[Nov 22 11:55:05] VERBOSE[1173] config.c: == Parsing '/etc/asterisk/logger.conf': [Nov 22 11:55:05] DEBUG[1173] config.c: Parsing /etc/asterisk/logger.conf [Nov 22 11:55:05] VERBOSE[1173] config.c: == Found [Nov 22 11:55:05] VERBOSE[1173] logger.c: Asterisk Queue Logger restarted [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> INVITE sip:150@192.168.10.70 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK7b748f352b774b231.c8f07a666c7b77df8 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=673d7b2f8c To: "150" Call-ID: 67985a870350a076 CSeq: 23316 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erika Musterfrau" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 53i/2.6.0.1008 Content-Type: application/sdp Content-Length: 620 v=0 o=MxSIP 0 0 IN IP4 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 0 [ 36]: INVITE sip:150@192.168.10.70 SIP/2.0 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK7b748f352b774b231.c8f07a666c7b77df8 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 3 [ 66]: From: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 4 [ 33]: To: "150" [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 6 [ 18]: CSeq: 23316 INVITE [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 9 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 53i/2.6.0.1008 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 13 [ 19]: Content-Length: 620 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 14 [ 0]: [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.201 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:11] DEBUG[1175] 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 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 21 [ 24]: a=silenceSupp:on - - - - [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 22 [ 20]: a=fmtp:18 annexb=yes [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Body 25 [ 10]: a=sendrecv [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: --- (14 headers 26 lines) --- [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: = Looking for Call ID: 67985a870350a076 (Checking From) --From tag 673d7b2f8c --To-tag [Nov 22 11:55:11] DEBUG[1175] acl.c: For destination '192.168.10.201', our source address is '192.168.10.70'. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Allocating new SIP dialog for 67985a870350a076 - INVITE (No RTP) [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -gruu- [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: gruu [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -path- [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: path [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: timer [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 22 11:55:11] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 22 11:55:11] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:55:11] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.201:5060 (no NAT) [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Initializing initreq for method INVITE - callid 67985a870350a076 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Using INVITE request as basis request - 67985a870350a076 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found peer 'phone1' for 'phone1' from 192.168.10.201:5060 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd800760' [Nov 22 11:55:11] DEBUG[1175] res_rtp_asterisk.c: Allocated port 14070 for RTP instance '0xd800760' [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: RTP instance '0xd800760' is setup and ready to go [Nov 22 11:55:11] DEBUG[1175] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd800760' [Nov 22 11:55:11] VERBOSE[1175] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 22 11:55:11] VERBOSE[1175] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Setting NAT on RTP to Off [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.201... UNSUPPORTED. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:55:11] DEBUG[1175] netsock2.c: Splitting '192.168.10.201' gives... [Nov 22 11:55:11] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '(null)'. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 18 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 18 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 106 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 106 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 107 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 113 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 110 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 110 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 111 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 111 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 112 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 112 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 98 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 98 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 97 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 97 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 115 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 115 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 96 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 9 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 9 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format G729 for ID 18 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format BV16 for ID 106 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format BV32 for ID 107 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 113 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 110 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 111 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 112 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format G726-16 for ID 98 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format G726-24 for ID 97 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format G726-32 for ID 115 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format G726-40 for ID 96 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format G722 for ID 9 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:on - - - -... UNSUPPORTED. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 9 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 18 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 97 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 98 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 106 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 110 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 111 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 112 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 113 on 0xb3380f78 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Incorporating payload 115 on 0xb3380f78 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8109f4c (ulaw|alaw|g726|slin|g729|speex|ilbc|g722|slin16|h263p|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:55:11] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd800760' [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 9 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 18 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 97 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 98 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 106 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 110 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 111 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 112 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 113 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] rtp_engine.c: Copying payload 115 from 0xb3380f78 to 0xd80090c [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Checking SIP call limits for device phone1 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Call from peer 'phone1' is 1 out of 2147483647 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: Looking for 150 in Standard (domain 192.168.10.70) [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: This channel will not be able to handle video. [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: build_route: Contact hop: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: list_route: hop: [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: SIP/phone1-0000000c: New call is still down.... Trying... [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK7b748f352b774b231.c8f07a666c7b77df8;received=192.168.10.201 From: "Erika Musterfrau" ;tag=673d7b2f8c To: "150" Call-ID: 67985a870350a076 CSeq: 23316 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK7b748f352b774b231.c8f07a666c7b77df8;received=192.168.10.201 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 2 [ 66]: From: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 3 [ 33]: To: "150" [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 5 [ 18]: CSeq: 23316 INVITE [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:55:11] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:55:11] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:55:11] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:55:11] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:55:11] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:55:11] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:55:11] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:55:11] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:55:11] DEBUG[1195] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone1-0000000c ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Erika Musterfrau AccountCode: Exten: 150 Context: Standard Uniqueid: 1290423311.12 [Nov 22 11:55:11] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-0000000c ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1290423311.12 [Nov 22 11:55:11] DEBUG[1244] pbx.c: Launching 'Dial' [Nov 22 11:55:11] VERBOSE[1244] pbx.c: -- Executing [150@Standard:1] Dial("SIP/phone1-0000000c", "SIP/phone2") in new stack [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Allocating new SIP dialog for 70896c3b41a848f6228ec11e1ab58977@192.168.10.70 - INVITE (No RTP) [Nov 22 11:55:11] DEBUG[1244] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd823478' [Nov 22 11:55:11] DEBUG[1244] res_rtp_asterisk.c: Allocated port 10632 for RTP instance '0xd823478' [Nov 22 11:55:11] DEBUG[1244] rtp_engine.c: RTP instance '0xd823478' is setup and ready to go [Nov 22 11:55:11] DEBUG[1244] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd823478' [Nov 22 11:55:11] VERBOSE[1244] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 22 11:55:11] VERBOSE[1244] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Setting NAT on RTP to Off [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 22 11:55:11] DEBUG[1244] acl.c: For destination '192.168.10.200', our source address is '192.168.10.70'. [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: This channel will not be able to handle video. [Nov 22 11:55:11] DEBUG[1244] rtp_engine.c: Seeded SDP of 'SIP/phone2-0000000d' with that of 'SIP/phone1-0000000c' [Nov 22 11:55:11] DEBUG[1244] channel.c: Not copying variable DIALEDTIME. [Nov 22 11:55:11] DEBUG[1244] channel.c: Not copying variable ANSWEREDTIME. [Nov 22 11:55:11] DEBUG[1244] channel.c: Not copying variable DIALEDPEERNAME. [Nov 22 11:55:11] DEBUG[1244] channel.c: Not copying variable DIALEDPEERNUMBER. [Nov 22 11:55:11] DEBUG[1244] channel.c: Not copying variable DIALSTATUS. [Nov 22 11:55:11] DEBUG[1244] channel.c: Not copying variable SIPCALLID. [Nov 22 11:55:11] DEBUG[1244] channel.c: Not copying variable SIPDOMAIN. [Nov 22 11:55:11] DEBUG[1244] channel.c: Not copying variable SIPURI. [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Outgoing Call for phone2 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Call to peer 'phone2' is 1 out of 2147483647 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:55:11] VERBOSE[1244] chan_sip.c: Audio is at 5060 [Nov 22 11:55:11] VERBOSE[1244] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:55:11] VERBOSE[1244] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:55:11] VERBOSE[1244] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Initializing initreq for method INVITE - callid 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1881b225 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 4 [ 50]: To: [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 6 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 9 [ 35]: Date: Mon, 22 Nov 2010 10:55:11 GMT [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:55:11] VERBOSE[1244] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.200:5060: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1881b225 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as73af062e To: Contact: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0-1 Date: Mon, 22 Nov 2010 10:55:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Erika Musterfrau" Content-Type: application/sdp Content-Length: 260 v=0 o=root 534818622 534818622 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 10632 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 --- [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1881b225 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 4 [ 50]: To: [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 6 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 9 [ 35]: Date: Mon, 22 Nov 2010 10:55:11 GMT [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 14 [ 19]: Content-Length: 260 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 15 [ 0]: [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 1 [ 47]: o=root 534818622 534818622 IN IP4 192.168.10.70 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 5 [ 29]: m=audio 10632 RTP/AVP 8 0 101 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #339 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:11] VERBOSE[1244] app_dial.c: -- Called phone2 [Nov 22 11:55:11] DEBUG[1195] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-0000000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 150 CallerIDName: Hans Muster AccountCode: Exten: Context: Standard Uniqueid: 1290423311.13 [Nov 22 11:55:11] DEBUG[1195] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone1-0000000c Destination: SIP/phone2-0000000d CallerIDNum: 100 CallerIDName: Erika Musterfrau UniqueID: 1290423311.12 DestUniqueID: 1290423311.13 Dialstring: phone2 [Nov 22 11:55:11] DEBUG[1195] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone2-0000000d CallerIDNum: 150 CallerIDName: Uniqueid: 1290423311.13 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Nov 22 11:55:11] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:11] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:11] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Nov 22 11:55:11] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '6' [Nov 22 11:55:11] DEBUG[1168] app_queue.c: Extension '100@Standard' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:11] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:11] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:11] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 1 [Nov 22 11:55:11] DEBUG[1168] app_queue.c: Extension '150@Standard' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:55:11] DEBUG[1194] 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. [Nov 22 11:55:11] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 8 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1881b225 From: "Erika Musterfrau" ;tag=as73af062e To: ;tag=3444016759 Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Server: Aastra 55i/2.6.0.66 Supported: gruu, path Content-Length: 0 <-------------> [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1881b225 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3444016759 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 8 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 22 11:55:11] VERBOSE[1175] chan_sip.c: --- (12 headers 0 lines) --- [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: = Looking for Call ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 (Checking To) --From tag as73af062e --To-tag 3444016759 [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: *** SIP TIMER: Cancelling retransmission #339 - INVITE (got response) [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '15c91da73e9014a058805f32700631ed@192.168.10.70' Request 102: Found [Nov 22 11:55:11] DEBUG[1175] chan_sip.c: SIP response 180 to standard invite [Nov 22 11:55:11] VERBOSE[1244] app_dial.c: -- SIP/phone2-0000000d is ringing [Nov 22 11:55:11] DEBUG[1244] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-0000000c' with that of 'SIP/phone2-0000000d' [Nov 22 11:55:11] VERBOSE[1244] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK7b748f352b774b231.c8f07a666c7b77df8;received=192.168.10.201 From: "Erika Musterfrau" ;tag=673d7b2f8c To: "150" ;tag=as2d91b5e0 Call-ID: 67985a870350a076 CSeq: 23316 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK7b748f352b774b231.c8f07a666c7b77df8;received=192.168.10.201 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 2 [ 66]: From: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 3 [ 48]: To: "150" ;tag=as2d91b5e0 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 4 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 5 [ 18]: CSeq: 23316 INVITE [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Header 11 [ 0]: [Nov 22 11:55:11] DEBUG[1244] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:55:11] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:11] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:11] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Nov 22 11:55:11] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '6' [Nov 22 11:55:11] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-0000000d ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 150 CallerIDName: Uniqueid: 1290423311.13 [Nov 22 11:55:11] DEBUG[1194] 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. [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1881b225 From: "Erika Musterfrau" ;tag=as73af062e To: ;tag=3444016759 Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Server: Aastra 55i/2.6.0.66 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 0 IN IP4 192.168.10.200 s=SIP Call c=IN IP4 192.168.10.200 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 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1881b225 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3444016759 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 8 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.200 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: = Looking for Call ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 (Checking To) --From tag as73af062e --To-tag 3444016759 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Acked pending invite 102 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Stopping retransmission on '15c91da73e9014a058805f32700631ed@192.168.10.70' of Request 102: Match Found [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: SIP response 200 to standard invite [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.200... UNSUPPORTED. [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:55:12] DEBUG[1175] netsock2.c: Splitting '192.168.10.200' gives... [Nov 22 11:55:12] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '(null)'. [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.200... OK. [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:55:12] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:55:12] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:55:12] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:55:12] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:55:12] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:55:12] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:55:12] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd823478' [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.200:3000 [Nov 22 11:55:12] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd823624 [Nov 22 11:55:12] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd823624 [Nov 22 11:55:12] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd823624 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: build_route: Contact hop: "" ;+sip.instance="" [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: list_route: hop: [Nov 22 11:55:12] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:55:12] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Strict routing enforced for session 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:55:12] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:55:12] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.200:5060: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4089c4bd Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as73af062e To: ;tag=3444016759 Contact: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4089c4bd [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=3444016759 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:12] VERBOSE[1244] app_dial.c: -- SIP/phone2-0000000d answered SIP/phone1-0000000c [Nov 22 11:55:12] DEBUG[1244] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-0000000c' with that of 'SIP/phone2-0000000d' [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: SIP answering channel: SIP/phone1-0000000c [Nov 22 11:55:12] DEBUG[1244] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Setting framing from config on incoming call [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:55:12] VERBOSE[1244] chan_sip.c: Audio is at 5060 [Nov 22 11:55:12] VERBOSE[1244] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:55:12] VERBOSE[1244] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:55:12] VERBOSE[1244] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:55:12] VERBOSE[1244] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK7b748f352b774b231.c8f07a666c7b77df8;received=192.168.10.201 From: "Erika Musterfrau" ;tag=673d7b2f8c To: "150" ;tag=as2d91b5e0 Call-ID: 67985a870350a076 CSeq: 23316 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "Hans Muster" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1729646775 1729646775 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 14070 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 <------------> [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK7b748f352b774b231.c8f07a666c7b77df8;received=192.168.10.201 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 2 [ 66]: From: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 3 [ 48]: To: "150" ;tag=as2d91b5e0 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 4 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 5 [ 18]: CSeq: 23316 INVITE [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 10 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 12 [ 19]: Content-Length: 262 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Header 13 [ 0]: [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 1 [ 49]: o=root 1729646775 1729646775 IN IP4 192.168.10.70 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 5 [ 29]: m=audio 14070 RTP/AVP 8 0 101 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #342 [Nov 22 11:55:12] DEBUG[1244] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:55:12] DEBUG[1244] features.c: bridge answer set, chan answer set [Nov 22 11:55:12] DEBUG[1244] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:55:12] DEBUG[1244] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:55:12] VERBOSE[1244] rtp_engine.c: -- Locally bridging SIP/phone1-0000000c and SIP/phone2-0000000d [Nov 22 11:55:12] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:12] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:12] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:55:12] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:55:12] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:12] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:12] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:55:12] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:55:12] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:55:12] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:55:12] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:55:12] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:55:12] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-0000000d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 150 CallerIDName: Uniqueid: 1290423311.13 [Nov 22 11:55:12] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-0000000c ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1290423311.12 [Nov 22 11:55:12] DEBUG[1195] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone2-0000000d Uniqueid: 1290423311.13 AccountCode: OldAccountCode: [Nov 22 11:55:12] DEBUG[1195] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone1-0000000c Channel2: SIP/phone2-0000000d Uniqueid1: 1290423311.12 Uniqueid2: 1290423311.13 CallerID1: 100 CallerID2: 150 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: INVITE [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '67985a870350a076' Method: INVITE [Nov 22 11:55:12] DEBUG[1168] app_queue.c: Extension '150@Standard' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:12] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:12] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:12] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:12] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 1 [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> ACK sip:150@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bKa5161b93ffb13d464.dad6e123c7fb89f3e Max-Forwards: 70 From: "Erika Musterfrau" ;tag=673d7b2f8c To: "150" ;tag=as2d91b5e0 Call-ID: 67985a870350a076 CSeq: 23316 ACK User-Agent: Aastra 53i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 0 [ 38]: ACK sip:150@192.168.10.70:5060 SIP/2.0 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bKa5161b93ffb13d464.dad6e123c7fb89f3e [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 3 [ 66]: From: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "150" ;tag=as2d91b5e0 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 6 [ 15]: CSeq: 23316 ACK [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 7 [ 33]: User-Agent: Aastra 53i/2.6.0.1008 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 22 11:55:12] VERBOSE[1175] chan_sip.c: --- (9 headers 0 lines) --- [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: = Looking for Call ID: 67985a870350a076 (Checking From) --From tag 673d7b2f8c --To-tag as2d91b5e0 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #342 [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Stopping retransmission on '67985a870350a076' of Response 23316: Match Found [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: INVITE [Nov 22 11:55:12] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '67985a870350a076' Method: ACK [Nov 22 11:55:13] DEBUG[1244] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:13] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.200:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 30236 SequenceNumberCycles: 0 IAJitter: 3 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Nov 22 11:55:13] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: INVITE [Nov 22 11:55:13] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '67985a870350a076' Method: ACK [Nov 22 11:55:14] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: INVITE [Nov 22 11:55:14] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '67985a870350a076' Method: ACK [Nov 22 11:55:15] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: INVITE [Nov 22 11:55:15] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '67985a870350a076' Method: ACK [Nov 22 11:55:16] DEBUG[1244] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:16] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.201:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 33554435 FractionLost: 3 PacketsLost: 2 HighestSequence: 26718 SequenceNumberCycles: 0 IAJitter: 15 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Nov 22 11:55:16] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: INVITE [Nov 22 11:55:16] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '67985a870350a076' Method: ACK [Nov 22 11:55:17] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: INVITE [Nov 22 11:55:17] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '67985a870350a076' Method: ACK [Nov 22 11:55:18] DEBUG[1244] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:18] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.200:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 150994956 FractionLost: 12 PacketsLost: 9 HighestSequence: 30426 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Nov 22 11:55:18] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: INVITE [Nov 22 11:55:18] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '67985a870350a076' Method: ACK [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> INVITE sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2fe60d4aa16c2b9d4.abada4d52b7f6367a Max-Forwards: 70 From: ;tag=3444016759 To: "Erika Musterfrau" ;tag=as73af062e Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 26430 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.6.0.66 Content-Type: application/sdp Content-Length: 620 v=0 o=MxSIP 0 1 IN IP4 192.168.10.200 s=SIP Call c=IN IP4 192.168.10.200 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:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendonly <-------------> [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 0 [ 41]: INVITE sip:100@192.168.10.70:5060 SIP/2.0 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2fe60d4aa16c2b9d4.abada4d52b7f6367a [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 3 [ 67]: From: ;tag=3444016759 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 6 [ 18]: CSeq: 26430 INVITE [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 9 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 13 [ 19]: Content-Length: 620 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 14 [ 0]: [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.200 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:19] DEBUG[1175] 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 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 21 [ 24]: a=silenceSupp:on - - - - [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 22 [ 20]: a=fmtp:18 annexb=yes [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 25 [ 10]: a=sendonly [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: --- (14 headers 26 lines) --- [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: = Looking for Call ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 (Checking From) --From tag 3444016759 --To-tag as73af062e [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -gruu- [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: gruu [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -path- [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: path [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: timer [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 22 11:55:19] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 22 11:55:19] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:55:19] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.200:5060 (no NAT) [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Initializing initreq for method INVITE - callid 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.200... UNSUPPORTED. [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:55:19] DEBUG[1175] netsock2.c: Splitting '192.168.10.200' gives... [Nov 22 11:55:19] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '(null)'. [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.200... OK. [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 18 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 18 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 106 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 106 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 107 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 113 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 110 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 110 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 111 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 111 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 112 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 112 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 98 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 98 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 97 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 97 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 115 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 115 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 96 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 9 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 9 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format G729 for ID 18 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format BV16 for ID 106 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format BV32 for ID 107 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 113 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 110 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 111 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 112 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format G726-16 for ID 98 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format G726-24 for ID 97 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format G726-32 for ID 115 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format G726-40 for ID 96 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format G722 for ID 9 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:on - - - -... UNSUPPORTED. [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED. [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 9 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 18 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 97 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 98 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 106 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 110 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 111 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 112 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 113 on 0xb3380f78 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Incorporating payload 115 on 0xb3380f78 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8109f4c (ulaw|alaw|g726|slin|g729|speex|ilbc|g722|slin16|h263p|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:55:19] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd823478' [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.200:3000 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 9 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 18 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 97 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 98 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 106 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 110 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 111 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 112 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 113 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] rtp_engine.c: Copying payload 115 from 0xb3380f78 to 0xd823624 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:55:19] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd823478' [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Got a SIP re-invite for call 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: SIP/phone2-0000000d: This call is UP.... [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2fe60d4aa16c2b9d4.abada4d52b7f6367a;received=192.168.10.200 From: ;tag=3444016759 To: "Erika Musterfrau" ;tag=as73af062e Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 26430 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2fe60d4aa16c2b9d4.abada4d52b7f6367a;received=192.168.10.200 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 2 [ 67]: From: ;tag=3444016759 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 5 [ 18]: CSeq: 26430 INVITE [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Setting framing from config on incoming call [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Audio is at 5060 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2fe60d4aa16c2b9d4.abada4d52b7f6367a;received=192.168.10.200 From: ;tag=3444016759 To: "Erika Musterfrau" ;tag=as73af062e Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 26430 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 534818622 534818623 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 10632 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=recvonly <------------> [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2fe60d4aa16c2b9d4.abada4d52b7f6367a;received=192.168.10.200 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 2 [ 67]: From: ;tag=3444016759 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 5 [ 18]: CSeq: 26430 INVITE [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 11 [ 19]: Content-Length: 260 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 12 [ 0]: [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 1 [ 47]: o=root 534818622 534818623 IN IP4 192.168.10.70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 5 [ 29]: m=audio 10632 RTP/AVP 8 0 101 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=recvonly [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #345 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: INVITE [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '67985a870350a076' Method: ACK [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:55:19] VERBOSE[1244] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/phone1-0000000c [Nov 22 11:55:19] DEBUG[1244] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:55:19] DEBUG[1195] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/phone1-0000000c UniqueID: 1290423311.12 Class: default [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:19] DEBUG[1244] channel.c: Set channel SIP/phone1-0000000c to write format slin [Nov 22 11:55:19] DEBUG[1244] res_musiconhold.c: SIP/phone1-0000000c Opened file 0 '/var/lib/asterisk/moh/iptam/iptam_moh' [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> ACK sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2fb75a4274378079b.5c413a31aaa333dc8 Max-Forwards: 70 From: ;tag=3444016759 To: "Erika Musterfrau" ;tag=as73af062e Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 26430 ACK User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 0 [ 38]: ACK sip:100@192.168.10.70:5060 SIP/2.0 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK2fb75a4274378079b.5c413a31aaa333dc8 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 3 [ 67]: From: ;tag=3444016759 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 6 [ 15]: CSeq: 26430 ACK [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 7 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 22 11:55:19] VERBOSE[1175] chan_sip.c: --- (9 headers 0 lines) --- [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: = Looking for Call ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 (Checking From) --From tag 3444016759 --To-tag as73af062e [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #345 [Nov 22 11:55:19] DEBUG[1175] chan_sip.c: Stopping retransmission on '15c91da73e9014a058805f32700631ed@192.168.10.70' of Response 26430: Match Found [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:19] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:19] DEBUG[1244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Nov 22 11:55:20] DEBUG[1195] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 192.168.10.201:3001 OurSSRC: 690870840 SentNTP: 1290423320.0944365568 SentRTP: 7200 SentPackets: 45 SentOctets: 7200 ReportBlock: FractionLost: 207 CumulativeLoss: 95 IAJitter: 0.0014 TheirLastSR: 277130772 DLSR: 3.5150 (sec) [Nov 22 11:55:20] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> REGISTER sip:192.168.10.70 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK6952b6bbac96f6188.fd1b9e56fe353f6a4 Max-Forwards: 70 From: ;tag=f54c2b9e49 To: Call-ID: 2f7e3545567b8d26 CSeq: 19006 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erika Musterfrau" ;+sip.instance="" Supported: gruu, path User-Agent: Aastra 53i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 0 [ 34]: REGISTER sip:192.168.10.70 SIP/2.0 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK6952b6bbac96f6188.fd1b9e56fe353f6a4 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 3 [ 47]: From: ;tag=f54c2b9e49 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 4 [ 30]: To: [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 2f7e3545567b8d26 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 6 [ 20]: CSeq: 19006 REGISTER [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 9 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 53i/2.6.0.1008 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 22 11:55:20] VERBOSE[1175] chan_sip.c: --- (13 headers 0 lines) --- [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: = Looking for Call ID: 2f7e3545567b8d26 (Checking From) --From tag f54c2b9e49 --To-tag [Nov 22 11:55:20] DEBUG[1175] acl.c: For destination '192.168.10.201', our source address is '192.168.10.70'. [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Allocating new SIP dialog for 2f7e3545567b8d26 - REGISTER (No RTP) [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Initializing initreq for method REGISTER - callid 2f7e3545567b8d26 [Nov 22 11:55:20] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:55:20] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:55:20] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.201:5060 (no NAT) [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 22 11:55:20] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:55:20] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:55:20] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK6952b6bbac96f6188.fd1b9e56fe353f6a4;received=192.168.10.201 From: ;tag=f54c2b9e49 To: ;tag=as2bb9e692 Call-ID: 2f7e3545567b8d26 CSeq: 19006 REGISTER Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 22 Nov 2010 10:55:20 GMT Content-Length: 0 <------------> [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK6952b6bbac96f6188.fd1b9e56fe353f6a4;received=192.168.10.201 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 2 [ 47]: From: ;tag=f54c2b9e49 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 3 [ 45]: To: ;tag=as2bb9e692 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 2f7e3545567b8d26 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 5 [ 20]: CSeq: 19006 REGISTER [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 11 [ 35]: Date: Mon, 22 Nov 2010 10:55:20 GMT [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:55:20] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog '2f7e3545567b8d26' in 32000 ms (Method: REGISTER) [Nov 22 11:55:20] DEBUG[1195] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone1 PeerStatus: Registered Address: 192.168.10.201:5060 [Nov 22 11:55:20] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:55:20] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:55:20] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:55:20] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:55:20] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Auto destroying SIP dialog '8b6423da4532bf09' [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: Destroying SIP dialog 8b6423da4532bf09 [Nov 22 11:55:20] VERBOSE[1175] chan_sip.c: Really destroying SIP dialog '8b6423da4532bf09' Method: REGISTER [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: ---------- SIP HISTORY for '8b6423da4532bf09' [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: * SIP Call [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: 001. Rx REGISTER / 3825 REGISTER / sip:192.168.10.70 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: 002. TxResp SIP/2.0 / 3825 REGISTER - 200 OK [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: 005. AutoDestroy 8b6423da4532bf09 [Nov 22 11:55:20] DEBUG[1175] chan_sip.c: ---------- END SIP HISTORY for '8b6423da4532bf09' [Nov 22 11:55:21] DEBUG[1244] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:21] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.201:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 59866 SequenceNumberCycles: 0 IAJitter: 46 LastSR: 51864.1342177280 DLSR: 1.4700(sec) RTT: 3(sec) [Nov 22 11:55:21] DEBUG[1244] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 2 instead [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> INVITE sip:180@192.168.10.70 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb1844290b91e02e39.7677c6599da12b644 Max-Forwards: 70 From: "" ;tag=cc6da1e7fe To: "180" Call-ID: 72df352e45db5411 CSeq: 16159 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.6.0.66 Content-Type: application/sdp Content-Length: 620 v=0 o=MxSIP 0 0 IN IP4 192.168.10.200 s=SIP Call c=IN IP4 192.168.10.200 t=0 0 m=audio 3002 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:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 0 [ 36]: INVITE sip:180@192.168.10.70 SIP/2.0 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb1844290b91e02e39.7677c6599da12b644 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "" ;tag=cc6da1e7fe [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 4 [ 33]: To: "180" [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 72df352e45db5411 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 6 [ 18]: CSeq: 16159 INVITE [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 9 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 13 [ 19]: Content-Length: 620 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 14 [ 0]: [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.200 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 5 [ 70]: m=audio 3002 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 21 [ 24]: a=silenceSupp:on - - - - [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 22 [ 20]: a=fmtp:18 annexb=yes [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Body 25 [ 10]: a=sendrecv [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: --- (14 headers 26 lines) --- [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: = Looking for Call ID: 72df352e45db5411 (Checking From) --From tag cc6da1e7fe --To-tag [Nov 22 11:55:22] DEBUG[1175] acl.c: For destination '192.168.10.200', our source address is '192.168.10.70'. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Allocating new SIP dialog for 72df352e45db5411 - INVITE (No RTP) [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -gruu- [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: gruu [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -path- [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: path [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: timer [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 22 11:55:22] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 22 11:55:22] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:55:22] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.200:5060 (no NAT) [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Initializing initreq for method INVITE - callid 72df352e45db5411 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Using INVITE request as basis request - 72df352e45db5411 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found peer 'phone2' for 'phone2' from 192.168.10.200:5060 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd7d9e30' [Nov 22 11:55:22] DEBUG[1175] res_rtp_asterisk.c: Allocated port 13320 for RTP instance '0xd7d9e30' [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: RTP instance '0xd7d9e30' is setup and ready to go [Nov 22 11:55:22] DEBUG[1175] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd7d9e30' [Nov 22 11:55:22] VERBOSE[1175] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 22 11:55:22] VERBOSE[1175] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Setting NAT on RTP to Off [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.200... UNSUPPORTED. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:55:22] DEBUG[1175] netsock2.c: Splitting '192.168.10.200' gives... [Nov 22 11:55:22] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '(null)'. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.200... OK. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 18 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 18 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 106 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 106 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 107 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 113 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 110 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 110 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 111 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 111 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 112 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 112 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 98 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 98 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 97 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 97 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 115 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 115 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 96 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 9 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 9 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format G729 for ID 18 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format BV16 for ID 106 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format BV32 for ID 107 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 113 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 110 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 111 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 112 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format G726-16 for ID 98 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format G726-24 for ID 97 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format G726-32 for ID 115 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format G726-40 for ID 96 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format G722 for ID 9 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:on - - - -... UNSUPPORTED. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 9 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 18 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 97 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 98 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 106 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 110 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 111 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 112 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 113 on 0xb3380f78 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Incorporating payload 115 on 0xb3380f78 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8109f4c (ulaw|alaw|g726|slin|g729|speex|ilbc|g722|slin16|h263p|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:55:22] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7d9e30' [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.200:3002 [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 9 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 18 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 97 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 98 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 106 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 110 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 111 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 112 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 113 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] rtp_engine.c: Copying payload 115 from 0xb3380f78 to 0xd7d9fdc [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Checking SIP call limits for device phone2 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Call from peer 'phone2' is 2 out of 2147483647 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: Looking for 180 in Standard (domain 192.168.10.70) [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: This channel will not be able to handle video. [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: build_route: Contact hop: "" ;+sip.instance="" [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: list_route: hop: [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: SIP/phone2-0000000e: New call is still down.... Trying... [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb1844290b91e02e39.7677c6599da12b644;received=192.168.10.200 From: "" ;tag=cc6da1e7fe To: "180" Call-ID: 72df352e45db5411 CSeq: 16159 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb1844290b91e02e39.7677c6599da12b644;received=192.168.10.200 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "" ;tag=cc6da1e7fe [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 3 [ 33]: To: "180" [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 72df352e45db5411 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 5 [ 18]: CSeq: 16159 INVITE [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:22] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:22] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:22] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:55:22] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:55:22] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:22] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:22] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:55:22] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:55:22] DEBUG[1195] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-0000000e ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 150 CallerIDName: Hans Muster AccountCode: Exten: 180 Context: Standard Uniqueid: 1290423322.14 [Nov 22 11:55:22] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-0000000e ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 150 CallerIDName: Hans Muster Uniqueid: 1290423322.14 [Nov 22 11:55:22] DEBUG[1245] pbx.c: Launching 'Dial' [Nov 22 11:55:22] VERBOSE[1245] pbx.c: -- Executing [180@Standard:1] Dial("SIP/phone2-0000000e", "SIP/phone3") in new stack [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Allocating new SIP dialog for 4d8aa31f16c054cb08720b982c4f39ec@192.168.10.70 - INVITE (No RTP) [Nov 22 11:55:22] DEBUG[1245] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd7d5000' [Nov 22 11:55:22] DEBUG[1245] res_rtp_asterisk.c: Allocated port 12996 for RTP instance '0xd7d5000' [Nov 22 11:55:22] DEBUG[1245] rtp_engine.c: RTP instance '0xd7d5000' is setup and ready to go [Nov 22 11:55:22] DEBUG[1245] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd7d5000' [Nov 22 11:55:22] VERBOSE[1245] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 22 11:55:22] VERBOSE[1245] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Setting NAT on RTP to Off [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 22 11:55:22] DEBUG[1245] acl.c: For destination '192.168.10.202', our source address is '192.168.10.70'. [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: This channel will not be able to handle video. [Nov 22 11:55:22] DEBUG[1245] rtp_engine.c: Seeded SDP of 'SIP/phone3-0000000f' with that of 'SIP/phone2-0000000e' [Nov 22 11:55:22] DEBUG[1245] channel.c: Not copying variable DIALEDTIME. [Nov 22 11:55:22] DEBUG[1245] channel.c: Not copying variable ANSWEREDTIME. [Nov 22 11:55:22] DEBUG[1245] channel.c: Not copying variable DIALEDPEERNAME. [Nov 22 11:55:22] DEBUG[1245] channel.c: Not copying variable DIALEDPEERNUMBER. [Nov 22 11:55:22] DEBUG[1245] channel.c: Not copying variable DIALSTATUS. [Nov 22 11:55:22] DEBUG[1245] channel.c: Not copying variable SIPCALLID. [Nov 22 11:55:22] DEBUG[1245] channel.c: Not copying variable SIPDOMAIN. [Nov 22 11:55:22] DEBUG[1245] channel.c: Not copying variable SIPURI. [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Outgoing Call for phone3 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Call to peer 'phone3' is 1 out of 2147483647 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:55:22] VERBOSE[1245] chan_sip.c: Audio is at 5060 [Nov 22 11:55:22] VERBOSE[1245] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:55:22] VERBOSE[1245] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:55:22] VERBOSE[1245] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Initializing initreq for method INVITE - callid 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2eb664e3 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 4 [ 50]: To: [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 6 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 9 [ 35]: Date: Mon, 22 Nov 2010 10:55:22 GMT [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:55:22] VERBOSE[1245] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.202:5060: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2eb664e3 Max-Forwards: 70 From: "Hans Muster" ;tag=as4f69ff21 To: Contact: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0-1 Date: Mon, 22 Nov 2010 10:55:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Hans Muster" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1706139730 1706139730 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 12996 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 --- [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2eb664e3 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 4 [ 50]: To: [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 6 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 9 [ 35]: Date: Mon, 22 Nov 2010 10:55:22 GMT [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 14 [ 19]: Content-Length: 262 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 15 [ 0]: [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 1 [ 49]: o=root 1706139730 1706139730 IN IP4 192.168.10.70 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 5 [ 29]: m=audio 12996 RTP/AVP 8 0 101 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #349 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:55:22] VERBOSE[1245] app_dial.c: -- Called phone3 [Nov 22 11:55:22] DEBUG[1195] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone3-0000000f ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 180 CallerIDName: Max Mustermann AccountCode: Exten: Context: Standard Uniqueid: 1290423322.15 [Nov 22 11:55:22] DEBUG[1195] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone2-0000000e Destination: SIP/phone3-0000000f CallerIDNum: 150 CallerIDName: Hans Muster UniqueID: 1290423322.14 DestUniqueID: 1290423322.15 Dialstring: phone3 [Nov 22 11:55:22] DEBUG[1195] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone3-0000000f CallerIDNum: 180 CallerIDName: Uniqueid: 1290423322.15 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Nov 22 11:55:22] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:55:22] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:55:22] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 6 (Ringing) [Nov 22 11:55:22] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '6' [Nov 22 11:55:22] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:22] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:22] DEBUG[1168] app_queue.c: Extension '180@Standard' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:55:22] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:55:22] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 8 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2eb664e3 From: "Hans Muster" ;tag=as4f69ff21 To: ;tag=684195326 Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="" Server: Aastra 51i/2.6.0.1008 Supported: gruu, path Content-Length: 0 <-------------> [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2eb664e3 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: ;tag=684195326 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 22 11:55:22] VERBOSE[1175] chan_sip.c: --- (12 headers 0 lines) --- [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: = Looking for Call ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 (Checking To) --From tag as4f69ff21 --To-tag 684195326 [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: *** SIP TIMER: Cancelling retransmission #349 - INVITE (got response) [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Request 102: Found [Nov 22 11:55:22] DEBUG[1175] chan_sip.c: SIP response 180 to standard invite [Nov 22 11:55:22] VERBOSE[1245] app_dial.c: -- SIP/phone3-0000000f is ringing [Nov 22 11:55:22] DEBUG[1245] rtp_engine.c: Setting early bridge SDP of 'SIP/phone2-0000000e' with that of 'SIP/phone3-0000000f' [Nov 22 11:55:22] VERBOSE[1245] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb1844290b91e02e39.7677c6599da12b644;received=192.168.10.200 From: "" ;tag=cc6da1e7fe To: "180" ;tag=as37e0aab8 Call-ID: 72df352e45db5411 CSeq: 16159 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb1844290b91e02e39.7677c6599da12b644;received=192.168.10.200 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 2 [ 50]: From: "" ;tag=cc6da1e7fe [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 3 [ 48]: To: "180" ;tag=as37e0aab8 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 4 [ 25]: Call-ID: 72df352e45db5411 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 5 [ 18]: CSeq: 16159 INVITE [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Header 11 [ 0]: [Nov 22 11:55:22] DEBUG[1245] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:22] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:55:22] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:55:22] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 6 (Ringing) [Nov 22 11:55:22] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '6' [Nov 22 11:55:22] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-0000000f ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 180 CallerIDName: Uniqueid: 1290423322.15 [Nov 22 11:55:22] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:55:22] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:22] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:23] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2eb664e3 From: "Hans Muster" ;tag=as4f69ff21 To: ;tag=684195326 Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="" Server: Aastra 51i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 0 IN IP4 192.168.10.202 s=SIP Call c=IN IP4 192.168.10.202 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 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2eb664e3 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: ;tag=684195326 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.202 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.202 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: = Looking for Call ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 (Checking To) --From tag as4f69ff21 --To-tag 684195326 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Acked pending invite 102 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Stopping retransmission on '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' of Request 102: Match Found [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: SIP response 200 to standard invite [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.202... UNSUPPORTED. [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:55:24] DEBUG[1175] netsock2.c: Splitting '192.168.10.202' gives... [Nov 22 11:55:24] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '(null)'. [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.202... OK. [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:55:24] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:55:24] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:55:24] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:55:24] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:55:24] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:55:24] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:55:24] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7d5000' [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.202:3000 [Nov 22 11:55:24] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd7d51ac [Nov 22 11:55:24] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd7d51ac [Nov 22 11:55:24] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd7d51ac [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: build_route: Contact hop: "Max Mustermann" ;+sip.instance="" [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: list_route: hop: [Nov 22 11:55:24] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:55:24] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Strict routing enforced for session 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:55:24] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:55:24] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.202:5060: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4684a33d Max-Forwards: 70 From: "Hans Muster" ;tag=as4f69ff21 To: ;tag=684195326 Contact: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4684a33d [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: ;tag=684195326 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:55:24] VERBOSE[1245] app_dial.c: -- SIP/phone3-0000000f answered SIP/phone2-0000000e [Nov 22 11:55:24] DEBUG[1245] rtp_engine.c: Setting early bridge SDP of 'SIP/phone2-0000000e' with that of 'SIP/phone3-0000000f' [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: SIP answering channel: SIP/phone2-0000000e [Nov 22 11:55:24] DEBUG[1245] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Setting framing from config on incoming call [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:55:24] VERBOSE[1245] chan_sip.c: Audio is at 5060 [Nov 22 11:55:24] VERBOSE[1245] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:55:24] VERBOSE[1245] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:55:24] VERBOSE[1245] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:55:24] VERBOSE[1245] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb1844290b91e02e39.7677c6599da12b644;received=192.168.10.200 From: "" ;tag=cc6da1e7fe To: "180" ;tag=as37e0aab8 Call-ID: 72df352e45db5411 CSeq: 16159 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "Max Mustermann" Content-Type: application/sdp Content-Length: 260 v=0 o=root 269116321 269116321 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 13320 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 <------------> [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb1844290b91e02e39.7677c6599da12b644;received=192.168.10.200 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 2 [ 50]: From: "" ;tag=cc6da1e7fe [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 3 [ 48]: To: "180" ;tag=as37e0aab8 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 4 [ 25]: Call-ID: 72df352e45db5411 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 5 [ 18]: CSeq: 16159 INVITE [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 10 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 12 [ 19]: Content-Length: 260 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Header 13 [ 0]: [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 1 [ 47]: o=root 269116321 269116321 IN IP4 192.168.10.70 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 5 [ 29]: m=audio 13320 RTP/AVP 8 0 101 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #352 [Nov 22 11:55:24] DEBUG[1245] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:24] DEBUG[1245] features.c: bridge answer set, chan answer set [Nov 22 11:55:24] DEBUG[1245] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:55:24] DEBUG[1245] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:55:24] VERBOSE[1245] rtp_engine.c: -- Locally bridging SIP/phone2-0000000e and SIP/phone3-0000000f [Nov 22 11:55:24] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:55:24] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:55:24] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 22 11:55:24] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '2' [Nov 22 11:55:24] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:55:24] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:55:24] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 22 11:55:24] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '2' [Nov 22 11:55:24] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:24] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:24] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:55:24] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:55:24] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-0000000f ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 180 CallerIDName: Uniqueid: 1290423322.15 [Nov 22 11:55:24] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-0000000e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 150 CallerIDName: Hans Muster Uniqueid: 1290423322.14 [Nov 22 11:55:24] DEBUG[1195] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone3-0000000f Uniqueid: 1290423322.15 AccountCode: OldAccountCode: [Nov 22 11:55:24] DEBUG[1195] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone2-0000000e Channel2: SIP/phone3-0000000f Uniqueid1: 1290423322.14 Uniqueid2: 1290423322.15 CallerID1: 150 CallerID2: 180 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '72df352e45db5411' Method: INVITE [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:24] DEBUG[1168] app_queue.c: Extension '180@Standard' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:24] DEBUG[1194] 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. [Nov 22 11:55:24] DEBUG[1194] 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. [Nov 22 11:55:24] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:24] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 1 [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> ACK sip:180@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK1d79d27f8e0e36377.2e0a4b98a8cc83691 Max-Forwards: 70 From: "" ;tag=cc6da1e7fe To: "180" ;tag=as37e0aab8 Call-ID: 72df352e45db5411 CSeq: 16159 ACK User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 0 [ 38]: ACK sip:180@192.168.10.70:5060 SIP/2.0 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK1d79d27f8e0e36377.2e0a4b98a8cc83691 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "" ;tag=cc6da1e7fe [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "180" ;tag=as37e0aab8 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 72df352e45db5411 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 6 [ 15]: CSeq: 16159 ACK [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 7 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 22 11:55:24] VERBOSE[1175] chan_sip.c: --- (9 headers 0 lines) --- [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: = Looking for Call ID: 72df352e45db5411 (Checking From) --From tag cc6da1e7fe --To-tag as37e0aab8 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #352 [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Stopping retransmission on '72df352e45db5411' of Response 16159: Match Found [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '72df352e45db5411' Method: ACK [Nov 22 11:55:24] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1244] res_rtp_asterisk.c: No remote address on RTP instance '0xd823478' so dropping frame [Nov 22 11:55:24] DEBUG[1245] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:24] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.202:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 16554 SequenceNumberCycles: 0 IAJitter: 1 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Nov 22 11:55:25] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '72df352e45db5411' Method: ACK [Nov 22 11:55:25] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:25] DEBUG[1195] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 192.168.10.201:3001 OurSSRC: 690870840 SentNTP: 1290423325.0947015680 SentRTP: 47040 SentPackets: 294 SentOctets: 47040 ReportBlock: FractionLost: 0 CumulativeLoss: 95 IAJitter: 0.0029 TheirLastSR: 277457797 DLSR: 3.5270 (sec) [Nov 22 11:55:26] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '72df352e45db5411' Method: ACK [Nov 22 11:55:26] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:26] DEBUG[1244] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:26] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.201:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 60114 SequenceNumberCycles: 0 IAJitter: 39 LastSR: 51869.0000000000 DLSR: 1.4600(sec) RTT: 2(sec) [Nov 22 11:55:27] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '72df352e45db5411' Method: ACK [Nov 22 11:55:27] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:28] DEBUG[1245] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:28] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.200:3003 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 67108871 FractionLost: 7 PacketsLost: 4 HighestSequence: 20227 SequenceNumberCycles: 0 IAJitter: 15 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Nov 22 11:55:28] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '72df352e45db5411' Method: ACK [Nov 22 11:55:28] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '72df352e45db5411' Method: ACK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> REFER sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4f7329b730f2975f2.ae720a53a1a847817 Max-Forwards: 70 From: ;tag=3444016759 To: "Erika Musterfrau" ;tag=as73af062e Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 26431 REFER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Refer-To: "180" Referred-By: Supported: gruu, path, timer User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 40]: REFER sip:100@192.168.10.70:5060 SIP/2.0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4f7329b730f2975f2.ae720a53a1a847817 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 67]: From: ;tag=3444016759 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 17]: CSeq: 26431 REFER [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [111]: Refer-To: "180" [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 11 [ 39]: Referred-By: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 12 [ 28]: Supported: gruu, path, timer [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 13 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: --- (15 headers 0 lines) --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: = Looking for Call ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 (Checking From) --From tag 3444016759 --To-tag as73af062e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Call 15c91da73e9014a058805f32700631ed@192.168.10.70 got a SIP call transfer from caller: (REFER)! [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 72df352e45db5411 F-tag: cc6da1e7fe T-tag: as37e0aab8 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: SIP transfer to extension 180@Standard by phone2@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: SIP attended transfer: Transferer channel SIP/phone2-0000000d, transferee channel SIP/phone1-0000000c [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/phone1-0000000c' [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Looking for callid 72df352e45db5411 (fromtag cc6da1e7fe totag as37e0aab8) [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Matched INCOMING call - their tag is cc6da1e7fe Our tag is as37e0aab8 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4f7329b730f2975f2.ae720a53a1a847817;received=192.168.10.200 From: ;tag=3444016759 To: "Erika Musterfrau" ;tag=as73af062e Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 26431 REFER Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 20]: SIP/2.0 202 Accepted [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK4f7329b730f2975f2.ae720a53a1a847817;received=192.168.10.200 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 67]: From: ;tag=3444016759 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 17]: CSeq: 26431 REFER [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: SIP attended transfer: trying to bridge SIP/phone2-0000000e and SIP/phone1-0000000c [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Sip transfer:-------------------- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: -- Transferer to PBX channel: SIP/phone2-0000000d State Up [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: -- Transferer to PBX second channel (target): SIP/phone2-0000000e State Up [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: -- Bridged call to transferee: SIP/phone1-0000000c State Up [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: -- Bridged call to transfer target: SIP/phone3-0000000f State Up [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: -- END Sip transfer:-------------------- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: SIP transfer: Four channels to handle [Nov 22 11:55:29] VERBOSE[1175] res_musiconhold.c: -- Stopped music on hold on SIP/phone1-0000000c [Nov 22 11:55:29] DEBUG[1175] channel.c: Set channel SIP/phone1-0000000c to write format alaw [Nov 22 11:55:29] DEBUG[1175] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: SIP transfer: trying to masquerade SIP/phone1-0000000c into SIP/phone2-0000000e [Nov 22 11:55:29] DEBUG[1175] channel.c: Planning to masquerade channel SIP/phone1-0000000c into the structure of SIP/phone2-0000000e [Nov 22 11:55:29] DEBUG[1175] channel.c: Done planning to masquerade channel SIP/phone1-0000000c into the structure of SIP/phone2-0000000e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Strict routing enforced for session 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:55:29] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:55:29] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.200:5060: NOTIFY sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK56104e6e Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as73af062e To: ;tag=3444016759 Contact: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 103 NOTIFY User-Agent: Asterisk PBX 1.8.0-1 Event: refer;id=26431 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 16 SIP/2.0 200 OK --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 59]: NOTIFY sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK56104e6e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=3444016759 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 103 NOTIFY [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [ 21]: Event: refer;id=26431 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [ 48]: Subscription-state: terminated;reason=noresource [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 11 [ 41]: Content-Type: message/sipfrag;version=2.0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 12 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 14 [ 18]: Content-Length: 16 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 15 [ 0]: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #355 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:29] DEBUG[1175] channel.c: Actually Masquerading SIP/phone1-0000000c(6) into the structure of SIP/phone2-0000000e(6) [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: SIP Fixup: New owner for dialogue 72df352e45db5411: SIP/phone1-0000000c (Old parent: SIP/phone1-0000000c) [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Hangup call SIP/phone1-0000000c, SIP callid 72df352e45db5411 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: update_call_counter(phone2) - decrement call limit counter on hangup [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Call from peer 'phone2' removed from call limit 2147483647 [Nov 22 11:55:29] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7d9e30' [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog '72df352e45db5411' in 32000 ms (Method: ACK) [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Strict routing enforced for session 72df352e45db5411 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:55:29] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:55:29] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.200:5060: BYE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1fbf6c51 Max-Forwards: 70 From: "180" ;tag=as37e0aab8 To: "" ;tag=cc6da1e7fe Call-ID: 72df352e45db5411 CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.0-1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 56]: BYE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1fbf6c51 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "180" ;tag=as37e0aab8 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "" ;tag=cc6da1e7fe [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 72df352e45db5411 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #357 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:29] DEBUG[1175] channel.c: Putting channel SIP/phone1-0000000c in alaw/alaw formats [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: SIP Fixup: New owner for dialogue 67985a870350a076: SIP/phone1-0000000c (Old parent: SIP/phone2-0000000e) [Nov 22 11:55:29] DEBUG[1175] channel.c: Released clone lock on 'SIP/phone2-0000000e' [Nov 22 11:55:29] DEBUG[1175] channel.c: Done Masquerading SIP/phone1-0000000c (6) [Nov 22 11:55:29] DEBUG[1175] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:55:29] DEBUG[1175] res_rtp_asterisk.c: Changing ssrc from 690870840 to 986557840 due to a source change [Nov 22 11:55:29] DEBUG[1175] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '72df352e45db5411' Method: ACK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:29] DEBUG[1245] rtp_engine.c: rtp-engine-local-bridge: Oooh, something is weird, backing out [Nov 22 11:55:29] VERBOSE[1245] rtp_engine.c: -- Locally bridging SIP/phone1-0000000c and SIP/phone3-0000000f [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Strict routing enforced for session 67985a870350a076 [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:55:29] DEBUG[1245] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:55:29] DEBUG[1245] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Audio is at 5060 [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Initializing already initialized SIP dialog 67985a870350a076 (presumably reinvite) [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK466684da [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as2d91b5e0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 6 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 11 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK466684da Max-Forwards: 70 From: "150" ;tag=as2d91b5e0 To: "Erika Musterfrau" ;tag=673d7b2f8c Contact: Call-ID: 67985a870350a076 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Max Mustermann" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1729646775 1729646776 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 14070 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 --- [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK466684da [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as2d91b5e0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 6 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 11 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 14 [ 0]: [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 1 [ 49]: o=root 1729646775 1729646776 IN IP4 192.168.10.70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 5 [ 29]: m=audio 14070 RTP/AVP 8 0 101 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #358 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Strict routing enforced for session 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:55:29] DEBUG[1245] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:55:29] DEBUG[1245] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Audio is at 5060 [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Initializing already initialized SIP dialog 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 (presumably reinvite) [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4c9e2823 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 4 [ 64]: To: ;tag=684195326 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 6 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 11 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:55:29] VERBOSE[1245] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.202:5060: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4c9e2823 Max-Forwards: 70 From: "Hans Muster" ;tag=as4f69ff21 To: ;tag=684195326 Contact: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Erika Musterfrau" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1706139730 1706139731 IN IP4 192.168.10.70 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.70 t=0 0 m=audio 12996 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 --- [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4c9e2823 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 4 [ 64]: To: ;tag=684195326 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 6 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 11 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Header 14 [ 0]: [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 1 [ 49]: o=root 1706139730 1706139731 IN IP4 192.168.10.70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 5 [ 29]: m=audio 12996 RTP/AVP 8 0 101 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #359 [Nov 22 11:55:29] DEBUG[1245] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Attended Channel: SIP/phone2-0000000d Uniqueid: 1290423311.13 SIP-Callid: 15c91da73e9014a058805f32700631ed@192.168.10.70 TargetChannel: SIP/phone2-0000000e TargetUniqueid: 1290423322.14 [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/phone1-0000000c UniqueID: 1290423311.12 [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/phone1-0000000c CloneState: Up Original: SIP/phone2-0000000e OriginalState: Up [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone1-0000000c Newname: SIP/phone1-0000000c Uniqueid: 1290423311.12 [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone2-0000000e Newname: SIP/phone1-0000000c Uniqueid: 1290423322.14 [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone1-0000000c Newname: SIP/phone2-0000000e Uniqueid: 1290423311.12 [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone1-0000000c CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1290423322.14 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/phone1-0000000c UniqueID: 1290423322.14 [Nov 22 11:55:29] DEBUG[1244] rtp_engine.c: rtp-engine-local-bridge: Oooh, something is weird, backing out [Nov 22 11:55:29] DEBUG[1244] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/phone2-0000000e, c1=SIP/phone2-0000000d, flags: Yes,Yes,No,No [Nov 22 11:55:29] DEBUG[1244] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:55:29] DEBUG[1244] channel.c: Bridge stops bridging channels SIP/phone2-0000000e and SIP/phone2-0000000d [Nov 22 11:55:29] DEBUG[1244] channel.c: Hanging up channel 'SIP/phone2-0000000d' [Nov 22 11:55:29] DEBUG[1244] chan_sip.c: update_call_counter(phone2) - decrement call limit counter on hangup [Nov 22 11:55:29] DEBUG[1244] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:55:29] DEBUG[1244] chan_sip.c: Call to peer 'phone2' removed from call limit 2147483647 [Nov 22 11:55:29] DEBUG[1244] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 15c91da73e9014a058805f32700631ed@192.168.10.70. [Nov 22 11:55:29] VERBOSE[1244] chan_sip.c: Scheduling destruction of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' in 32000 ms (Method: REFER) [Nov 22 11:55:29] DEBUG[1244] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 22 11:55:29] DEBUG[1244] pbx.c: Spawn extension (Standard,150,1) exited non-zero on 'SIP/phone2-0000000e' [Nov 22 11:55:29] VERBOSE[1244] pbx.c: == Spawn extension (Standard, 150, 1) exited non-zero on 'SIP/phone2-0000000e' [Nov 22 11:55:29] DEBUG[1244] channel.c: Soft-Hanging up channel 'SIP/phone2-0000000e' [Nov 22 11:55:29] DEBUG[1244] channel.c: Hanging up zombie 'SIP/phone2-0000000e' [Nov 22 11:55:29] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:29] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:29] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:55:29] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:55:29] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:29] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:29] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:55:29] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:55:29] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:29] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:29] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:55:29] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:55:29] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:55:29] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:55:29] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:55:29] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone2-0000000e Channel2: SIP/phone2-0000000d Uniqueid1: 1290423311.12 Uniqueid2: 1290423311.13 CallerID1: 150 CallerID2: 150 [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 100 Destination: 150 DestinationContext: Standard CallerID: "Erika Musterfrau" <100> Channel: SIP/phone1-0000000c DestinationChannel: SIP/phone2-0000000d LastApplication: Dial LastData: SIP/phone2 StartTime: 2010-11-22 11:55:11 AnswerTime: 2010-11-22 11:55:12 EndTime: 2010-11-22 11:55:29 Duration: 18 BillableSeconds: 17 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1290423311.12 UserField: [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-0000000d Uniqueid: 1290423311.13 CallerIDNum: 150 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone2-0000000e UniqueID: 1290423311.12 DialStatus: ANSWER [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-0000000e Uniqueid: 1290423311.12 CallerIDNum: 150 CallerIDName: Hans Muster Cause: 16 Cause-txt: Normal Clearing [Nov 22 11:55:29] DEBUG[1168] app_queue.c: Extension '150@Standard' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:55:29] DEBUG[1194] 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. [Nov 22 11:55:29] DEBUG[1194] 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. [Nov 22 11:55:29] DEBUG[1194] 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. [Nov 22 11:55:29] DEBUG[1194] 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. [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 0 [Nov 22 11:55:29] DEBUG[1245] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Nov 22 11:55:29] DEBUG[1245] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK466684da From: "150" ;tag=as2d91b5e0 To: "Erika Musterfrau" ;tag=673d7b2f8c Call-ID: 67985a870350a076 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erika Musterfrau" ;+sip.instance="" Server: Aastra 53i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 1 IN IP4 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK466684da [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "150" ;tag=as2d91b5e0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.201 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: = Looking for Call ID: 67985a870350a076 (Checking To) --From tag as2d91b5e0 --To-tag 673d7b2f8c [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Acked pending invite 102 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #358 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Stopping retransmission on '67985a870350a076' of Request 102: Match Found [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 67985a870350a076 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.201... UNSUPPORTED. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:55:29] DEBUG[1175] netsock2.c: Splitting '192.168.10.201' gives... [Nov 22 11:55:29] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '(null)'. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:55:29] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd800760' [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd80090c [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd80090c [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd80090c [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:55:29] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:55:29] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Strict routing enforced for session 67985a870350a076 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:55:29] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:55:29] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.201:5060: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK77747430 Max-Forwards: 70 From: "150" ;tag=as2d91b5e0 To: "Erika Musterfrau" ;tag=673d7b2f8c Contact: Call-ID: 67985a870350a076 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK77747430 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as2d91b5e0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK56104e6e From: "Erika Musterfrau" ;tag=as73af062e To: ;tag=3444016759 Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 103 NOTIFY Server: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK56104e6e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3444016759 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 103 NOTIFY [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: --- (8 headers 0 lines) --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: = Looking for Call ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 (Checking To) --From tag as73af062e --To-tag 3444016759 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #355 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Stopping retransmission on '15c91da73e9014a058805f32700631ed@192.168.10.70' of Request 103: Match Found [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:29] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:55:29] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:55:29] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:55:29] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> BYE sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb010d85539eb4e93f.c70a451a3ac5929dc Max-Forwards: 70 From: ;tag=3444016759 To: "Erika Musterfrau" ;tag=as73af062e Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 26432 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Supported: gruu, path, timer User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 38]: BYE sip:100@192.168.10.70:5060 SIP/2.0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb010d85539eb4e93f.c70a451a3ac5929dc [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 67]: From: ;tag=3444016759 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 15]: CSeq: 26432 BYE [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [ 28]: Supported: gruu, path, timer [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: --- (12 headers 0 lines) --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: = Looking for Call ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 (Checking From) --From tag 3444016759 --To-tag as73af062e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Initializing initreq for method BYE - callid 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:55:29] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.200:5060 (no NAT) [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Setting SIP_ALREADYGONE on dialog 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd823478' [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' in 32000 ms (Method: BYE) [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Received bye, no owner, selfdestruct soon. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb010d85539eb4e93f.c70a451a3ac5929dc;received=192.168.10.200 From: ;tag=3444016759 To: "Erika Musterfrau" ;tag=as73af062e Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 CSeq: 26432 BYE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb010d85539eb4e93f.c70a451a3ac5929dc;received=192.168.10.200 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 67]: From: ;tag=3444016759 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as73af062e [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 15]: CSeq: 26432 BYE [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:29] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4c9e2823 From: "Hans Muster" ;tag=as4f69ff21 To: ;tag=684195326 Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="" Server: Aastra 51i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 1 IN IP4 192.168.10.202 s=SIP Call c=IN IP4 192.168.10.202 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 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4c9e2823 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: ;tag=684195326 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.202 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.202 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: = Looking for Call ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 (Checking To) --From tag as4f69ff21 --To-tag 684195326 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Acked pending invite 103 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #359 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Stopping retransmission on '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' of Request 103: Match Found [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.202... UNSUPPORTED. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:55:29] DEBUG[1175] netsock2.c: Splitting '192.168.10.202' gives... [Nov 22 11:55:29] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '(null)'. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.202... OK. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:55:29] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7d5000' [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.202:3000 [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd7d51ac [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd7d51ac [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd7d51ac [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:55:29] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:55:29] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Strict routing enforced for session 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:55:29] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:55:29] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.202:5060: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5c812845 Max-Forwards: 70 From: "Hans Muster" ;tag=as4f69ff21 To: ;tag=684195326 Contact: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5c812845 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: ;tag=684195326 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:29] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:55:29] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:55:29] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 22 11:55:29] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '2' [Nov 22 11:55:29] DEBUG[1194] 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. [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1fbf6c51 From: "180" ;tag=as37e0aab8 To: "" ;tag=cc6da1e7fe Call-ID: 72df352e45db5411 CSeq: 102 BYE Server: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1fbf6c51 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "180" ;tag=as37e0aab8 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 3 [ 48]: To: "" ;tag=cc6da1e7fe [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 72df352e45db5411 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 6 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: --- (8 headers 0 lines) --- [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: = Looking for Call ID: 72df352e45db5411 (Checking To) --From tag as37e0aab8 --To-tag cc6da1e7fe [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #357 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Stopping retransmission on '72df352e45db5411' of Request 102: Match Found [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Destroying SIP dialog 72df352e45db5411 [Nov 22 11:55:29] VERBOSE[1175] chan_sip.c: Really destroying SIP dialog '72df352e45db5411' Method: ACK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: ---------- SIP HISTORY for '72df352e45db5411' [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: * SIP Call [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 001. Rx INVITE / 16159 INVITE / sip:180@192.168.10.70 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 002. NewChan Channel SIP/phone2-0000000e - from 72df352e45db5411 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 003. TxResp SIP/2.0 / 16159 INVITE - 100 Trying [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 004. TxResp SIP/2.0 / 16159 INVITE - 180 Ringing [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 005. ConnectedLine Called party is now Max Mustermann <180> [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 006. TxRespRel SIP/2.0 / 16159 INVITE - 200 OK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 007. Rx ACK / 16159 ACK / sip:180@192.168.10.70:5060 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 008. Masq Old channel: SIP/phone1-0000000c [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 009. Masq (cont) ...new owner: SIP/phone1-0000000c [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 010. Hangup Cause Normal Clearing [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 011. SchedDestroy 32000 ms [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 012. RTCPaudio Quality:ssrc=989253218;themssrc=1232357725;lp=0;rxjitter=0.0000 [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 013. TxReqRel BYE / 102 BYE - BYE [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 014. Rx SIP/2.0 / 102 BYE / 200 OK [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: 015. NeedDestroy Setting needdestroy because transaction completed [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: ---------- END SIP HISTORY for '72df352e45db5411' [Nov 22 11:55:29] DEBUG[1175] rtp_engine.c: Destroyed RTP instance '0xd7d9e30' [Nov 22 11:55:29] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:29] DEBUG[1245] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:29] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.202:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 50299 SequenceNumberCycles: 0 IAJitter: 9 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Nov 22 11:55:30] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:30] DEBUG[1195] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 192.168.10.201:3001 OurSSRC: 986557840 SentNTP: 1290423330.0920506368 SentRTP: 79840 SentPackets: 499 SentOctets: 79840 ReportBlock: FractionLost: 0 CumulativeLoss: 95 IAJitter: 0.0020 TheirLastSR: 277784822 DLSR: 3.5300 (sec) [Nov 22 11:55:31] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:31] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:31] DEBUG[1195] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 192.168.10.202:3001 OurSSRC: 1531786074 SentNTP: 1290423331.2305187840 SentRTP: 1178772984 SentPackets: 75 SentOctets: 12000 ReportBlock: FractionLost: 228 CumulativeLoss: 92 IAJitter: 0.0000 TheirLastSR: 276628111 DLSR: 1.8750 (sec) [Nov 22 11:55:31] DEBUG[1245] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:31] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.201:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 67108881 FractionLost: 17 PacketsLost: 4 HighestSequence: 20321 SequenceNumberCycles: 0 IAJitter: 13 LastSR: 51874.2147483648 DLSR: 1.4600(sec) RTT: 7(sec) [Nov 22 11:55:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:33] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:34] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:34] DEBUG[1245] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:55:34] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.202:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 50475 SequenceNumberCycles: 0 IAJitter: 4 LastSR: 51875.0805306368 DLSR: 3.1000(sec) RTT: 10(sec) [Nov 22 11:55:35] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: INVITE [Nov 22 11:55:36] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> BYE sip:150@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK5de637b042b6b3979.4aa5af548512d8d2a Max-Forwards: 70 From: ;tag=684195326 To: "Hans Muster" ;tag=as4f69ff21 Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 CSeq: 10025 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Supported: gruu, path, timer User-Agent: Aastra 51i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 0 [ 38]: BYE sip:150@192.168.10.70:5060 SIP/2.0 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK5de637b042b6b3979.4aa5af548512d8d2a [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 3 [ 66]: From: ;tag=684195326 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 4 [ 56]: To: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 6 [ 15]: CSeq: 10025 BYE [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 9 [ 28]: Supported: gruu, path, timer [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 10 [ 33]: User-Agent: Aastra 51i/2.6.0.1008 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 22 11:55:36] VERBOSE[1175] chan_sip.c: --- (12 headers 0 lines) --- [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: = Looking for Call ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 (Checking From) --From tag 684195326 --To-tag as4f69ff21 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Initializing initreq for method BYE - callid 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:36] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:55:36] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:55:36] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.202:5060 (no NAT) [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Setting SIP_ALREADYGONE on dialog 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:36] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7d5000' [Nov 22 11:55:36] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' in 32000 ms (Method: BYE) [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Received bye, issuing owner hangup [Nov 22 11:55:36] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK5de637b042b6b3979.4aa5af548512d8d2a;received=192.168.10.202 From: ;tag=684195326 To: "Hans Muster" ;tag=as4f69ff21 Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 CSeq: 10025 BYE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK5de637b042b6b3979.4aa5af548512d8d2a;received=192.168.10.202 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 2 [ 66]: From: ;tag=684195326 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 3 [ 56]: To: "Hans Muster" ;tag=as4f69ff21 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 5 [ 15]: CSeq: 10025 BYE [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: BYE [Nov 22 11:55:36] DEBUG[1245] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a hangup [Nov 22 11:55:36] DEBUG[1245] channel.c: Returning from native bridge, channels: SIP/phone1-0000000c, SIP/phone3-0000000f [Nov 22 11:55:36] DEBUG[1245] channel.c: Hanging up channel 'SIP/phone3-0000000f' [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Hangup call SIP/phone3-0000000f, SIP callid 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: update_call_counter(phone3) - decrement call limit counter on hangup [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Call to peer 'phone3' removed from call limit 2147483647 [Nov 22 11:55:36] DEBUG[1245] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7d5000' [Nov 22 11:55:36] DEBUG[1245] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 22 11:55:36] DEBUG[1245] pbx.c: Spawn extension (Standard,180,1) exited non-zero on 'SIP/phone1-0000000c' [Nov 22 11:55:36] VERBOSE[1245] pbx.c: == Spawn extension (Standard, 180, 1) exited non-zero on 'SIP/phone1-0000000c' [Nov 22 11:55:36] DEBUG[1245] channel.c: Soft-Hanging up channel 'SIP/phone1-0000000c' [Nov 22 11:55:36] DEBUG[1245] channel.c: Hanging up channel 'SIP/phone1-0000000c' [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Hangup call SIP/phone1-0000000c, SIP callid 67985a870350a076 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: update_call_counter(phone1) - decrement call limit counter on hangup [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Updating call counter for incoming call [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Call from peer 'phone1' removed from call limit 2147483647 [Nov 22 11:55:36] DEBUG[1245] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd800760' [Nov 22 11:55:36] VERBOSE[1245] chan_sip.c: Scheduling destruction of SIP dialog '67985a870350a076' in 32000 ms (Method: ACK) [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Strict routing enforced for session 67985a870350a076 [Nov 22 11:55:36] VERBOSE[1245] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:55:36] DEBUG[1245] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:55:36] DEBUG[1245] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:55:36] VERBOSE[1245] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:55:36] VERBOSE[1245] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: BYE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4fa7d8b0 Max-Forwards: 70 From: "150" ;tag=as2d91b5e0 To: "Erika Musterfrau" ;tag=673d7b2f8c Call-ID: 67985a870350a076 CSeq: 103 BYE User-Agent: Asterisk PBX 1.8.0-1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 0 [ 56]: BYE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4fa7d8b0 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as2d91b5e0 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 5 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 6 [ 13]: CSeq: 103 BYE [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Header 11 [ 0]: [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #364 [Nov 22 11:55:36] DEBUG[1245] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:55:36] DEBUG[1195] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone1-0000000c Channel2: SIP/phone3-0000000f Uniqueid1: 1290423322.14 Uniqueid2: 1290423322.15 CallerID1: 100 CallerID2: 180 [Nov 22 11:55:36] DEBUG[1195] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 150 Destination: 180 DestinationContext: Standard CallerID: "Hans Muster" <150> Channel: SIP/phone2-0000000e DestinationChannel: SIP/phone3-0000000f LastApplication: Dial LastData: SIP/phone3 StartTime: 2010-11-22 11:55:22 AnswerTime: 2010-11-22 11:55:24 EndTime: 2010-11-22 11:55:36 Duration: 14 BillableSeconds: 12 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1290423322.14 UserField: [Nov 22 11:55:36] DEBUG[1195] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone3-0000000f Uniqueid: 1290423322.15 CallerIDNum: 180 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Nov 22 11:55:36] DEBUG[1195] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone1-0000000c UniqueID: 1290423322.14 DialStatus: ANSWER [Nov 22 11:55:36] DEBUG[1195] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone1-0000000c Uniqueid: 1290423322.14 CallerIDNum: 100 CallerIDName: Erika Musterfrau Cause: 16 Cause-txt: Normal Clearing [Nov 22 11:55:36] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:55:36] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:55:36] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 22 11:55:36] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '1' [Nov 22 11:55:36] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:55:36] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:55:36] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 22 11:55:36] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '1' [Nov 22 11:55:36] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:55:36] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:55:36] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 22 11:55:36] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '1' [Nov 22 11:55:36] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:55:36] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:55:36] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 22 11:55:36] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '1' [Nov 22 11:55:36] DEBUG[1168] app_queue.c: Extension '180@Standard' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:55:36] DEBUG[1168] app_queue.c: Extension '100@Standard' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:55:36] DEBUG[1194] 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. [Nov 22 11:55:36] DEBUG[1194] 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. [Nov 22 11:55:36] DEBUG[1194] 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. [Nov 22 11:55:36] DEBUG[1194] 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. [Nov 22 11:55:36] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 0 [Nov 22 11:55:36] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 0 [Nov 22 11:55:36] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4fa7d8b0 From: "150" ;tag=as2d91b5e0 To: "Erika Musterfrau" ;tag=673d7b2f8c Call-ID: 67985a870350a076 CSeq: 103 BYE Server: Aastra 53i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4fa7d8b0 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "150" ;tag=as2d91b5e0 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: "Erika Musterfrau" ;tag=673d7b2f8c [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 67985a870350a076 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 6 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 22 11:55:36] VERBOSE[1175] chan_sip.c: --- (8 headers 0 lines) --- [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: = Looking for Call ID: 67985a870350a076 (Checking To) --From tag as2d91b5e0 --To-tag 673d7b2f8c [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #364 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Stopping retransmission on '67985a870350a076' of Request 103: Match Found [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: Destroying SIP dialog 67985a870350a076 [Nov 22 11:55:36] VERBOSE[1175] chan_sip.c: Really destroying SIP dialog '67985a870350a076' Method: ACK [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: ---------- SIP HISTORY for '67985a870350a076' [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: * SIP Call [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 001. Rx INVITE / 23316 INVITE / sip:150@192.168.10.70 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 002. NewChan Channel SIP/phone1-0000000c - from 67985a870350a076 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 003. TxResp SIP/2.0 / 23316 INVITE - 100 Trying [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 004. TxResp SIP/2.0 / 23316 INVITE - 180 Ringing [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 005. ConnectedLine Called party is now Hans Muster <150> [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 006. TxRespRel SIP/2.0 / 23316 INVITE - 200 OK [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 007. Rx ACK / 23316 ACK / sip:150@192.168.10.70:5060 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 008. Masq Old channel: SIP/phone2-0000000e [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 009. Masq (cont) ...new owner: SIP/phone1-0000000c [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 010. ConnectedLine Called party is now Max Mustermann <180> [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 011. TxReqRel INVITE / 102 INVITE - INVITE [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 012. Rx SIP/2.0 / 102 INVITE / 200 OK [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 013. TxReq ACK / 102 ACK - ACK [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 014. Hangup Cause Normal Clearing [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 015. SchedDestroy 32000 ms [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 016. RTCPaudio Quality:ssrc=986557840;themssrc=503322279;lp=95;rxjitter=0.0019 [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 017. TxReqRel BYE / 103 BYE - BYE [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 018. Rx SIP/2.0 / 103 BYE / 200 OK [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: 019. NeedDestroy Setting needdestroy because received 200 response [Nov 22 11:55:36] DEBUG[1175] chan_sip.c: ---------- END SIP HISTORY for '67985a870350a076' [Nov 22 11:55:36] DEBUG[1175] rtp_engine.c: Destroyed RTP instance '0xd800760' [Nov 22 11:55:48] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> REGISTER sip:192.168.10.70 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKe581140975ca75a86.e10aa3fb7f79be2d2 Max-Forwards: 70 From: ;tag=ef592dfd6b To: Call-ID: bddd1f752853e839 CSeq: 9009 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="" Supported: gruu, path User-Agent: Aastra 51i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 0 [ 34]: REGISTER sip:192.168.10.70 SIP/2.0 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKe581140975ca75a86.e10aa3fb7f79be2d2 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 3 [ 47]: From: ;tag=ef592dfd6b [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 4 [ 30]: To: [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: bddd1f752853e839 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 6 [ 19]: CSeq: 9009 REGISTER [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 9 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 51i/2.6.0.1008 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 22 11:55:48] VERBOSE[1175] chan_sip.c: --- (13 headers 0 lines) --- [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: = Looking for Call ID: bddd1f752853e839 (Checking From) --From tag ef592dfd6b --To-tag [Nov 22 11:55:48] DEBUG[1175] acl.c: For destination '192.168.10.202', our source address is '192.168.10.70'. [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Allocating new SIP dialog for bddd1f752853e839 - REGISTER (No RTP) [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Initializing initreq for method REGISTER - callid bddd1f752853e839 [Nov 22 11:55:48] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:55:48] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:55:48] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.202:5060 (no NAT) [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 22 11:55:48] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:55:48] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:55:48] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKe581140975ca75a86.e10aa3fb7f79be2d2;received=192.168.10.202 From: ;tag=ef592dfd6b To: ;tag=as2645d8f1 Call-ID: bddd1f752853e839 CSeq: 9009 REGISTER Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 22 Nov 2010 10:55:48 GMT Content-Length: 0 <------------> [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKe581140975ca75a86.e10aa3fb7f79be2d2;received=192.168.10.202 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 2 [ 47]: From: ;tag=ef592dfd6b [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 3 [ 45]: To: ;tag=as2645d8f1 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: bddd1f752853e839 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 5 [ 19]: CSeq: 9009 REGISTER [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 11 [ 35]: Date: Mon, 22 Nov 2010 10:55:48 GMT [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:55:48] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:55:48] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog 'bddd1f752853e839' in 32000 ms (Method: REGISTER) [Nov 22 11:55:48] DEBUG[1195] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone3 PeerStatus: Registered Address: 192.168.10.202:5060 [Nov 22 11:55:48] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:55:48] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:55:48] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 22 11:55:48] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '1' [Nov 22 11:55:48] DEBUG[1194] 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. [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: Auto destroying SIP dialog '2f7e3545567b8d26' [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: Destroying SIP dialog 2f7e3545567b8d26 [Nov 22 11:55:52] VERBOSE[1175] chan_sip.c: Really destroying SIP dialog '2f7e3545567b8d26' Method: REGISTER [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: ---------- SIP HISTORY for '2f7e3545567b8d26' [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: * SIP Call [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: 001. Rx REGISTER / 19006 REGISTER / sip:192.168.10.70 [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: 002. TxResp SIP/2.0 / 19006 REGISTER - 200 OK [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: 005. AutoDestroy 2f7e3545567b8d26 [Nov 22 11:55:52] DEBUG[1175] chan_sip.c: ---------- END SIP HISTORY for '2f7e3545567b8d26' [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: Auto destroying SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: Destroying SIP dialog 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:56:01] VERBOSE[1175] chan_sip.c: Really destroying SIP dialog '15c91da73e9014a058805f32700631ed@192.168.10.70' Method: BYE [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: Call to peer 'phone2' removed from call limit 2147483647 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: This call did not properly clean up call limits. Call ID 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: ---------- SIP HISTORY for '15c91da73e9014a058805f32700631ed@192.168.10.70' [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: * SIP Call [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 001. NewChan Channel SIP/phone2-0000000d - from 15c91da73e9014a058805f327006 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 200 OK [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 005. TxReq ACK / 102 ACK - ACK [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 006. Rx INVITE / 26430 INVITE / sip:100@192.168.10.70:5060 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 007. Hold INVITE [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 008. ReInv Re-invite received [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 009. TxResp SIP/2.0 / 26430 INVITE - 100 Trying [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 010. TxRespRel SIP/2.0 / 26430 INVITE - 200 OK [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 011. Rx ACK / 26430 ACK / sip:100@192.168.10.70:5060 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 012. Rx REFER / 26431 REFER / sip:100@192.168.10.70:5060 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 013. TxResp SIP/2.0 / 26431 REFER - 202 Accepted [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 014. Xfer Refer accepted [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 015. TxReqRel NOTIFY / 103 NOTIFY - NOTIFY [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 016. Xfer Refer succeeded [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 017. SchedDestroy 32000 ms [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 018. Rx SIP/2.0 / 103 NOTIFY / 200 OK [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 019. Rx BYE / 26432 BYE / sip:100@192.168.10.70:5060 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 020. RTCPaudio Quality:ssrc=1060254541;themssrc=2136089674;lp=0;rxjitter=0.000 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 021. RTCPaudioJitter Quality:minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 022. RTCPaudioLoss Quality:minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.00000 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 023. RTCPaudioRTT Quality:minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrt [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 024. CancelDestroy [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 025. SchedDestroy 32000 ms [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 026. TxResp SIP/2.0 / 26432 BYE - 200 OK [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: 027. AutoDestroy 15c91da73e9014a058805f32700631ed@192.168.10.70 [Nov 22 11:56:01] DEBUG[1175] chan_sip.c: ---------- END SIP HISTORY for '15c91da73e9014a058805f32700631ed@192.168.10.70' [Nov 22 11:56:01] DEBUG[1175] rtp_engine.c: Destroyed RTP instance '0xd823478' [Nov 22 11:56:01] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:56:01] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:56:01] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:56:01] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:56:01] DEBUG[1194] 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. [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: Auto destroying SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: Destroying SIP dialog 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:56:08] VERBOSE[1175] chan_sip.c: Really destroying SIP dialog '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' Method: BYE [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: ---------- SIP HISTORY for '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: * SIP Call [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 001. NewChan Channel SIP/phone3-0000000f - from 09c3fc3a2c0f19510efef690474d [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 200 OK [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 005. TxReq ACK / 102 ACK - ACK [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 006. ConnectedLine Calling party is now Erika Musterfrau <100> [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 007. TxReqRel INVITE / 103 INVITE - INVITE [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 008. Rx SIP/2.0 / 103 INVITE / 200 OK [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 009. TxReq ACK / 103 ACK - ACK [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 010. Rx BYE / 10025 BYE / sip:150@192.168.10.70:5060 [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 011. RTCPaudio Quality:ssrc=1531786074;themssrc=1262245749;lp=92;rxjitter=0.00 [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 012. RTCPaudioJitter Quality:minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0 [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 013. RTCPaudioLoss Quality:minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.00000 [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 014. RTCPaudioRTT Quality:minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrt [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 015. SchedDestroy 32000 ms [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 016. TxResp SIP/2.0 / 10025 BYE - 200 OK [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 017. Hangup Cause Normal Clearing [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: 018. AutoDestroy 09c3fc3a2c0f19510efef690474d8f36@192.168.10.70 [Nov 22 11:56:08] DEBUG[1175] chan_sip.c: ---------- END SIP HISTORY for '09c3fc3a2c0f19510efef690474d8f36@192.168.10.70' [Nov 22 11:56:08] DEBUG[1175] rtp_engine.c: Destroyed RTP instance '0xd7d5000' [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: Auto destroying SIP dialog 'bddd1f752853e839' [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: Destroying SIP dialog bddd1f752853e839 [Nov 22 11:56:20] VERBOSE[1175] chan_sip.c: Really destroying SIP dialog 'bddd1f752853e839' Method: REGISTER [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: ---------- SIP HISTORY for 'bddd1f752853e839' [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: * SIP Call [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: 001. Rx REGISTER / 9009 REGISTER / sip:192.168.10.70 [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: 002. TxResp SIP/2.0 / 9009 REGISTER - 200 OK [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: 005. AutoDestroy bddd1f752853e839 [Nov 22 11:56:20] DEBUG[1175] chan_sip.c: ---------- END SIP HISTORY for 'bddd1f752853e839' [Nov 22 11:56:33] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> REGISTER sip:192.168.10.70 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKa6e1f32af25c3da0f.ad214b1e5c403a693 Max-Forwards: 70 From: ;tag=e9b34a8110 To: Call-ID: 8b6423da4532bf09 CSeq: 3826 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Supported: gruu, path User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 0 [ 34]: REGISTER sip:192.168.10.70 SIP/2.0 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKa6e1f32af25c3da0f.ad214b1e5c403a693 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 3 [ 47]: From: ;tag=e9b34a8110 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 4 [ 30]: To: [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 8b6423da4532bf09 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 6 [ 19]: CSeq: 3826 REGISTER [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 9 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 22 11:56:33] VERBOSE[1175] chan_sip.c: --- (13 headers 0 lines) --- [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: = Looking for Call ID: 8b6423da4532bf09 (Checking From) --From tag e9b34a8110 --To-tag [Nov 22 11:56:33] DEBUG[1175] acl.c: For destination '192.168.10.200', our source address is '192.168.10.70'. [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Allocating new SIP dialog for 8b6423da4532bf09 - REGISTER (No RTP) [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Initializing initreq for method REGISTER - callid 8b6423da4532bf09 [Nov 22 11:56:33] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:56:33] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:56:33] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.200:5060 (no NAT) [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 22 11:56:33] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:56:33] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:56:33] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKa6e1f32af25c3da0f.ad214b1e5c403a693;received=192.168.10.200 From: ;tag=e9b34a8110 To: ;tag=as0f882a1e Call-ID: 8b6423da4532bf09 CSeq: 3826 REGISTER Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 22 Nov 2010 10:56:33 GMT Content-Length: 0 <------------> [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKa6e1f32af25c3da0f.ad214b1e5c403a693;received=192.168.10.200 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 2 [ 47]: From: ;tag=e9b34a8110 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 3 [ 45]: To: ;tag=as0f882a1e [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 8b6423da4532bf09 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 5 [ 19]: CSeq: 3826 REGISTER [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 11 [ 35]: Date: Mon, 22 Nov 2010 10:56:33 GMT [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:56:33] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:56:33] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog '8b6423da4532bf09' in 32000 ms (Method: REGISTER) [Nov 22 11:56:33] DEBUG[1195] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone2 PeerStatus: Registered Address: 192.168.10.200:5060 [Nov 22 11:56:33] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:56:33] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:56:33] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:56:33] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:56:33] DEBUG[1194] 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.