[Nov 4 19:32:01] VERBOSE[3292] config.c: == Parsing '/etc/asterisk/logger.conf': [Nov 4 19:32:01] DEBUG[3292] config.c: Parsing /etc/asterisk/logger.conf [Nov 4 19:32:01] VERBOSE[3292] config.c: == Found [Nov 4 19:32:01] VERBOSE[3292] logger.c: Asterisk Queue Logger restarted [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog '8e82bf14271b4964' [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: Destroying SIP dialog 8e82bf14271b4964 [Nov 4 19:32:04] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog '8e82bf14271b4964' Method: REGISTER [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for '8e82bf14271b4964' [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: 001. Rx REGISTER / 27611 REGISTER / sip:192.168.10.70:5060 [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: 002. TxResp SIP/2.0 / 27611 REGISTER - 200 OK [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: 005. AutoDestroy 8e82bf14271b4964 [Nov 4 19:32:04] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for '8e82bf14271b4964' [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.204:5060 ---> INVITE sip:180@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK821ecd5c786e3218b.913fe1391d4cb8bf8 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=7694b19ea5 To: "180" Call-ID: 16199ba5d692a456 CSeq: 13990 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erika Musterfrau" ;+sip.instance="";isfocus 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 192.168.10.204 s=SIP Call c=IN IP4 192.168.10.204 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 0 [ 41]: INVITE sip:180@192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK821ecd5c786e3218b.913fe1391d4cb8bf8 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 3 [ 71]: From: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 4 [ 38]: To: "180" [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 6 [ 18]: CSeq: 13990 INVITE [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 9 [146]: Contact: "Erika Musterfrau" ;+sip.instance="";isfocus [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 53i/2.6.0.1008 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 13 [ 19]: Content-Length: 620 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 14 [ 0]: [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.204 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:20] DEBUG[2885] 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 4 19:32:20] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 21 [ 25]: a=silenceSupp:off - - - - [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 22 [ 19]: a=fmtp:18 annexb=no [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Body 25 [ 10]: a=sendrecv [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: --- (14 headers 26 lines) --- [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: = Looking for Call ID: 16199ba5d692a456 (Checking From) --From tag 7694b19ea5 --To-tag [Nov 4 19:32:20] DEBUG[2885] acl.c: For destination '192.168.10.204', our source address is '192.168.10.70'. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Allocating new SIP dialog for 16199ba5d692a456 - INVITE (No RTP) [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -gruu- [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: gruu [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -path- [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: path [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: timer [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 4 19:32:20] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 4 19:32:20] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:20] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.204:5060 (no NAT) [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Initializing initreq for method INVITE - callid 16199ba5d692a456 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Using INVITE request as basis request - 16199ba5d692a456 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found peer 'phone1' for 'phone1' from 192.168.10.204:5060 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcfed320' [Nov 4 19:32:20] DEBUG[2885] res_rtp_asterisk.c: Allocated port 13090 for RTP instance '0xcfed320' [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: RTP instance '0xcfed320' is setup and ready to go [Nov 4 19:32:20] DEBUG[2885] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcfed320' [Nov 4 19:32:20] VERBOSE[2885] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 4 19:32:20] VERBOSE[2885] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Setting NAT on RTP to Off [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.204... UNSUPPORTED. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:20] DEBUG[2885] netsock2.c: Splitting '192.168.10.204' gives... [Nov 4 19:32:20] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '(null)'. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 18 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 18 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 106 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 106 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 107 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 113 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 110 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 110 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 111 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 111 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 112 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 112 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 98 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 98 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 97 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 97 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 115 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 115 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 96 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 9 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 9 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format G729 for ID 18 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format BV16 for ID 106 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format BV32 for ID 107 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format L16 for ID 113 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 110 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 111 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format L16 for ID 112 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format G726-16 for ID 98 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format G726-24 for ID 97 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format G726-32 for ID 115 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format G726-40 for ID 96 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format G722 for ID 9 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 9 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 18 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 97 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 98 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 106 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 110 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 111 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 112 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 113 on 0xb3507f88 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Incorporating payload 115 on 0xb3507f88 [Nov 4 19:32:20] VERBOSE[2885] 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 4 19:32:20] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:20] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfed320' [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.204:3000 [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 9 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 18 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 97 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 98 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 106 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 110 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 111 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 112 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 113 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] rtp_engine.c: Copying payload 115 from 0xb3507f88 to 0xcfed4cc [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Checking SIP call limits for device phone1 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Updating call counter for incoming call [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Call from peer 'phone1' is 1 out of 2147483647 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: Looking for 180 in Standard (domain 192.168.10.70:5060) [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: This channel will not be able to handle video. [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: build_route: Contact hop: "Erika Musterfrau" ;+sip.instance="";isfocus [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: list_route: hop: [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: SIP/phone1-00000000: New call is still down.... Trying... [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.204:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK821ecd5c786e3218b.913fe1391d4cb8bf8;received=192.168.10.204 From: "Erika Musterfrau" ;tag=7694b19ea5 To: "180" Call-ID: 16199ba5d692a456 CSeq: 13990 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 4 19:32:20] DEBUG[2885] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK821ecd5c786e3218b.913fe1391d4cb8bf8;received=192.168.10.204 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 2 [ 71]: From: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 3 [ 38]: To: "180" [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 5 [ 18]: CSeq: 13990 INVITE [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 9 [ 37]: Contact: [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 11 [ 0]: [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:20] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:20] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:20] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 4 19:32:20] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '2' [Nov 4 19:32:20] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:20] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:20] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 4 19:32:20] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '2' [Nov 4 19:32:20] DEBUG[2931] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone1-00000000 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Erika Musterfrau AccountCode: Exten: 180 Context: Standard Uniqueid: 1288895540.0 [Nov 4 19:32:20] DEBUG[2931] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-00000000 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1288895540.0 [Nov 4 19:32:20] DEBUG[3356] pbx.c: Launching 'Dial' [Nov 4 19:32:20] VERBOSE[3356] pbx.c: -- Executing [180@Standard:1] Dial("SIP/phone1-00000000", "SIP/phone3") in new stack [Nov 4 19:32:20] DEBUG[2879] 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 4 19:32:20] DEBUG[2904] 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 4 19:32:20] DEBUG[2904] 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 4 19:32:20] DEBUG[2931] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 1 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Allocating new SIP dialog for 6d3d4bc015078b713f7d31c66567717d@192.168.10.70 - INVITE (No RTP) [Nov 4 19:32:20] DEBUG[3356] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcfe79a8' [Nov 4 19:32:20] DEBUG[3356] res_rtp_asterisk.c: Allocated port 12552 for RTP instance '0xcfe79a8' [Nov 4 19:32:20] DEBUG[3356] rtp_engine.c: RTP instance '0xcfe79a8' is setup and ready to go [Nov 4 19:32:20] DEBUG[3356] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcfe79a8' [Nov 4 19:32:20] VERBOSE[3356] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 4 19:32:20] VERBOSE[3356] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Setting NAT on RTP to Off [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 4 19:32:20] DEBUG[3356] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: This channel will not be able to handle video. [Nov 4 19:32:20] DEBUG[3356] rtp_engine.c: Seeded SDP of 'SIP/phone3-00000001' with that of 'SIP/phone1-00000000' [Nov 4 19:32:20] DEBUG[3356] channel.c: Not copying variable DIALEDTIME. [Nov 4 19:32:20] DEBUG[3356] channel.c: Not copying variable ANSWEREDTIME. [Nov 4 19:32:20] DEBUG[3356] channel.c: Not copying variable DIALEDPEERNAME. [Nov 4 19:32:20] DEBUG[3356] channel.c: Not copying variable DIALEDPEERNUMBER. [Nov 4 19:32:20] DEBUG[3356] channel.c: Not copying variable DIALSTATUS. [Nov 4 19:32:20] DEBUG[3356] channel.c: Not copying variable SIPCALLID. [Nov 4 19:32:20] DEBUG[3356] channel.c: Not copying variable SIPDOMAIN. [Nov 4 19:32:20] DEBUG[3356] channel.c: Not copying variable SIPURI. [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Outgoing Call for phone3 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Call to peer 'phone3' is 1 out of 2147483647 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 4 19:32:20] VERBOSE[3356] chan_sip.c: Audio is at 5060 [Nov 4 19:32:20] VERBOSE[3356] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:20] VERBOSE[3356] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:20] VERBOSE[3356] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Initializing initreq for method INVITE - callid 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2fe1bca1 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 4 [ 50]: To: [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 9 [ 35]: Date: Thu, 04 Nov 2010 18:32:20 GMT [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:20] VERBOSE[3356] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:5060: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2fe1bca1 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as23e81265 To: Contact: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0-1 Date: Thu, 04 Nov 2010 18:32:20 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 396862845 396862845 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 12552 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 4 19:32:20] DEBUG[3356] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2fe1bca1 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 4 [ 50]: To: [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 9 [ 35]: Date: Thu, 04 Nov 2010 18:32:20 GMT [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 14 [ 19]: Content-Length: 260 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 1 [ 47]: o=root 396862845 396862845 IN IP4 192.168.10.70 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 5 [ 29]: m=audio 12552 RTP/AVP 8 0 101 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #248 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:20] VERBOSE[3356] app_dial.c: -- Called phone3 [Nov 4 19:32:20] DEBUG[2931] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone3-00000001 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 180 CallerIDName: Max Mustermann AccountCode: Exten: Context: Standard Uniqueid: 1288895540.1 [Nov 4 19:32:20] DEBUG[2931] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone1-00000000 Destination: SIP/phone3-00000001 CallerIDNum: 100 CallerIDName: Erika Musterfrau UniqueID: 1288895540.0 DestUniqueID: 1288895540.1 Dialstring: phone3 [Nov 4 19:32:20] DEBUG[2931] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone3-00000001 CallerIDNum: 180 CallerIDName: Uniqueid: 1288895540.1 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Nov 4 19:32:20] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:20] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:20] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 6 (Ringing) [Nov 4 19:32:20] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '6' [Nov 4 19:32:20] DEBUG[2879] 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 4 19:32:20] DEBUG[2904] 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 4 19:32:20] DEBUG[2931] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 8 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2fe1bca1 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Call-ID: 62c2b7e359a754d65a4182b429640262@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 55i/2.6.0.66 Supported: gruu, path Content-Length: 0 <-------------> [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2fe1bca1 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 4 19:32:20] VERBOSE[2885] chan_sip.c: --- (12 headers 0 lines) --- [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: = Looking for Call ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 (Checking To) --From tag as23e81265 --To-tag 1288348833 [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: *** SIP TIMER: Cancelling retransmission #248 - INVITE (got response) [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '62c2b7e359a754d65a4182b429640262@192.168.10.70' Request 102: Found [Nov 4 19:32:20] DEBUG[2885] chan_sip.c: SIP response 180 to standard invite [Nov 4 19:32:20] VERBOSE[3356] app_dial.c: -- SIP/phone3-00000001 is ringing [Nov 4 19:32:20] DEBUG[3356] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000000' with that of 'SIP/phone3-00000001' [Nov 4 19:32:20] VERBOSE[3356] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.204:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK821ecd5c786e3218b.913fe1391d4cb8bf8;received=192.168.10.204 From: "Erika Musterfrau" ;tag=7694b19ea5 To: "180" ;tag=as6cca4433 Call-ID: 16199ba5d692a456 CSeq: 13990 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 4 19:32:20] DEBUG[3356] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK821ecd5c786e3218b.913fe1391d4cb8bf8;received=192.168.10.204 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 2 [ 71]: From: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 3 [ 53]: To: "180" ;tag=as6cca4433 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 4 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 5 [ 18]: CSeq: 13990 INVITE [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 9 [ 37]: Contact: [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Header 11 [ 0]: [Nov 4 19:32:20] DEBUG[3356] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:20] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:20] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:20] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 6 (Ringing) [Nov 4 19:32:20] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '6' [Nov 4 19:32:20] DEBUG[2931] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000001 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 180 CallerIDName: Uniqueid: 1288895540.1 [Nov 4 19:32:20] DEBUG[2904] 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 4 19:32:22] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2fe1bca1 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Call-ID: 62c2b7e359a754d65a4182b429640262@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 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 192.168.10.203 s=SIP Call c=IN IP4 192.168.10.203 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 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2fe1bca1 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.203 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: = Looking for Call ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 (Checking To) --From tag as23e81265 --To-tag 1288348833 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Acked pending invite 102 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Stopping retransmission on '62c2b7e359a754d65a4182b429640262@192.168.10.70' of Request 102: Match Found [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: SIP response 200 to standard invite [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.203... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.203' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '(null)'. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:22] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfe79a8' [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.203:3000 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfe7b54 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfe7b54 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfe7b54 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: build_route: Contact hop: "Max Mustermann" ;+sip.instance="" [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: list_route: hop: [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Strict routing enforced for session 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.203:5060 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.203:5060: ACK sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK6f166486 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Contact: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK6f166486 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:22] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:22] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:22] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 4 19:32:22] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '2' [Nov 4 19:32:22] VERBOSE[3356] app_dial.c: -- SIP/phone3-00000001 answered SIP/phone1-00000000 [Nov 4 19:32:22] DEBUG[3356] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000000' with that of 'SIP/phone3-00000001' [Nov 4 19:32:22] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:22] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:22] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 4 19:32:22] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '2' [Nov 4 19:32:22] DEBUG[2931] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000001 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 180 CallerIDName: Uniqueid: 1288895540.1 [Nov 4 19:32:22] DEBUG[2879] 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 4 19:32:22] DEBUG[2904] 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 4 19:32:22] DEBUG[2931] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 1 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: SIP answering channel: SIP/phone1-00000000 [Nov 4 19:32:22] DEBUG[3356] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Setting framing from config on incoming call [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Audio is at 5060 [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.204:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK821ecd5c786e3218b.913fe1391d4cb8bf8;received=192.168.10.204 From: "Erika Musterfrau" ;tag=7694b19ea5 To: "180" ;tag=as6cca4433 Call-ID: 16199ba5d692a456 CSeq: 13990 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: 262 v=0 o=root 1104495279 1104495279 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 13090 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 4 19:32:22] DEBUG[3356] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK821ecd5c786e3218b.913fe1391d4cb8bf8;received=192.168.10.204 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 2 [ 71]: From: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 3 [ 53]: To: "180" ;tag=as6cca4433 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 4 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 5 [ 18]: CSeq: 13990 INVITE [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 9 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 10 [ 66]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 12 [ 19]: Content-Length: 262 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 1 [ 49]: o=root 1104495279 1104495279 IN IP4 192.168.10.70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 5 [ 29]: m=audio 13090 RTP/AVP 8 0 101 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #251 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:22] DEBUG[3356] features.c: bridge answer set, chan answer set [Nov 4 19:32:22] DEBUG[3356] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 4 19:32:22] DEBUG[3356] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 4 19:32:22] VERBOSE[3356] rtp_engine.c: -- Remotely bridging SIP/phone1-00000000 and SIP/phone3-00000001 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Deferring reinvite on SIP '16199ba5d692a456' - It's audio will be redirected to IP 192.168.10.203:3000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Sending reinvite on SIP '62c2b7e359a754d65a4182b429640262@192.168.10.70' - It's audio soon redirected to IP 192.168.10.204:3000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Strict routing enforced for session 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:22] DEBUG[3356] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:22] DEBUG[3356] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: set_destination: set destination to 192.168.10.203:5060 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Audio is at 5060 [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Initializing already initialized SIP dialog 62c2b7e359a754d65a4182b429640262@192.168.10.70 (presumably reinvite) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3de3d948 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 4 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:5060: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3de3d948 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Contact: Call-ID: 62c2b7e359a754d65a4182b429640262@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 X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Erika Musterfrau" Content-Type: application/sdp Content-Length: 261 v=0 o=root 396862845 396862846 IN IP4 192.168.10.204 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.204 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3de3d948 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 4 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 14 [ 19]: Content-Length: 261 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 1 [ 48]: o=root 396862845 396862846 IN IP4 192.168.10.204 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #252 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:22] DEBUG[2904] 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 4 19:32:22] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:22] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:22] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 4 19:32:22] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '2' [Nov 4 19:32:22] DEBUG[2931] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-00000000 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1288895540.0 [Nov 4 19:32:22] DEBUG[2931] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone3-00000001 Uniqueid: 1288895540.1 AccountCode: OldAccountCode: [Nov 4 19:32:22] DEBUG[2931] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone1-00000000 Channel2: SIP/phone3-00000001 Uniqueid1: 1288895540.0 Uniqueid2: 1288895540.1 CallerID1: 100 CallerID2: 180 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: INVITE [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:22] DEBUG[2904] 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 4 19:32:22] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3de3d948 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Call-ID: 62c2b7e359a754d65a4182b429640262@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 55i/2.6.0.66 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 1 IN IP4 192.168.10.203 s=SIP Call c=IN IP4 192.168.10.203 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 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3de3d948 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.203 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: = Looking for Call ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 (Checking To) --From tag as23e81265 --To-tag 1288348833 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Acked pending invite 103 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #252 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Stopping retransmission on '62c2b7e359a754d65a4182b429640262@192.168.10.70' of Request 103: Match Found [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: SIP response 200 to RE-invite on outgoing call 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.203... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.203' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '(null)'. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:22] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfe79a8' [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.203:3000 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfe7b54 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfe7b54 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfe7b54 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Strict routing enforced for session 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.203:5060 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.203:5060: ACK sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0af91855 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Contact: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0af91855 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: INVITE [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:22] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:22] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:22] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 4 19:32:22] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '2' [Nov 4 19:32:22] DEBUG[2904] 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 4 19:32:22] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.204:5060 ---> ACK sip:180@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK30302f709e4ef947d.83eea797973ab6849 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=7694b19ea5 To: "180" ;tag=as6cca4433 Call-ID: 16199ba5d692a456 CSeq: 13990 ACK User-Agent: Aastra 53i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 38]: ACK sip:180@192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK30302f709e4ef947d.83eea797973ab6849 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 71]: From: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 53]: To: "180" ;tag=as6cca4433 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 15]: CSeq: 13990 ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 33]: User-Agent: Aastra 53i/2.6.0.1008 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: --- (9 headers 0 lines) --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: = Looking for Call ID: 16199ba5d692a456 (Checking From) --From tag 7694b19ea5 --To-tag as6cca4433 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #251 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Stopping retransmission on '16199ba5d692a456' of Response 13990: Match Found [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Sending pending reinvite on '16199ba5d692a456' [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Strict routing enforced for session 16199ba5d692a456 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.204:5060 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Audio is at 5060 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Initializing already initialized SIP dialog 16199ba5d692a456 (presumably reinvite) [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1f10d234 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 12 [ 66]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.204:5060: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1f10d234 Max-Forwards: 70 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Contact: Call-ID: 16199ba5d692a456 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 X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Max Mustermann" Content-Type: application/sdp Content-Length: 263 v=0 o=root 1104495279 1104495280 IN IP4 192.168.10.203 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.203 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1f10d234 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 12 [ 66]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 14 [ 19]: Content-Length: 263 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 1 [ 50]: o=root 1104495279 1104495280 IN IP4 192.168.10.203 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #253 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.204:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1f10d234 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Call-ID: 16199ba5d692a456 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="";isfocus 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 192.168.10.204 s=SIP Call c=IN IP4 192.168.10.204 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 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1f10d234 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [146]: Contact: "Erika Musterfrau" ;+sip.instance="";isfocus [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.204 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: = Looking for Call ID: 16199ba5d692a456 (Checking To) --From tag as6cca4433 --To-tag 7694b19ea5 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Acked pending invite 102 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #253 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Stopping retransmission on '16199ba5d692a456' of Request 102: Match Found [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: SIP response 200 to RE-invite on outgoing call 16199ba5d692a456 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.204... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.204' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '(null)'. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:22] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfed320' [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.204:3000 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Updating call counter for incoming call [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Strict routing enforced for session 16199ba5d692a456 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.204:5060 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.204:5060: ACK sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK6054b8e7 Max-Forwards: 70 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Contact: Call-ID: 16199ba5d692a456 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK6054b8e7 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:22] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:22] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:22] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 4 19:32:22] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '2' [Nov 4 19:32:22] DEBUG[3356] rtp_engine.c: Oooh, 'SIP/phone1-00000000' changed end address to 192.168.10.204:3000 (format unknown) [Nov 4 19:32:22] DEBUG[3356] rtp_engine.c: Oooh, 'SIP/phone1-00000000' was 192.168.10.204:3000/(format unknown) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Sending reinvite on SIP '62c2b7e359a754d65a4182b429640262@192.168.10.70' - It's audio soon redirected to IP 192.168.10.204:3000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Strict routing enforced for session 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:22] DEBUG[3356] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:22] DEBUG[3356] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: set_destination: set destination to 192.168.10.203:5060 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Audio is at 5060 [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Initializing already initialized SIP dialog 62c2b7e359a754d65a4182b429640262@192.168.10.70 (presumably reinvite) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK75557039 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 4 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] VERBOSE[3356] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:5060: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK75557039 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Contact: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Erika Musterfrau" Content-Type: application/sdp Content-Length: 261 v=0 o=root 396862845 396862847 IN IP4 192.168.10.204 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.204 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK75557039 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 4 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 14 [ 19]: Content-Length: 261 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 1 [ 48]: o=root 396862845 396862847 IN IP4 192.168.10.204 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #254 [Nov 4 19:32:22] DEBUG[3356] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:22] DEBUG[2904] 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 4 19:32:22] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK75557039 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 104 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 55i/2.6.0.66 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 2 IN IP4 192.168.10.203 s=SIP Call c=IN IP4 192.168.10.203 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 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK75557039 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 2 IN IP4 192.168.10.203 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: = Looking for Call ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 (Checking To) --From tag as23e81265 --To-tag 1288348833 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Acked pending invite 104 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #254 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Stopping retransmission on '62c2b7e359a754d65a4182b429640262@192.168.10.70' of Request 104: Match Found [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: SIP response 200 to RE-invite on outgoing call 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.10.203... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.203' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '(null)'. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:22] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfe79a8' [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.203:3000 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfe7b54 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfe7b54 [Nov 4 19:32:22] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfe7b54 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Strict routing enforced for session 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:22] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:22] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.203:5060 [Nov 4 19:32:22] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.203:5060: ACK sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1df95400 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Contact: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1df95400 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 104 ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:22] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:22] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:22] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 4 19:32:22] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '2' [Nov 4 19:32:22] DEBUG[2904] 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 4 19:32:23] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:23] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:24] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:24] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> INVITE sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKbc3586f46cd7325d1.a1e7c13a0f21cb777 Max-Forwards: 70 From: ;tag=1288348833 To: "Erika Musterfrau" ;tag=as23e81265 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 29393 INVITE 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, timer, 100rel, replaces User-Agent: Aastra 55i/2.6.0.66 Content-Type: application/sdp Content-Length: 620 v=0 o=MxSIP 0 3 IN IP4 192.168.10.203 s=SIP Call c=IN IP4 192.168.10.203 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 a=ptime:30 a=sendonly <-------------> [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 0 [ 41]: INVITE sip:100@192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKbc3586f46cd7325d1.a1e7c13a0f21cb777 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 3 [ 67]: From: ;tag=1288348833 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 5 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 6 [ 18]: CSeq: 29393 INVITE [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 9 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 13 [ 19]: Content-Length: 620 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 14 [ 0]: [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.203 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:25] DEBUG[2885] 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 4 19:32:25] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 21 [ 25]: a=silenceSupp:off - - - - [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 22 [ 19]: a=fmtp:18 annexb=no [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 25 [ 10]: a=sendonly [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: --- (14 headers 26 lines) --- [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: = Looking for Call ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 (Checking From) --From tag 1288348833 --To-tag as23e81265 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -gruu- [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: gruu [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -path- [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: path [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: timer [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 4 19:32:25] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 4 19:32:25] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:25] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.203:5060 (no NAT) [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Initializing initreq for method INVITE - callid 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.10.203... UNSUPPORTED. [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:25] DEBUG[2885] netsock2.c: Splitting '192.168.10.203' gives... [Nov 4 19:32:25] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '(null)'. [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 18 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 18 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 106 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 106 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 107 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 113 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 110 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 110 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 111 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 111 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 112 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 112 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 98 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 98 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 97 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 97 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 115 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 115 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 96 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 9 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 9 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format G729 for ID 18 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format BV16 for ID 106 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format BV32 for ID 107 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format L16 for ID 113 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 110 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 111 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format L16 for ID 112 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format G726-16 for ID 98 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format G726-24 for ID 97 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format G726-32 for ID 115 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format G726-40 for ID 96 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format G722 for ID 9 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 9 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 18 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 97 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 98 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 106 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 110 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 111 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 112 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 113 on 0xb3507f88 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Incorporating payload 115 on 0xb3507f88 [Nov 4 19:32:25] VERBOSE[2885] 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 4 19:32:25] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:25] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfe79a8' [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.203:3000 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 9 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 18 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 97 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 98 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 106 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 110 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 111 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 112 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 113 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] rtp_engine.c: Copying payload 115 from 0xb3507f88 to 0xcfe7b54 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:25] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfe79a8' [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Got a SIP re-invite for call 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: SIP/phone3-00000001: This call is UP.... [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKbc3586f46cd7325d1.a1e7c13a0f21cb777;received=192.168.10.203 From: ;tag=1288348833 To: "Erika Musterfrau" ;tag=as23e81265 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 29393 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 4 19:32:25] DEBUG[2885] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKbc3586f46cd7325d1.a1e7c13a0f21cb777;received=192.168.10.203 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 2 [ 67]: From: ;tag=1288348833 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 5 [ 18]: CSeq: 29393 INVITE [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 9 [ 37]: Contact: [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 11 [ 0]: [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Setting framing from config on incoming call [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Audio is at 5060 [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:25] VERBOSE[2885] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKbc3586f46cd7325d1.a1e7c13a0f21cb777;received=192.168.10.203 From: ;tag=1288348833 To: "Erika Musterfrau" ;tag=as23e81265 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 29393 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: 261 v=0 o=root 396862845 396862848 IN IP4 192.168.10.204 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.204 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly <------------> [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKbc3586f46cd7325d1.a1e7c13a0f21cb777;received=192.168.10.203 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 2 [ 67]: From: ;tag=1288348833 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 5 [ 18]: CSeq: 29393 INVITE [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 9 [ 37]: Contact: [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 11 [ 19]: Content-Length: 261 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Header 12 [ 0]: [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 1 [ 48]: o=root 396862845 396862848 IN IP4 192.168.10.204 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=recvonly [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #255 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Sending reinvite on SIP '16199ba5d692a456' - It's audio soon redirected to IP 192.168.10.70:5060 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Strict routing enforced for session 16199ba5d692a456 [Nov 4 19:32:25] VERBOSE[3356] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:25] DEBUG[3356] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:25] DEBUG[3356] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:25] VERBOSE[3356] chan_sip.c: set_destination: set destination to 192.168.10.204:5060 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 19:32:25] VERBOSE[3356] chan_sip.c: Audio is at 5060 [Nov 4 19:32:25] VERBOSE[3356] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:25] VERBOSE[3356] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:25] VERBOSE[3356] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Initializing already initialized SIP dialog 16199ba5d692a456 (presumably reinvite) [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0f3d61e0 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 12 [ 66]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:25] VERBOSE[3356] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.204:5060: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0f3d61e0 Max-Forwards: 70 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Contact: Call-ID: 16199ba5d692a456 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 X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Max Mustermann" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1104495279 1104495281 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 13090 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 4 19:32:25] DEBUG[3356] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0f3d61e0 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 12 [ 66]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 14 [ 19]: Content-Length: 262 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 1 [ 49]: o=root 1104495279 1104495281 IN IP4 192.168.10.70 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 5 [ 29]: m=audio 13090 RTP/AVP 8 0 101 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #256 [Nov 4 19:32:25] DEBUG[3356] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:25] DEBUG[3356] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 4 19:32:25] VERBOSE[3356] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/phone1-00000000 [Nov 4 19:32:25] DEBUG[3356] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Nov 4 19:32:25] DEBUG[3356] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:25] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: INVITE [Nov 4 19:32:25] DEBUG[2931] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/phone1-00000000 UniqueID: 1288895540.0 Class: default [Nov 4 19:32:25] DEBUG[3356] channel.c: Set channel SIP/phone1-00000000 to write format slin [Nov 4 19:32:25] DEBUG[3356] res_musiconhold.c: SIP/phone1-00000000 Opened file 0 '/var/lib/asterisk/moh/iptam/iptam_moh' [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xcfed320' [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> ACK sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK341af2721c60ae2b6.9dfbe5954c82517f1 Max-Forwards: 70 From: ;tag=1288348833 To: "Erika Musterfrau" ;tag=as23e81265 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 29393 ACK User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 0 [ 38]: ACK sip:100@192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK341af2721c60ae2b6.9dfbe5954c82517f1 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 3 [ 67]: From: ;tag=1288348833 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 5 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 6 [ 15]: CSeq: 29393 ACK [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 7 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: --- (9 headers 0 lines) --- [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: = Looking for Call ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 (Checking From) --From tag 1288348833 --To-tag as23e81265 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #255 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Stopping retransmission on '62c2b7e359a754d65a4182b429640262@192.168.10.70' of Response 29393: Match Found [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.204:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0f3d61e0 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Call-ID: 16199ba5d692a456 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erika Musterfrau" ;+sip.instance="";isfocus Server: Aastra 53i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 2 IN IP4 192.168.10.204 s=SIP Call c=IN IP4 192.168.10.204 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 4 19:32:26] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0f3d61e0 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 2 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 3 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 8 [146]: Contact: "Erika Musterfrau" ;+sip.instance="";isfocus [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 2 IN IP4 192.168.10.204 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: = Looking for Call ID: 16199ba5d692a456 (Checking To) --From tag as6cca4433 --To-tag 7694b19ea5 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Acked pending invite 103 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #256 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Stopping retransmission on '16199ba5d692a456' of Request 103: Match Found [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: SIP response 200 to RE-invite on outgoing call 16199ba5d692a456 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.10.204... UNSUPPORTED. [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:26] DEBUG[2885] netsock2.c: Splitting '192.168.10.204' gives... [Nov 4 19:32:26] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '(null)'. [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:26] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:26] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:26] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:26] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:26] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:26] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:26] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfed320' [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.204:3000 [Nov 4 19:32:26] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:26] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:26] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Updating call counter for incoming call [Nov 4 19:32:26] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:26] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Strict routing enforced for session 16199ba5d692a456 [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:26] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:26] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.204:5060 [Nov 4 19:32:26] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.204:5060: ACK sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK46d5df9d Max-Forwards: 70 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Contact: Call-ID: 16199ba5d692a456 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK46d5df9d [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:26] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:26] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:26] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:26] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 4 19:32:26] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '2' [Nov 4 19:32:26] DEBUG[2904] 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 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:26] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:27] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:27] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:28] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:28] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:29] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:29] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 4 19:32:30] DEBUG[2931] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.204:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 37381 SequenceNumberCycles: 0 IAJitter: 7 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> INVITE sip:150@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK2a8635994610f690f.0978f0c19125b5d49 Max-Forwards: 70 From: "Max Mustermann" ;tag=0f7a6e4d38 To: "150" Call-ID: d35e65a68bab1563 CSeq: 4843 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="";isfocus 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 192.168.10.203 s=SIP Call c=IN IP4 192.168.10.203 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:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 0 [ 41]: INVITE sip:150@192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK2a8635994610f690f.0978f0c19125b5d49 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 3 [ 69]: From: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 4 [ 38]: To: "150" [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 6 [ 17]: CSeq: 4843 INVITE [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 9 [144]: Contact: "Max Mustermann" ;+sip.instance="";isfocus [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 13 [ 19]: Content-Length: 620 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 14 [ 0]: [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.203 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:30] DEBUG[2885] 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 4 19:32:30] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 21 [ 25]: a=silenceSupp:off - - - - [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 22 [ 19]: a=fmtp:18 annexb=no [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Body 25 [ 10]: a=sendrecv [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: --- (14 headers 26 lines) --- [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: = Looking for Call ID: d35e65a68bab1563 (Checking From) --From tag 0f7a6e4d38 --To-tag [Nov 4 19:32:30] DEBUG[2885] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Allocating new SIP dialog for d35e65a68bab1563 - INVITE (No RTP) [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -gruu- [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: gruu [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -path- [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: path [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: timer [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 4 19:32:30] DEBUG[2885] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 4 19:32:30] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:30] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.203:5060 (no NAT) [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Initializing initreq for method INVITE - callid d35e65a68bab1563 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Using INVITE request as basis request - d35e65a68bab1563 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found peer 'phone3' for 'phone3' from 192.168.10.203:5060 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd006238' [Nov 4 19:32:30] DEBUG[2885] res_rtp_asterisk.c: Allocated port 10044 for RTP instance '0xd006238' [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: RTP instance '0xd006238' is setup and ready to go [Nov 4 19:32:30] DEBUG[2885] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd006238' [Nov 4 19:32:30] VERBOSE[2885] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 4 19:32:30] VERBOSE[2885] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Setting NAT on RTP to Off [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.203... UNSUPPORTED. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:30] DEBUG[2885] netsock2.c: Splitting '192.168.10.203' gives... [Nov 4 19:32:30] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '(null)'. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 18 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 18 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 106 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 106 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 107 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 113 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 110 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 110 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 111 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 111 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 112 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 112 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 98 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 98 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 97 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 97 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 115 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 115 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 96 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 9 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 9 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format G729 for ID 18 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format BV16 for ID 106 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format BV32 for ID 107 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format L16 for ID 113 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 110 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 111 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format L16 for ID 112 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format G726-16 for ID 98 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format G726-24 for ID 97 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format G726-32 for ID 115 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format G726-40 for ID 96 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format G722 for ID 9 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 9 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 18 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 97 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 98 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 106 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 110 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 111 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 112 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 113 on 0xb3507f88 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Incorporating payload 115 on 0xb3507f88 [Nov 4 19:32:30] VERBOSE[2885] 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 4 19:32:30] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:30] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd006238' [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.203:3002 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 9 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 18 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 97 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 98 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 106 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 110 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 111 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 112 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 113 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] rtp_engine.c: Copying payload 115 from 0xb3507f88 to 0xd0063e4 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Checking SIP call limits for device phone3 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Updating call counter for incoming call [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Call from peer 'phone3' is 2 out of 2147483647 [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: Looking for 150 in Standard (domain 192.168.10.70:5060) [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: This channel will not be able to handle video. [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: build_route: Contact hop: "Max Mustermann" ;+sip.instance="";isfocus [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: list_route: hop: [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: SIP/phone3-00000002: New call is still down.... Trying... [Nov 4 19:32:30] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK2a8635994610f690f.0978f0c19125b5d49;received=192.168.10.203 From: "Max Mustermann" ;tag=0f7a6e4d38 To: "150" Call-ID: d35e65a68bab1563 CSeq: 4843 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 4 19:32:30] DEBUG[2885] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK2a8635994610f690f.0978f0c19125b5d49;received=192.168.10.203 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 2 [ 69]: From: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 3 [ 38]: To: "150" [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 5 [ 17]: CSeq: 4843 INVITE [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 9 [ 37]: Contact: [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Header 11 [ 0]: [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:30] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:30] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 4 19:32:30] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '2' [Nov 4 19:32:30] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:30] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:30] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 4 19:32:30] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '2' [Nov 4 19:32:30] DEBUG[2931] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone3-00000002 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 180 CallerIDName: Max Mustermann AccountCode: Exten: 150 Context: Standard Uniqueid: 1288895550.2 [Nov 4 19:32:30] DEBUG[2931] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000002 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 180 CallerIDName: Max Mustermann Uniqueid: 1288895550.2 [Nov 4 19:32:30] DEBUG[3357] pbx.c: Launching 'Dial' [Nov 4 19:32:30] VERBOSE[3357] pbx.c: -- Executing [150@Standard:1] Dial("SIP/phone3-00000002", "SIP/phone2") in new stack [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Allocating new SIP dialog for 18b537555fa5e9716bd72adf4be8b4c3@192.168.10.70 - INVITE (No RTP) [Nov 4 19:32:30] DEBUG[3357] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcfea2e8' [Nov 4 19:32:30] DEBUG[3357] res_rtp_asterisk.c: Allocated port 13048 for RTP instance '0xcfea2e8' [Nov 4 19:32:30] DEBUG[3357] rtp_engine.c: RTP instance '0xcfea2e8' is setup and ready to go [Nov 4 19:32:30] DEBUG[3357] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcfea2e8' [Nov 4 19:32:30] VERBOSE[3357] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 4 19:32:30] VERBOSE[3357] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Setting NAT on RTP to Off [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 4 19:32:30] DEBUG[3357] acl.c: For destination '192.168.10.205', our source address is '192.168.10.70'. [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: This channel will not be able to handle video. [Nov 4 19:32:30] DEBUG[3357] rtp_engine.c: Seeded SDP of 'SIP/phone2-00000003' with that of 'SIP/phone3-00000002' [Nov 4 19:32:30] DEBUG[3357] channel.c: Not copying variable DIALEDTIME. [Nov 4 19:32:30] DEBUG[3357] channel.c: Not copying variable ANSWEREDTIME. [Nov 4 19:32:30] DEBUG[3357] channel.c: Not copying variable DIALEDPEERNAME. [Nov 4 19:32:30] DEBUG[3357] channel.c: Not copying variable DIALEDPEERNUMBER. [Nov 4 19:32:30] DEBUG[3357] channel.c: Not copying variable DIALSTATUS. [Nov 4 19:32:30] DEBUG[3357] channel.c: Not copying variable SIPCALLID. [Nov 4 19:32:30] DEBUG[3357] channel.c: Not copying variable SIPDOMAIN. [Nov 4 19:32:30] DEBUG[3357] channel.c: Not copying variable SIPURI. [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Outgoing Call for phone2 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Call to peer 'phone2' is 1 out of 2147483647 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 4 19:32:30] VERBOSE[3357] chan_sip.c: Audio is at 5060 [Nov 4 19:32:30] VERBOSE[3357] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:30] VERBOSE[3357] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:30] VERBOSE[3357] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Initializing initreq for method INVITE - callid 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK590aaa34 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 4 [ 50]: To: [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 9 [ 35]: Date: Thu, 04 Nov 2010 18:32:30 GMT [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:30] VERBOSE[3357] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.205:5060: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK590aaa34 Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0-1 Date: Thu, 04 Nov 2010 18:32:30 GMT 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: 258 v=0 o=root 81777966 81777966 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 13048 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 4 19:32:30] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK590aaa34 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 4 [ 50]: To: [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 9 [ 35]: Date: Thu, 04 Nov 2010 18:32:30 GMT [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 14 [ 19]: Content-Length: 258 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 1 [ 45]: o=root 81777966 81777966 IN IP4 192.168.10.70 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 5 [ 29]: m=audio 13048 RTP/AVP 8 0 101 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #259 [Nov 4 19:32:30] DEBUG[3357] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:30] VERBOSE[3357] app_dial.c: -- Called phone2 [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[2931] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-00000003 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 150 CallerIDName: Hans Muster AccountCode: Exten: Context: Standard Uniqueid: 1288895550.3 [Nov 4 19:32:30] DEBUG[2931] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone3-00000002 Destination: SIP/phone2-00000003 CallerIDNum: 180 CallerIDName: Max Mustermann UniqueID: 1288895550.2 DestUniqueID: 1288895550.3 Dialstring: phone2 [Nov 4 19:32:30] DEBUG[2931] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone2-00000003 CallerIDNum: 150 CallerIDName: Uniqueid: 1288895550.3 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:30] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:30] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:32:30] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:32:30] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Nov 4 19:32:30] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '6' [Nov 4 19:32:30] DEBUG[2904] 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 4 19:32:30] DEBUG[2904] 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 4 19:32:30] DEBUG[2879] 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 4 19:32:30] DEBUG[2904] 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 4 19:32:30] DEBUG[2931] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 8 [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:30] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:31] DEBUG[2931] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 192.168.10.204:3001 OurSSRC: 521077756 SentNTP: 1288895551.0067313664 SentRTP: 40160 SentPackets: 251 SentOctets: 40160 ReportBlock: FractionLost: 256 CumulativeLoss: 1 IAJitter: 0.0000 TheirLastSR: 241169859 DLSR: 0.7520 (sec) [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK590aaa34 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@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: "Hans Muster" ;+sip.instance="" Server: Aastra 51i/2.6.0.1008 Supported: gruu, path Content-Length: 0 <-------------> [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK590aaa34 [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 2 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 8 [133]: Contact: "Hans Muster" ;+sip.instance="" [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 4 19:32:31] VERBOSE[2885] chan_sip.c: --- (12 headers 0 lines) --- [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: = Looking for Call ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (Checking To) --From tag as33b155b8 --To-tag 3198768735 [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: *** SIP TIMER: Cancelling retransmission #259 - INVITE (got response) [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Request 102: Found [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: SIP response 180 to standard invite [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:31] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:31] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:32:31] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:32:31] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Nov 4 19:32:31] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '6' [Nov 4 19:32:31] DEBUG[2931] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000003 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 150 CallerIDName: Uniqueid: 1288895550.3 [Nov 4 19:32:31] DEBUG[2904] 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 4 19:32:31] VERBOSE[3357] app_dial.c: -- SIP/phone2-00000003 is ringing [Nov 4 19:32:31] DEBUG[3357] rtp_engine.c: Setting early bridge SDP of 'SIP/phone3-00000002' with that of 'SIP/phone2-00000003' [Nov 4 19:32:31] VERBOSE[3357] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK2a8635994610f690f.0978f0c19125b5d49;received=192.168.10.203 From: "Max Mustermann" ;tag=0f7a6e4d38 To: "150" ;tag=as456f70bd Call-ID: d35e65a68bab1563 CSeq: 4843 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 4 19:32:31] DEBUG[3357] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK2a8635994610f690f.0978f0c19125b5d49;received=192.168.10.203 [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 2 [ 69]: From: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 3 [ 53]: To: "150" ;tag=as456f70bd [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 4 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 5 [ 17]: CSeq: 4843 INVITE [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 9 [ 37]: Contact: [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Header 11 [ 0]: [Nov 4 19:32:31] DEBUG[3357] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:31] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK590aaa34 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@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: "Hans Muster" ;+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 192.168.10.205 s=SIP Call c=IN IP4 192.168.10.205 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 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK590aaa34 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [133]: Contact: "Hans Muster" ;+sip.instance="" [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.205 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.205 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: = Looking for Call ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (Checking To) --From tag as33b155b8 --To-tag 3198768735 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Acked pending invite 102 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Stopping retransmission on '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' of Request 102: Match Found [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: SIP response 200 to standard invite [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.205... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.205' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '(null)'. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.205... OK. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:32] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfea2e8' [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.205:3000 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfea494 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfea494 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfea494 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: build_route: Contact hop: "Hans Muster" ;+sip.instance="" [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: list_route: hop: [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Strict routing enforced for session 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.205:5060 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.205:5060: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK14b889cf Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK14b889cf [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:32] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:32:32] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:32:32] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 4 19:32:32] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '2' [Nov 4 19:32:32] VERBOSE[3357] app_dial.c: -- SIP/phone2-00000003 answered SIP/phone3-00000002 [Nov 4 19:32:32] DEBUG[3357] rtp_engine.c: Setting early bridge SDP of 'SIP/phone3-00000002' with that of 'SIP/phone2-00000003' [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: SIP answering channel: SIP/phone3-00000002 [Nov 4 19:32:32] DEBUG[3357] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Setting framing from config on incoming call [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: Audio is at 5060 [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK2a8635994610f690f.0978f0c19125b5d49;received=192.168.10.203 From: "Max Mustermann" ;tag=0f7a6e4d38 To: "150" ;tag=as456f70bd Call-ID: d35e65a68bab1563 CSeq: 4843 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 1175340312 1175340312 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 10044 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 4 19:32:32] DEBUG[3357] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK2a8635994610f690f.0978f0c19125b5d49;received=192.168.10.203 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 2 [ 69]: From: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 3 [ 53]: To: "150" ;tag=as456f70bd [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 4 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 5 [ 17]: CSeq: 4843 INVITE [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 9 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 10 [ 63]: P-Asserted-Identity: "Hans Muster" [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 12 [ 19]: Content-Length: 262 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 1 [ 49]: o=root 1175340312 1175340312 IN IP4 192.168.10.70 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 5 [ 29]: m=audio 10044 RTP/AVP 8 0 101 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #262 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:32] DEBUG[3357] features.c: bridge answer set, chan answer set [Nov 4 19:32:32] DEBUG[3357] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 4 19:32:32] DEBUG[3357] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 4 19:32:32] VERBOSE[3357] rtp_engine.c: -- Remotely bridging SIP/phone3-00000002 and SIP/phone2-00000003 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Deferring reinvite on SIP 'd35e65a68bab1563' - It's audio will be redirected to IP 192.168.10.205:3000 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Sending reinvite on SIP '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' - It's audio soon redirected to IP 192.168.10.203:3002 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Strict routing enforced for session 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:32] DEBUG[3357] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:32] DEBUG[3357] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: set_destination: set destination to 192.168.10.205:5060 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: Audio is at 5060 [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Initializing already initialized SIP dialog 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (presumably reinvite) [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4d900bee [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] VERBOSE[3357] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.205:5060: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4d900bee Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@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 X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Max Mustermann" Content-Type: application/sdp Content-Length: 259 v=0 o=root 81777966 81777967 IN IP4 192.168.10.203 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.203 t=0 0 m=audio 3002 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 4 19:32:32] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4d900bee [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 14 [ 19]: Content-Length: 259 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 1 [ 46]: o=root 81777966 81777967 IN IP4 192.168.10.203 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 5 [ 28]: m=audio 3002 RTP/AVP 8 0 101 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #263 [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[2931] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000003 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 150 CallerIDName: Uniqueid: 1288895550.3 [Nov 4 19:32:32] DEBUG[2931] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000002 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 180 CallerIDName: Max Mustermann Uniqueid: 1288895550.2 [Nov 4 19:32:32] DEBUG[2931] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone2-00000003 Uniqueid: 1288895550.3 AccountCode: OldAccountCode: [Nov 4 19:32:32] DEBUG[2931] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone3-00000002 Channel2: SIP/phone2-00000003 Uniqueid1: 1288895550.2 Uniqueid2: 1288895550.3 CallerID1: 180 CallerID2: 150 [Nov 4 19:32:32] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:32:32] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:32:32] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 4 19:32:32] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '2' [Nov 4 19:32:32] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:32] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:32] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 4 19:32:32] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '2' [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'd35e65a68bab1563' Method: INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:32] DEBUG[2879] 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 4 19:32:32] DEBUG[2904] 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 4 19:32:32] DEBUG[2931] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 1 [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[2904] 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 4 19:32:32] DEBUG[2904] 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 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> ACK sip:150@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK0718607904015b4c8.92cd70c37dab7637e Max-Forwards: 70 From: "Max Mustermann" ;tag=0f7a6e4d38 To: "150" ;tag=as456f70bd Call-ID: d35e65a68bab1563 CSeq: 4843 ACK User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 38]: ACK sip:150@192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK0718607904015b4c8.92cd70c37dab7637e [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 69]: From: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 53]: To: "150" ;tag=as456f70bd [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 14]: CSeq: 4843 ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: --- (9 headers 0 lines) --- [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: = Looking for Call ID: d35e65a68bab1563 (Checking From) --From tag 0f7a6e4d38 --To-tag as456f70bd [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #262 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Stopping retransmission on 'd35e65a68bab1563' of Response 4843: Match Found [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Sending pending reinvite on 'd35e65a68bab1563' [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Strict routing enforced for session d35e65a68bab1563 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.203:5060 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Audio is at 5060 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Initializing already initialized SIP dialog d35e65a68bab1563 (presumably reinvite) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK364f1b31 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "150" ;tag=as456f70bd [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 67]: To: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Hans Muster" [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:5060: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK364f1b31 Max-Forwards: 70 From: "150" ;tag=as456f70bd To: "Max Mustermann" ;tag=0f7a6e4d38 Contact: Call-ID: d35e65a68bab1563 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 X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Hans Muster" Content-Type: application/sdp Content-Length: 263 v=0 o=root 1175340312 1175340313 IN IP4 192.168.10.205 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.205 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK364f1b31 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "150" ;tag=as456f70bd [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 67]: To: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Hans Muster" [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 14 [ 19]: Content-Length: 263 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 1 [ 50]: o=root 1175340312 1175340313 IN IP4 192.168.10.205 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.205 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #264 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'd35e65a68bab1563' Method: ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK364f1b31 From: "150" ;tag=as456f70bd To: "Max Mustermann" ;tag=0f7a6e4d38 Call-ID: d35e65a68bab1563 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="";isfocus Server: Aastra 55i/2.6.0.66 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 1 IN IP4 192.168.10.203 s=SIP Call c=IN IP4 192.168.10.203 t=0 0 m=audio 3002 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 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK364f1b31 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 55]: From: "150" ;tag=as456f70bd [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 67]: To: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [144]: Contact: "Max Mustermann" ;+sip.instance="";isfocus [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.203 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3002 RTP/AVP 8 0 101 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: = Looking for Call ID: d35e65a68bab1563 (Checking To) --From tag as456f70bd --To-tag 0f7a6e4d38 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Acked pending invite 102 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #264 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Stopping retransmission on 'd35e65a68bab1563' of Request 102: Match Found [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: SIP response 200 to RE-invite on outgoing call d35e65a68bab1563 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.203... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.203' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '(null)'. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:32] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd006238' [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.203:3002 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xd0063e4 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xd0063e4 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xd0063e4 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Updating call counter for incoming call [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Strict routing enforced for session d35e65a68bab1563 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.203:5060 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.203:5060: ACK sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2bab25a3 Max-Forwards: 70 From: "150" ;tag=as456f70bd To: "Max Mustermann" ;tag=0f7a6e4d38 Contact: Call-ID: d35e65a68bab1563 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2bab25a3 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "150" ;tag=as456f70bd [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 67]: To: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'd35e65a68bab1563' Method: ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:32] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:32] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:32] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 4 19:32:32] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '2' [Nov 4 19:32:32] DEBUG[3357] rtp_engine.c: Oooh, 'SIP/phone3-00000002' changed end address to 192.168.10.203:3002 (format unknown) [Nov 4 19:32:32] DEBUG[3357] rtp_engine.c: Oooh, 'SIP/phone3-00000002' was 192.168.10.203:3002/(format unknown) [Nov 4 19:32:32] DEBUG[3357] chan_sip.c: Deferring reinvite on SIP '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' - It's audio will be redirected to IP 192.168.10.203:3002 [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4d900bee From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@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: "Hans Muster" ;+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 192.168.10.205 s=SIP Call c=IN IP4 192.168.10.205 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 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4d900bee [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [133]: Contact: "Hans Muster" ;+sip.instance="" [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.205 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.205 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: = Looking for Call ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (Checking To) --From tag as33b155b8 --To-tag 3198768735 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Acked pending invite 103 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #263 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Stopping retransmission on '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' of Request 103: Match Found [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: SIP response 200 to RE-invite on outgoing call 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.205... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.205' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '(null)'. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.205... OK. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:32] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfea2e8' [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.205:3000 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfea494 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfea494 [Nov 4 19:32:32] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfea494 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Strict routing enforced for session 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.205:5060 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.205:5060: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK71d36582 Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK71d36582 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Sending pending reinvite on '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Strict routing enforced for session 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:32] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:32] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.205:5060 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Audio is at 5060 [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Initializing already initialized SIP dialog 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (presumably reinvite) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1452bdd0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] VERBOSE[2885] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.205:5060: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1452bdd0 Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Max Mustermann" Content-Type: application/sdp Content-Length: 259 v=0 o=root 81777966 81777968 IN IP4 192.168.10.203 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.203 t=0 0 m=audio 3002 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 4 19:32:32] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1452bdd0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 14 [ 19]: Content-Length: 259 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 1 [ 46]: o=root 81777966 81777968 IN IP4 192.168.10.203 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.203 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3002 RTP/AVP 8 0 101 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #265 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'd35e65a68bab1563' Method: ACK [Nov 4 19:32:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:32] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:32:32] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:32:32] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 4 19:32:32] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '2' [Nov 4 19:32:32] DEBUG[2904] 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 4 19:32:32] DEBUG[2904] 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 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:32] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1452bdd0 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Hans Muster" ;+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 2 IN IP4 192.168.10.205 s=SIP Call c=IN IP4 192.168.10.205 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 4 19:32:33] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1452bdd0 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 2 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 8 [133]: Contact: "Hans Muster" ;+sip.instance="" [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 2 IN IP4 192.168.10.205 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.205 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: = Looking for Call ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (Checking To) --From tag as33b155b8 --To-tag 3198768735 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Acked pending invite 104 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #265 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Stopping retransmission on '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' of Request 104: Match Found [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: SIP response 200 to RE-invite on outgoing call 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.10.205... UNSUPPORTED. [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:33] DEBUG[2885] netsock2.c: Splitting '192.168.10.205' gives... [Nov 4 19:32:33] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '(null)'. [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.205... OK. [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:33] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:33] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:33] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:33] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:33] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:33] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:33] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfea2e8' [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.205:3000 [Nov 4 19:32:33] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfea494 [Nov 4 19:32:33] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfea494 [Nov 4 19:32:33] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfea494 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:33] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:33] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Strict routing enforced for session 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:33] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:33] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.205:5060 [Nov 4 19:32:33] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.205:5060: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK32ef9425 Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK32ef9425 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 104 ACK [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'd35e65a68bab1563' Method: ACK [Nov 4 19:32:33] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:33] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:32:33] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:32:33] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 4 19:32:33] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '2' [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[2904] 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 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:33] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:34] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:34] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'd35e65a68bab1563' Method: ACK [Nov 4 19:32:34] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:34] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:35] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:35] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'd35e65a68bab1563' Method: ACK [Nov 4 19:32:35] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 4 19:32:35] DEBUG[2931] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.204:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 37631 SequenceNumberCycles: 0 IAJitter: 12 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:35] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'd35e65a68bab1563' Method: ACK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: ACK [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 192.168.10.204:3001 OurSSRC: 521077756 SentNTP: 1288895556.0021016576 SentRTP: 80160 SentPackets: 501 SentOctets: 80160 ReportBlock: FractionLost: 0 CumulativeLoss: 1 IAJitter: 0.0000 TheirLastSR: 241496883 DLSR: 0.7510 (sec) [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> REFER sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK6fbd6d9e62c353a93.b74bcb398d660b047 Max-Forwards: 70 From: ;tag=1288348833 To: "Erika Musterfrau" ;tag=as23e81265 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 29394 REFER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="" Refer-To: "150" Referred-By: Supported: gruu, path, timer User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 40]: REFER sip:100@192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK6fbd6d9e62c353a93.b74bcb398d660b047 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 67]: From: ;tag=1288348833 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 17]: CSeq: 29394 REFER [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [116]: Refer-To: "150" [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 39]: Referred-By: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 12 [ 28]: Supported: gruu, path, timer [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 13 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (15 headers 0 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 (Checking From) --From tag 1288348833 --To-tag as23e81265 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Call 62c2b7e359a754d65a4182b429640262@192.168.10.70 got a SIP call transfer from caller: (REFER)! [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Attended transfer: Will use Replace-Call-ID : d35e65a68bab1563 F-tag: 0f7a6e4d38 T-tag: as456f70bd [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: SIP transfer to extension 150@Standard by phone3@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP attended transfer: Transferer channel SIP/phone3-00000001, transferee channel SIP/phone1-00000000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/phone1-00000000' [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Looking for callid d35e65a68bab1563 (fromtag 0f7a6e4d38 totag as456f70bd) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Matched INCOMING call - their tag is 0f7a6e4d38 Our tag is as456f70bd [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK6fbd6d9e62c353a93.b74bcb398d660b047;received=192.168.10.203 From: ;tag=1288348833 To: "Erika Musterfrau" ;tag=as23e81265 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 29394 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 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 20]: SIP/2.0 202 Accepted [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK6fbd6d9e62c353a93.b74bcb398d660b047;received=192.168.10.203 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 67]: From: ;tag=1288348833 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 17]: CSeq: 29394 REFER [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP attended transfer: trying to bridge SIP/phone3-00000002 and SIP/phone1-00000000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Sip transfer:-------------------- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: -- Transferer to PBX channel: SIP/phone3-00000001 State Up [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: -- Transferer to PBX second channel (target): SIP/phone3-00000002 State Up [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: -- Bridged call to transferee: SIP/phone1-00000000 State Up [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: -- Bridged call to transfer target: SIP/phone2-00000003 State Up [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: -- END Sip transfer:-------------------- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP transfer: Four channels to handle [Nov 4 19:32:36] VERBOSE[2885] res_musiconhold.c: -- Stopped music on hold on SIP/phone1-00000000 [Nov 4 19:32:36] DEBUG[2885] channel.c: Set channel SIP/phone1-00000000 to write format alaw [Nov 4 19:32:36] DEBUG[2885] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP transfer: trying to masquerade SIP/phone1-00000000 into SIP/phone3-00000002 [Nov 4 19:32:36] DEBUG[2885] channel.c: Planning to masquerade channel SIP/phone1-00000000 into the structure of SIP/phone3-00000002 [Nov 4 19:32:36] DEBUG[2885] channel.c: Done planning to masquerade channel SIP/phone1-00000000 into the structure of SIP/phone3-00000002 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Strict routing enforced for session 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.203:5060 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:5060: NOTIFY sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK68d38663 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Contact: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 105 NOTIFY User-Agent: Asterisk PBX 1.8.0-1 Event: refer;id=29394 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 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 59]: NOTIFY sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK68d38663 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 105 NOTIFY [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 21]: Event: refer;id=29394 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 48]: Subscription-state: terminated;reason=noresource [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 41]: Content-Type: message/sipfrag;version=2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 12 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 14 [ 18]: Content-Length: 16 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #266 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:36] DEBUG[2885] channel.c: Actually Masquerading SIP/phone1-00000000(6) into the structure of SIP/phone3-00000002(6) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Sending reinvite on SIP 'd35e65a68bab1563' - It's audio soon redirected to IP 192.168.10.70:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Strict routing enforced for session d35e65a68bab1563 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.203:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Audio is at 5060 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Initializing already initialized SIP dialog d35e65a68bab1563 (presumably reinvite) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5fc3c822 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "150" ;tag=as456f70bd [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 67]: To: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 12 [ 66]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.203:5060: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5fc3c822 Max-Forwards: 70 From: "150" ;tag=as456f70bd To: "Max Mustermann" ;tag=0f7a6e4d38 Contact: Call-ID: d35e65a68bab1563 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 X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Max Mustermann" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1175340312 1175340314 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 10044 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 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.203:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5fc3c822 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "150" ;tag=as456f70bd [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 67]: To: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 12 [ 66]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 14 [ 19]: Content-Length: 262 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 1 [ 49]: o=root 1175340312 1175340314 IN IP4 192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 5 [ 29]: m=audio 10044 RTP/AVP 8 0 101 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #267 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP Fixup: New owner for dialogue d35e65a68bab1563: SIP/phone1-00000000 (Old parent: SIP/phone1-00000000) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Hangup call SIP/phone1-00000000, SIP callid d35e65a68bab1563 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: update_call_counter(phone3) - decrement call limit counter on hangup [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Updating call counter for incoming call [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Call from peer 'phone3' removed from call limit 2147483647 [Nov 4 19:32:36] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd006238' [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog 'd35e65a68bab1563' in 32000 ms (Method: ACK) [Nov 4 19:32:36] DEBUG[2885] channel.c: Putting channel SIP/phone1-00000000 in alaw/alaw formats [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP Fixup: New owner for dialogue 16199ba5d692a456: SIP/phone1-00000000 (Old parent: SIP/phone3-00000002) [Nov 4 19:32:36] DEBUG[2885] channel.c: Released clone lock on 'SIP/phone3-00000002' [Nov 4 19:32:36] DEBUG[2885] channel.c: Done Masquerading SIP/phone1-00000000 (6) [Nov 4 19:32:36] DEBUG[2885] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 4 19:32:36] DEBUG[2885] res_rtp_asterisk.c: Changing ssrc from 521077756 to 1336532803 due to a source change [Nov 4 19:32:36] DEBUG[2885] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'd35e65a68bab1563' Method: ACK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: REFER [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Attended Channel: SIP/phone3-00000001 Uniqueid: 1288895540.1 SIP-Callid: 62c2b7e359a754d65a4182b429640262@192.168.10.70 TargetChannel: SIP/phone3-00000002 TargetUniqueid: 1288895550.2 [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/phone1-00000000 UniqueID: 1288895540.0 [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/phone1-00000000 CloneState: Up Original: SIP/phone3-00000002 OriginalState: Up [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone1-00000000 Newname: SIP/phone1-00000000 Uniqueid: 1288895540.0 [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone3-00000002 Newname: SIP/phone1-00000000 Uniqueid: 1288895550.2 [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone1-00000000 Newname: SIP/phone3-00000002 Uniqueid: 1288895540.0 [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone1-00000000 CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1288895550.2 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/phone1-00000000 UniqueID: 1288895550.2 [Nov 4 19:32:36] DEBUG[3356] rtp_engine.c: Oooh, something is weird, backing out [Nov 4 19:32:36] WARNING[3356] rtp_engine.c: Channel 'SIP/phone3-00000002' failed to break RTP bridge [Nov 4 19:32:36] DEBUG[3356] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/phone3-00000002, c1=SIP/phone3-00000001, flags: Yes,Yes,No,No [Nov 4 19:32:36] DEBUG[3356] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 4 19:32:36] DEBUG[3356] channel.c: Bridge stops bridging channels SIP/phone3-00000002 and SIP/phone3-00000001 [Nov 4 19:32:36] DEBUG[3356] channel.c: Hanging up channel 'SIP/phone3-00000001' [Nov 4 19:32:36] DEBUG[3356] chan_sip.c: update_call_counter(phone3) - decrement call limit counter on hangup [Nov 4 19:32:36] DEBUG[3356] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:36] DEBUG[3356] chan_sip.c: Call to peer 'phone3' removed from call limit 2147483647 [Nov 4 19:32:36] DEBUG[3356] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 62c2b7e359a754d65a4182b429640262@192.168.10.70. [Nov 4 19:32:36] VERBOSE[3356] chan_sip.c: Scheduling destruction of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' in 32000 ms (Method: REFER) [Nov 4 19:32:36] DEBUG[3356] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 4 19:32:36] DEBUG[3356] pbx.c: Spawn extension (Standard,180,1) exited non-zero on 'SIP/phone3-00000002' [Nov 4 19:32:36] VERBOSE[3356] pbx.c: == Spawn extension (Standard, 180, 1) exited non-zero on 'SIP/phone3-00000002' [Nov 4 19:32:36] DEBUG[3356] channel.c: Soft-Hanging up channel 'SIP/phone3-00000002' [Nov 4 19:32:36] DEBUG[3356] channel.c: Hanging up zombie 'SIP/phone3-00000002' [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Strict routing enforced for session 16199ba5d692a456 [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[3357] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:36] DEBUG[3357] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: set_destination: set destination to 192.168.10.204:5060 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Audio is at 5060 [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Initializing already initialized SIP dialog 16199ba5d692a456 (presumably reinvite) [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2f92dcf0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 11 [ 63]: P-Asserted-Identity: "Hans Muster" [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.204:5060: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2f92dcf0 Max-Forwards: 70 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Contact: Call-ID: 16199ba5d692a456 CSeq: 104 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: "Hans Muster" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1104495279 1104495282 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 13090 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 4 19:32:36] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2f92dcf0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 11 [ 63]: P-Asserted-Identity: "Hans Muster" [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 14 [ 0]: [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 1 [ 49]: o=root 1104495279 1104495282 IN IP4 192.168.10.70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 5 [ 29]: m=audio 13090 RTP/AVP 8 0 101 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #270 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:36] DEBUG[3357] rtp_engine.c: Oooh, something is weird, backing out [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Sending reinvite on SIP '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' - It's audio soon redirected to IP 192.168.10.70:5060 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Strict routing enforced for session 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[3357] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:36] DEBUG[3357] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: set_destination: set destination to 192.168.10.205:5060 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Audio is at 5060 [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Initializing already initialized SIP dialog 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (presumably reinvite) [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2611663b [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.205:5060: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2611663b Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Max Mustermann" Content-Type: application/sdp Content-Length: 258 v=0 o=root 81777966 81777969 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 13048 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 4 19:32:36] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2611663b [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 14 [ 19]: Content-Length: 258 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 1 [ 45]: o=root 81777966 81777969 IN IP4 192.168.10.70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 5 [ 29]: m=audio 13048 RTP/AVP 8 0 101 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #271 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:36] VERBOSE[3357] rtp_engine.c: -- Remotely bridging SIP/phone1-00000000 and SIP/phone2-00000003 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Deferring reinvite on SIP '16199ba5d692a456' - It's audio will be redirected to IP 192.168.10.205:3000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Deferring reinvite on SIP '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' - It's audio will be redirected to IP 192.168.10.204:3000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Strict routing enforced for session 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[3357] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:36] DEBUG[3357] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: set_destination: set destination to 192.168.10.205:5060 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Audio is at 5060 [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Initializing already initialized SIP dialog 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (presumably reinvite) [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK38677cd0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 106 INVITE [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 11 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.205:5060: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK38677cd0 Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 106 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: 259 v=0 o=root 81777966 81777970 IN IP4 192.168.10.204 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.204 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK38677cd0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 7 [ 16]: CSeq: 106 INVITE [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 11 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 13 [ 19]: Content-Length: 259 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Header 14 [ 0]: [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 1 [ 46]: o=root 81777966 81777970 IN IP4 192.168.10.204 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #272 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '1' [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '1' [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '1' [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '1' [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone3-00000002 Channel2: SIP/phone3-00000001 Uniqueid1: 1288895540.0 Uniqueid2: 1288895540.1 CallerID1: 180 CallerID2: 180 [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 100 Destination: 180 DestinationContext: Standard CallerID: "Erika Musterfrau" <100> Channel: SIP/phone1-00000000 DestinationChannel: SIP/phone3-00000001 LastApplication: Dial LastData: SIP/phone3 StartTime: 2010-11-04 19:32:20 AnswerTime: 2010-11-04 19:32:22 EndTime: 2010-11-04 19:32:36 Duration: 16 BillableSeconds: 14 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1288895540.0 UserField: [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone3-00000001 Uniqueid: 1288895540.1 CallerIDNum: 180 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone3-00000002 UniqueID: 1288895540.0 DialStatus: ANSWER [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone3-00000002 Uniqueid: 1288895540.0 CallerIDNum: 180 CallerIDName: Max Mustermann Cause: 16 Cause-txt: Normal Clearing [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] DEBUG[2879] 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 4 19:32:36] DEBUG[2904] 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 4 19:32:36] DEBUG[2904] 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 4 19:32:36] DEBUG[2904] 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 4 19:32:36] DEBUG[2904] 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 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 0 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.204:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2f92dcf0 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Call-ID: 16199ba5d692a456 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erika Musterfrau" ;+sip.instance="";isfocus Server: Aastra 53i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 3 IN IP4 192.168.10.204 s=SIP Call c=IN IP4 192.168.10.204 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 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2f92dcf0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [146]: Contact: "Erika Musterfrau" ;+sip.instance="";isfocus [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.204 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: 16199ba5d692a456 (Checking To) --From tag as6cca4433 --To-tag 7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Acked pending invite 104 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #270 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Stopping retransmission on '16199ba5d692a456' of Request 104: Match Found [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP response 200 to RE-invite on outgoing call 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.10.204... UNSUPPORTED. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.204' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '(null)'. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:36] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfed320' [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.204:3000 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: We have an owner, now see if we need to change this call [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Updating call counter for incoming call [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Strict routing enforced for session 16199ba5d692a456 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.204:5060 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.204:5060: ACK sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4c425214 Max-Forwards: 70 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Contact: Call-ID: 16199ba5d692a456 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4c425214 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 104 ACK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Sending pending reinvite on '16199ba5d692a456' [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Strict routing enforced for session 16199ba5d692a456 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.204:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Audio is at 5060 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Initializing already initialized SIP dialog 16199ba5d692a456 (presumably reinvite) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK23adea2c [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Hans Muster" [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.204:5060: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK23adea2c Max-Forwards: 70 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Contact: Call-ID: 16199ba5d692a456 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Hans Muster" Content-Type: application/sdp Content-Length: 263 v=0 o=root 1104495279 1104495283 IN IP4 192.168.10.205 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.205 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK23adea2c [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Hans Muster" [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 14 [ 19]: Content-Length: 263 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 15 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 1 [ 50]: o=root 1104495279 1104495283 IN IP4 192.168.10.205 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.205 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #273 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '2' [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK68d38663 From: "Erika Musterfrau" ;tag=as23e81265 To: ;tag=1288348833 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 105 NOTIFY Server: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK68d38663 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=1288348833 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 105 NOTIFY [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (8 headers 0 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 (Checking To) --From tag as23e81265 --To-tag 1288348833 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #266 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Stopping retransmission on '62c2b7e359a754d65a4182b429640262@192.168.10.70' of Request 105: Match Found [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] DEBUG[2904] 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 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> BYE sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK44f90f1e77cac8ee6.e58e562c0f903e7fa Max-Forwards: 70 From: ;tag=1288348833 To: "Erika Musterfrau" ;tag=as23e81265 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 29395 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 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 38]: BYE sip:100@192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK44f90f1e77cac8ee6.e58e562c0f903e7fa [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 67]: From: ;tag=1288348833 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 15]: CSeq: 29395 BYE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 28]: Supported: gruu, path, timer [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (12 headers 0 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 (Checking From) --From tag 1288348833 --To-tag as23e81265 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Initializing initreq for method BYE - callid 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.203:5060 (no NAT) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Setting SIP_ALREADYGONE on dialog 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfe79a8' [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' in 32000 ms (Method: BYE) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Received bye, no owner, selfdestruct soon. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK44f90f1e77cac8ee6.e58e562c0f903e7fa;received=192.168.10.203 From: ;tag=1288348833 To: "Erika Musterfrau" ;tag=as23e81265 Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 CSeq: 29395 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 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK44f90f1e77cac8ee6.e58e562c0f903e7fa;received=192.168.10.203 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 67]: From: ;tag=1288348833 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as23e81265 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 15]: CSeq: 29395 BYE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK38677cd0 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 106 INVITE Retry-After: 10 Server: Aastra 51i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 33]: SIP/2.0 500 Internal Server Error [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK38677cd0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 106 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 15]: Retry-After: 10 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (9 headers 0 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (Checking To) --From tag as33b155b8 --To-tag 3198768735 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Acked pending invite 106 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #272 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Stopping retransmission on '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' of Request 106: Match Found [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: -- Got SIP response 500 "Internal Server Error" back from 192.168.10.205:5060 [Nov 4 19:32:36] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfea2e8' [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Strict routing enforced for session 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.205:5060 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.205:5060: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK38677cd0 Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 106 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK38677cd0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 106 ACK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Setting SIP_ALREADYGONE on dialog 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] DEBUG[3357] rtp_engine.c: Got a FRAME_CONTROL (8) frame on channel SIP/phone2-00000003 [Nov 4 19:32:36] DEBUG[3357] channel.c: Returning from native bridge, channels: SIP/phone1-00000000, SIP/phone2-00000003 [Nov 4 19:32:36] DEBUG[3357] channel.c: Hanging up channel 'SIP/phone2-00000003' [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Hangup call SIP/phone2-00000003, SIP callid 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: update_call_counter(phone2) - decrement call limit counter on hangup [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Call to peer 'phone2' removed from call limit 2147483647 [Nov 4 19:32:36] DEBUG[3357] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfea2e8' [Nov 4 19:32:36] DEBUG[3357] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 4 19:32:36] DEBUG[3357] pbx.c: Spawn extension (Standard,150,1) exited non-zero on 'SIP/phone1-00000000' [Nov 4 19:32:36] VERBOSE[3357] pbx.c: == Spawn extension (Standard, 150, 1) exited non-zero on 'SIP/phone1-00000000' [Nov 4 19:32:36] DEBUG[3357] channel.c: Soft-Hanging up channel 'SIP/phone1-00000000' [Nov 4 19:32:36] DEBUG[3357] channel.c: Hanging up channel 'SIP/phone1-00000000' [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Hangup call SIP/phone1-00000000, SIP callid 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: update_call_counter(phone1) - decrement call limit counter on hangup [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Updating call counter for incoming call [Nov 4 19:32:36] DEBUG[3357] chan_sip.c: Call from peer 'phone1' removed from call limit 2147483647 [Nov 4 19:32:36] DEBUG[3357] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfed320' [Nov 4 19:32:36] VERBOSE[3357] chan_sip.c: Scheduling destruction of SIP dialog '16199ba5d692a456' in 32000 ms (Method: ACK) [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone1-00000000 Channel2: SIP/phone2-00000003 Uniqueid1: 1288895550.2 Uniqueid2: 1288895550.3 CallerID1: 100 CallerID2: 150 [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 180 Destination: 150 DestinationContext: Standard CallerID: "Max Mustermann" <180> Channel: SIP/phone3-00000002 DestinationChannel: SIP/phone2-00000003 LastApplication: Dial LastData: SIP/phone2 StartTime: 2010-11-04 19:32:30 AnswerTime: 2010-11-04 19:32:32 EndTime: 2010-11-04 19:32:36 Duration: 6 BillableSeconds: 4 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1288895550.2 UserField: [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-00000003 Uniqueid: 1288895550.3 CallerIDNum: 150 CallerIDName: Cause: 38 Cause-txt: Network out of order [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone1-00000000 UniqueID: 1288895550.2 DialStatus: ANSWER [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone1-00000000 Uniqueid: 1288895550.2 CallerIDNum: 100 CallerIDName: Erika Musterfrau Cause: 38 Cause-txt: Network out of order [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2611663b From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 105 INVITE Server: Aastra 51i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2611663b [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (8 headers 0 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (Checking To) --From tag as33b155b8 --To-tag 3198768735 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: *** SIP TIMER: Cancelling retransmission #271 - INVITE (got response) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Request 105: Found [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP response 100 to standard invite [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Got response on call that is already terminated: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (ignoring) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2611663b From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Hans Muster" ;+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 3 IN IP4 192.168.10.205 s=SIP Call c=IN IP4 192.168.10.205 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 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2611663b [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [133]: Contact: "Hans Muster" ;+sip.instance="" [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.205 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.205 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (Checking To) --From tag as33b155b8 --To-tag 3198768735 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Stopping retransmission on '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' of Request 105: Match Found [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP response 200 to standard invite [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Got response on call that is already terminated: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (ignoring) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.204:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK23adea2c From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Call-ID: 16199ba5d692a456 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erika Musterfrau" ;+sip.instance="";isfocus Server: Aastra 53i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 4 IN IP4 192.168.10.204 s=SIP Call c=IN IP4 192.168.10.204 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 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK23adea2c [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [146]: Contact: "Erika Musterfrau" ;+sip.instance="";isfocus [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 4 IN IP4 192.168.10.204 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.204 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: 16199ba5d692a456 (Checking To) --From tag as6cca4433 --To-tag 7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Acked pending invite 105 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #273 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Stopping retransmission on '16199ba5d692a456' of Request 105: Match Found [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP response 200 to standard invite [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP o=MxSIP 0 4 IN IP4 192.168.10.204... UNSUPPORTED. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.204' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '(null)'. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found RTP audio format 8 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Setting payload 8 based on m type on 0xb3508588 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found RTP audio format 0 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Setting payload 0 based on m type on 0xb3508588 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found RTP audio format 101 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Setting payload 101 based on m type on 0xb3508588 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Incorporating payload 0 on 0xb3508588 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Incorporating payload 8 on 0xb3508588 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Incorporating payload 101 on 0xb3508588 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 4 19:32:36] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfed320' [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Peer audio RTP is at port 192.168.10.204:3000 [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Copying payload 0 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Copying payload 8 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:36] DEBUG[2885] rtp_engine.c: Copying payload 101 from 0xb3508588 to 0xcfed4cc [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Updating call counter for incoming call [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: build_route: Retaining previous route: [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Strict routing enforced for session 16199ba5d692a456 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.204:5060 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.204:5060: ACK sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3f289bb9 Max-Forwards: 70 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Contact: Call-ID: 16199ba5d692a456 CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3f289bb9 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 105 ACK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Strict routing enforced for session 16199ba5d692a456 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:36] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:36] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.204:5060 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.204:5060: BYE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0920f9e5 Max-Forwards: 70 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Call-ID: 16199ba5d692a456 CSeq: 106 BYE User-Agent: Asterisk PBX 1.8.0-1 X-Asterisk-HangupCause: Network out of order X-Asterisk-HangupCauseCode: 38 Content-Length: 0 --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 56]: BYE sip:phone1@192.168.10.204:5060;transport=udp SIP/2.0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0920f9e5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 13]: CSeq: 106 BYE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 8 [ 44]: X-Asterisk-HangupCause: Network out of order [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 38 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 11 [ 0]: [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #276 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog '16199ba5d692a456' in 32000 ms (Method: ACK) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5fc3c822 From: "150" ;tag=as456f70bd To: "Max Mustermann" ;tag=0f7a6e4d38 Call-ID: d35e65a68bab1563 CSeq: 103 INVITE Server: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK5fc3c822 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 55]: From: "150" ;tag=as456f70bd [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 67]: To: "Max Mustermann" ;tag=0f7a6e4d38 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: d35e65a68bab1563 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (8 headers 0 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: d35e65a68bab1563 (Checking To) --From tag as456f70bd --To-tag 0f7a6e4d38 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: *** SIP TIMER: Cancelling retransmission #267 - INVITE (got response) [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on 'd35e65a68bab1563' Request 103: Found [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: SIP response 100 to standard invite [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '1' [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '1' [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '1' [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '1' [Nov 4 19:32:36] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:36] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:36] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 4 19:32:36] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '1' [Nov 4 19:32:36] DEBUG[2879] 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 4 19:32:36] DEBUG[2879] 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 4 19:32:36] DEBUG[2904] 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 4 19:32:36] DEBUG[2904] 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 4 19:32:36] DEBUG[2904] 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 4 19:32:36] DEBUG[2904] 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 4 19:32:36] DEBUG[2904] 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 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 0 [Nov 4 19:32:36] DEBUG[2931] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 0 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.204:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0920f9e5 From: "180" ;tag=as6cca4433 To: "Erika Musterfrau" ;tag=7694b19ea5 Call-ID: 16199ba5d692a456 CSeq: 106 BYE Server: Aastra 53i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0920f9e5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 2 [ 55]: From: "180" ;tag=as6cca4433 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 3 [ 69]: To: "Erika Musterfrau" ;tag=7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 16199ba5d692a456 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 5 [ 13]: CSeq: 106 BYE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 6 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 4 19:32:36] VERBOSE[2885] chan_sip.c: --- (8 headers 0 lines) --- [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: = Looking for Call ID: 16199ba5d692a456 (Checking To) --From tag as6cca4433 --To-tag 7694b19ea5 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #276 [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Stopping retransmission on '16199ba5d692a456' of Request 106: Match Found [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:37] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2611663b From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Hans Muster" ;+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 3 IN IP4 192.168.10.205 s=SIP Call c=IN IP4 192.168.10.205 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 4 19:32:37] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2611663b [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 2 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 3 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 8 [133]: Contact: "Hans Muster" ;+sip.instance="" [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 0 [ 3]: v=0 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.205 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.205 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 4 19:32:37] VERBOSE[2885] chan_sip.c: --- (13 headers 12 lines) --- [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: = Looking for Call ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (Checking To) --From tag as33b155b8 --To-tag 3198768735 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Stopping retransmission on '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' of Request 105: Match Not Found [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Strict routing enforced for session 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:37] VERBOSE[2885] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 4 19:32:37] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:32:37] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:32:37] VERBOSE[2885] chan_sip.c: set_destination: set destination to 192.168.10.205:5060 [Nov 4 19:32:37] VERBOSE[2885] chan_sip.c: Transmitting (no NAT) to 192.168.10.205:5060: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1e299326 Max-Forwards: 70 From: "Max Mustermann" ;tag=as33b155b8 To: ;tag=3198768735 Contact: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.205:5060;transport=udp SIP/2.0 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1e299326 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 3 [ 61]: From: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 4 [ 65]: To: ;tag=3198768735 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 5 [ 37]: Contact: [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 6 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 7 [ 13]: CSeq: 105 ACK [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:37] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:38] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:38] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:39] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:39] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:40] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:40] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:41] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:41] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:42] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:42] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:43] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:43] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:44] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:44] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:45] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:45] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:46] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:46] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:47] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:47] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:48] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:48] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:49] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:49] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:50] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:50] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:51] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:51] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:52] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:52] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:53] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> REGISTER sip:192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK3395c73904461f2a9.abf8dbae7894f2de4 Max-Forwards: 70 From: ;tag=a6e2ad8dab To: Call-ID: 09a9c19478904dae CSeq: 9282 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 55i/2.6.0.66 Aastra-Line: 1 Content-Length: 0 <-------------> [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 0 [ 39]: REGISTER sip:192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK3395c73904461f2a9.abf8dbae7894f2de4 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 3 [ 52]: From: ;tag=a6e2ad8dab [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 4 [ 35]: To: [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: 09a9c19478904dae [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 6 [ 19]: CSeq: 9282 REGISTER [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 9 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 12 [ 14]: Aastra-Line: 1 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Nov 4 19:32:53] VERBOSE[2885] chan_sip.c: --- (14 headers 0 lines) --- [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: = Looking for Call ID: 09a9c19478904dae (Checking From) --From tag a6e2ad8dab --To-tag [Nov 4 19:32:53] DEBUG[2885] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Allocating new SIP dialog for 09a9c19478904dae - REGISTER (No RTP) [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Initializing initreq for method REGISTER - callid 09a9c19478904dae [Nov 4 19:32:53] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:53] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:53] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.203:5060 (no NAT) [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 4 19:32:53] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:32:53] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:32:53] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK3395c73904461f2a9.abf8dbae7894f2de4;received=192.168.10.203 From: ;tag=a6e2ad8dab To: ;tag=as04a2f4af Call-ID: 09a9c19478904dae CSeq: 9282 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: Thu, 04 Nov 2010 18:32:53 GMT Content-Length: 0 <------------> [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bK3395c73904461f2a9.abf8dbae7894f2de4;received=192.168.10.203 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 2 [ 52]: From: ;tag=a6e2ad8dab [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 3 [ 50]: To: ;tag=as04a2f4af [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 09a9c19478904dae [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 5 [ 19]: CSeq: 9282 REGISTER [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 11 [ 35]: Date: Thu, 04 Nov 2010 18:32:53 GMT [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:32:53] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog '09a9c19478904dae' in 32000 ms (Method: REGISTER) [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:53] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:53] DEBUG[2931] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone3 PeerStatus: Registered Address: 192.168.10.203:5060 [Nov 4 19:32:53] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:32:53] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:32:53] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 4 19:32:53] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '1' [Nov 4 19:32:53] DEBUG[2904] 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 4 19:32:54] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:54] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:55] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:55] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:56] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:56] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:57] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:57] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:58] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.204:5060 ---> REGISTER sip:192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKa613dbab70deb0853.cd762482fc4021e3c Max-Forwards: 70 From: ;tag=ff85eb4207 To: Call-ID: ae5660b1f5e6ab8f CSeq: 6875 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 Aastra-Line: 1 Content-Length: 0 <-------------> [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 0 [ 39]: REGISTER sip:192.168.10.70:5060 SIP/2.0 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKa613dbab70deb0853.cd762482fc4021e3c [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 3 [ 52]: From: ;tag=ff85eb4207 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 4 [ 35]: To: [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: ae5660b1f5e6ab8f [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 6 [ 19]: CSeq: 6875 REGISTER [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 9 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 53i/2.6.0.1008 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 12 [ 14]: Aastra-Line: 1 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Nov 4 19:32:58] VERBOSE[2885] chan_sip.c: --- (14 headers 0 lines) --- [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: = Looking for Call ID: ae5660b1f5e6ab8f (Checking From) --From tag ff85eb4207 --To-tag [Nov 4 19:32:58] DEBUG[2885] acl.c: For destination '192.168.10.204', our source address is '192.168.10.70'. [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Allocating new SIP dialog for ae5660b1f5e6ab8f - REGISTER (No RTP) [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Initializing initreq for method REGISTER - callid ae5660b1f5e6ab8f [Nov 4 19:32:58] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:58] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:58] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.204:5060 (no NAT) [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 4 19:32:58] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:32:58] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:32:58] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.204:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKa613dbab70deb0853.cd762482fc4021e3c;received=192.168.10.204 From: ;tag=ff85eb4207 To: ;tag=as407b0fff Call-ID: ae5660b1f5e6ab8f CSeq: 6875 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: Thu, 04 Nov 2010 18:32:58 GMT Content-Length: 0 <------------> [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKa613dbab70deb0853.cd762482fc4021e3c;received=192.168.10.204 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 2 [ 52]: From: ;tag=ff85eb4207 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 3 [ 50]: To: ;tag=as407b0fff [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: ae5660b1f5e6ab8f [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 5 [ 19]: CSeq: 6875 REGISTER [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 11 [ 35]: Date: Thu, 04 Nov 2010 18:32:58 GMT [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:32:58] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog 'ae5660b1f5e6ab8f' in 32000 ms (Method: REGISTER) [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:58] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:32:58] DEBUG[2931] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone1 PeerStatus: Registered Address: 192.168.10.204:5060 [Nov 4 19:32:58] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:32:58] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:32:58] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 4 19:32:58] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '1' [Nov 4 19:32:58] DEBUG[2904] 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 4 19:32:59] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:32:59] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:00] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:00] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:01] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:01] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:02] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:02] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:03] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:03] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:04] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:04] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:05] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:05] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:06] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:06] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:07] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:07] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Destroying SIP dialog 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:33:08] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog '62c2b7e359a754d65a4182b429640262@192.168.10.70' Method: BYE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Call to peer 'phone3' removed from call limit 2147483647 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: This call did not properly clean up call limits. Call ID 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for '62c2b7e359a754d65a4182b429640262@192.168.10.70' [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 001. NewChan Channel SIP/phone3-00000001 - from 62c2b7e359a754d65a4182b42964 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 005. TxReq ACK / 102 ACK - ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 006. ReInv Re-invite sent [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 007. TxReqRel INVITE / 103 INVITE - INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 008. Rx SIP/2.0 / 103 INVITE / 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 009. TxReq ACK / 103 ACK - ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 010. ReInv Re-invite sent [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 011. TxReqRel INVITE / 104 INVITE - INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 012. Rx SIP/2.0 / 104 INVITE / 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 013. TxReq ACK / 104 ACK - ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 014. Rx INVITE / 29393 INVITE / sip:100@192.168.10.70:5060 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 015. Hold INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 016. ReInv Re-invite received [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 017. TxResp SIP/2.0 / 29393 INVITE - 100 Trying [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 018. TxRespRel SIP/2.0 / 29393 INVITE - 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 019. Rx ACK / 29393 ACK / sip:100@192.168.10.70:5060 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 020. Rx REFER / 29394 REFER / sip:100@192.168.10.70:5060 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 021. TxResp SIP/2.0 / 29394 REFER - 202 Accepted [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 022. Xfer Refer accepted [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 023. TxReqRel NOTIFY / 105 NOTIFY - NOTIFY [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 024. Xfer Refer succeeded [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 025. SchedDestroy 32000 ms [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 026. Rx SIP/2.0 / 105 NOTIFY / 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 027. Rx BYE / 29395 BYE / sip:100@192.168.10.70:5060 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 028. RTCPaudio Quality:ssrc=41096550;themssrc=0;lp=0;rxjitter=0.000000;rxcount [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 029. RTCPaudioJitter Quality:minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 030. RTCPaudioLoss Quality:minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.00000 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 031. RTCPaudioRTT Quality:minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrt [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 032. CancelDestroy [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 033. SchedDestroy 32000 ms [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 034. TxResp SIP/2.0 / 29395 BYE - 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 035. AutoDestroy 62c2b7e359a754d65a4182b429640262@192.168.10.70 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for '62c2b7e359a754d65a4182b429640262@192.168.10.70' [Nov 4 19:33:08] DEBUG[2885] rtp_engine.c: Destroyed RTP instance '0xcfe79a8' [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:08] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:33:08] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:33:08] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 4 19:33:08] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '1' [Nov 4 19:33:08] DEBUG[2904] 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 4 19:33:08] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog '16199ba5d692a456' [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Destroying SIP dialog 16199ba5d692a456 [Nov 4 19:33:08] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog '16199ba5d692a456' Method: ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for '16199ba5d692a456' [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 001. Rx INVITE / 13990 INVITE / sip:180@192.168.10.70:5060 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 002. NewChan Channel SIP/phone1-00000000 - from 16199ba5d692a456 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 003. TxResp SIP/2.0 / 13990 INVITE - 100 Trying [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 004. TxResp SIP/2.0 / 13990 INVITE - 180 Ringing [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 005. ConnectedLine Called party is now Max Mustermann <180> [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 006. TxRespRel SIP/2.0 / 13990 INVITE - 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 007. Rx ACK / 13990 ACK / sip:180@192.168.10.70:5060 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 008. ReInv Re-invite sent [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 009. TxReqRel INVITE / 102 INVITE - INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 010. Rx SIP/2.0 / 102 INVITE / 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 011. TxReq ACK / 102 ACK - ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 012. ReInv Re-invite sent [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 013. TxReqRel INVITE / 103 INVITE - INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 014. Rx SIP/2.0 / 103 INVITE / 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 015. TxReq ACK / 103 ACK - ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 016. Masq Old channel: SIP/phone3-00000002 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 017. Masq (cont) ...new owner: SIP/phone1-00000000 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 018. ConnectedLine Calling party is now Hans Muster <150> [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 019. TxReqRel INVITE / 104 INVITE - INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 020. Rx SIP/2.0 / 104 INVITE / 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 021. TxReq ACK / 104 ACK - ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 022. ReInv Re-invite sent [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 023. TxReqRel INVITE / 105 INVITE - INVITE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 024. Hangup Cause Network out of order [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 025. SchedDestroy 32000 ms [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 026. CancelDestroy [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 027. Rx SIP/2.0 / 105 INVITE / 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 028. TxReq ACK / 105 ACK - ACK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 029. TxReqRel BYE / 106 BYE - BYE [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 030. SchedDestroy 32000 ms [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 031. Rx SIP/2.0 / 106 BYE / 200 OK [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 032. NeedDestroy Setting needdestroy because received 200 response [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: 033. AutoDestroy 16199ba5d692a456 [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for '16199ba5d692a456' [Nov 4 19:33:08] DEBUG[2885] rtp_engine.c: Destroyed RTP instance '0xcfed320' [Nov 4 19:33:08] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:09] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:10] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:11] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:12] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:13] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:14] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:15] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: INVITE [Nov 4 19:33:16] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> BYE sip:180@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bK46b51c3f8b8f28800.85af1f68fd704337a Max-Forwards: 70 From: ;tag=3198768735 To: "Max Mustermann" ;tag=as33b155b8 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 31074 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 4 19:33:16] DEBUG[2885] chan_sip.c: Header 0 [ 38]: BYE sip:180@192.168.10.70:5060 SIP/2.0 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bK46b51c3f8b8f28800.85af1f68fd704337a [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 3 [ 67]: From: ;tag=3198768735 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 4 [ 59]: To: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 5 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 6 [ 15]: CSeq: 31074 BYE [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 9 [ 28]: Supported: gruu, path, timer [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 10 [ 33]: User-Agent: Aastra 51i/2.6.0.1008 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 4 19:33:16] VERBOSE[2885] chan_sip.c: --- (12 headers 0 lines) --- [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: = Looking for Call ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 (Checking From) --From tag 3198768735 --To-tag as33b155b8 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Initializing initreq for method BYE - callid 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:33:16] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:33:16] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:33:16] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.205:5060 (no NAT) [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Setting SIP_ALREADYGONE on dialog 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:33:16] DEBUG[2885] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcfea2e8' [Nov 4 19:33:16] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' in 32000 ms (Method: BYE) [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Received bye, no owner, selfdestruct soon. [Nov 4 19:33:16] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bK46b51c3f8b8f28800.85af1f68fd704337a;received=192.168.10.205 From: ;tag=3198768735 To: "Max Mustermann" ;tag=as33b155b8 Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 CSeq: 31074 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 4 19:33:16] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bK46b51c3f8b8f28800.85af1f68fd704337a;received=192.168.10.205 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 2 [ 67]: From: ;tag=3198768735 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 3 [ 59]: To: "Max Mustermann" ;tag=as33b155b8 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 4 [ 55]: Call-ID: 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 5 [ 15]: CSeq: 31074 BYE [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Header 10 [ 0]: [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:33:16] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:17] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> REGISTER sip:192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bKa60ed7bc280128760.1289f172982e46b82 Max-Forwards: 70 From: ;tag=f24870da00 To: Call-ID: 8e82bf14271b4964 CSeq: 27612 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Hans Muster" ;+sip.instance="" Supported: gruu, path User-Agent: Aastra 51i/2.6.0.1008 Aastra-Line: 1 Content-Length: 0 <-------------> [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 0 [ 39]: REGISTER sip:192.168.10.70:5060 SIP/2.0 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bKa60ed7bc280128760.1289f172982e46b82 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 3 [ 52]: From: ;tag=f24870da00 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 4 [ 35]: To: [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: 8e82bf14271b4964 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 6 [ 20]: CSeq: 27612 REGISTER [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 9 [133]: Contact: "Hans Muster" ;+sip.instance="" [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 51i/2.6.0.1008 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 12 [ 14]: Aastra-Line: 1 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Nov 4 19:33:17] VERBOSE[2885] chan_sip.c: --- (14 headers 0 lines) --- [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: = Looking for Call ID: 8e82bf14271b4964 (Checking From) --From tag f24870da00 --To-tag [Nov 4 19:33:17] DEBUG[2885] acl.c: For destination '192.168.10.205', our source address is '192.168.10.70'. [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Allocating new SIP dialog for 8e82bf14271b4964 - REGISTER (No RTP) [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Initializing initreq for method REGISTER - callid 8e82bf14271b4964 [Nov 4 19:33:17] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:33:17] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:33:17] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.205:5060 (no NAT) [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 4 19:33:17] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:33:17] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:33:17] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bKa60ed7bc280128760.1289f172982e46b82;received=192.168.10.205 From: ;tag=f24870da00 To: ;tag=as6b7ee42c Call-ID: 8e82bf14271b4964 CSeq: 27612 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: Thu, 04 Nov 2010 18:33:17 GMT Content-Length: 0 <------------> [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bKa60ed7bc280128760.1289f172982e46b82;received=192.168.10.205 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 2 [ 52]: From: ;tag=f24870da00 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 3 [ 50]: To: ;tag=as6b7ee42c [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 8e82bf14271b4964 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 5 [ 20]: CSeq: 27612 REGISTER [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 11 [ 35]: Date: Thu, 04 Nov 2010 18:33:17 GMT [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:33:17] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog '8e82bf14271b4964' in 32000 ms (Method: REGISTER) [Nov 4 19:33:17] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:17] DEBUG[2931] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone2 PeerStatus: Registered Address: 192.168.10.205:5060 [Nov 4 19:33:17] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:33:17] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:33:17] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 4 19:33:17] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '1' [Nov 4 19:33:17] DEBUG[2904] 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 4 19:33:18] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:19] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:20] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:21] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:22] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:23] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:24] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog '09a9c19478904dae' [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: Destroying SIP dialog 09a9c19478904dae [Nov 4 19:33:25] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog '09a9c19478904dae' Method: REGISTER [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for '09a9c19478904dae' [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: 001. Rx REGISTER / 9282 REGISTER / sip:192.168.10.70:5060 [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: 002. TxResp SIP/2.0 / 9282 REGISTER - 200 OK [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: 005. AutoDestroy 09a9c19478904dae [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for '09a9c19478904dae' [Nov 4 19:33:25] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:26] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:28] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:29] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog 'ae5660b1f5e6ab8f' [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: Destroying SIP dialog ae5660b1f5e6ab8f [Nov 4 19:33:30] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog 'ae5660b1f5e6ab8f' Method: REGISTER [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for 'ae5660b1f5e6ab8f' [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: 001. Rx REGISTER / 6875 REGISTER / sip:192.168.10.70:5060 [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: 002. TxResp SIP/2.0 / 6875 REGISTER - 200 OK [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: 005. AutoDestroy ae5660b1f5e6ab8f [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for 'ae5660b1f5e6ab8f' [Nov 4 19:33:30] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:31] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:32] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:33] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:34] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:35] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:36] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:37] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:38] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:39] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:40] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:41] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:42] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:43] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:44] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:45] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:46] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:47] DEBUG[2885] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: Destroying SIP dialog 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:33:48] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' Method: BYE [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 001. NewChan Channel SIP/phone2-00000003 - from 6987ee01797fdbf6241da1440ab8 [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 200 OK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 005. TxReq ACK / 102 ACK - ACK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 006. ReInv Re-invite sent [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 007. TxReqRel INVITE / 103 INVITE - INVITE [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 008. Rx SIP/2.0 / 103 INVITE / 200 OK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 009. TxReq ACK / 103 ACK - ACK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 010. ReInv Re-invite sent [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 011. TxReqRel INVITE / 104 INVITE - INVITE [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 012. Rx SIP/2.0 / 104 INVITE / 200 OK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 013. TxReq ACK / 104 ACK - ACK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 014. ReInv Re-invite sent [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 015. TxReqRel INVITE / 105 INVITE - INVITE [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 016. ConnectedLine Calling party is now Erika Musterfrau <100> [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 017. TxReqRel INVITE / 106 INVITE - INVITE [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 018. Rx SIP/2.0 / 106 INVITE / 500 Internal Server Error [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 019. TxReq ACK / 106 ACK - ACK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 020. Hangup Cause Network out of order [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 021. NeedDestroy Setting needdestroy because hangup [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 022. Rx SIP/2.0 / 105 INVITE / 100 Trying [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 023. Rx SIP/2.0 / 105 INVITE / 200 OK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 024. Rx SIP/2.0 / 105 INVITE / 200 OK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 025. TxReq ACK / 105 ACK - ACK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 026. Ignore Ignoring this retransmit [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 027. Rx BYE / 31074 BYE / sip:180@192.168.10.70:5060 [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 028. RTCPaudio Quality:ssrc=1928928681;themssrc=0;lp=0;rxjitter=0.000000;rxcou [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 029. RTCPaudioJitter Quality:minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0 [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 030. RTCPaudioLoss Quality:minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.00000 [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 031. RTCPaudioRTT Quality:minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrt [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 032. SchedDestroy 32000 ms [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 033. TxResp SIP/2.0 / 31074 BYE - 200 OK [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: 034. AutoDestroy 6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70 [Nov 4 19:33:48] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for '6987ee01797fdbf6241da1440ab8a7c1@192.168.10.70' [Nov 4 19:33:48] DEBUG[2885] rtp_engine.c: Destroyed RTP instance '0xcfea2e8' [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog '8e82bf14271b4964' [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: Destroying SIP dialog 8e82bf14271b4964 [Nov 4 19:33:49] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog '8e82bf14271b4964' Method: REGISTER [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for '8e82bf14271b4964' [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: 001. Rx REGISTER / 27612 REGISTER / sip:192.168.10.70:5060 [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: 002. TxResp SIP/2.0 / 27612 REGISTER - 200 OK [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: 005. AutoDestroy 8e82bf14271b4964 [Nov 4 19:33:49] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for '8e82bf14271b4964' [Nov 4 19:34:39] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.203:5060 ---> REGISTER sip:192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKe3d542ea306751be1.295f49e98b1b8b7d1 Max-Forwards: 70 From: ;tag=a6e2ad8dab To: Call-ID: 09a9c19478904dae CSeq: 9283 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 55i/2.6.0.66 Aastra-Line: 1 Content-Length: 0 <-------------> [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 0 [ 39]: REGISTER sip:192.168.10.70:5060 SIP/2.0 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKe3d542ea306751be1.295f49e98b1b8b7d1 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 3 [ 52]: From: ;tag=a6e2ad8dab [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 4 [ 35]: To: [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: 09a9c19478904dae [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 6 [ 19]: CSeq: 9283 REGISTER [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 9 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 12 [ 14]: Aastra-Line: 1 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Nov 4 19:34:39] VERBOSE[2885] chan_sip.c: --- (14 headers 0 lines) --- [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: = Looking for Call ID: 09a9c19478904dae (Checking From) --From tag a6e2ad8dab --To-tag [Nov 4 19:34:39] DEBUG[2885] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Allocating new SIP dialog for 09a9c19478904dae - REGISTER (No RTP) [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Initializing initreq for method REGISTER - callid 09a9c19478904dae [Nov 4 19:34:39] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:34:39] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:34:39] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.203:5060 (no NAT) [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 4 19:34:39] DEBUG[2885] netsock2.c: Splitting '192.168.10.203:5060' gives... [Nov 4 19:34:39] DEBUG[2885] netsock2.c: ...host '192.168.10.203' and port '5060'. [Nov 4 19:34:39] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKe3d542ea306751be1.295f49e98b1b8b7d1;received=192.168.10.203 From: ;tag=a6e2ad8dab To: ;tag=as72a46d1e Call-ID: 09a9c19478904dae CSeq: 9283 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: Thu, 04 Nov 2010 18:34:39 GMT Content-Length: 0 <------------> [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.203:5060;branch=z9hG4bKe3d542ea306751be1.295f49e98b1b8b7d1;received=192.168.10.203 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 2 [ 52]: From: ;tag=a6e2ad8dab [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 3 [ 50]: To: ;tag=as72a46d1e [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 09a9c19478904dae [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 5 [ 19]: CSeq: 9283 REGISTER [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 11 [ 35]: Date: Thu, 04 Nov 2010 18:34:39 GMT [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:34:39] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:5060 [Nov 4 19:34:39] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog '09a9c19478904dae' in 32000 ms (Method: REGISTER) [Nov 4 19:34:39] DEBUG[2931] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone3 PeerStatus: Registered Address: 192.168.10.203:5060 [Nov 4 19:34:39] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 4 19:34:39] DEBUG[2878] chan_sip.c: Checking device state for peer phone3 [Nov 4 19:34:39] DEBUG[2878] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 4 19:34:39] DEBUG[2878] devicestate.c: device 'SIP/phone3' state '1' [Nov 4 19:34:39] DEBUG[2904] 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 4 19:34:43] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.204:5060 ---> REGISTER sip:192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK8a92c08057b4a0dc0.8fffaf06ed6ea28bd Max-Forwards: 70 From: ;tag=ff85eb4207 To: Call-ID: ae5660b1f5e6ab8f CSeq: 6876 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 Aastra-Line: 1 Content-Length: 0 <-------------> [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 0 [ 39]: REGISTER sip:192.168.10.70:5060 SIP/2.0 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK8a92c08057b4a0dc0.8fffaf06ed6ea28bd [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 3 [ 52]: From: ;tag=ff85eb4207 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 4 [ 35]: To: [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: ae5660b1f5e6ab8f [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 6 [ 19]: CSeq: 6876 REGISTER [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 9 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 53i/2.6.0.1008 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 12 [ 14]: Aastra-Line: 1 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Nov 4 19:34:43] VERBOSE[2885] chan_sip.c: --- (14 headers 0 lines) --- [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: = Looking for Call ID: ae5660b1f5e6ab8f (Checking From) --From tag ff85eb4207 --To-tag [Nov 4 19:34:43] DEBUG[2885] acl.c: For destination '192.168.10.204', our source address is '192.168.10.70'. [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Allocating new SIP dialog for ae5660b1f5e6ab8f - REGISTER (No RTP) [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Initializing initreq for method REGISTER - callid ae5660b1f5e6ab8f [Nov 4 19:34:43] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:34:43] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:34:43] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.204:5060 (no NAT) [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 4 19:34:43] DEBUG[2885] netsock2.c: Splitting '192.168.10.204:5060' gives... [Nov 4 19:34:43] DEBUG[2885] netsock2.c: ...host '192.168.10.204' and port '5060'. [Nov 4 19:34:43] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.204:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK8a92c08057b4a0dc0.8fffaf06ed6ea28bd;received=192.168.10.204 From: ;tag=ff85eb4207 To: ;tag=as46c4d84f Call-ID: ae5660b1f5e6ab8f CSeq: 6876 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: Thu, 04 Nov 2010 18:34:43 GMT Content-Length: 0 <------------> [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK8a92c08057b4a0dc0.8fffaf06ed6ea28bd;received=192.168.10.204 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 2 [ 52]: From: ;tag=ff85eb4207 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 3 [ 50]: To: ;tag=as46c4d84f [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: ae5660b1f5e6ab8f [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 5 [ 19]: CSeq: 6876 REGISTER [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 11 [ 35]: Date: Thu, 04 Nov 2010 18:34:43 GMT [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:34:43] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.204:5060 [Nov 4 19:34:43] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog 'ae5660b1f5e6ab8f' in 32000 ms (Method: REGISTER) [Nov 4 19:34:43] DEBUG[2931] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone1 PeerStatus: Registered Address: 192.168.10.204:5060 [Nov 4 19:34:43] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 4 19:34:43] DEBUG[2878] chan_sip.c: Checking device state for peer phone1 [Nov 4 19:34:43] DEBUG[2878] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 4 19:34:43] DEBUG[2878] devicestate.c: device 'SIP/phone1' state '1' [Nov 4 19:34:43] DEBUG[2904] 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 4 19:35:02] VERBOSE[2885] chan_sip.c: <--- SIP read from UDP:192.168.10.205:5060 ---> REGISTER sip:192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bK4cbadfe99181d6a2b.d290c165262a2ba90 Max-Forwards: 70 From: ;tag=f24870da00 To: Call-ID: 8e82bf14271b4964 CSeq: 27613 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Hans Muster" ;+sip.instance="" Supported: gruu, path User-Agent: Aastra 51i/2.6.0.1008 Aastra-Line: 1 Content-Length: 0 <-------------> [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 0 [ 39]: REGISTER sip:192.168.10.70:5060 SIP/2.0 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bK4cbadfe99181d6a2b.d290c165262a2ba90 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 3 [ 52]: From: ;tag=f24870da00 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 4 [ 35]: To: [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 5 [ 25]: Call-ID: 8e82bf14271b4964 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 6 [ 20]: CSeq: 27613 REGISTER [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 9 [133]: Contact: "Hans Muster" ;+sip.instance="" [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 51i/2.6.0.1008 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 12 [ 14]: Aastra-Line: 1 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Nov 4 19:35:02] VERBOSE[2885] chan_sip.c: --- (14 headers 0 lines) --- [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: = Looking for Call ID: 8e82bf14271b4964 (Checking From) --From tag f24870da00 --To-tag [Nov 4 19:35:02] DEBUG[2885] acl.c: For destination '192.168.10.205', our source address is '192.168.10.70'. [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Allocating new SIP dialog for 8e82bf14271b4964 - REGISTER (No RTP) [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Initializing initreq for method REGISTER - callid 8e82bf14271b4964 [Nov 4 19:35:02] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:35:02] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:35:02] VERBOSE[2885] chan_sip.c: Sending to 192.168.10.205:5060 (no NAT) [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 4 19:35:02] DEBUG[2885] netsock2.c: Splitting '192.168.10.205:5060' gives... [Nov 4 19:35:02] DEBUG[2885] netsock2.c: ...host '192.168.10.205' and port '5060'. [Nov 4 19:35:02] VERBOSE[2885] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bK4cbadfe99181d6a2b.d290c165262a2ba90;received=192.168.10.205 From: ;tag=f24870da00 To: ;tag=as37548642 Call-ID: 8e82bf14271b4964 CSeq: 27613 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: Thu, 04 Nov 2010 18:35:02 GMT Content-Length: 0 <------------> [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.205:5060;branch=z9hG4bK4cbadfe99181d6a2b.d290c165262a2ba90;received=192.168.10.205 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 2 [ 52]: From: ;tag=f24870da00 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 3 [ 50]: To: ;tag=as37548642 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 4 [ 25]: Call-ID: 8e82bf14271b4964 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 5 [ 20]: CSeq: 27613 REGISTER [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 11 [ 35]: Date: Thu, 04 Nov 2010 18:35:02 GMT [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Header 13 [ 0]: [Nov 4 19:35:02] DEBUG[2885] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.205:5060 [Nov 4 19:35:02] VERBOSE[2885] chan_sip.c: Scheduling destruction of SIP dialog '8e82bf14271b4964' in 32000 ms (Method: REGISTER) [Nov 4 19:35:02] DEBUG[2931] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone2 PeerStatus: Registered Address: 192.168.10.205:5060 [Nov 4 19:35:02] DEBUG[2878] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 4 19:35:02] DEBUG[2878] chan_sip.c: Checking device state for peer phone2 [Nov 4 19:35:02] DEBUG[2878] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 4 19:35:02] DEBUG[2878] devicestate.c: device 'SIP/phone2' state '1' [Nov 4 19:35:02] DEBUG[2904] 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 4 19:35:11] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog '09a9c19478904dae' [Nov 4 19:35:11] DEBUG[2885] chan_sip.c: Destroying SIP dialog 09a9c19478904dae [Nov 4 19:35:11] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog '09a9c19478904dae' Method: REGISTER [Nov 4 19:35:11] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for '09a9c19478904dae' [Nov 4 19:35:11] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:35:11] DEBUG[2885] chan_sip.c: 001. Rx REGISTER / 9283 REGISTER / sip:192.168.10.70:5060 [Nov 4 19:35:11] DEBUG[2885] chan_sip.c: 002. TxResp SIP/2.0 / 9283 REGISTER - 200 OK [Nov 4 19:35:11] DEBUG[2885] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 4 19:35:11] DEBUG[2885] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 4 19:35:11] DEBUG[2885] chan_sip.c: 005. AutoDestroy 09a9c19478904dae [Nov 4 19:35:11] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for '09a9c19478904dae' [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog 'ae5660b1f5e6ab8f' [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: Destroying SIP dialog ae5660b1f5e6ab8f [Nov 4 19:35:15] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog 'ae5660b1f5e6ab8f' Method: REGISTER [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for 'ae5660b1f5e6ab8f' [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: 001. Rx REGISTER / 6876 REGISTER / sip:192.168.10.70:5060 [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: 002. TxResp SIP/2.0 / 6876 REGISTER - 200 OK [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: 005. AutoDestroy ae5660b1f5e6ab8f [Nov 4 19:35:15] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for 'ae5660b1f5e6ab8f' [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: Auto destroying SIP dialog '8e82bf14271b4964' [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: Destroying SIP dialog 8e82bf14271b4964 [Nov 4 19:35:34] VERBOSE[2885] chan_sip.c: Really destroying SIP dialog '8e82bf14271b4964' Method: REGISTER [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: ---------- SIP HISTORY for '8e82bf14271b4964' [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: * SIP Call [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: 001. Rx REGISTER / 27613 REGISTER / sip:192.168.10.70:5060 [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: 002. TxResp SIP/2.0 / 27613 REGISTER - 200 OK [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: 005. AutoDestroy 8e82bf14271b4964 [Nov 4 19:35:34] DEBUG[2885] chan_sip.c: ---------- END SIP HISTORY for '8e82bf14271b4964'