=root 1166121089 1166121090 IN IP4 192.168.10.209 s=Asterisk PBX 1.8.5-2 c=IN IP4 192.168.10.209 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=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK11c13298;rport [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 3 [ 54]: From: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 4 [ 54]: To: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 5 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 6 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 14 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 16 [ 19]: Content-Length: 263 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 17 [ 0]: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 1 [ 50]: o=root 1166121089 1166121090 IN IP4 192.168.10.209 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.5-2 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.209 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #176 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Strict routing enforced for session 848bff54a41b66ee [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:41] DEBUG[5291] netsock2.c: Splitting '192.168.10.209:5060' into... [Aug 15 18:38:41] DEBUG[5291] netsock2.c: ...host '192.168.10.209' and port '5060'. [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: set_destination: set destination to 192.168.10.209:5060 [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.209:5060: UPDATE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK4e701d10 Max-Forwards: 70 From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Contact: Call-ID: 848bff54a41b66ee CSeq: 102 UPDATE User-Agent: IPTAM PBX P-Asserted-Identity: "User 4" X-Asterisk-rpid-update: Yes Content-Length: 0 --- [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 0 [ 59]: UPDATE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK4e701d10 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 3 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 4 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 6 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 102 UPDATE [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 9 [ 58]: P-Asserted-Identity: "User 4" [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 10 [ 27]: X-Asterisk-rpid-update: Yes [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 12 [ 0]: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #177 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Trying to put 'UPDATE sip:' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:38:41] DEBUG[5291] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Aug 15 18:38:41] DEBUG[5291] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Aug 15 18:38:41] DEBUG[5291] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x9062008' [Aug 15 18:38:41] DEBUG[5291] channel.c: Dropping duplicate answer! [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Pickup Privilege: call,all Channel: SIP/phone4-00000006 TargetChannel: SIP/phone1-00000004 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/phone4-00000006 CloneState: Up Original: SIP/phone1-00000004 OriginalState: Ringing [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone4-00000006 Newname: SIP/phone4-00000006 Uniqueid: 1313426321.6 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone1-00000004 Newname: SIP/phone4-00000006 Uniqueid: 1313426307.4 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone4-00000006 Newname: SIP/phone1-00000004 Uniqueid: 1313426321.6 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone4-00000006 CallerIDNum: 104 CallerIDName: User 4 Uniqueid: 1313426307.4 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone1-00000004 Uniqueid: 1313426321.6 CallerIDNum: 101 CallerIDName: User 1 ConnectedLineNum: 103 ConnectedLineName: User 3 Cause: 16 Cause-txt: Normal Clearing [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-00000005 Uniqueid: 1313426307.5 CallerIDNum: 102 CallerIDName: User 2 ConnectedLineNum: 103 ConnectedLineName: User 3 Cause: 26 Cause-txt: Unknown [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000003 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 103 CallerIDName: User 3 ConnectedLineNum: 101 ConnectedLineName: User 1 Uniqueid: 1313426307.3 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone4-00000006 Uniqueid: 1313426307.4 AccountCode: OldAccountCode: [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone3-00000003 Channel2: SIP/phone4-00000006 Uniqueid1: 1313426307.3 Uniqueid2: 1313426307.4 CallerID1: 103 CallerID2: 104 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone1 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone1' state '1' [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone1 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone1' state '1' [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone2 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone2' state '1' [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone3 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone3' state '2' [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: LINKEDID_END AccountCode: CallerIDnum: 104 CallerIDname: User 4 CallerIDani: 104 CallerIDrdnis: CallerIDdnid: *8 Exten: *8 Context: Standard Channel: SIP/phone4-00000006 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.6 LinkedID: 1313426321.6 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 101 CallerIDname: User 1 CallerIDani: 101 CallerIDrdnis: CallerIDdnid: Exten: *8 Context: Standard Channel: SIP/phone1-00000004 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.6 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_END AccountCode: CallerIDnum: 101 CallerIDname: User 1 CallerIDani: 101 CallerIDrdnis: CallerIDdnid: Exten: *8 Context: Standard Channel: SIP/phone1-00000004 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.6 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 102 CallerIDname: User 2 CallerIDani: 102 CallerIDrdnis: CallerIDdnid: Exten: 106 Context: Standard Channel: SIP/phone2-00000005 Application: AppDial AppData: (Outgoing Line) EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426307.5 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: ANSWER AccountCode: CallerIDnum: 103 CallerIDname: User 3 CallerIDani: 103 CallerIDrdnis: CallerIDdnid: 106 Exten: 106 Context: Standard Channel: SIP/phone3-00000003 Application: Dial AppData: SIP/phone1&SIP/phone2 EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426307.3 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: BRIDGE_START AccountCode: CallerIDnum: 103 CallerIDname: User 3 CallerIDani: 103 CallerIDrdnis: CallerIDdnid: 106 Exten: 106 Context: Standard Channel: SIP/phone3-00000003 Application: Dial AppData: SIP/phone1&SIP/phone2 EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426307.3 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5241] app_queue.c: Extension '101@_extensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5241] chan_sip.c: Strict routing enforced for session 3c2670e67e76-t62560t3kg7w [Aug 15 18:38:41] VERBOSE[5241] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:41] DEBUG[5241] netsock2.c: Splitting '192.168.10.203:2053' into... [Aug 15 18:38:41] DEBUG[5241] netsock2.c: ...host '192.168.10.203' and port '2053'. [Aug 15 18:38:41] VERBOSE[5241] chan_sip.c: set_destination: set destination to 192.168.10.203:2053 [Aug 15 18:38:41] VERBOSE[5241] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:2053: NOTIFY sip:phone5@192.168.10.203:2053;line=ixa0bhmu SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3361ef55;rport Max-Forwards: 70 From: ;tag=as08d89766 To: ;tag=gij4ob7fro Contact: Call-ID: 3c2670e67e76-t62560t3kg7w CSeq: 104 NOTIFY User-Agent: IPTAM PBX Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 201 terminated --- [Aug 15 18:38:41] DEBUG[5241] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #179 [Aug 15 18:38:41] DEBUG[5241] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.203:2053 [Aug 15 18:38:41] VERBOSE[5241] chan_sip.c: == Extension Changed 101[_extensions] new state Idle for Notify User phone5 [Aug 15 18:38:41] DEBUG[5241] app_queue.c: Extension '102@_extensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5241] chan_sip.c: Strict routing enforced for session 3c2670e6785d-xzn3ch211ngt [Aug 15 18:38:41] VERBOSE[5241] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:41] DEBUG[5241] netsock2.c: Splitting '192.168.10.203:2053' into... [Aug 15 18:38:41] DEBUG[5241] netsock2.c: ...host '192.168.10.203' and port '2053'. [Aug 15 18:38:41] VERBOSE[5241] chan_sip.c: set_destination: set destination to 192.168.10.203:2053 [Aug 15 18:38:41] VERBOSE[5241] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:2053: NOTIFY sip:phone5@192.168.10.203:2053;line=ixa0bhmu SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK5d983b8f;rport Max-Forwards: 70 From: ;tag=as193adead To: ;tag=86yhofjdv1 Contact: Call-ID: 3c2670e6785d-xzn3ch211ngt CSeq: 104 NOTIFY User-Agent: IPTAM PBX Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 201 terminated --- [Aug 15 18:38:41] DEBUG[5241] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #180 [Aug 15 18:38:41] DEBUG[5241] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.203:2053 [Aug 15 18:38:41] VERBOSE[5241] chan_sip.c: == Extension Changed 102[_extensions] new state Idle for Notify User phone5 [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 101 Context: _extensions Hint: SIP/phone1 Status: 0 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 102 Context: _extensions Hint: SIP/phone2 Status: 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.207:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 From: "User 3" ;tag=as1426c059 To: ;tag=2el7dj9iuz Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 CANCEL Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=2el7dj9iuz [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 CANCEL [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (7 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 (Checking To) --From tag as1426c059 --To-tag 2el7dj9iuz [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Acked pending invite 102 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #170 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' of Request 102: Match Found [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.207:2048 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 From: "User 3" ;tag=as1426c059 To: ;tag=2el7dj9iuz Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 INVITE Contact: ;reg-id=1 Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 30]: SIP/2.0 487 Request Terminated [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=2el7dj9iuz [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (8 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 (Checking To) --From tag as1426c059 --To-tag 2el7dj9iuz [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' of Request 102: Match Found [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: SIP response 487 to standard invite [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Transmitting (no NAT) to 192.168.10.207:2048: ACK sip:phone1@192.168.10.207:2048;line=4dsger5q SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 Max-Forwards: 70 From: "User 3" ;tag=as1426c059 To: ;tag=2el7dj9iuz Contact: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 CSeq: 102 ACK User-Agent: IPTAM PBX Content-Length: 0 --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.207:2048;line=4dsger5q SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK26a85e96 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 53]: From: "User 3" ;tag=as1426c059 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 65]: To: ;tag=2el7dj9iuz [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 55]: Call-ID: 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.207:2048 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Updating call counter for outgoing call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Call to peer 'phone1' removed from call limit 2147483647 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Setting SIP_ALREADYGONE on dialog 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Destroying SIP dialog 453246b6506d5ebe1d40824369b95d3f@192.168.10.75 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '453246b6506d5ebe1d40824369b95d3f@192.168.10.75' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Destroyed RTP instance '0x9052e60' [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone1 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone1' state '1' [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] DEBUG[5293] channel.c: Planning to masquerade channel SIP/phone5-00000007 into the structure of SIP/phone2-00000005 [Aug 15 18:38:41] DEBUG[5293] channel.c: Done planning to masquerade channel SIP/phone5-00000007 into the structure of SIP/phone2-00000005 [Aug 15 18:38:41] DEBUG[5293] channel.c: Actually Masquerading SIP/phone5-00000007(6) into the structure of SIP/phone2-00000005(5) [Aug 15 18:38:41] ERROR[5293] astobj2.c: user_data is NULL [Aug 15 18:38:41] ERROR[5293] astobj2.c: user_data is NULL [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: Asked to hangup channel that was not connected [Aug 15 18:38:41] DEBUG[5293] channel.c: Putting channel SIP/phone5-00000007 in alaw/alaw formats [Aug 15 18:38:41] DEBUG[5293] chan_sip.c: SIP Fixup: New owner for dialogue 3c2671a0557b-e3xeyovy6nas: SIP/phone5-00000007 (Old parent: SIP/phone2-00000005) [Aug 15 18:38:41] DEBUG[5293] channel.c: Released clone lock on 'SIP/phone2-00000005' [Aug 15 18:38:41] DEBUG[5293] channel.c: Done Masquerading SIP/phone5-00000007 (6) [Aug 15 18:38:41] DEBUG[5293] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Aug 15 18:38:41] DEBUG[5293] channel.c: Hanging up zombie 'SIP/phone2-00000005' [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Pickup Privilege: call,all Channel: SIP/phone5-00000007 TargetChannel: SIP/phone2-00000005 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/phone5-00000007 CloneState: Up Original: SIP/phone2-00000005 OriginalState: Ringing [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone5-00000007 Newname: SIP/phone5-00000007 Uniqueid: 1313426321.7 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone2-00000005 Newname: SIP/phone5-00000007 Uniqueid: 1313426307.5 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone5-00000007 Newname: SIP/phone2-00000005 Uniqueid: 1313426321.7 [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone5-00000007 CallerIDNum: 105 CallerIDName: User 5 Uniqueid: 1313426307.5 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-00000005 Uniqueid: 1313426321.7 CallerIDNum: 102 CallerIDName: User 2 ConnectedLineNum: 103 ConnectedLineName: User 3 Cause: 16 Cause-txt: Normal Clearing [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone2 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone2' state '1' [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: LINKEDID_END AccountCode: CallerIDnum: 105 CallerIDname: User 5 CallerIDani: 105 CallerIDrdnis: CallerIDdnid: *8 Exten: *8 Context: Standard Channel: SIP/phone5-00000007 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.7 LinkedID: 1313426321.7 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 102 CallerIDname: User 2 CallerIDani: 102 CallerIDrdnis: CallerIDdnid: Exten: *8 Context: Standard Channel: SIP/phone2-00000005 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.7 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_END AccountCode: CallerIDnum: 102 CallerIDname: User 2 CallerIDani: 102 CallerIDrdnis: CallerIDdnid: Exten: *8 Context: Standard Channel: SIP/phone2-00000005 Application: AppData: EventTime: 2011-08-15 18:38:41 AMAFlags: DOCUMENTATION UniqueID: 1313426321.7 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2053 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3361ef55;rport=5060 From: ;tag=as08d89766 To: ;tag=gij4ob7fro Call-ID: 3c2670e67e76-t62560t3kg7w CSeq: 104 NOTIFY Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3361ef55;rport=5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 55]: From: ;tag=as08d89766 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 45]: To: ;tag=gij4ob7fro [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2670e67e76-t62560t3kg7w [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 104 NOTIFY [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (7 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2670e67e76-t62560t3kg7w (Checking To) --From tag as08d89766 --To-tag gij4ob7fro [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Acked pending invite 104 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #179 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '3c2670e67e76-t62560t3kg7w' of Request 104: Match Found [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.201:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK11c13298;rport=5060 From: ;tag=as5c196cbb To: "User 4" ;tag=vo1tbz2k84 Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 102 INVITE Contact: ;reg-id=1 Require: timer Session-Expires: 1800;refresher=uas User-Agent: snom360/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 248 v=0 o=root 2087804736 2087804737 IN IP4 192.168.10.201 s=call c=IN IP4 192.168.10.201 t=0 0 m=audio 52952 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 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK11c13298;rport=5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 54]: From: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 54]: To: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 14]: Require: timer [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 26]: User-Agent: snom360/7.3.30 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 14 [ 19]: Content-Length: 248 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 15 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 1 [ 50]: o=root 2087804736 2087804737 IN IP4 192.168.10.201 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 2 [ 6]: s=call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 5 [ 29]: m=audio 52952 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 pcma/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (15 headers 12 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671b15914-fot3bvnvp7v5 (Checking To) --From tag as5c196cbb --To-tag vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Acked pending invite 102 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #176 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '3c2671b15914-fot3bvnvp7v5' of Request 102: Match Found [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP o=root 2087804736 2087804737 IN IP4 192.168.10.201... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.201' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port ''. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 8 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 8 based on m type on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 0 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 0 based on m type on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 101 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 101 based on m type on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format pcma for ID 8 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format pcmu for ID 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 0 on 0xb616c588 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 8 on 0xb616c588 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 101 on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 15 18:38:41] DEBUG[5261] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9081ba0' [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Peer audio RTP is at port 192.168.10.201:52952 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 0 from 0xb616c588 to 0x9081d4c [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 8 from 0xb616c588 to 0x9081d4c [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 101 from 0xb616c588 to 0x9081d4c [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: We have an owner, now see if we need to change this call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Updating call counter for incoming call [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Strict routing enforced for session 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: set_destination: set destination to 192.168.10.201:2051 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Transmitting (no NAT) to 192.168.10.201:2051: ACK sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK578e28a9;rport Max-Forwards: 70 From: ;tag=as5c196cbb To: "User 4" ;tag=vo1tbz2k84 Contact: Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 102 ACK User-Agent: IPTAM PBX Content-Length: 0 --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 56]: ACK sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK578e28a9;rport [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 54]: From: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 54]: To: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone4 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone4 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone4 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone4' state '2' [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone4' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.208:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 From: "User 3" ;tag=as59eca913 To: ;tag=1190011780 Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 CSeq: 102 CANCEL Server: Aastra 51i/2.6.0.2019 Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as59eca913 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=1190011780 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 CANCEL [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 29]: Server: Aastra 51i/2.6.0.2019 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (8 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 (Checking To) --From tag as59eca913 --To-tag 1190011780 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Acked pending invite 102 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #173 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '38a269487dde38005bbd0a773e00a253@192.168.10.75' of Request 102: Match Found [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.208:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 From: "User 3" ;tag=as59eca913 To: ;tag=1190011780 Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 CSeq: 102 INVITE Server: Aastra 51i/2.6.0.2019 Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 30]: SIP/2.0 487 Request Terminated [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 53]: From: "User 3" ;tag=as59eca913 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 65]: To: ;tag=1190011780 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 55]: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 29]: Server: Aastra 51i/2.6.0.2019 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (8 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 (Checking To) --From tag as59eca913 --To-tag 1190011780 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '38a269487dde38005bbd0a773e00a253@192.168.10.75' of Request 102: Match Found [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: SIP response 487 to standard invite [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Transmitting (no NAT) to 192.168.10.208:5060: ACK sip:phone2@192.168.10.208:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 Max-Forwards: 70 From: "User 3" ;tag=as59eca913 To: ;tag=1190011780 Contact: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 CSeq: 102 ACK User-Agent: IPTAM PBX Content-Length: 0 --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.208:5060;transport=udp SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK20162828 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 53]: From: "User 3" ;tag=as59eca913 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 65]: To: ;tag=1190011780 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 55]: Call-ID: 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.208:5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Updating call counter for outgoing call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Call to peer 'phone2' removed from call limit 2147483647 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Setting SIP_ALREADYGONE on dialog 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Destroying SIP dialog 38a269487dde38005bbd0a773e00a253@192.168.10.75 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '38a269487dde38005bbd0a773e00a253@192.168.10.75' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Destroyed RTP instance '0x9078ed8' [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.203:2053 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK5d983b8f;rport=5060 From: ;tag=as193adead To: ;tag=86yhofjdv1 Call-ID: 3c2670e6785d-xzn3ch211ngt CSeq: 104 NOTIFY Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK5d983b8f;rport=5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 55]: From: ;tag=as193adead [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 45]: To: ;tag=86yhofjdv1 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2670e6785d-xzn3ch211ngt [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 104 NOTIFY [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (7 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2670e6785d-xzn3ch211ngt (Checking To) --From tag as193adead --To-tag 86yhofjdv1 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Acked pending invite 104 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #180 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '3c2670e6785d-xzn3ch211ngt' of Request 104: Match Found [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone2 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone2' state '1' [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.209:5060 ---> ACK sip:106@192.168.10.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bKc170025512b44fbe0 Max-Forwards: 70 From: "user 3" ;tag=f1865c1211 To: ;tag=as5b35b701 Call-ID: 848bff54a41b66ee CSeq: 24500 ACK User-Agent: Aastra 53i/3.2.1.43 Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 38]: ACK sip:106@192.168.10.75:5060 SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bKc170025512b44fbe0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 61]: From: "user 3" ;tag=f1865c1211 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 58]: To: ;tag=as5b35b701 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 15]: CSeq: 24500 ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 31]: User-Agent: Aastra 53i/3.2.1.43 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (9 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 848bff54a41b66ee (Checking From) --From tag f1865c1211 --To-tag as5b35b701 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #175 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '848bff54a41b66ee' of Response 24500: Match Found [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Sending pending reinvite on '848bff54a41b66ee' [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Strict routing enforced for session 848bff54a41b66ee [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.209:5060' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port '5060'. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: set_destination: set destination to 192.168.10.209:5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Audio is at 5060 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: -- Done with adding codecs to SDP [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Initializing already initialized SIP dialog 848bff54a41b66ee (presumably reinvite) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK14372fee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "User 4" [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.209:5060: INVITE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK14372fee Max-Forwards: 70 From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Contact: Call-ID: 848bff54a41b66ee CSeq: 103 INVITE User-Agent: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "User 4" Content-Type: application/sdp Content-Length: 262 v=0 o=root 217296799 217296800 IN IP4 192.168.10.201 s=Asterisk PBX 1.8.5-2 c=IN IP4 192.168.10.201 t=0 0 m=audio 52952 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 --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK14372fee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "User 4" [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 14 [ 19]: Content-Length: 262 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 15 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 1 [ 48]: o=root 217296799 217296800 IN IP4 192.168.10.201 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.5-2 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 5 [ 29]: m=audio 52952 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #182 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.209:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK4e701d10 From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Call-ID: 848bff54a41b66ee CSeq: 102 UPDATE Contact: "user 3" ;+sip.instance="" Server: Aastra 53i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK4e701d10 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 102 UPDATE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [128]: Contact: "user 3" ;+sip.instance="" [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 27]: Server: Aastra 53i/3.2.1.43 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 15]: Supported: path [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (10 headers 0 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 848bff54a41b66ee (Checking To) --From tag as5b35b701 --To-tag f1865c1211 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #177 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '848bff54a41b66ee' of Request 102: Match Found [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.209:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK14372fee From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Call-ID: 848bff54a41b66ee CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "user 3" ;+sip.instance="" Server: Aastra 53i/3.2.1.43 Supported: path, replaces Content-Type: application/sdp Content-Length: 296 v=0 o=MxSIP 0 2 IN IP4 192.168.10.209 s=SIP Call c=IN IP4 192.168.10.209 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=rtcp:3001 IN IP4 192.168.10.209 a=sendrecv <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK14372fee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [128]: Contact: "user 3" ;+sip.instance="" [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 27]: Server: Aastra 53i/3.2.1.43 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 25]: Supported: path, replaces [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 12 [ 19]: Content-Length: 296 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 13 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 2 IN IP4 192.168.10.209 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 2 [ 10]: s=SIP Call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.209 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 12 [ 33]: a=rtcp:3001 IN IP4 192.168.10.209 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 13 [ 10]: a=sendrecv [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (13 headers 14 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 848bff54a41b66ee (Checking To) --From tag as5b35b701 --To-tag f1865c1211 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Acked pending invite 103 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #182 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '848bff54a41b66ee' of Request 103: Match Found [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: SIP response 200 to RE-invite on outgoing call 848bff54a41b66ee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.10.209... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.209' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port ''. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.209... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 8 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 8 based on m type on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 0 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 0 based on m type on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 101 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 101 based on m type on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtcp:3001 IN IP4 192.168.10.209... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 0 on 0xb616c588 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 8 on 0xb616c588 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 101 on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 15 18:38:41] DEBUG[5261] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9062008' [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Peer audio RTP is at port 192.168.10.209:3000 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 0 from 0xb616c588 to 0x90621b4 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 8 from 0xb616c588 to 0x90621b4 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 101 from 0xb616c588 to 0x90621b4 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: We have an owner, now see if we need to change this call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Updating call counter for incoming call [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.209:5060' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port '5060'. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Strict routing enforced for session 848bff54a41b66ee [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.209:5060' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port '5060'. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: set_destination: set destination to 192.168.10.209:5060 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Transmitting (no NAT) to 192.168.10.209:5060: ACK sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3bcf475d Max-Forwards: 70 From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Contact: Call-ID: 848bff54a41b66ee CSeq: 103 ACK User-Agent: IPTAM PBX Content-Length: 0 --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK3bcf475d [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone3 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone3' state '2' [Aug 15 18:38:41] DEBUG[5291] rtp_engine.c: Oooh, 'SIP/phone3-00000003' changed end address to 192.168.10.209:3000 (format unknown) [Aug 15 18:38:41] DEBUG[5291] rtp_engine.c: Oooh, 'SIP/phone3-00000003' was 192.168.10.209:3000/(format unknown) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Sending reinvite on SIP '3c2671b15914-fot3bvnvp7v5' - It's audio soon redirected to IP 192.168.10.209:3000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Strict routing enforced for session 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:41] DEBUG[5291] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:41] DEBUG[5291] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: set_destination: set destination to 192.168.10.201:2051 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Audio is at 5060 [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: -- Done with adding codecs to SDP [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Initializing already initialized SIP dialog 3c2671b15914-fot3bvnvp7v5 (presumably reinvite) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK703b625e;rport [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 3 [ 54]: From: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 4 [ 54]: To: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 5 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 6 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 14 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] VERBOSE[5291] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:2051: INVITE sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK703b625e;rport Max-Forwards: 70 From: ;tag=as5c196cbb To: "User 4" ;tag=vo1tbz2k84 Contact: Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 103 INVITE User-Agent: IPTAM PBX Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "User 3" Content-Type: application/sdp Content-Length: 263 v=0 o=root 1166121089 1166121091 IN IP4 192.168.10.209 s=Asterisk PBX 1.8.5-2 c=IN IP4 192.168.10.209 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=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK703b625e;rport [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 3 [ 54]: From: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 4 [ 54]: To: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 5 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 6 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 14 [ 53]: P-Asserted-Identity: "User 3" [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 16 [ 19]: Content-Length: 263 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Header 17 [ 0]: [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 1 [ 50]: o=root 1166121089 1166121091 IN IP4 192.168.10.209 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.5-2 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.209 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #183 [Aug 15 18:38:41] DEBUG[5291] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.201:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK703b625e;rport=5060 From: ;tag=as5c196cbb To: "User 4" ;tag=vo1tbz2k84 Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 103 INVITE Contact: ;reg-id=1 Require: timer Session-Expires: 1800;refresher=uas User-Agent: snom360/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 248 v=0 o=root 2087804736 2087804738 IN IP4 192.168.10.201 s=call c=IN IP4 192.168.10.201 t=0 0 m=audio 52952 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 <-------------> [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK703b625e;rport=5060 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 54]: From: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 54]: To: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 14]: Require: timer [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 35]: Session-Expires: 1800;refresher=uas [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 26]: User-Agent: snom360/7.3.30 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 11 [ 42]: Allow-Events: talk, hold, refer, call-info [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 12 [ 47]: Supported: timer, 100rel, replaces, from-change [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 14 [ 19]: Content-Length: 248 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 15 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 1 [ 50]: o=root 2087804736 2087804738 IN IP4 192.168.10.201 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 2 [ 6]: s=call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 5 [ 29]: m=audio 52952 RTP/AVP 8 0 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 pcma/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 pcmu/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: --- (15 headers 12 lines) --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671b15914-fot3bvnvp7v5 (Checking To) --From tag as5c196cbb --To-tag vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Acked pending invite 103 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #183 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Stopping retransmission on '3c2671b15914-fot3bvnvp7v5' of Request 103: Match Found [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP o=root 2087804736 2087804738 IN IP4 192.168.10.201... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.201' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port ''. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 8 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 8 based on m type on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 0 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 0 based on m type on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found RTP audio format 101 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Setting payload 101 based on m type on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format pcma for ID 8 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format pcmu for ID 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 0 on 0xb616c588 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 8 on 0xb616c588 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Incorporating payload 101 on 0xb616c588 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 15 18:38:41] DEBUG[5261] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9081ba0' [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Peer audio RTP is at port 192.168.10.201:52952 [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 0 from 0xb616c588 to 0x9081d4c [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 8 from 0xb616c588 to 0x9081d4c [Aug 15 18:38:41] DEBUG[5261] rtp_engine.c: Copying payload 101 from 0xb616c588 to 0x9081d4c [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: We have an owner, now see if we need to change this call [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Updating call counter for incoming call [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Strict routing enforced for session 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:41] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:41] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: set_destination: set destination to 192.168.10.201:2051 [Aug 15 18:38:41] VERBOSE[5261] chan_sip.c: Transmitting (no NAT) to 192.168.10.201:2051: ACK sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK05458ab1;rport Max-Forwards: 70 From: ;tag=as5c196cbb To: "User 4" ;tag=vo1tbz2k84 Contact: Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 103 ACK User-Agent: IPTAM PBX Content-Length: 0 --- [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 0 [ 56]: ACK sip:phone4@192.168.10.201:2051;line=8kn400vr SIP/2.0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK05458ab1;rport [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 3 [ 54]: From: ;tag=as5c196cbb [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 4 [ 54]: To: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 5 [ 36]: Contact: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 6 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Header 10 [ 0]: [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:41] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:41] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone4 [Aug 15 18:38:41] DEBUG[5240] chan_sip.c: Checking device state for peer phone4 [Aug 15 18:38:41] DEBUG[5240] devicestate.c: Changing state for SIP/phone4 - state 2 (In use) [Aug 15 18:38:41] DEBUG[5240] devicestate.c: device 'SIP/phone4' state '2' [Aug 15 18:38:41] DEBUG[5267] app_queue.c: Device 'SIP/phone4' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 15 18:38:42] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:42] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.201:2051 ---> SUBSCRIBE sip:102@192.168.10.75;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-736a0l7zvi60;rport From: ;tag=n1hsg8w80p To: Call-ID: 3c2671b7445e-izh5olpsn160 CSeq: 4 SUBSCRIBE Max-Forwards: 70 Contact: ;reg-id=1 Event: dialog Accept: application/dialog-info+xml User-Agent: snom360/7.3.30 Expires: 3600 Content-Length: 0 <-------------> [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 0 [ 50]: SUBSCRIBE sip:102@192.168.10.75;user=phone SIP/2.0 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-736a0l7zvi60;rport [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 2 [ 47]: From: ;tag=n1hsg8w80p [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 3 [ 38]: To: [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b7445e-izh5olpsn160 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 5 [ 17]: CSeq: 4 SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 8 [ 13]: Event: dialog [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 9 [ 35]: Accept: application/dialog-info+xml [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 10 [ 26]: User-Agent: snom360/7.3.30 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 11 [ 13]: Expires: 3600 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: --- (13 headers 0 lines) --- [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671b7445e-izh5olpsn160 (Checking From) --From tag n1hsg8w80p --To-tag [Aug 15 18:38:43] DEBUG[5261] acl.c: For destination '192.168.10.201', our source address is '192.168.10.75'. [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Allocating new SIP dialog for 3c2671b7445e-izh5olpsn160 - SUBSCRIBE (No RTP) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Creating new subscription [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 3c2671b7445e-izh5olpsn160 [Aug 15 18:38:43] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:43] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.201:2051 (no NAT) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: list_route: hop: [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: No matching peer for 'katrin' from '192.168.10.201:2051' [Aug 15 18:38:43] NOTICE[5261] chan_sip.c: Sending fake auth rejection for device ;tag=n1hsg8w80p [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-736a0l7zvi60;received=192.168.10.201;rport=2051 From: ;tag=n1hsg8w80p To: ;tag=as5301ef7a Call-ID: 3c2671b7445e-izh5olpsn160 CSeq: 4 SUBSCRIBE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="458f3a72" Content-Length: 0 <------------> [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-736a0l7zvi60;received=192.168.10.201;rport=2051 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 2 [ 47]: From: ;tag=n1hsg8w80p [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 3 [ 53]: To: ;tag=as5301ef7a [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b7445e-izh5olpsn160 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 5 [ 17]: CSeq: 4 SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="458f3a72" [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 11 [ 0]: [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Scheduling destruction of SIP dialog '3c2671b7445e-izh5olpsn160' in 32000 ms (Method: SUBSCRIBE) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Destroying SIP dialog 3c2671b7445e-izh5olpsn160 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '3c2671b7445e-izh5olpsn160' Method: SUBSCRIBE [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.201:2051 ---> SUBSCRIBE sip:101@192.168.10.75;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-md79eea510gl;rport From: ;tag=xr1md614v4 To: Call-ID: 3c2671b74885-zj0og4zzn04v CSeq: 4 SUBSCRIBE Max-Forwards: 70 Contact: ;reg-id=1 Event: dialog Accept: application/dialog-info+xml User-Agent: snom360/7.3.30 Expires: 3600 Content-Length: 0 <-------------> [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 0 [ 50]: SUBSCRIBE sip:101@192.168.10.75;user=phone SIP/2.0 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-md79eea510gl;rport [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 2 [ 47]: From: ;tag=xr1md614v4 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 3 [ 38]: To: [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b74885-zj0og4zzn04v [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 5 [ 17]: CSeq: 4 SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 8 [ 13]: Event: dialog [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 9 [ 35]: Accept: application/dialog-info+xml [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 10 [ 26]: User-Agent: snom360/7.3.30 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 11 [ 13]: Expires: 3600 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: --- (13 headers 0 lines) --- [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671b74885-zj0og4zzn04v (Checking From) --From tag xr1md614v4 --To-tag [Aug 15 18:38:43] DEBUG[5261] acl.c: For destination '192.168.10.201', our source address is '192.168.10.75'. [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Allocating new SIP dialog for 3c2671b74885-zj0og4zzn04v - SUBSCRIBE (No RTP) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Creating new subscription [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 3c2671b74885-zj0og4zzn04v [Aug 15 18:38:43] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:43] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.201:2051 (no NAT) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: list_route: hop: [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: No matching peer for 'katrin' from '192.168.10.201:2051' [Aug 15 18:38:43] NOTICE[5261] chan_sip.c: Sending fake auth rejection for device ;tag=xr1md614v4 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-md79eea510gl;received=192.168.10.201;rport=2051 From: ;tag=xr1md614v4 To: ;tag=as195d33d1 Call-ID: 3c2671b74885-zj0og4zzn04v CSeq: 4 SUBSCRIBE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5445f0a5" Content-Length: 0 <------------> [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-md79eea510gl;received=192.168.10.201;rport=2051 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 2 [ 47]: From: ;tag=xr1md614v4 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 3 [ 53]: To: ;tag=as195d33d1 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b74885-zj0og4zzn04v [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 5 [ 17]: CSeq: 4 SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5445f0a5" [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 11 [ 0]: [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Scheduling destruction of SIP dialog '3c2671b74885-zj0og4zzn04v' in 32000 ms (Method: SUBSCRIBE) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Destroying SIP dialog 3c2671b74885-zj0og4zzn04v [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '3c2671b74885-zj0og4zzn04v' Method: SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.201:2051 ---> SUBSCRIBE sip:102@192.168.10.75;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-750pb08ukxb8;rport From: ;tag=n1hsg8w80p To: Call-ID: 3c2671b7445e-izh5olpsn160 CSeq: 5 SUBSCRIBE Max-Forwards: 70 Contact: ;reg-id=1 Event: dialog Accept: application/dialog-info+xml User-Agent: snom360/7.3.30 Authorization: Digest username="phone4",realm="asterisk",nonce="458f3a72",uri="sip:102@192.168.10.75;user=phone",response="439813379270cd52b76af1d2e9bfbed9",algorithm=MD5 Expires: 3600 Content-Length: 0 <-------------> [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 0 [ 50]: SUBSCRIBE sip:102@192.168.10.75;user=phone SIP/2.0 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-750pb08ukxb8;rport [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 2 [ 47]: From: ;tag=n1hsg8w80p [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 3 [ 38]: To: [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b7445e-izh5olpsn160 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 5 [ 17]: CSeq: 5 SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 8 [ 13]: Event: dialog [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 9 [ 35]: Accept: application/dialog-info+xml [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 10 [ 26]: User-Agent: snom360/7.3.30 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 11 [170]: Authorization: Digest username="phone4",realm="asterisk",nonce="458f3a72",uri="sip:102@192.168.10.75;user=phone",response="439813379270cd52b76af1d2e9bfbed9",algorithm=MD5 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 12 [ 13]: Expires: 3600 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: --- (14 headers 0 lines) --- [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671b7445e-izh5olpsn160 (Checking From) --From tag n1hsg8w80p --To-tag [Aug 15 18:38:43] DEBUG[5261] acl.c: For destination '192.168.10.201', our source address is '192.168.10.75'. [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Allocating new SIP dialog for 3c2671b7445e-izh5olpsn160 - SUBSCRIBE (No RTP) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Creating new subscription [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 3c2671b7445e-izh5olpsn160 [Aug 15 18:38:43] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:43] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.201:2051 (no NAT) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: list_route: hop: [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: No matching peer for 'katrin' from '192.168.10.201:2051' [Aug 15 18:38:43] NOTICE[5261] chan_sip.c: Sending fake auth rejection for device ;tag=n1hsg8w80p [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-750pb08ukxb8;received=192.168.10.201;rport=2051 From: ;tag=n1hsg8w80p To: ;tag=as02684382 Call-ID: 3c2671b7445e-izh5olpsn160 CSeq: 5 SUBSCRIBE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="566eb138" Content-Length: 0 <------------> [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-750pb08ukxb8;received=192.168.10.201;rport=2051 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 2 [ 47]: From: ;tag=n1hsg8w80p [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 3 [ 53]: To: ;tag=as02684382 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b7445e-izh5olpsn160 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 5 [ 17]: CSeq: 5 SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="566eb138" [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 11 [ 0]: [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Scheduling destruction of SIP dialog '3c2671b7445e-izh5olpsn160' in 32000 ms (Method: SUBSCRIBE) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Destroying SIP dialog 3c2671b7445e-izh5olpsn160 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '3c2671b7445e-izh5olpsn160' Method: SUBSCRIBE [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.201:2051 ---> SUBSCRIBE sip:101@192.168.10.75;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-uggyb2vjw9hq;rport From: ;tag=xr1md614v4 To: Call-ID: 3c2671b74885-zj0og4zzn04v CSeq: 5 SUBSCRIBE Max-Forwards: 70 Contact: ;reg-id=1 Event: dialog Accept: application/dialog-info+xml User-Agent: snom360/7.3.30 Authorization: Digest username="phone4",realm="asterisk",nonce="5445f0a5",uri="sip:101@192.168.10.75;user=phone",response="5134fef0664572af71a5874adc414c1e",algorithm=MD5 Expires: 3600 Content-Length: 0 <-------------> [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 0 [ 50]: SUBSCRIBE sip:101@192.168.10.75;user=phone SIP/2.0 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-uggyb2vjw9hq;rport [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 2 [ 47]: From: ;tag=xr1md614v4 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 3 [ 38]: To: [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b74885-zj0og4zzn04v [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 5 [ 17]: CSeq: 5 SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 8 [ 13]: Event: dialog [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 9 [ 35]: Accept: application/dialog-info+xml [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 10 [ 26]: User-Agent: snom360/7.3.30 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 11 [170]: Authorization: Digest username="phone4",realm="asterisk",nonce="5445f0a5",uri="sip:101@192.168.10.75;user=phone",response="5134fef0664572af71a5874adc414c1e",algorithm=MD5 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 12 [ 13]: Expires: 3600 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: --- (14 headers 0 lines) --- [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671b74885-zj0og4zzn04v (Checking From) --From tag xr1md614v4 --To-tag [Aug 15 18:38:43] DEBUG[5261] acl.c: For destination '192.168.10.201', our source address is '192.168.10.75'. [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Allocating new SIP dialog for 3c2671b74885-zj0og4zzn04v - SUBSCRIBE (No RTP) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Creating new subscription [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 3c2671b74885-zj0og4zzn04v [Aug 15 18:38:43] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:43] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.201:2051 (no NAT) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: list_route: hop: [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: No matching peer for 'katrin' from '192.168.10.201:2051' [Aug 15 18:38:43] NOTICE[5261] chan_sip.c: Sending fake auth rejection for device ;tag=xr1md614v4 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-uggyb2vjw9hq;received=192.168.10.201;rport=2051 From: ;tag=xr1md614v4 To: ;tag=as0dc2f049 Call-ID: 3c2671b74885-zj0og4zzn04v CSeq: 5 SUBSCRIBE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2875a06a" Content-Length: 0 <------------> [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-uggyb2vjw9hq;received=192.168.10.201;rport=2051 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 2 [ 47]: From: ;tag=xr1md614v4 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 3 [ 53]: To: ;tag=as0dc2f049 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b74885-zj0og4zzn04v [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 5 [ 17]: CSeq: 5 SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2875a06a" [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Header 11 [ 0]: [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Scheduling destruction of SIP dialog '3c2671b74885-zj0og4zzn04v' in 32000 ms (Method: SUBSCRIBE) [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Destroying SIP dialog 3c2671b74885-zj0og4zzn04v [Aug 15 18:38:43] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '3c2671b74885-zj0og4zzn04v' Method: SUBSCRIBE [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:43] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:44] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:44] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:45] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:45] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:46] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:46] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:46] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:46] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:46] DEBUG[5268] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To: 192.168.10.209:3001 OurSSRC: 1433741529 SentNTP: 1313426326.1895436288 SentRTP: 3496808 SentPackets: 1 SentOctets: 160 ReportBlock: FractionLost: 256 CumulativeLoss: 1 IAJitter: 0.0000 TheirLastSR: 0 DLSR: 17.3420 (sec) [Aug 15 18:38:46] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:46] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:47] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:47] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:48] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:48] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:49] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:49] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:50] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:50] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: ACK [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.201:2051 ---> BYE sip:*8@192.168.10.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-qeow5twvohni;rport From: "User 4" ;tag=vo1tbz2k84 To: ;tag=as5c196cbb Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 2 BYE Max-Forwards: 70 Contact: ;reg-id=1 User-Agent: snom360/7.3.30 RTP-RxStat: Total_Rx_Pkts=493,Rx_Pkts=493,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=508,Tx_Pkts=508,Remote_Tx_Pkts=1668 Content-Length: 0 <-------------> [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 0 [ 37]: BYE sip:*8@192.168.10.75:5060 SIP/2.0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-qeow5twvohni;rport [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 2 [ 56]: From: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 3 [ 52]: To: ;tag=as5c196cbb [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 5 [ 11]: CSeq: 2 BYE [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 7 [ 64]: Contact: ;reg-id=1 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 8 [ 26]: User-Agent: snom360/7.3.30 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 9 [ 78]: RTP-RxStat: Total_Rx_Pkts=493,Rx_Pkts=493,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 10 [ 61]: RTP-TxStat: Total_Tx_Pkts=508,Tx_Pkts=508,Remote_Tx_Pkts=1668 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: --- (12 headers 0 lines) --- [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3c2671b15914-fot3bvnvp7v5 (Checking From) --From tag vo1tbz2k84 --To-tag as5c196cbb [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Initializing initreq for method BYE - callid 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:51] DEBUG[5261] netsock2.c: Splitting '192.168.10.201:2051' into... [Aug 15 18:38:51] DEBUG[5261] netsock2.c: ...host '192.168.10.201' and port '2051'. [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.201:2051 (no NAT) [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:51] DEBUG[5261] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9081ba0' [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Session timer stopped: 162 - 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Scheduling destruction of SIP dialog '3c2671b15914-fot3bvnvp7v5' in 32000 ms (Method: BYE) [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Received bye, issuing owner hangup [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-qeow5twvohni;received=192.168.10.201;rport=2051 From: "User 4" ;tag=vo1tbz2k84 To: ;tag=as5c196cbb Call-ID: 3c2671b15914-fot3bvnvp7v5 CSeq: 2 BYE Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 1 [ 99]: Via: SIP/2.0/UDP 192.168.10.201:2051;branch=z9hG4bK-qeow5twvohni;received=192.168.10.201;rport=2051 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 2 [ 56]: From: "User 4" ;tag=vo1tbz2k84 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 3 [ 52]: To: ;tag=as5c196cbb [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 4 [ 34]: Call-ID: 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 5 [ 11]: CSeq: 2 BYE [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 10 [ 0]: [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.201:2051 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: BYE [Aug 15 18:38:51] DEBUG[5291] rtp_engine.c: Oooh, got a hangup [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Sending reinvite on SIP '848bff54a41b66ee' - It's audio soon redirected to IP 192.168.10.75:5060 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Strict routing enforced for session 848bff54a41b66ee [Aug 15 18:38:51] VERBOSE[5291] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:51] DEBUG[5291] netsock2.c: Splitting '192.168.10.209:5060' into... [Aug 15 18:38:51] DEBUG[5291] netsock2.c: ...host '192.168.10.209' and port '5060'. [Aug 15 18:38:51] VERBOSE[5291] chan_sip.c: set_destination: set destination to 192.168.10.209:5060 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Aug 15 18:38:51] VERBOSE[5291] chan_sip.c: Audio is at 5060 [Aug 15 18:38:51] VERBOSE[5291] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Aug 15 18:38:51] VERBOSE[5291] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Aug 15 18:38:51] VERBOSE[5291] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: -- Done with adding codecs to SDP [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Initializing already initialized SIP dialog 848bff54a41b66ee (presumably reinvite) [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK73a8e2e8 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 3 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 4 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 6 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "User 4" [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:51] VERBOSE[5291] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.209:5060: INVITE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK73a8e2e8 Max-Forwards: 70 From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Contact: Call-ID: 848bff54a41b66ee CSeq: 104 INVITE User-Agent: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "User 4" Content-Type: application/sdp Content-Length: 260 v=0 o=root 217296799 217296801 IN IP4 192.168.10.75 s=Asterisk PBX 1.8.5-2 c=IN IP4 192.168.10.75 t=0 0 m=audio 15448 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 --- [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK73a8e2e8 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 3 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 4 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 6 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "User 4" [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 14 [ 19]: Content-Length: 260 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Header 15 [ 0]: [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 1 [ 47]: o=root 217296799 217296801 IN IP4 192.168.10.75 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.5-2 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.75 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 5 [ 29]: m=audio 15448 RTP/AVP 8 0 101 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Body 11 [ 10]: a=sendrecv [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #189 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:38:51] DEBUG[5291] channel.c: Returning from native bridge, channels: SIP/phone3-00000003, SIP/phone4-00000006 [Aug 15 18:38:51] DEBUG[5291] pbx.c: Launching 'NoOp' [Aug 15 18:38:51] VERBOSE[5291] pbx.c: -- Executing [h@Standard:1] NoOp("SIP/phone3-00000003", "") in new stack [Aug 15 18:38:51] DEBUG[5291] channel.c: Hanging up channel 'SIP/phone4-00000006' [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Hangup call SIP/phone4-00000006, SIP callid 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: update_call_counter(phone4) - decrement call limit counter on hangup [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Updating call counter for incoming call [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Call from peer 'phone4' removed from call limit 2147483647 [Aug 15 18:38:51] DEBUG[5291] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9081ba0' [Aug 15 18:38:51] DEBUG[5291] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Aug 15 18:38:51] DEBUG[5291] pbx.c: Spawn extension (Standard,106,1) exited non-zero on 'SIP/phone3-00000003' [Aug 15 18:38:51] VERBOSE[5291] pbx.c: == Spawn extension (Standard, 106, 1) exited non-zero on 'SIP/phone3-00000003' [Aug 15 18:38:51] DEBUG[5291] channel.c: Soft-Hanging up channel 'SIP/phone3-00000003' [Aug 15 18:38:51] DEBUG[5291] channel.c: Hanging up channel 'SIP/phone3-00000003' [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Hangup call SIP/phone3-00000003, SIP callid 848bff54a41b66ee [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: update_call_counter(phone3) - decrement call limit counter on hangup [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Updating call counter for incoming call [Aug 15 18:38:51] DEBUG[5291] chan_sip.c: Call from peer 'phone3' removed from call limit 2147483647 [Aug 15 18:38:51] DEBUG[5291] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9062008' [Aug 15 18:38:51] VERBOSE[5291] chan_sip.c: Scheduling destruction of SIP dialog '848bff54a41b66ee' in 32000 ms (Method: ACK) [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone3-00000003 Channel2: SIP/phone4-00000006 Uniqueid1: 1313426307.3 Uniqueid2: 1313426307.4 CallerID1: 103 CallerID2: 104 [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 103 Destination: 106 DestinationContext: Standard CallerID: "User 3" <103> Channel: SIP/phone3-00000003 DestinationChannel: SIP/phone4-00000006 LastApplication: Dial LastData: SIP/phone1&SIP/phone2 StartTime: 2011-08-15 18:38:27 AnswerTime: 2011-08-15 18:38:41 EndTime: 2011-08-15 18:38:51 Duration: 24 BillableSeconds: 10 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1313426307.3 UserField: [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone4-00000006 Uniqueid: 1313426307.4 CallerIDNum: 104 CallerIDName: User 4 ConnectedLineNum: 103 ConnectedLineName: User 3 Cause: 16 Cause-txt: Normal Clearing [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone3-00000003 UniqueID: 1313426307.3 DialStatus: ANSWER [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone3-00000003 Uniqueid: 1313426307.3 CallerIDNum: 103 CallerIDName: User 3 ConnectedLineNum: 104 ConnectedLineName: User 4 Cause: 16 Cause-txt: Normal Clearing [Aug 15 18:38:51] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone4 [Aug 15 18:38:51] DEBUG[5240] chan_sip.c: Checking device state for peer phone4 [Aug 15 18:38:51] DEBUG[5240] devicestate.c: Changing state for SIP/phone4 - state 1 (Not in use) [Aug 15 18:38:51] DEBUG[5240] devicestate.c: device 'SIP/phone4' state '1' [Aug 15 18:38:51] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone4 [Aug 15 18:38:51] DEBUG[5240] chan_sip.c: Checking device state for peer phone4 [Aug 15 18:38:51] DEBUG[5240] devicestate.c: Changing state for SIP/phone4 - state 1 (Not in use) [Aug 15 18:38:51] DEBUG[5240] devicestate.c: device 'SIP/phone4' state '1' [Aug 15 18:38:51] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Aug 15 18:38:51] DEBUG[5240] chan_sip.c: Checking device state for peer phone3 [Aug 15 18:38:51] DEBUG[5240] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Aug 15 18:38:51] DEBUG[5240] devicestate.c: device 'SIP/phone3' state '1' [Aug 15 18:38:51] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Aug 15 18:38:51] DEBUG[5240] chan_sip.c: Checking device state for peer phone3 [Aug 15 18:38:51] DEBUG[5240] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Aug 15 18:38:51] DEBUG[5240] devicestate.c: device 'SIP/phone3' state '1' [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: BRIDGE_END AccountCode: CallerIDnum: 103 CallerIDname: User 3 CallerIDani: 103 CallerIDrdnis: CallerIDdnid: 106 Exten: 106 Context: Standard Channel: SIP/phone3-00000003 Application: Dial AppData: SIP/phone1&SIP/phone2 EventTime: 2011-08-15 18:38:51 AMAFlags: DOCUMENTATION UniqueID: 1313426307.3 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 104 CallerIDname: User 4 CallerIDani: 104 CallerIDrdnis: CallerIDdnid: *8 Exten: Context: Standard Channel: SIP/phone4-00000006 Application: AppDial AppData: (Outgoing Line) EventTime: 2011-08-15 18:38:51 AMAFlags: DOCUMENTATION UniqueID: 1313426307.4 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_END AccountCode: CallerIDnum: 104 CallerIDname: User 4 CallerIDani: 104 CallerIDrdnis: CallerIDdnid: *8 Exten: Context: Standard Channel: SIP/phone4-00000006 Application: AppDial AppData: (Outgoing Line) EventTime: 2011-08-15 18:38:51 AMAFlags: DOCUMENTATION UniqueID: 1313426307.4 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: APP_END AccountCode: CallerIDnum: 103 CallerIDname: User 3 CallerIDani: 103 CallerIDrdnis: CallerIDdnid: 106 Exten: 106 Context: Standard Channel: SIP/phone3-00000003 Application: Dial AppData: SIP/phone1&SIP/phone2 EventTime: 2011-08-15 18:38:51 AMAFlags: DOCUMENTATION UniqueID: 1313426307.3 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: HANGUP AccountCode: CallerIDnum: 103 CallerIDname: User 3 CallerIDani: 103 CallerIDrdnis: CallerIDdnid: 106 Exten: 106 Context: Standard Channel: SIP/phone3-00000003 Application: AppData: EventTime: 2011-08-15 18:38:51 AMAFlags: DOCUMENTATION UniqueID: 1313426307.3 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: CEL Privilege: call,all EventName: CHAN_END AccountCode: CallerIDnum: 103 CallerIDname: User 3 CallerIDani: 103 CallerIDrdnis: CallerIDdnid: 106 Exten: 106 Context: Standard Channel: SIP/phone3-00000003 Application: AppData: EventTime: 2011-08-15 18:38:51 AMAFlags: DOCUMENTATION UniqueID: 1313426307.3 LinkedID: 1313426307.3 Userfield: Peer: [Aug 15 18:38:51] DEBUG[5241] app_queue.c: Extension '104@_extensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:51] DEBUG[5241] app_queue.c: Extension '103@_extensions' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:51] DEBUG[5267] app_queue.c: Device 'SIP/phone4' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:51] DEBUG[5267] app_queue.c: Device 'SIP/phone4' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:51] DEBUG[5267] app_queue.c: Device 'SIP/phone3' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:51] DEBUG[5267] app_queue.c: Device 'SIP/phone3' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 104 Context: _extensions Hint: SIP/phone4 Status: 0 [Aug 15 18:38:51] DEBUG[5268] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 103 Context: _extensions Hint: SIP/phone3 Status: 0 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.209:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK73a8e2e8 From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Call-ID: 848bff54a41b66ee CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "user 3" ;+sip.instance="" Server: Aastra 53i/3.2.1.43 Supported: path, replaces Content-Type: application/sdp Content-Length: 296 v=0 o=MxSIP 0 3 IN IP4 192.168.10.209 s=SIP Call c=IN IP4 192.168.10.209 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=rtcp:3001 IN IP4 192.168.10.209 a=sendrecv <-------------> [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK73a8e2e8 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 2 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 3 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 4 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 8 [128]: Contact: "user 3" ;+sip.instance="" [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 9 [ 27]: Server: Aastra 53i/3.2.1.43 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 10 [ 25]: Supported: path, replaces [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 12 [ 19]: Content-Length: 296 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 13 [ 0]: [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 0 [ 3]: v=0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.209 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 2 [ 10]: s=SIP Call [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.209 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 4 [ 5]: t=0 0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 12 [ 33]: a=rtcp:3001 IN IP4 192.168.10.209 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Body 13 [ 10]: a=sendrecv [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: --- (13 headers 14 lines) --- [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: = Looking for Call ID: 848bff54a41b66ee (Checking To) --From tag as5b35b701 --To-tag f1865c1211 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Acked pending invite 104 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #189 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Stopping retransmission on '848bff54a41b66ee' of Request 104: Match Found [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: SIP response 200 to standard invite [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.10.209... UNSUPPORTED. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Aug 15 18:38:51] DEBUG[5261] netsock2.c: Splitting '192.168.10.209' into... [Aug 15 18:38:51] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port ''. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.209... OK. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Found RTP audio format 8 [Aug 15 18:38:51] DEBUG[5261] rtp_engine.c: Setting payload 8 based on m type on 0xb616c588 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Found RTP audio format 0 [Aug 15 18:38:51] DEBUG[5261] rtp_engine.c: Setting payload 0 based on m type on 0xb616c588 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Found RTP audio format 101 [Aug 15 18:38:51] DEBUG[5261] rtp_engine.c: Setting payload 101 based on m type on 0xb616c588 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Found audio description format PCMA for ID 8 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Found audio description format PCMU for ID 0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Found audio description format telephone-event for ID 101 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=rtcp:3001 IN IP4 192.168.10.209... UNSUPPORTED. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Aug 15 18:38:51] DEBUG[5261] rtp_engine.c: Incorporating payload 0 on 0xb616c588 [Aug 15 18:38:51] DEBUG[5261] rtp_engine.c: Incorporating payload 8 on 0xb616c588 [Aug 15 18:38:51] DEBUG[5261] rtp_engine.c: Incorporating payload 101 on 0xb616c588 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 15 18:38:51] DEBUG[5261] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9062008' [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Peer audio RTP is at port 192.168.10.209:3000 [Aug 15 18:38:51] DEBUG[5261] rtp_engine.c: Copying payload 0 from 0xb616c588 to 0x90621b4 [Aug 15 18:38:51] DEBUG[5261] rtp_engine.c: Copying payload 8 from 0xb616c588 to 0x90621b4 [Aug 15 18:38:51] DEBUG[5261] rtp_engine.c: Copying payload 101 from 0xb616c588 to 0x90621b4 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Updating call counter for incoming call [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: build_route: Retaining previous route: [Aug 15 18:38:51] DEBUG[5261] netsock2.c: Splitting '192.168.10.209:5060' into... [Aug 15 18:38:51] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port '5060'. [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Strict routing enforced for session 848bff54a41b66ee [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:51] DEBUG[5261] netsock2.c: Splitting '192.168.10.209:5060' into... [Aug 15 18:38:51] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port '5060'. [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: set_destination: set destination to 192.168.10.209:5060 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Transmitting (no NAT) to 192.168.10.209:5060: ACK sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK62bdaabf Max-Forwards: 70 From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Contact: Call-ID: 848bff54a41b66ee CSeq: 104 ACK User-Agent: IPTAM PBX Content-Length: 0 --- [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK62bdaabf [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 3 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 4 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 5 [ 37]: Contact: [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 6 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 7 [ 13]: CSeq: 104 ACK [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 8 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 10 [ 0]: [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Strict routing enforced for session 848bff54a41b66ee [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: set_destination: Parsing for address/port to send to [Aug 15 18:38:51] DEBUG[5261] netsock2.c: Splitting '192.168.10.209:5060' into... [Aug 15 18:38:51] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port '5060'. [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: set_destination: set destination to 192.168.10.209:5060 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.209:5060: BYE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK6acf18fe Max-Forwards: 70 From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Call-ID: 848bff54a41b66ee CSeq: 105 BYE User-Agent: IPTAM PBX X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 0 [ 56]: BYE sip:phone3@192.168.10.209:5060;transport=udp SIP/2.0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK6acf18fe [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 3 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 4 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 5 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 6 [ 13]: CSeq: 105 BYE [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 7 [ 21]: User-Agent: IPTAM PBX [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Header 11 [ 0]: [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #191 [Aug 15 18:38:51] DEBUG[5261] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:38:51] VERBOSE[5261] chan_sip.c: Scheduling destruction of SIP dialog '848bff54a41b66ee' in 32000 ms (Method: ACK) [Aug 15 18:38:51] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Aug 15 18:38:51] DEBUG[5240] chan_sip.c: Checking device state for peer phone3 [Aug 15 18:38:51] DEBUG[5240] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Aug 15 18:38:51] DEBUG[5240] devicestate.c: device 'SIP/phone3' state '1' [Aug 15 18:38:51] DEBUG[5267] app_queue.c: Device 'SIP/phone3' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:38:52] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.209:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK6acf18fe From: ;tag=as5b35b701 To: "user 3" ;tag=f1865c1211 Call-ID: 848bff54a41b66ee CSeq: 105 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 53i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.75:5060;branch=z9hG4bK6acf18fe [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 2 [ 60]: From: ;tag=as5b35b701 [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 3 [ 59]: To: "user 3" ;tag=f1865c1211 [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 4 [ 25]: Call-ID: 848bff54a41b66ee [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 5 [ 13]: CSeq: 105 BYE [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 8 [ 27]: Server: Aastra 53i/3.2.1.43 [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 9 [ 15]: Supported: path [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Aug 15 18:38:52] VERBOSE[5261] chan_sip.c: --- (11 headers 0 lines) --- [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: = Looking for Call ID: 848bff54a41b66ee (Checking To) --From tag as5b35b701 --To-tag f1865c1211 [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #191 [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Stopping retransmission on '848bff54a41b66ee' of Request 105: Match Found [Aug 15 18:38:52] DEBUG[5261] chan_sip.c: Destroying SIP dialog 848bff54a41b66ee [Aug 15 18:38:52] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '848bff54a41b66ee' Method: ACK [Aug 15 18:38:52] DEBUG[5261] rtp_engine.c: Destroyed RTP instance '0x9062008' [Aug 15 18:39:15] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.209:5060 ---> REGISTER sip:192.168.10.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK14d3e6408f0de8edd Max-Forwards: 70 From: "user 3" ;tag=ee5bf79915 To: "user 3" Call-ID: 3b76f1b91f8170df CSeq: 11635 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "user 3" ;+sip.instance="" Supported: path, gruu User-Agent: Aastra 53i/3.2.1.43 Aastra-Line: 1 Content-Length: 0 <-------------> [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 0 [ 39]: REGISTER sip:192.168.10.75:5060 SIP/2.0 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK14d3e6408f0de8edd [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 3 [ 61]: From: "user 3" ;tag=ee5bf79915 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 4 [ 44]: To: "user 3" [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 5 [ 25]: Call-ID: 3b76f1b91f8170df [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 6 [ 20]: CSeq: 11635 REGISTER [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 9 [128]: Contact: "user 3" ;+sip.instance="" [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 10 [ 21]: Supported: path, gruu [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 53i/3.2.1.43 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 12 [ 14]: Aastra-Line: 1 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 15 18:39:15] VERBOSE[5261] chan_sip.c: --- (14 headers 0 lines) --- [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: = Looking for Call ID: 3b76f1b91f8170df (Checking From) --From tag ee5bf79915 --To-tag [Aug 15 18:39:15] DEBUG[5261] acl.c: For destination '192.168.10.209', our source address is '192.168.10.75'. [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Allocating new SIP dialog for 3b76f1b91f8170df - REGISTER (No RTP) [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Initializing initreq for method REGISTER - callid 3b76f1b91f8170df [Aug 15 18:39:15] DEBUG[5261] netsock2.c: Splitting '192.168.10.209' into... [Aug 15 18:39:15] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port ''. [Aug 15 18:39:15] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.209:5060 (no NAT) [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Store REGISTER's Contact header for call routing. [Aug 15 18:39:15] DEBUG[5261] netsock2.c: Splitting '192.168.10.209:5060' into... [Aug 15 18:39:15] DEBUG[5261] netsock2.c: ...host '192.168.10.209' and port '5060'. [Aug 15 18:39:15] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.209:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK14d3e6408f0de8edd;received=192.168.10.209 From: "user 3" ;tag=ee5bf79915 To: "user 3" ;tag=as6224626f Call-ID: 3b76f1b91f8170df CSeq: 11635 REGISTER Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 15 Aug 2011 16:39:15 GMT Content-Length: 0 <------------> [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.10.209;branch=z9hG4bK14d3e6408f0de8edd;received=192.168.10.209 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 2 [ 61]: From: "user 3" ;tag=ee5bf79915 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 3 [ 59]: To: "user 3" ;tag=as6224626f [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 4 [ 25]: Call-ID: 3b76f1b91f8170df [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 5 [ 20]: CSeq: 11635 REGISTER [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 9 [ 12]: Expires: 120 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 11 [ 35]: Date: Mon, 15 Aug 2011 16:39:15 GMT [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Header 13 [ 0]: [Aug 15 18:39:15] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.209:5060 [Aug 15 18:39:15] VERBOSE[5261] chan_sip.c: Scheduling destruction of SIP dialog '3b76f1b91f8170df' in 32000 ms (Method: REGISTER) [Aug 15 18:39:15] DEBUG[5268] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone3 PeerStatus: Registered Address: 192.168.10.209:5060 [Aug 15 18:39:15] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Aug 15 18:39:15] DEBUG[5240] chan_sip.c: Checking device state for peer phone3 [Aug 15 18:39:15] DEBUG[5240] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Aug 15 18:39:15] DEBUG[5240] devicestate.c: device 'SIP/phone3' state '1' [Aug 15 18:39:15] DEBUG[5267] app_queue.c: Device 'SIP/phone3' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:39:18] VERBOSE[5271] asterisk.c: -- Remote UNIX connection disconnected [Aug 15 18:39:23] DEBUG[5261] chan_sip.c: Auto destroying SIP dialog '3c2671b15914-fot3bvnvp7v5' [Aug 15 18:39:23] DEBUG[5261] chan_sip.c: Destroying SIP dialog 3c2671b15914-fot3bvnvp7v5 [Aug 15 18:39:23] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '3c2671b15914-fot3bvnvp7v5' Method: BYE [Aug 15 18:39:23] DEBUG[5261] rtp_engine.c: Destroyed RTP instance '0x9081ba0' [Aug 15 18:39:46] VERBOSE[5261] chan_sip.c: <--- SIP read from UDP:192.168.10.208:5060 ---> REGISTER sip:192.168.10.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.208:5060;branch=z9hG4bK11592f41bd44d46b4.94c2baee5f101f9d1 Max-Forwards: 70 From: ;tag=0da8a8a0d0 To: Call-ID: 6b06460c523fb0c1 CSeq: 641 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "User 2" ;+sip.instance="" Supported: gruu, path User-Agent: Aastra 51i/2.6.0.2019 Aastra-Line: 1 Content-Length: 0 <-------------> [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 0 [ 39]: REGISTER sip:192.168.10.75:5060 SIP/2.0 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.208:5060;branch=z9hG4bK11592f41bd44d46b4.94c2baee5f101f9d1 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 3 [ 52]: From: ;tag=0da8a8a0d0 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 4 [ 35]: To: [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 5 [ 25]: Call-ID: 6b06460c523fb0c1 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 6 [ 18]: CSeq: 641 REGISTER [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 9 [128]: Contact: "User 2" ;+sip.instance="" [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 51i/2.6.0.2019 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 12 [ 14]: Aastra-Line: 1 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Aug 15 18:39:46] VERBOSE[5261] chan_sip.c: --- (14 headers 0 lines) --- [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: = Looking for Call ID: 6b06460c523fb0c1 (Checking From) --From tag 0da8a8a0d0 --To-tag [Aug 15 18:39:46] DEBUG[5261] acl.c: For destination '192.168.10.208', our source address is '192.168.10.75'. [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.75:5060 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Allocating new SIP dialog for 6b06460c523fb0c1 - REGISTER (No RTP) [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Initializing initreq for method REGISTER - callid 6b06460c523fb0c1 [Aug 15 18:39:46] DEBUG[5261] netsock2.c: Splitting '192.168.10.208:5060' into... [Aug 15 18:39:46] DEBUG[5261] netsock2.c: ...host '192.168.10.208' and port '5060'. [Aug 15 18:39:46] VERBOSE[5261] chan_sip.c: Sending to 192.168.10.208:5060 (no NAT) [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Store REGISTER's Contact header for call routing. [Aug 15 18:39:46] DEBUG[5261] netsock2.c: Splitting '192.168.10.208:5060' into... [Aug 15 18:39:46] DEBUG[5261] netsock2.c: ...host '192.168.10.208' and port '5060'. [Aug 15 18:39:46] VERBOSE[5261] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.208:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.208:5060;branch=z9hG4bK11592f41bd44d46b4.94c2baee5f101f9d1;received=192.168.10.208 From: ;tag=0da8a8a0d0 To: ;tag=as0fb0ae51 Call-ID: 6b06460c523fb0c1 CSeq: 641 REGISTER Server: IPTAM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 15 Aug 2011 16:39:46 GMT Content-Length: 0 <------------> [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.208:5060;branch=z9hG4bK11592f41bd44d46b4.94c2baee5f101f9d1;received=192.168.10.208 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 2 [ 52]: From: ;tag=0da8a8a0d0 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 3 [ 50]: To: ;tag=as0fb0ae51 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 4 [ 25]: Call-ID: 6b06460c523fb0c1 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 5 [ 18]: CSeq: 641 REGISTER [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 6 [ 17]: Server: IPTAM PBX [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 9 [ 12]: Expires: 120 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 11 [ 35]: Date: Mon, 15 Aug 2011 16:39:46 GMT [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Header 13 [ 0]: [Aug 15 18:39:46] DEBUG[5261] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.208:5060 [Aug 15 18:39:46] VERBOSE[5261] chan_sip.c: Scheduling destruction of SIP dialog '6b06460c523fb0c1' in 32000 ms (Method: REGISTER) [Aug 15 18:39:46] DEBUG[5268] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone2 PeerStatus: Registered Address: 192.168.10.208:5060 [Aug 15 18:39:46] DEBUG[5240] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Aug 15 18:39:46] DEBUG[5240] chan_sip.c: Checking device state for peer phone2 [Aug 15 18:39:46] DEBUG[5240] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Aug 15 18:39:46] DEBUG[5240] devicestate.c: device 'SIP/phone2' state '1' [Aug 15 18:39:46] DEBUG[5267] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Aug 15 18:39:47] DEBUG[5261] chan_sip.c: Auto destroying SIP dialog '3b76f1b91f8170df' [Aug 15 18:39:47] DEBUG[5261] chan_sip.c: Destroying SIP dialog 3b76f1b91f8170df [Aug 15 18:39:47] VERBOSE[5261] chan_sip.c: Really destroying SIP dialog '3b76f1b91f8170df' Method: REGISTER