[Nov 22 11:57:21] VERBOSE[1173] config.c: == Parsing '/etc/asterisk/logger.conf': [Nov 22 11:57:21] DEBUG[1173] config.c: Parsing /etc/asterisk/logger.conf [Nov 22 11:57:21] VERBOSE[1173] config.c: == Found [Nov 22 11:57:21] VERBOSE[1173] logger.c: Asterisk Queue Logger restarted [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> INVITE sip:150@192.168.10.70 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK580e2f25c1d8b0320.6c1622570ffd6b6aa Max-Forwards: 70 From: "Erika Musterfrau" ;tag=bdabd584b8 To: "150" Call-ID: 8bd610aef94fb8be CSeq: 4433 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erika Musterfrau" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 53i/2.6.0.1008 Content-Type: application/sdp Content-Length: 620 v=0 o=MxSIP 0 0 IN IP4 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 0 [ 36]: INVITE sip:150@192.168.10.70 SIP/2.0 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK580e2f25c1d8b0320.6c1622570ffd6b6aa [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 3 [ 66]: From: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 4 [ 33]: To: "150" [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 6 [ 17]: CSeq: 4433 INVITE [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 9 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 53i/2.6.0.1008 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 13 [ 19]: Content-Length: 620 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 14 [ 0]: [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.201 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 5 [ 70]: m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 21 [ 24]: a=silenceSupp:on - - - - [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 22 [ 20]: a=fmtp:18 annexb=yes [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Body 25 [ 10]: a=sendrecv [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: --- (14 headers 26 lines) --- [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: = Looking for Call ID: 8bd610aef94fb8be (Checking From) --From tag bdabd584b8 --To-tag [Nov 22 11:57:30] DEBUG[1175] acl.c: For destination '192.168.10.201', our source address is '192.168.10.70'. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Allocating new SIP dialog for 8bd610aef94fb8be - INVITE (No RTP) [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -gruu- [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: gruu [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -path- [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: path [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: timer [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 22 11:57:30] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 22 11:57:30] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:30] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.201:5060 (no NAT) [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Initializing initreq for method INVITE - callid 8bd610aef94fb8be [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Using INVITE request as basis request - 8bd610aef94fb8be [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found peer 'phone1' for 'phone1' from 192.168.10.201:5060 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd829cd8' [Nov 22 11:57:30] DEBUG[1175] res_rtp_asterisk.c: Allocated port 12072 for RTP instance '0xd829cd8' [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: RTP instance '0xd829cd8' is setup and ready to go [Nov 22 11:57:30] DEBUG[1175] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd829cd8' [Nov 22 11:57:30] VERBOSE[1175] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 22 11:57:30] VERBOSE[1175] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Setting NAT on RTP to Off [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.201... UNSUPPORTED. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:30] DEBUG[1175] netsock2.c: Splitting '192.168.10.201' gives... [Nov 22 11:57:30] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '(null)'. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 18 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 18 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 106 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 106 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 107 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 113 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 110 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 110 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 111 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 111 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 112 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 112 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 98 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 98 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 97 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 97 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 115 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 115 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 96 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 9 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 9 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format G729 for ID 18 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format BV16 for ID 106 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format BV32 for ID 107 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 113 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 110 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 111 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 112 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format G726-16 for ID 98 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format G726-24 for ID 97 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format G726-32 for ID 115 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format G726-40 for ID 96 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format G722 for ID 9 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:on - - - -... UNSUPPORTED. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 9 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 18 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 97 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 98 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 106 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 110 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 111 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 112 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 113 on 0xb3380f78 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Incorporating payload 115 on 0xb3380f78 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8109f4c (ulaw|alaw|g726|slin|g729|speex|ilbc|g722|slin16|h263p|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:30] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd829cd8' [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 9 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 18 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 97 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 98 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 106 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 110 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 111 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 112 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 113 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] rtp_engine.c: Copying payload 115 from 0xb3380f78 to 0xd829e84 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Checking SIP call limits for device phone1 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Call from peer 'phone1' is 1 out of 2147483647 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: Looking for 150 in Standard (domain 192.168.10.70) [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: This channel will not be able to handle video. [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: build_route: Contact hop: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: list_route: hop: [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: SIP/phone1-00000010: New call is still down.... Trying... [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK580e2f25c1d8b0320.6c1622570ffd6b6aa;received=192.168.10.201 From: "Erika Musterfrau" ;tag=bdabd584b8 To: "150" Call-ID: 8bd610aef94fb8be CSeq: 4433 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK580e2f25c1d8b0320.6c1622570ffd6b6aa;received=192.168.10.201 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 2 [ 66]: From: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 3 [ 33]: To: "150" [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 5 [ 17]: CSeq: 4433 INVITE [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:30] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:57:30] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:57:30] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:57:30] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:57:30] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:57:30] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:57:30] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:57:30] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:57:30] DEBUG[1195] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone1-00000010 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Erika Musterfrau AccountCode: Exten: 150 Context: Standard Uniqueid: 1290423450.16 [Nov 22 11:57:30] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-00000010 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1290423450.16 [Nov 22 11:57:30] DEBUG[1252] pbx.c: Launching 'Dial' [Nov 22 11:57:30] VERBOSE[1252] pbx.c: -- Executing [150@Standard:1] Dial("SIP/phone1-00000010", "SIP/phone2") in new stack [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Allocating new SIP dialog for 3b0ccf311617b7597141c55456d02509@192.168.10.70 - INVITE (No RTP) [Nov 22 11:57:30] DEBUG[1252] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd809e70' [Nov 22 11:57:30] DEBUG[1252] res_rtp_asterisk.c: Allocated port 14344 for RTP instance '0xd809e70' [Nov 22 11:57:30] DEBUG[1252] rtp_engine.c: RTP instance '0xd809e70' is setup and ready to go [Nov 22 11:57:30] DEBUG[1252] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd809e70' [Nov 22 11:57:30] VERBOSE[1252] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 22 11:57:30] VERBOSE[1252] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Setting NAT on RTP to Off [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 22 11:57:30] DEBUG[1252] acl.c: For destination '192.168.10.200', our source address is '192.168.10.70'. [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: This channel will not be able to handle video. [Nov 22 11:57:30] DEBUG[1252] rtp_engine.c: Seeded SDP of 'SIP/phone2-00000011' with that of 'SIP/phone1-00000010' [Nov 22 11:57:30] DEBUG[1252] channel.c: Not copying variable DIALEDTIME. [Nov 22 11:57:30] DEBUG[1252] channel.c: Not copying variable ANSWEREDTIME. [Nov 22 11:57:30] DEBUG[1252] channel.c: Not copying variable DIALEDPEERNAME. [Nov 22 11:57:30] DEBUG[1252] channel.c: Not copying variable DIALEDPEERNUMBER. [Nov 22 11:57:30] DEBUG[1252] channel.c: Not copying variable DIALSTATUS. [Nov 22 11:57:30] DEBUG[1252] channel.c: Not copying variable SIPCALLID. [Nov 22 11:57:30] DEBUG[1252] channel.c: Not copying variable SIPDOMAIN. [Nov 22 11:57:30] DEBUG[1252] channel.c: Not copying variable SIPURI. [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Outgoing Call for phone2 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Call to peer 'phone2' is 1 out of 2147483647 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:57:30] VERBOSE[1252] chan_sip.c: Audio is at 5060 [Nov 22 11:57:30] VERBOSE[1252] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:30] VERBOSE[1252] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:30] VERBOSE[1252] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Initializing initreq for method INVITE - callid 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4efc38af [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 4 [ 50]: To: [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 9 [ 35]: Date: Mon, 22 Nov 2010 10:57:30 GMT [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:30] VERBOSE[1252] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.200:5060: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4efc38af Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as45ace4d4 To: Contact: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0-1 Date: Mon, 22 Nov 2010 10:57:30 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: 262 v=0 o=root 1496546433 1496546433 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 14344 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4efc38af [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 4 [ 50]: To: [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 9 [ 35]: Date: Mon, 22 Nov 2010 10:57:30 GMT [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 14 [ 19]: Content-Length: 262 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 1 [ 49]: o=root 1496546433 1496546433 IN IP4 192.168.10.70 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 5 [ 29]: m=audio 14344 RTP/AVP 8 0 101 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #384 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:30] VERBOSE[1252] app_dial.c: -- Called phone2 [Nov 22 11:57:30] DEBUG[1195] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-00000011 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 150 CallerIDName: Hans Muster AccountCode: Exten: Context: Standard Uniqueid: 1290423450.17 [Nov 22 11:57:30] DEBUG[1195] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone1-00000010 Destination: SIP/phone2-00000011 CallerIDNum: 100 CallerIDName: Erika Musterfrau UniqueID: 1290423450.16 DestUniqueID: 1290423450.17 Dialstring: phone2 [Nov 22 11:57:30] DEBUG[1195] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone2-00000011 CallerIDNum: 150 CallerIDName: Uniqueid: 1290423450.17 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Nov 22 11:57:30] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:30] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:30] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Nov 22 11:57:30] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '6' [Nov 22 11:57:30] DEBUG[1168] app_queue.c: Extension '100@Standard' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:30] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:30] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:30] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 1 [Nov 22 11:57:30] DEBUG[1168] app_queue.c: Extension '150@Standard' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:57:30] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:57:30] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 8 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4efc38af From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Server: Aastra 55i/2.6.0.66 Supported: gruu, path Content-Length: 0 <-------------> [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4efc38af [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 8 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 22 11:57:30] VERBOSE[1175] chan_sip.c: --- (12 headers 0 lines) --- [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: = Looking for Call ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (Checking To) --From tag as45ace4d4 --To-tag 1768984948 [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: *** SIP TIMER: Cancelling retransmission #384 - INVITE (got response) [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Request 102: Found [Nov 22 11:57:30] DEBUG[1175] chan_sip.c: SIP response 180 to standard invite [Nov 22 11:57:30] VERBOSE[1252] app_dial.c: -- SIP/phone2-00000011 is ringing [Nov 22 11:57:30] DEBUG[1252] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000010' with that of 'SIP/phone2-00000011' [Nov 22 11:57:30] VERBOSE[1252] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.201:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK580e2f25c1d8b0320.6c1622570ffd6b6aa;received=192.168.10.201 From: "Erika Musterfrau" ;tag=bdabd584b8 To: "150" ;tag=as37527f23 Call-ID: 8bd610aef94fb8be CSeq: 4433 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK580e2f25c1d8b0320.6c1622570ffd6b6aa;received=192.168.10.201 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 2 [ 66]: From: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 3 [ 48]: To: "150" ;tag=as37527f23 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 4 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 5 [ 17]: CSeq: 4433 INVITE [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Header 11 [ 0]: [Nov 22 11:57:30] DEBUG[1252] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:30] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:30] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:30] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Nov 22 11:57:30] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '6' [Nov 22 11:57:30] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000011 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 150 CallerIDName: Uniqueid: 1290423450.17 [Nov 22 11:57:30] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4efc38af From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Server: Aastra 55i/2.6.0.66 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 0 IN IP4 192.168.10.200 s=SIP Call c=IN IP4 192.168.10.200 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4efc38af [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 8 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.200 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: = Looking for Call ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (Checking To) --From tag as45ace4d4 --To-tag 1768984948 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Acked pending invite 102 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Stopping retransmission on '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' of Request 102: Match Found [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: SIP response 200 to standard invite [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.200... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.200' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '(null)'. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.200... OK. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:32] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd809e70' [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.200:3000 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd80a01c [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd80a01c [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd80a01c [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: build_route: Contact hop: "" ;+sip.instance="" [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: list_route: hop: [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Strict routing enforced for session 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.200:5060: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK66a596bc Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Contact: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK66a596bc [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:32] VERBOSE[1252] app_dial.c: -- SIP/phone2-00000011 answered SIP/phone1-00000010 [Nov 22 11:57:32] DEBUG[1252] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000010' with that of 'SIP/phone2-00000011' [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: SIP answering channel: SIP/phone1-00000010 [Nov 22 11:57:32] DEBUG[1252] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Setting framing from config on incoming call [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Audio is at 5060 [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK580e2f25c1d8b0320.6c1622570ffd6b6aa;received=192.168.10.201 From: "Erika Musterfrau" ;tag=bdabd584b8 To: "150" ;tag=as37527f23 Call-ID: 8bd610aef94fb8be CSeq: 4433 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 1247526371 1247526371 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 12072 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bK580e2f25c1d8b0320.6c1622570ffd6b6aa;received=192.168.10.201 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 2 [ 66]: From: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 3 [ 48]: To: "150" ;tag=as37527f23 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 4 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 5 [ 17]: CSeq: 4433 INVITE [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 10 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 12 [ 19]: Content-Length: 262 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 1 [ 49]: o=root 1247526371 1247526371 IN IP4 192.168.10.70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 5 [ 29]: m=audio 12072 RTP/AVP 8 0 101 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #387 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:32] DEBUG[1252] features.c: bridge answer set, chan answer set [Nov 22 11:57:32] DEBUG[1252] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:57:32] DEBUG[1252] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:57:32] VERBOSE[1252] rtp_engine.c: -- Remotely bridging SIP/phone1-00000010 and SIP/phone2-00000011 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Deferring reinvite on SIP '8bd610aef94fb8be' - It's audio will be redirected to IP 192.168.10.200:3000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Sending reinvite on SIP '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' - It's audio soon redirected to IP 192.168.10.201:3000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Strict routing enforced for session 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:32] DEBUG[1252] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:32] DEBUG[1252] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Audio is at 5060 [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Initializing already initialized SIP dialog 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (presumably reinvite) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK79c14137 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 4 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.200:5060: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK79c14137 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Contact: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@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: 263 v=0 o=root 1496546433 1496546434 IN IP4 192.168.10.201 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK79c14137 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 4 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 14 [ 19]: Content-Length: 263 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 1 [ 50]: o=root 1496546433 1496546434 IN IP4 192.168.10.201 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #388 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:32] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:32] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:32] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:57:32] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:57:32] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:32] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:32] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:57:32] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:57:32] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:57:32] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:57:32] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:57:32] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:57:32] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000011 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 150 CallerIDName: Uniqueid: 1290423450.17 [Nov 22 11:57:32] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-00000010 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1290423450.16 [Nov 22 11:57:32] DEBUG[1195] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone2-00000011 Uniqueid: 1290423450.17 AccountCode: OldAccountCode: [Nov 22 11:57:32] DEBUG[1195] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone1-00000010 Channel2: SIP/phone2-00000011 Uniqueid1: 1290423450.16 Uniqueid2: 1290423450.17 CallerID1: 100 CallerID2: 150 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: INVITE [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:32] DEBUG[1168] app_queue.c: Extension '150@Standard' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:32] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:32] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:32] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:32] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 1 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK79c14137 From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@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: "" ;+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 192.168.10.200 s=SIP Call c=IN IP4 192.168.10.200 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK79c14137 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 8 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.200 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: = Looking for Call ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (Checking To) --From tag as45ace4d4 --To-tag 1768984948 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Acked pending invite 103 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #388 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Stopping retransmission on '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' of Request 103: Match Found [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.200... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.200' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '(null)'. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.200... OK. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:32] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd809e70' [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.200:3000 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd80a01c [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd80a01c [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd80a01c [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Strict routing enforced for session 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.200:5060: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2f0e8d3c Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Contact: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2f0e8d3c [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: INVITE [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:32] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:32] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:32] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:57:32] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:57:32] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> ACK sip:150@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bKf7d31dd40e2f326aa.04d75523e645928a3 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=bdabd584b8 To: "150" ;tag=as37527f23 Call-ID: 8bd610aef94fb8be CSeq: 4433 ACK User-Agent: Aastra 53i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 0 [ 38]: ACK sip:150@192.168.10.70:5060 SIP/2.0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.201:5060;branch=z9hG4bKf7d31dd40e2f326aa.04d75523e645928a3 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 3 [ 66]: From: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "150" ;tag=as37527f23 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 6 [ 14]: CSeq: 4433 ACK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 7 [ 33]: User-Agent: Aastra 53i/2.6.0.1008 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: --- (9 headers 0 lines) --- [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: = Looking for Call ID: 8bd610aef94fb8be (Checking From) --From tag bdabd584b8 --To-tag as37527f23 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #387 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Stopping retransmission on '8bd610aef94fb8be' of Response 4433: Match Found [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Sending pending reinvite on '8bd610aef94fb8be' [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Strict routing enforced for session 8bd610aef94fb8be [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Audio is at 5060 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Initializing already initialized SIP dialog 8bd610aef94fb8be (presumably reinvite) [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK071a3191 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK071a3191 Max-Forwards: 70 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Contact: Call-ID: 8bd610aef94fb8be 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 1247526371 1247526372 IN IP4 192.168.10.200 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.200 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK071a3191 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 14 [ 19]: Content-Length: 263 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 1 [ 50]: o=root 1247526371 1247526372 IN IP4 192.168.10.200 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #389 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK071a3191 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Call-ID: 8bd610aef94fb8be CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erika Musterfrau" ;+sip.instance="" Server: Aastra 53i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 1 IN IP4 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK071a3191 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 8 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.201 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: = Looking for Call ID: 8bd610aef94fb8be (Checking To) --From tag as37527f23 --To-tag bdabd584b8 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Acked pending invite 102 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #389 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Stopping retransmission on '8bd610aef94fb8be' of Request 102: Match Found [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 8bd610aef94fb8be [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.201... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.201' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '(null)'. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:32] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd829cd8' [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:32] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Strict routing enforced for session 8bd610aef94fb8be [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:32] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:32] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:57:32] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.201:5060: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK63e78cc0 Max-Forwards: 70 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Contact: Call-ID: 8bd610aef94fb8be CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK63e78cc0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:32] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:57:32] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:57:32] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:57:32] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:57:32] DEBUG[1252] rtp_engine.c: Oooh, 'SIP/phone1-00000010' changed end address to 192.168.10.201:3000 (format unknown) [Nov 22 11:57:32] DEBUG[1252] rtp_engine.c: Oooh, 'SIP/phone1-00000010' was 192.168.10.201:3000/(format unknown) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Sending reinvite on SIP '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' - It's audio soon redirected to IP 192.168.10.201:3000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Strict routing enforced for session 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:32] DEBUG[1252] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:32] DEBUG[1252] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Audio is at 5060 [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Initializing already initialized SIP dialog 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (presumably reinvite) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0cb5c941 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 4 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] VERBOSE[1252] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.200:5060: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0cb5c941 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Contact: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@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: 263 v=0 o=root 1496546433 1496546435 IN IP4 192.168.10.201 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0cb5c941 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 4 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 12 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 14 [ 19]: Content-Length: 263 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 1 [ 50]: o=root 1496546433 1496546435 IN IP4 192.168.10.201 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #390 [Nov 22 11:57:32] DEBUG[1252] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:32] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:32] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0cb5c941 From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@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: "" ;+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 192.168.10.200 s=SIP Call c=IN IP4 192.168.10.200 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0cb5c941 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 8 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 2 IN IP4 192.168.10.200 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: = Looking for Call ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (Checking To) --From tag as45ace4d4 --To-tag 1768984948 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Acked pending invite 104 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #390 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Stopping retransmission on '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' of Request 104: Match Found [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.10.200... UNSUPPORTED. [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:33] DEBUG[1175] netsock2.c: Splitting '192.168.10.200' gives... [Nov 22 11:57:33] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '(null)'. [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.200... OK. [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:33] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:33] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:33] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:33] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:33] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:33] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:33] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd809e70' [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.200:3000 [Nov 22 11:57:33] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd80a01c [Nov 22 11:57:33] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd80a01c [Nov 22 11:57:33] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd80a01c [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:33] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:33] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Strict routing enforced for session 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:33] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:33] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.200:5060: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK12aed5ee Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Contact: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK12aed5ee [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 104 ACK [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:33] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:33] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:33] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:57:33] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:57:33] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> REGISTER sip:192.168.10.70 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK9ba124d10a6d27901.9a51108da54cd8541 Max-Forwards: 70 From: ;tag=ef592dfd6b To: Call-ID: bddd1f752853e839 CSeq: 9010 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="" Supported: gruu, path User-Agent: Aastra 51i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 0 [ 34]: REGISTER sip:192.168.10.70 SIP/2.0 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK9ba124d10a6d27901.9a51108da54cd8541 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 3 [ 47]: From: ;tag=ef592dfd6b [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 4 [ 30]: To: [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: bddd1f752853e839 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 6 [ 19]: CSeq: 9010 REGISTER [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 9 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 11 [ 33]: User-Agent: Aastra 51i/2.6.0.1008 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: --- (13 headers 0 lines) --- [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: = Looking for Call ID: bddd1f752853e839 (Checking From) --From tag ef592dfd6b --To-tag [Nov 22 11:57:33] DEBUG[1175] acl.c: For destination '192.168.10.202', our source address is '192.168.10.70'. [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Allocating new SIP dialog for bddd1f752853e839 - REGISTER (No RTP) [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Initializing initreq for method REGISTER - callid bddd1f752853e839 [Nov 22 11:57:33] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:33] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.202:5060 (no NAT) [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Store REGISTER's Contact header for call routing. [Nov 22 11:57:33] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:33] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK9ba124d10a6d27901.9a51108da54cd8541;received=192.168.10.202 From: ;tag=ef592dfd6b To: ;tag=as3c6e1936 Call-ID: bddd1f752853e839 CSeq: 9010 REGISTER Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 120 Contact: ;expires=120 Date: Mon, 22 Nov 2010 10:57:33 GMT Content-Length: 0 <------------> [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bK9ba124d10a6d27901.9a51108da54cd8541;received=192.168.10.202 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 2 [ 47]: From: ;tag=ef592dfd6b [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 3 [ 45]: To: ;tag=as3c6e1936 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: bddd1f752853e839 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 5 [ 19]: CSeq: 9010 REGISTER [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 9 [ 12]: Expires: 120 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 10 [ 67]: Contact: ;expires=120 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 11 [ 35]: Date: Mon, 22 Nov 2010 10:57:33 GMT [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:33] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog 'bddd1f752853e839' in 32000 ms (Method: REGISTER) [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:33] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:33] DEBUG[1195] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone3 PeerStatus: Registered Address: 192.168.10.202:5060 [Nov 22 11:57:33] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:57:33] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:57:33] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 22 11:57:33] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '1' [Nov 22 11:57:33] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:34] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:34] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:35] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:35] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> INVITE sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKf7dc0505d99b5194b.4a59710ad03d3406a Max-Forwards: 70 From: ;tag=1768984948 To: "Erika Musterfrau" ;tag=as45ace4d4 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 29237 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.6.0.66 Content-Type: application/sdp Content-Length: 620 v=0 o=MxSIP 0 3 IN IP4 192.168.10.200 s=SIP Call c=IN IP4 192.168.10.200 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendonly <-------------> [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 0 [ 41]: INVITE sip:100@192.168.10.70:5060 SIP/2.0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKf7dc0505d99b5194b.4a59710ad03d3406a [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 3 [ 67]: From: ;tag=1768984948 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 6 [ 18]: CSeq: 29237 INVITE [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 9 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 13 [ 19]: Content-Length: 620 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 14 [ 0]: [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.200 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 5 [ 70]: m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 21 [ 24]: a=silenceSupp:on - - - - [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 22 [ 20]: a=fmtp:18 annexb=yes [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 25 [ 10]: a=sendonly [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: --- (14 headers 26 lines) --- [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: = Looking for Call ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (Checking From) --From tag 1768984948 --To-tag as45ace4d4 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -gruu- [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: gruu [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -path- [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: path [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: timer [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 22 11:57:36] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 22 11:57:36] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:36] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.200:5060 (no NAT) [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Initializing initreq for method INVITE - callid 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.10.200... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] netsock2.c: Splitting '192.168.10.200' gives... [Nov 22 11:57:36] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '(null)'. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.200... OK. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 18 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 18 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 106 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 106 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 107 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 113 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 110 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 110 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 111 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 111 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 112 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 112 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 98 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 98 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 97 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 97 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 115 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 115 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 96 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 9 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 9 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format G729 for ID 18 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format BV16 for ID 106 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format BV32 for ID 107 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 113 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 110 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 111 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 112 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format G726-16 for ID 98 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format G726-24 for ID 97 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format G726-32 for ID 115 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format G726-40 for ID 96 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format G722 for ID 9 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:on - - - -... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 9 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 18 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 97 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 98 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 106 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 110 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 111 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 112 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 113 on 0xb3380f78 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 115 on 0xb3380f78 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8109f4c (ulaw|alaw|g726|slin|g729|speex|ilbc|g722|slin16|h263p|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:36] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd809e70' [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.200:3000 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 9 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 18 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 97 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 98 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 106 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 110 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 111 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 112 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 113 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 115 from 0xb3380f78 to 0xd80a01c [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:36] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd809e70' [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Got a SIP re-invite for call 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: SIP/phone2-00000011: This call is UP.... [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKf7dc0505d99b5194b.4a59710ad03d3406a;received=192.168.10.200 From: ;tag=1768984948 To: "Erika Musterfrau" ;tag=as45ace4d4 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 29237 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKf7dc0505d99b5194b.4a59710ad03d3406a;received=192.168.10.200 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 2 [ 67]: From: ;tag=1768984948 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 5 [ 18]: CSeq: 29237 INVITE [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Setting framing from config on incoming call [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Audio is at 5060 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKf7dc0505d99b5194b.4a59710ad03d3406a;received=192.168.10.200 From: ;tag=1768984948 To: "Erika Musterfrau" ;tag=as45ace4d4 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 29237 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: 263 v=0 o=root 1496546433 1496546436 IN IP4 192.168.10.201 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly <------------> [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKf7dc0505d99b5194b.4a59710ad03d3406a;received=192.168.10.200 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 2 [ 67]: From: ;tag=1768984948 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 5 [ 18]: CSeq: 29237 INVITE [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 11 [ 19]: Content-Length: 263 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 12 [ 0]: [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 1 [ 50]: o=root 1496546433 1496546436 IN IP4 192.168.10.201 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=recvonly [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #393 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Sending reinvite on SIP '8bd610aef94fb8be' - It's audio soon redirected to IP 192.168.10.70:5060 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Strict routing enforced for session 8bd610aef94fb8be [Nov 22 11:57:36] VERBOSE[1252] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:36] DEBUG[1252] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:36] DEBUG[1252] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:36] VERBOSE[1252] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:57:36] VERBOSE[1252] chan_sip.c: Audio is at 5060 [Nov 22 11:57:36] VERBOSE[1252] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:36] VERBOSE[1252] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:36] VERBOSE[1252] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Initializing already initialized SIP dialog 8bd610aef94fb8be (presumably reinvite) [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK05480cba [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:36] VERBOSE[1252] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK05480cba Max-Forwards: 70 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Contact: Call-ID: 8bd610aef94fb8be 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: "Hans Muster" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1247526371 1247526373 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 12072 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK05480cba [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 14 [ 19]: Content-Length: 262 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 1 [ 49]: o=root 1247526371 1247526373 IN IP4 192.168.10.70 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 5 [ 29]: m=audio 12072 RTP/AVP 8 0 101 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #394 [Nov 22 11:57:36] DEBUG[1252] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:36] DEBUG[1252] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:57:36] VERBOSE[1252] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/phone1-00000010 [Nov 22 11:57:36] DEBUG[1252] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Nov 22 11:57:36] DEBUG[1252] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: INVITE [Nov 22 11:57:36] DEBUG[1195] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/phone1-00000010 UniqueID: 1290423450.16 Class: default [Nov 22 11:57:36] DEBUG[1252] channel.c: Set channel SIP/phone1-00000010 to write format slin [Nov 22 11:57:36] DEBUG[1252] res_musiconhold.c: SIP/phone1-00000010 Opened file 0 '/var/lib/asterisk/moh/iptam/iptam_moh' [Nov 22 11:57:36] DEBUG[1252] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Nov 22 11:57:36] DEBUG[1252] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Nov 22 11:57:36] DEBUG[1252] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xd829cd8' [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> ACK sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK5b726c985545d3cc9.9248521d168e4f7ea Max-Forwards: 70 From: ;tag=1768984948 To: "Erika Musterfrau" ;tag=as45ace4d4 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 29237 ACK User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 0 [ 38]: ACK sip:100@192.168.10.70:5060 SIP/2.0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK5b726c985545d3cc9.9248521d168e4f7ea [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 3 [ 67]: From: ;tag=1768984948 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 6 [ 15]: CSeq: 29237 ACK [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 7 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: --- (9 headers 0 lines) --- [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: = Looking for Call ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (Checking From) --From tag 1768984948 --To-tag as45ace4d4 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #393 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Stopping retransmission on '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' of Response 29237: Match Found [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK05480cba From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Call-ID: 8bd610aef94fb8be 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="" 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 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK05480cba [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 8 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 2 IN IP4 192.168.10.201 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: = Looking for Call ID: 8bd610aef94fb8be (Checking To) --From tag as37527f23 --To-tag bdabd584b8 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Acked pending invite 103 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #394 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Stopping retransmission on '8bd610aef94fb8be' of Request 103: Match Found [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 8bd610aef94fb8be [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.10.201... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] netsock2.c: Splitting '192.168.10.201' gives... [Nov 22 11:57:36] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '(null)'. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:36] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd829cd8' [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:36] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:57:36] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:36] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Strict routing enforced for session 8bd610aef94fb8be [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:36] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:36] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:57:36] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.201:5060: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK40b54ba6 Max-Forwards: 70 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Contact: Call-ID: 8bd610aef94fb8be CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK40b54ba6 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:36] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:36] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:57:36] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:57:36] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:57:36] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:57:36] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: Auto destroying SIP dialog '2f7e3545567b8d26' [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: Destroying SIP dialog 2f7e3545567b8d26 [Nov 22 11:57:37] VERBOSE[1175] chan_sip.c: Really destroying SIP dialog '2f7e3545567b8d26' Method: REGISTER [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: ---------- SIP HISTORY for '2f7e3545567b8d26' [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: * SIP Call [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: 001. Rx REGISTER / 19007 REGISTER / sip:192.168.10.70 [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: 002. TxResp SIP/2.0 / 19007 REGISTER - 200 OK [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: 005. AutoDestroy 2f7e3545567b8d26 [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: ---------- END SIP HISTORY for '2f7e3545567b8d26' [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:37] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:38] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:38] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:39] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:39] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> INVITE sip:180@192.168.10.70 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK088b6ae1453bdb43c.602b98344631a3ba5 Max-Forwards: 70 From: "" ;tag=5ba829929a To: "180" Call-ID: 2e4ec62ed2ac174d CSeq: 6007 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.6.0.66 Content-Type: application/sdp Content-Length: 620 v=0 o=MxSIP 0 0 IN IP4 192.168.10.200 s=SIP Call c=IN IP4 192.168.10.200 t=0 0 m=audio 3002 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 0 [ 36]: INVITE sip:180@192.168.10.70 SIP/2.0 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK088b6ae1453bdb43c.602b98344631a3ba5 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "" ;tag=5ba829929a [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 4 [ 33]: To: "180" [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 6 [ 17]: CSeq: 6007 INVITE [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 9 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 11 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 13 [ 19]: Content-Length: 620 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 14 [ 0]: [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.200 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 5 [ 70]: m=audio 3002 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 21 [ 24]: a=silenceSupp:on - - - - [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 22 [ 20]: a=fmtp:18 annexb=yes [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Body 25 [ 10]: a=sendrecv [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: --- (14 headers 26 lines) --- [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: = Looking for Call ID: 2e4ec62ed2ac174d (Checking From) --From tag 5ba829929a --To-tag [Nov 22 11:57:40] DEBUG[1175] acl.c: For destination '192.168.10.200', our source address is '192.168.10.70'. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Allocating new SIP dialog for 2e4ec62ed2ac174d - INVITE (No RTP) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -gruu- [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: gruu [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -path- [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: path [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -timer- [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: timer [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -100rel- [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: 100rel [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Found SIP option: -replaces- [Nov 22 11:57:40] DEBUG[1175] sip/reqresp_parser.c: Matched SIP option: replaces [Nov 22 11:57:40] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:40] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.200:5060 (no NAT) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Initializing initreq for method INVITE - callid 2e4ec62ed2ac174d [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Using INVITE request as basis request - 2e4ec62ed2ac174d [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found peer 'phone2' for 'phone2' from 192.168.10.200:5060 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd805070' [Nov 22 11:57:40] DEBUG[1175] res_rtp_asterisk.c: Allocated port 11490 for RTP instance '0xd805070' [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: RTP instance '0xd805070' is setup and ready to go [Nov 22 11:57:40] DEBUG[1175] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd805070' [Nov 22 11:57:40] VERBOSE[1175] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 22 11:57:40] VERBOSE[1175] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Setting NAT on RTP to Off [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.200... UNSUPPORTED. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:40] DEBUG[1175] netsock2.c: Splitting '192.168.10.200' gives... [Nov 22 11:57:40] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '(null)'. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.200... OK. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 18 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 18 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 106 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 106 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 107 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 113 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 110 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 110 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 111 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 111 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 112 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 112 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 98 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 98 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 97 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 97 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 115 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 115 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 96 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 9 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 9 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format G729 for ID 18 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format BV16 for ID 106 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format BV32 for ID 107 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 113 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 110 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 111 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format L16 for ID 112 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format G726-16 for ID 98 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format G726-24 for ID 97 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format G726-32 for ID 115 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format G726-40 for ID 96 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format G722 for ID 9 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:on - - - -... UNSUPPORTED. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 9 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 18 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 97 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 98 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 106 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 110 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 111 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 112 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 113 on 0xb3380f78 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Incorporating payload 115 on 0xb3380f78 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8109f4c (ulaw|alaw|g726|slin|g729|speex|ilbc|g722|slin16|h263p|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:40] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd805070' [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.200:3002 [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 9 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 18 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 97 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 98 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 106 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 110 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 111 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 112 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 113 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] rtp_engine.c: Copying payload 115 from 0xb3380f78 to 0xd80521c [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Checking SIP call limits for device phone2 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Call from peer 'phone2' is 2 out of 2147483647 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: Looking for 180 in Standard (domain 192.168.10.70) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: This channel will not be able to handle video. [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: build_route: Contact hop: "" ;+sip.instance="" [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: list_route: hop: [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: SIP/phone2-00000012: New call is still down.... Trying... [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK088b6ae1453bdb43c.602b98344631a3ba5;received=192.168.10.200 From: "" ;tag=5ba829929a To: "180" Call-ID: 2e4ec62ed2ac174d CSeq: 6007 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK088b6ae1453bdb43c.602b98344631a3ba5;received=192.168.10.200 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "" ;tag=5ba829929a [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 3 [ 33]: To: "180" [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 5 [ 17]: CSeq: 6007 INVITE [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:40] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:40] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:40] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:57:40] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:57:40] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:40] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:40] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:57:40] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:57:40] DEBUG[1195] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-00000012 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 150 CallerIDName: Hans Muster AccountCode: Exten: 180 Context: Standard Uniqueid: 1290423460.18 [Nov 22 11:57:40] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000012 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 150 CallerIDName: Hans Muster Uniqueid: 1290423460.18 [Nov 22 11:57:40] DEBUG[1253] pbx.c: Launching 'Dial' [Nov 22 11:57:40] VERBOSE[1253] pbx.c: -- Executing [180@Standard:1] Dial("SIP/phone2-00000012", "SIP/phone3") in new stack [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Allocating new SIP dialog for 256e4fec77275bcd33acab2921ca33cb@192.168.10.70 - INVITE (No RTP) [Nov 22 11:57:40] DEBUG[1253] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd7fdab8' [Nov 22 11:57:40] DEBUG[1253] res_rtp_asterisk.c: Allocated port 14208 for RTP instance '0xd7fdab8' [Nov 22 11:57:40] DEBUG[1253] rtp_engine.c: RTP instance '0xd7fdab8' is setup and ready to go [Nov 22 11:57:40] DEBUG[1253] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd7fdab8' [Nov 22 11:57:40] VERBOSE[1253] netsock2.c: == Using SIP RTP TOS bits 184 [Nov 22 11:57:40] VERBOSE[1253] netsock2.c: == Using SIP RTP CoS mark 5 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Setting NAT on RTP to Off [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Nov 22 11:57:40] DEBUG[1253] acl.c: For destination '192.168.10.202', our source address is '192.168.10.70'. [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: This channel will not be able to handle video. [Nov 22 11:57:40] DEBUG[1253] rtp_engine.c: Seeded SDP of 'SIP/phone3-00000013' with that of 'SIP/phone2-00000012' [Nov 22 11:57:40] DEBUG[1253] channel.c: Not copying variable DIALEDTIME. [Nov 22 11:57:40] DEBUG[1253] channel.c: Not copying variable ANSWEREDTIME. [Nov 22 11:57:40] DEBUG[1253] channel.c: Not copying variable DIALEDPEERNAME. [Nov 22 11:57:40] DEBUG[1253] channel.c: Not copying variable DIALEDPEERNUMBER. [Nov 22 11:57:40] DEBUG[1253] channel.c: Not copying variable DIALSTATUS. [Nov 22 11:57:40] DEBUG[1253] channel.c: Not copying variable SIPCALLID. [Nov 22 11:57:40] DEBUG[1253] channel.c: Not copying variable SIPDOMAIN. [Nov 22 11:57:40] DEBUG[1253] channel.c: Not copying variable SIPURI. [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Outgoing Call for phone3 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Call to peer 'phone3' is 1 out of 2147483647 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:57:40] VERBOSE[1253] chan_sip.c: Audio is at 5060 [Nov 22 11:57:40] VERBOSE[1253] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:40] VERBOSE[1253] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:40] VERBOSE[1253] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Initializing initreq for method INVITE - callid 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK746d4dec [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 4 [ 50]: To: [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 9 [ 35]: Date: Mon, 22 Nov 2010 10:57:40 GMT [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:40] VERBOSE[1253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.202:5060: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK746d4dec Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0-1 Date: Mon, 22 Nov 2010 10:57:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Hans Muster" Content-Type: application/sdp Content-Length: 260 v=0 o=root 395103105 395103105 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 14208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK746d4dec [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 4 [ 50]: To: [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 9 [ 35]: Date: Mon, 22 Nov 2010 10:57:40 GMT [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 14 [ 19]: Content-Length: 260 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 1 [ 47]: o=root 395103105 395103105 IN IP4 192.168.10.70 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 5 [ 29]: m=audio 14208 RTP/AVP 8 0 101 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #397 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:40] VERBOSE[1253] app_dial.c: -- Called phone3 [Nov 22 11:57:40] DEBUG[1195] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone3-00000013 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 180 CallerIDName: Max Mustermann AccountCode: Exten: Context: Standard Uniqueid: 1290423460.19 [Nov 22 11:57:40] DEBUG[1195] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone2-00000012 Destination: SIP/phone3-00000013 CallerIDNum: 150 CallerIDName: Hans Muster UniqueID: 1290423460.18 DestUniqueID: 1290423460.19 Dialstring: phone3 [Nov 22 11:57:40] DEBUG[1195] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone3-00000013 CallerIDNum: 180 CallerIDName: Uniqueid: 1290423460.19 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:40] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:57:40] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:57:40] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 6 (Ringing) [Nov 22 11:57:40] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '6' [Nov 22 11:57:40] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:40] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:40] DEBUG[1168] app_queue.c: Extension '180@Standard' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:57:40] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:57:40] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 8 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK746d4dec From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="" Server: Aastra 51i/2.6.0.1008 Supported: gruu, path Content-Length: 0 <-------------> [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK746d4dec [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 22 11:57:40] VERBOSE[1175] chan_sip.c: --- (12 headers 0 lines) --- [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: = Looking for Call ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (Checking To) --From tag as1a46d292 --To-tag 3092808172 [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: *** SIP TIMER: Cancelling retransmission #397 - INVITE (got response) [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '43691a3867af054e1f075d71580a7d02@192.168.10.70' Request 102: Found [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: SIP response 180 to standard invite [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:40] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:40] VERBOSE[1253] app_dial.c: -- SIP/phone3-00000013 is ringing [Nov 22 11:57:40] DEBUG[1253] rtp_engine.c: Setting early bridge SDP of 'SIP/phone2-00000012' with that of 'SIP/phone3-00000013' [Nov 22 11:57:40] VERBOSE[1253] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK088b6ae1453bdb43c.602b98344631a3ba5;received=192.168.10.200 From: "" ;tag=5ba829929a To: "180" ;tag=as5a7bf510 Call-ID: 2e4ec62ed2ac174d CSeq: 6007 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK088b6ae1453bdb43c.602b98344631a3ba5;received=192.168.10.200 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 2 [ 50]: From: "" ;tag=5ba829929a [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 3 [ 48]: To: "180" ;tag=as5a7bf510 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 4 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 5 [ 17]: CSeq: 6007 INVITE [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Header 11 [ 0]: [Nov 22 11:57:40] DEBUG[1253] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:40] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:57:40] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:57:40] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 6 (Ringing) [Nov 22 11:57:40] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '6' [Nov 22 11:57:40] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000013 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 180 CallerIDName: Uniqueid: 1290423460.19 [Nov 22 11:57:40] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1195] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.201:3001 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 39820 SequenceNumberCycles: 0 IAJitter: 43 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:40] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1195] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 192.168.10.201:3001 OurSSRC: 2092856043 SentNTP: 1290423461.0286588928 SentRTP: 40160 SentPackets: 251 SentOctets: 40160 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0000 TheirLastSR: 286577131 DLSR: 0.2000 (sec) [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK746d4dec From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="" Server: Aastra 51i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 0 IN IP4 192.168.10.202 s=SIP Call c=IN IP4 192.168.10.202 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK746d4dec [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 0 IN IP4 192.168.10.202 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.202 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: = Looking for Call ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (Checking To) --From tag as1a46d292 --To-tag 3092808172 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Acked pending invite 102 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Stopping retransmission on '43691a3867af054e1f075d71580a7d02@192.168.10.70' of Request 102: Match Found [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: SIP response 200 to standard invite [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 192.168.10.202... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.202' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '(null)'. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.202... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:41] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7fdab8' [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.202:3000 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd7fdc64 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd7fdc64 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd7fdc64 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: build_route: Contact hop: "Max Mustermann" ;+sip.instance="" [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: list_route: hop: [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Strict routing enforced for session 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.202:5060: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK18eb8258 Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK18eb8258 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] VERBOSE[1253] app_dial.c: -- SIP/phone3-00000013 answered SIP/phone2-00000012 [Nov 22 11:57:41] DEBUG[1253] rtp_engine.c: Setting early bridge SDP of 'SIP/phone2-00000012' with that of 'SIP/phone3-00000013' [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: SIP answering channel: SIP/phone2-00000012 [Nov 22 11:57:41] DEBUG[1253] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Setting framing from config on incoming call [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Audio is at 5060 [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK088b6ae1453bdb43c.602b98344631a3ba5;received=192.168.10.200 From: "" ;tag=5ba829929a To: "180" ;tag=as5a7bf510 Call-ID: 2e4ec62ed2ac174d CSeq: 6007 INVITE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "Max Mustermann" Content-Type: application/sdp Content-Length: 260 v=0 o=root 296130322 296130322 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 11490 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK088b6ae1453bdb43c.602b98344631a3ba5;received=192.168.10.200 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 2 [ 50]: From: "" ;tag=5ba829929a [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 3 [ 48]: To: "180" ;tag=as5a7bf510 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 4 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 5 [ 17]: CSeq: 6007 INVITE [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 10 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 12 [ 19]: Content-Length: 260 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 1 [ 47]: o=root 296130322 296130322 IN IP4 192.168.10.70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 5 [ 29]: m=audio 11490 RTP/AVP 8 0 101 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #400 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:41] DEBUG[1253] features.c: bridge answer set, chan answer set [Nov 22 11:57:41] DEBUG[1253] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:57:41] DEBUG[1253] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:57:41] VERBOSE[1253] rtp_engine.c: -- Remotely bridging SIP/phone2-00000012 and SIP/phone3-00000013 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Deferring reinvite on SIP '2e4ec62ed2ac174d' - It's audio will be redirected to IP 192.168.10.202:3000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Sending reinvite on SIP '43691a3867af054e1f075d71580a7d02@192.168.10.70' - It's audio soon redirected to IP 192.168.10.200:3002 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Strict routing enforced for session 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:41] DEBUG[1253] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:41] DEBUG[1253] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Audio is at 5060 [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Initializing already initialized SIP dialog 43691a3867af054e1f075d71580a7d02@192.168.10.70 (presumably reinvite) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7fecdb10 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.202:5060: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7fecdb10 Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@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: "Hans Muster" Content-Type: application/sdp Content-Length: 261 v=0 o=root 395103105 395103106 IN IP4 192.168.10.200 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.200 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 22 11:57:41] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7fecdb10 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 14 [ 19]: Content-Length: 261 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 1 [ 48]: o=root 395103105 395103106 IN IP4 192.168.10.200 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 5 [ 28]: m=audio 3002 RTP/AVP 8 0 101 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #401 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:41] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:57:41] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:57:41] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 22 11:57:41] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '2' [Nov 22 11:57:41] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:57:41] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:57:41] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 22 11:57:41] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '2' [Nov 22 11:57:41] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:41] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:41] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:57:41] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:57:41] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000013 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 180 CallerIDName: Uniqueid: 1290423460.19 [Nov 22 11:57:41] DEBUG[1195] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000012 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 150 CallerIDName: Hans Muster Uniqueid: 1290423460.18 [Nov 22 11:57:41] DEBUG[1195] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone3-00000013 Uniqueid: 1290423460.19 AccountCode: OldAccountCode: [Nov 22 11:57:41] DEBUG[1195] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone2-00000012 Channel2: SIP/phone3-00000013 Uniqueid1: 1290423460.18 Uniqueid2: 1290423460.19 CallerID1: 150 CallerID2: 180 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2e4ec62ed2ac174d' Method: INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:41] DEBUG[1168] app_queue.c: Extension '180@Standard' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:41] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:41] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:41] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:41] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 1 [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> ACK sip:180@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb3105ff394ff73109.c12ae6097dee4b8cc Max-Forwards: 70 From: "" ;tag=5ba829929a To: "180" ;tag=as5a7bf510 Call-ID: 2e4ec62ed2ac174d CSeq: 6007 ACK User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 38]: ACK sip:180@192.168.10.70:5060 SIP/2.0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKb3105ff394ff73109.c12ae6097dee4b8cc [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "" ;tag=5ba829929a [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "180" ;tag=as5a7bf510 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 14]: CSeq: 6007 ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: --- (9 headers 0 lines) --- [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: = Looking for Call ID: 2e4ec62ed2ac174d (Checking From) --From tag 5ba829929a --To-tag as5a7bf510 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #400 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Stopping retransmission on '2e4ec62ed2ac174d' of Response 6007: Match Found [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Sending pending reinvite on '2e4ec62ed2ac174d' [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Strict routing enforced for session 2e4ec62ed2ac174d [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Audio is at 5060 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Initializing already initialized SIP dialog 2e4ec62ed2ac174d (presumably reinvite) [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4ae4a129 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "180" ;tag=as5a7bf510 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "" ;tag=5ba829929a [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.200:5060: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4ae4a129 Max-Forwards: 70 From: "180" ;tag=as5a7bf510 To: "" ;tag=5ba829929a Contact: Call-ID: 2e4ec62ed2ac174d 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: 261 v=0 o=root 296130322 296130323 IN IP4 192.168.10.202 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.202 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4ae4a129 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "180" ;tag=as5a7bf510 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "" ;tag=5ba829929a [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 14 [ 19]: Content-Length: 261 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 1 [ 48]: o=root 296130322 296130323 IN IP4 192.168.10.202 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.202 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #402 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2e4ec62ed2ac174d' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7fecdb10 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Max Mustermann" ;+sip.instance="" Server: Aastra 51i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 1 IN IP4 192.168.10.202 s=SIP Call c=IN IP4 192.168.10.202 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7fecdb10 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.202 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.202 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: = Looking for Call ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (Checking To) --From tag as1a46d292 --To-tag 3092808172 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Acked pending invite 103 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #401 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Stopping retransmission on '43691a3867af054e1f075d71580a7d02@192.168.10.70' of Request 103: Match Found [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.202... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.202' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '(null)'. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.202... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:41] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7fdab8' [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.202:3000 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd7fdc64 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd7fdc64 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd7fdc64 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Strict routing enforced for session 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.202:5060: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK224a7b78 Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK224a7b78 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 103 ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2e4ec62ed2ac174d' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:41] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:57:41] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:57:41] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 22 11:57:41] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '2' [Nov 22 11:57:41] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4ae4a129 From: "180" ;tag=as5a7bf510 To: "" ;tag=5ba829929a Call-ID: 2e4ec62ed2ac174d CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Server: Aastra 55i/2.6.0.66 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 1 IN IP4 192.168.10.200 s=SIP Call c=IN IP4 192.168.10.200 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 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4ae4a129 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "180" ;tag=as5a7bf510 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 48]: To: "" ;tag=5ba829929a [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 9 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 1 IN IP4 192.168.10.200 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3002 RTP/AVP 8 0 101 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: = Looking for Call ID: 2e4ec62ed2ac174d (Checking To) --From tag as5a7bf510 --To-tag 5ba829929a [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Acked pending invite 102 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #402 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Stopping retransmission on '2e4ec62ed2ac174d' of Request 102: Match Found [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 2e4ec62ed2ac174d [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.10.200... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.200' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '(null)'. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.200... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:41] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd805070' [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.200:3002 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd80521c [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd80521c [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd80521c [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Strict routing enforced for session 2e4ec62ed2ac174d [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.200:5060: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK032e2f23 Max-Forwards: 70 From: "180" ;tag=as5a7bf510 To: "" ;tag=5ba829929a Contact: Call-ID: 2e4ec62ed2ac174d CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK032e2f23 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "180" ;tag=as5a7bf510 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "" ;tag=5ba829929a [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2e4ec62ed2ac174d' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:41] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:41] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:41] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Nov 22 11:57:41] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '2' [Nov 22 11:57:41] DEBUG[1253] rtp_engine.c: Oooh, 'SIP/phone2-00000012' changed end address to 192.168.10.200:3002 (format unknown) [Nov 22 11:57:41] DEBUG[1253] rtp_engine.c: Oooh, 'SIP/phone2-00000012' was 192.168.10.200:3002/(format unknown) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Sending reinvite on SIP '43691a3867af054e1f075d71580a7d02@192.168.10.70' - It's audio soon redirected to IP 192.168.10.200:3002 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Strict routing enforced for session 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:41] DEBUG[1253] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:41] DEBUG[1253] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Audio is at 5060 [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Initializing already initialized SIP dialog 43691a3867af054e1f075d71580a7d02@192.168.10.70 (presumably reinvite) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3b13365b [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] VERBOSE[1253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.202:5060: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3b13365b Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@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: "Hans Muster" Content-Type: application/sdp Content-Length: 261 v=0 o=root 395103105 395103107 IN IP4 192.168.10.200 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.200 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 22 11:57:41] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3b13365b [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 14 [ 19]: Content-Length: 261 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 1 [ 48]: o=root 395103105 395103107 IN IP4 192.168.10.200 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.200 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 5 [ 28]: m=audio 3002 RTP/AVP 8 0 101 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #403 [Nov 22 11:57:41] DEBUG[1253] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2e4ec62ed2ac174d' Method: ACK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:41] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3b13365b From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Call-ID: 43691a3867af054e1f075d71580a7d02@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 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 192.168.10.202 s=SIP Call c=IN IP4 192.168.10.202 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK3b13365b [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 2 IN IP4 192.168.10.202 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.202 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: = Looking for Call ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (Checking To) --From tag as1a46d292 --To-tag 3092808172 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Acked pending invite 104 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #403 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Stopping retransmission on '43691a3867af054e1f075d71580a7d02@192.168.10.70' of Request 104: Match Found [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.10.202... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:41] DEBUG[1175] netsock2.c: Splitting '192.168.10.202' gives... [Nov 22 11:57:41] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '(null)'. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.202... OK. [Nov 22 11:57:41] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:41] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:41] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:42] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:42] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:42] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:42] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:42] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:42] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7fdab8' [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.202:3000 [Nov 22 11:57:42] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd7fdc64 [Nov 22 11:57:42] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd7fdc64 [Nov 22 11:57:42] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd7fdc64 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:42] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:42] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Strict routing enforced for session 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:42] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:42] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:57:42] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.202:5060: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0047adba Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK0047adba [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 104 ACK [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2e4ec62ed2ac174d' Method: ACK [Nov 22 11:57:42] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:42] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:57:42] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:57:42] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Nov 22 11:57:42] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '2' [Nov 22 11:57:42] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:42] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:43] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:43] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:43] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2e4ec62ed2ac174d' Method: ACK [Nov 22 11:57:43] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:43] DEBUG[1252] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2e4ec62ed2ac174d' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: ACK [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> REFER sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKeb6fcad5af3c6b9cb.a5ac0929c11f12c77 Max-Forwards: 70 From: ;tag=1768984948 To: "Erika Musterfrau" ;tag=as45ace4d4 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 29238 REFER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "" ;+sip.instance="" Refer-To: "180" Referred-By: Supported: gruu, path, timer User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 40]: REFER sip:100@192.168.10.70:5060 SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKeb6fcad5af3c6b9cb.a5ac0929c11f12c77 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 67]: From: ;tag=1768984948 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 17]: CSeq: 29238 REFER [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [122]: Contact: "" ;+sip.instance="" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [111]: Refer-To: "180" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 39]: Referred-By: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 28]: Supported: gruu, path, timer [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (15 headers 0 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (Checking From) --From tag 1768984948 --To-tag as45ace4d4 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Call 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 got a SIP call transfer from caller: (REFER)! [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 2e4ec62ed2ac174d F-tag: 5ba829929a T-tag: as5a7bf510 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: SIP transfer to extension 180@Standard by phone2@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP attended transfer: Transferer channel SIP/phone2-00000011, transferee channel SIP/phone1-00000010 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/phone1-00000010' [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Looking for callid 2e4ec62ed2ac174d (fromtag 5ba829929a totag as5a7bf510) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Matched INCOMING call - their tag is 5ba829929a Our tag is as5a7bf510 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKeb6fcad5af3c6b9cb.a5ac0929c11f12c77;received=192.168.10.200 From: ;tag=1768984948 To: "Erika Musterfrau" ;tag=as45ace4d4 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 29238 REFER Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 20]: SIP/2.0 202 Accepted [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bKeb6fcad5af3c6b9cb.a5ac0929c11f12c77;received=192.168.10.200 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 67]: From: ;tag=1768984948 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 17]: CSeq: 29238 REFER [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP attended transfer: trying to bridge SIP/phone2-00000012 and SIP/phone1-00000010 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Sip transfer:-------------------- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: -- Transferer to PBX channel: SIP/phone2-00000011 State Up [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: -- Transferer to PBX second channel (target): SIP/phone2-00000012 State Up [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: -- Bridged call to transferee: SIP/phone1-00000010 State Up [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: -- Bridged call to transfer target: SIP/phone3-00000013 State Up [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: -- END Sip transfer:-------------------- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP transfer: Four channels to handle [Nov 22 11:57:44] VERBOSE[1175] res_musiconhold.c: -- Stopped music on hold on SIP/phone1-00000010 [Nov 22 11:57:44] DEBUG[1175] channel.c: Set channel SIP/phone1-00000010 to write format alaw [Nov 22 11:57:44] DEBUG[1175] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP transfer: trying to masquerade SIP/phone1-00000010 into SIP/phone2-00000012 [Nov 22 11:57:44] DEBUG[1175] channel.c: Planning to masquerade channel SIP/phone1-00000010 into the structure of SIP/phone2-00000012 [Nov 22 11:57:44] DEBUG[1175] channel.c: Done planning to masquerade channel SIP/phone1-00000010 into the structure of SIP/phone2-00000012 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Strict routing enforced for session 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.200:5060: NOTIFY sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK02d33600 Max-Forwards: 70 From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Contact: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 105 NOTIFY User-Agent: Asterisk PBX 1.8.0-1 Event: refer;id=29238 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 16 SIP/2.0 200 OK --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 59]: NOTIFY sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK02d33600 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 105 NOTIFY [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 21]: Event: refer;id=29238 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 48]: Subscription-state: terminated;reason=noresource [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 41]: Content-Type: message/sipfrag;version=2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 14 [ 18]: Content-Length: 16 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #404 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:44] DEBUG[1175] channel.c: Actually Masquerading SIP/phone1-00000010(6) into the structure of SIP/phone2-00000012(6) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Sending reinvite on SIP '2e4ec62ed2ac174d' - It's audio soon redirected to IP 192.168.10.70:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Strict routing enforced for session 2e4ec62ed2ac174d [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.200:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Audio is at 5060 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Initializing already initialized SIP dialog 2e4ec62ed2ac174d (presumably reinvite) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2b94d2ab [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "180" ;tag=as5a7bf510 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "" ;tag=5ba829929a [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.200:5060: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2b94d2ab Max-Forwards: 70 From: "180" ;tag=as5a7bf510 To: "" ;tag=5ba829929a Contact: Call-ID: 2e4ec62ed2ac174d 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: "Hans Muster" Content-Type: application/sdp Content-Length: 260 v=0 o=root 296130322 296130324 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 11490 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 59]: INVITE sip:phone2@192.168.10.200:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2b94d2ab [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "180" ;tag=as5a7bf510 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 48]: To: "" ;tag=5ba829929a [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 14 [ 19]: Content-Length: 260 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 1 [ 47]: o=root 296130322 296130324 IN IP4 192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 5 [ 29]: m=audio 11490 RTP/AVP 8 0 101 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #405 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP Fixup: New owner for dialogue 2e4ec62ed2ac174d: SIP/phone1-00000010 (Old parent: SIP/phone1-00000010) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Hangup call SIP/phone1-00000010, SIP callid 2e4ec62ed2ac174d [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: update_call_counter(phone2) - decrement call limit counter on hangup [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Call from peer 'phone2' removed from call limit 2147483647 [Nov 22 11:57:44] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd805070' [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog '2e4ec62ed2ac174d' in 32000 ms (Method: ACK) [Nov 22 11:57:44] DEBUG[1175] channel.c: Putting channel SIP/phone1-00000010 in alaw/alaw formats [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP Fixup: New owner for dialogue 8bd610aef94fb8be: SIP/phone1-00000010 (Old parent: SIP/phone2-00000012) [Nov 22 11:57:44] DEBUG[1175] channel.c: Released clone lock on 'SIP/phone2-00000012' [Nov 22 11:57:44] DEBUG[1175] channel.c: Done Masquerading SIP/phone1-00000010 (6) [Nov 22 11:57:44] DEBUG[1175] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Nov 22 11:57:44] DEBUG[1175] res_rtp_asterisk.c: Changing ssrc from 2092856043 to 1657650883 due to a source change [Nov 22 11:57:44] DEBUG[1175] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2e4ec62ed2ac174d' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' Method: REFER [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Attended Channel: SIP/phone2-00000011 Uniqueid: 1290423450.17 SIP-Callid: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 TargetChannel: SIP/phone2-00000012 TargetUniqueid: 1290423460.18 [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/phone1-00000010 UniqueID: 1290423450.16 [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/phone1-00000010 CloneState: Up Original: SIP/phone2-00000012 OriginalState: Up [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone1-00000010 Newname: SIP/phone1-00000010 Uniqueid: 1290423450.16 [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone2-00000012 Newname: SIP/phone1-00000010 Uniqueid: 1290423460.18 [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone1-00000010 Newname: SIP/phone2-00000012 Uniqueid: 1290423450.16 [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone1-00000010 CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1290423460.18 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/phone1-00000010 UniqueID: 1290423460.18 [Nov 22 11:57:44] DEBUG[1252] rtp_engine.c: Oooh, something is weird, backing out [Nov 22 11:57:44] WARNING[1252] rtp_engine.c: Channel 'SIP/phone2-00000012' failed to break RTP bridge [Nov 22 11:57:44] DEBUG[1252] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/phone2-00000012, c1=SIP/phone2-00000011, flags: Yes,Yes,No,No [Nov 22 11:57:44] DEBUG[1252] res_rtp_asterisk.c: Setting the marker bit due to a source update [Nov 22 11:57:44] DEBUG[1252] channel.c: Bridge stops bridging channels SIP/phone2-00000012 and SIP/phone2-00000011 [Nov 22 11:57:44] DEBUG[1252] channel.c: Hanging up channel 'SIP/phone2-00000011' [Nov 22 11:57:44] DEBUG[1252] chan_sip.c: update_call_counter(phone2) - decrement call limit counter on hangup [Nov 22 11:57:44] DEBUG[1252] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:44] DEBUG[1252] chan_sip.c: Call to peer 'phone2' removed from call limit 2147483647 [Nov 22 11:57:44] DEBUG[1252] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70. [Nov 22 11:57:44] VERBOSE[1252] chan_sip.c: Scheduling destruction of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' in 32000 ms (Method: REFER) [Nov 22 11:57:44] DEBUG[1252] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 22 11:57:44] DEBUG[1252] pbx.c: Spawn extension (Standard,150,1) exited non-zero on 'SIP/phone2-00000012' [Nov 22 11:57:44] VERBOSE[1252] pbx.c: == Spawn extension (Standard, 150, 1) exited non-zero on 'SIP/phone2-00000012' [Nov 22 11:57:44] DEBUG[1252] channel.c: Soft-Hanging up channel 'SIP/phone2-00000012' [Nov 22 11:57:44] DEBUG[1252] channel.c: Hanging up zombie 'SIP/phone2-00000012' [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Strict routing enforced for session 8bd610aef94fb8be [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1253] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:44] DEBUG[1253] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Audio is at 5060 [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Initializing already initialized SIP dialog 8bd610aef94fb8be (presumably reinvite) [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4abe4226 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 11 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4abe4226 Max-Forwards: 70 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Contact: Call-ID: 8bd610aef94fb8be 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: "Max Mustermann" Content-Type: application/sdp Content-Length: 262 v=0 o=root 1247526371 1247526374 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 12072 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4abe4226 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 11 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 13 [ 19]: Content-Length: 262 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 14 [ 0]: [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 1 [ 49]: o=root 1247526371 1247526374 IN IP4 192.168.10.70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 5 [ 29]: m=audio 12072 RTP/AVP 8 0 101 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #408 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:44] DEBUG[1253] rtp_engine.c: Oooh, something is weird, backing out [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Sending reinvite on SIP '43691a3867af054e1f075d71580a7d02@192.168.10.70' - It's audio soon redirected to IP 192.168.10.70:5060 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Strict routing enforced for session 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1253] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:44] DEBUG[1253] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Audio is at 5060 [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Initializing already initialized SIP dialog 43691a3867af054e1f075d71580a7d02@192.168.10.70 (presumably reinvite) [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1b65f629 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.202:5060: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1b65f629 Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@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: "Hans Muster" Content-Type: application/sdp Content-Length: 260 v=0 o=root 395103105 395103108 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 14208 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1b65f629 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 12 [ 58]: P-Asserted-Identity: "Hans Muster" [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 14 [ 19]: Content-Length: 260 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 1 [ 47]: o=root 395103105 395103108 IN IP4 192.168.10.70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.10.70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 5 [ 29]: m=audio 14208 RTP/AVP 8 0 101 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #409 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:44] VERBOSE[1253] rtp_engine.c: -- Remotely bridging SIP/phone1-00000010 and SIP/phone3-00000013 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Deferring reinvite on SIP '8bd610aef94fb8be' - It's audio will be redirected to IP 192.168.10.202:3000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Deferring reinvite on SIP '43691a3867af054e1f075d71580a7d02@192.168.10.70' - It's audio will be redirected to IP 192.168.10.201:3000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Strict routing enforced for session 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1253] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:44] DEBUG[1253] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Audio is at 5060 [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Initializing already initialized SIP dialog 43691a3867af054e1f075d71580a7d02@192.168.10.70 (presumably reinvite) [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7eb4cf28 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 106 INVITE [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 11 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.202:5060: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7eb4cf28 Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@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: 261 v=0 o=root 395103105 395103109 IN IP4 192.168.10.201 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 0 [ 59]: INVITE sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7eb4cf28 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 7 [ 16]: CSeq: 106 INVITE [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 11 [ 63]: P-Asserted-Identity: "Erika Musterfrau" [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 13 [ 19]: Content-Length: 261 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Header 14 [ 0]: [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 1 [ 48]: o=root 395103105 395103109 IN IP4 192.168.10.201 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #410 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone2 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone2' state '1' [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone2-00000012 Channel2: SIP/phone2-00000011 Uniqueid1: 1290423450.16 Uniqueid2: 1290423450.17 CallerID1: 150 CallerID2: 150 [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 100 Destination: 150 DestinationContext: Standard CallerID: "Erika Musterfrau" <100> Channel: SIP/phone1-00000010 DestinationChannel: SIP/phone2-00000011 LastApplication: Dial LastData: SIP/phone2 StartTime: 2010-11-22 11:57:30 AnswerTime: 2010-11-22 11:57:32 EndTime: 2010-11-22 11:57:44 Duration: 14 BillableSeconds: 12 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1290423450.16 UserField: [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-00000011 Uniqueid: 1290423450.17 CallerIDNum: 150 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone2-00000012 UniqueID: 1290423450.16 DialStatus: ANSWER [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-00000012 Uniqueid: 1290423450.16 CallerIDNum: 150 CallerIDName: Hans Muster Cause: 16 Cause-txt: Normal Clearing [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] DEBUG[1168] app_queue.c: Extension '150@Standard' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 0 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4abe4226 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Call-ID: 8bd610aef94fb8be 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="" 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 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK4abe4226 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.201 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 8bd610aef94fb8be (Checking To) --From tag as37527f23 --To-tag bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Acked pending invite 104 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #408 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Stopping retransmission on '8bd610aef94fb8be' of Request 104: Match Found [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP response 200 to RE-invite on outgoing call 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.10.201... UNSUPPORTED. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.201' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '(null)'. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:44] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd829cd8' [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: We have an owner, now see if we need to change this call [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Strict routing enforced for session 8bd610aef94fb8be [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.201:5060: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK447684fc Max-Forwards: 70 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Contact: Call-ID: 8bd610aef94fb8be CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK447684fc [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 104 ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Sending pending reinvite on '8bd610aef94fb8be' [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Strict routing enforced for session 8bd610aef94fb8be [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** Our native-bridge filtered capablity: 0xc (ulaw|alaw) [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Audio is at 5060 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: -- Done with adding codecs to SDP [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Initializing already initialized SIP dialog 8bd610aef94fb8be (presumably reinvite) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK19c53b7f [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK19c53b7f Max-Forwards: 70 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Contact: Call-ID: 8bd610aef94fb8be 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: 263 v=0 o=root 1247526371 1247526375 IN IP4 192.168.10.202 s=Asterisk PBX 1.8.0-1 c=IN IP4 192.168.10.202 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 59]: INVITE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK19c53b7f [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 61]: P-Asserted-Identity: "Max Mustermann" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 14 [ 19]: Content-Length: 263 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 15 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 1 [ 50]: o=root 1247526371 1247526375 IN IP4 192.168.10.202 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.202 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #411 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '2' [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK02d33600 From: "Erika Musterfrau" ;tag=as45ace4d4 To: ;tag=1768984948 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 105 NOTIFY Server: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK02d33600 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 63]: From: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=1768984948 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 105 NOTIFY [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (8 headers 0 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (Checking To) --From tag as45ace4d4 --To-tag 1768984948 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #404 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Stopping retransmission on '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' of Request 105: Match Found [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> BYE sip:100@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK94c49eefee330019a.e66972e74c83e2bc6 Max-Forwards: 70 From: ;tag=1768984948 To: "Erika Musterfrau" ;tag=as45ace4d4 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 29239 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Supported: gruu, path, timer User-Agent: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 38]: BYE sip:100@192.168.10.70:5060 SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK94c49eefee330019a.e66972e74c83e2bc6 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 67]: From: ;tag=1768984948 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 61]: To: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 15]: CSeq: 29239 BYE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 28]: Supported: gruu, path, timer [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 31]: User-Agent: Aastra 55i/2.6.0.66 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (12 headers 0 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 (Checking From) --From tag 1768984948 --To-tag as45ace4d4 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Initializing initreq for method BYE - callid 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.200:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.200' and port '5060'. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.200:5060 (no NAT) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd809e70' [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog '4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70' in 32000 ms (Method: BYE) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Received bye, no owner, selfdestruct soon. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK94c49eefee330019a.e66972e74c83e2bc6;received=192.168.10.200 From: ;tag=1768984948 To: "Erika Musterfrau" ;tag=as45ace4d4 Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 CSeq: 29239 BYE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK94c49eefee330019a.e66972e74c83e2bc6;received=192.168.10.200 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 67]: From: ;tag=1768984948 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 61]: To: "Erika Musterfrau" ;tag=as45ace4d4 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 4e823f1b246ad18b58efb4e00b9f6c74@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 15]: CSeq: 29239 BYE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.200:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7eb4cf28 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 106 INVITE Retry-After: 9 Server: Aastra 51i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 33]: SIP/2.0 500 Internal Server Error [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7eb4cf28 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 106 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 14]: Retry-After: 9 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (9 headers 0 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (Checking To) --From tag as1a46d292 --To-tag 3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Acked pending invite 106 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #410 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Stopping retransmission on '43691a3867af054e1f075d71580a7d02@192.168.10.70' of Request 106: Match Found [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: -- Got SIP response 500 "Internal Server Error" back from 192.168.10.202:5060 [Nov 22 11:57:44] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7fdab8' [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Strict routing enforced for session 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.202:5060: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7eb4cf28 Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 106 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK7eb4cf28 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 106 ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Setting SIP_ALREADYGONE on dialog 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] DEBUG[1253] rtp_engine.c: Got a FRAME_CONTROL (8) frame on channel SIP/phone3-00000013 [Nov 22 11:57:44] DEBUG[1253] channel.c: Returning from native bridge, channels: SIP/phone1-00000010, SIP/phone3-00000013 [Nov 22 11:57:44] DEBUG[1253] channel.c: Hanging up channel 'SIP/phone3-00000013' [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Hangup call SIP/phone3-00000013, SIP callid 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: update_call_counter(phone3) - decrement call limit counter on hangup [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Updating call counter for outgoing call [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Call to peer 'phone3' removed from call limit 2147483647 [Nov 22 11:57:44] DEBUG[1253] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7fdab8' [Nov 22 11:57:44] DEBUG[1253] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 22 11:57:44] DEBUG[1253] pbx.c: Spawn extension (Standard,180,1) exited non-zero on 'SIP/phone1-00000010' [Nov 22 11:57:44] VERBOSE[1253] pbx.c: == Spawn extension (Standard, 180, 1) exited non-zero on 'SIP/phone1-00000010' [Nov 22 11:57:44] DEBUG[1253] channel.c: Soft-Hanging up channel 'SIP/phone1-00000010' [Nov 22 11:57:44] DEBUG[1253] channel.c: Hanging up channel 'SIP/phone1-00000010' [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Hangup call SIP/phone1-00000010, SIP callid 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: update_call_counter(phone1) - decrement call limit counter on hangup [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Updating call counter for incoming call [Nov 22 11:57:44] DEBUG[1253] chan_sip.c: Call from peer 'phone1' removed from call limit 2147483647 [Nov 22 11:57:44] DEBUG[1253] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd829cd8' [Nov 22 11:57:44] VERBOSE[1253] chan_sip.c: Scheduling destruction of SIP dialog '8bd610aef94fb8be' in 32000 ms (Method: ACK) [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone1-00000010 Channel2: SIP/phone3-00000013 Uniqueid1: 1290423460.18 Uniqueid2: 1290423460.19 CallerID1: 100 CallerID2: 180 [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 150 Destination: 180 DestinationContext: Standard CallerID: "Hans Muster" <150> Channel: SIP/phone2-00000012 DestinationChannel: SIP/phone3-00000013 LastApplication: Dial LastData: SIP/phone3 StartTime: 2010-11-22 11:57:40 AnswerTime: 2010-11-22 11:57:41 EndTime: 2010-11-22 11:57:44 Duration: 4 BillableSeconds: 3 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1290423460.18 UserField: [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone3-00000013 Uniqueid: 1290423460.19 CallerIDNum: 180 CallerIDName: Cause: 38 Cause-txt: Network out of order [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone1-00000010 UniqueID: 1290423460.18 DialStatus: ANSWER [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone1-00000010 Uniqueid: 1290423460.18 CallerIDNum: 100 CallerIDName: Erika Musterfrau Cause: 38 Cause-txt: Network out of order [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1b65f629 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 105 INVITE Server: Aastra 51i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1b65f629 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (8 headers 0 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (Checking To) --From tag as1a46d292 --To-tag 3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: *** SIP TIMER: Cancelling retransmission #409 - INVITE (got response) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '43691a3867af054e1f075d71580a7d02@192.168.10.70' Request 105: Found [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP response 100 to standard invite [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Got response on call that is already terminated: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (ignoring) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1b65f629 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Call-ID: 43691a3867af054e1f075d71580a7d02@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: "Max Mustermann" ;+sip.instance="" Server: Aastra 51i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 3 IN IP4 192.168.10.202 s=SIP Call c=IN IP4 192.168.10.202 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1b65f629 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.202 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.202 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (Checking To) --From tag as1a46d292 --To-tag 3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Stopping retransmission on '43691a3867af054e1f075d71580a7d02@192.168.10.70' of Request 105: Match Found [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP response 200 to standard invite [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Got response on call that is already terminated: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (ignoring) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK19c53b7f From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Call-ID: 8bd610aef94fb8be 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="" 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 192.168.10.201 s=SIP Call c=IN IP4 192.168.10.201 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK19c53b7f [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [138]: Contact: "Erika Musterfrau" ;+sip.instance="" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 4 IN IP4 192.168.10.201 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.201 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 8bd610aef94fb8be (Checking To) --From tag as37527f23 --To-tag bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Acked pending invite 105 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #411 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Stopping retransmission on '8bd610aef94fb8be' of Request 105: Match Found [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP response 200 to standard invite [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP o=MxSIP 0 4 IN IP4 192.168.10.201... UNSUPPORTED. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.201' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '(null)'. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.201... OK. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found RTP audio format 8 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Setting payload 8 based on m type on 0xb3381578 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found RTP audio format 0 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Setting payload 0 based on m type on 0xb3381578 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found RTP audio format 101 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Setting payload 101 based on m type on 0xb3381578 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found audio description format PCMA for ID 8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Incorporating payload 0 on 0xb3381578 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Incorporating payload 8 on 0xb3381578 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Incorporating payload 101 on 0xb3381578 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 22 11:57:44] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd829cd8' [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Peer audio RTP is at port 192.168.10.201:3000 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Copying payload 0 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Copying payload 8 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:44] DEBUG[1175] rtp_engine.c: Copying payload 101 from 0xb3381578 to 0xd829e84 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Updating call counter for incoming call [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: build_route: Retaining previous route: [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Strict routing enforced for session 8bd610aef94fb8be [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.201:5060: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK00e04f01 Max-Forwards: 70 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Contact: Call-ID: 8bd610aef94fb8be CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK00e04f01 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 105 ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Strict routing enforced for session 8bd610aef94fb8be [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.201:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.201' and port '5060'. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.201:5060 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.201:5060: BYE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK39b96121 Max-Forwards: 70 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Call-ID: 8bd610aef94fb8be 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 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 56]: BYE sip:phone1@192.168.10.201:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK39b96121 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 13]: CSeq: 106 BYE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 44]: X-Asterisk-HangupCause: Network out of order [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 38 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #414 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.201:5060 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog '8bd610aef94fb8be' in 32000 ms (Method: ACK) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '1' [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone3 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone3' state '1' [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '1' [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '1' [Nov 22 11:57:44] DEBUG[1167] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Nov 22 11:57:44] DEBUG[1167] chan_sip.c: Checking device state for peer phone1 [Nov 22 11:57:44] DEBUG[1167] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Nov 22 11:57:44] DEBUG[1167] devicestate.c: device 'SIP/phone1' state '1' [Nov 22 11:57:44] DEBUG[1168] app_queue.c: Extension '180@Standard' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1168] app_queue.c: Extension '100@Standard' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone3' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1194] app_queue.c: Device 'SIP/phone1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 0 [Nov 22 11:57:44] DEBUG[1195] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 0 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.200:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2b94d2ab From: "180" ;tag=as5a7bf510 To: "" ;tag=5ba829929a Call-ID: 2e4ec62ed2ac174d CSeq: 103 INVITE Server: Aastra 55i/2.6.0.66 Content-Length: 0 <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK2b94d2ab [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "180" ;tag=as5a7bf510 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 48]: To: "" ;tag=5ba829929a [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 2e4ec62ed2ac174d [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 27]: Server: Aastra 55i/2.6.0.66 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (8 headers 0 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 2e4ec62ed2ac174d (Checking To) --From tag as5a7bf510 --To-tag 5ba829929a [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: *** SIP TIMER: Cancelling retransmission #405 - INVITE (got response) [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2e4ec62ed2ac174d' Request 103: Found [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: SIP response 100 to standard invite [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK39b96121 From: "150" ;tag=as37527f23 To: "Erika Musterfrau" ;tag=bdabd584b8 Call-ID: 8bd610aef94fb8be CSeq: 106 BYE Server: Aastra 53i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK39b96121 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 50]: From: "150" ;tag=as37527f23 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 64]: To: "Erika Musterfrau" ;tag=bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 25]: Call-ID: 8bd610aef94fb8be [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 13]: CSeq: 106 BYE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 29]: Server: Aastra 53i/2.6.0.1008 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (8 headers 0 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 8bd610aef94fb8be (Checking To) --From tag as37527f23 --To-tag bdabd584b8 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #414 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Stopping retransmission on '8bd610aef94fb8be' of Request 106: Match Found [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1b65f629 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Call-ID: 43691a3867af054e1f075d71580a7d02@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: "Max Mustermann" ;+sip.instance="" Server: Aastra 51i/2.6.0.1008 Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 234 v=0 o=MxSIP 0 3 IN IP4 192.168.10.202 s=SIP Call c=IN IP4 192.168.10.202 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK1b65f629 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [136]: Contact: "Max Mustermann" ;+sip.instance="" [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 29]: Server: Aastra 51i/2.6.0.1008 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 38]: Supported: gruu, path, timer, replaces [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 12 [ 19]: Content-Length: 234 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 13 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 0 [ 3]: v=0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 1 [ 33]: o=MxSIP 0 3 IN IP4 192.168.10.202 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 2 [ 10]: s=SIP Call [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.10.202 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 4 [ 5]: t=0 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 5 [ 28]: m=audio 3000 RTP/AVP 8 0 101 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Body 11 [ 10]: a=sendrecv [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: --- (13 headers 12 lines) --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: = Looking for Call ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (Checking To) --From tag as1a46d292 --To-tag 3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Stopping retransmission on '43691a3867af054e1f075d71580a7d02@192.168.10.70' of Request 105: Match Not Found [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Strict routing enforced for session 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: Parsing for address/port to send to [Nov 22 11:57:44] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:44] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: set_destination: set destination to 192.168.10.202:5060 [Nov 22 11:57:44] VERBOSE[1175] chan_sip.c: Transmitting (no NAT) to 192.168.10.202:5060: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK58ce6718 Max-Forwards: 70 From: "Hans Muster" ;tag=as1a46d292 To: ;tag=3092808172 Contact: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.0-1 Content-Length: 0 --- [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 0 [ 56]: ACK sip:phone3@192.168.10.202:5060;transport=udp SIP/2.0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.10.70:5060;branch=z9hG4bK58ce6718 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 3 [ 58]: From: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 4 [ 65]: To: ;tag=3092808172 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 5 [ 37]: Contact: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 6 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 7 [ 13]: CSeq: 105 ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.0-1 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:44] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:45] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:45] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:46] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:46] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:47] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:47] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:48] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:48] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:49] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:49] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:50] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:50] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:51] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:51] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:52] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:52] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:53] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:53] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:54] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:54] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:55] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:55] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: INVITE [Nov 22 11:57:56] VERBOSE[1175] chan_sip.c: <--- SIP read from UDP:192.168.10.202:5060 ---> BYE sip:150@192.168.10.70:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKd8a467852cdb2855a.8e1c59d6df2de8a8f Max-Forwards: 70 From: ;tag=3092808172 To: "Hans Muster" ;tag=as1a46d292 Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 20119 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Supported: gruu, path, timer User-Agent: Aastra 51i/2.6.0.1008 Content-Length: 0 <-------------> [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 0 [ 38]: BYE sip:150@192.168.10.70:5060 SIP/2.0 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKd8a467852cdb2855a.8e1c59d6df2de8a8f [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 3 [ 67]: From: ;tag=3092808172 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 4 [ 56]: To: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 5 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 6 [ 15]: CSeq: 20119 BYE [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 7 [ 87]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 9 [ 28]: Supported: gruu, path, timer [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 10 [ 33]: User-Agent: Aastra 51i/2.6.0.1008 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Nov 22 11:57:56] VERBOSE[1175] chan_sip.c: --- (12 headers 0 lines) --- [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: = Looking for Call ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 (Checking From) --From tag 3092808172 --To-tag as1a46d292 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Initializing initreq for method BYE - callid 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:56] DEBUG[1175] netsock2.c: Splitting '192.168.10.202:5060' gives... [Nov 22 11:57:56] DEBUG[1175] netsock2.c: ...host '192.168.10.202' and port '5060'. [Nov 22 11:57:56] VERBOSE[1175] chan_sip.c: Sending to 192.168.10.202:5060 (no NAT) [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Setting SIP_ALREADYGONE on dialog 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:56] DEBUG[1175] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd7fdab8' [Nov 22 11:57:56] VERBOSE[1175] chan_sip.c: Scheduling destruction of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' in 32000 ms (Method: BYE) [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Received bye, no owner, selfdestruct soon. [Nov 22 11:57:56] VERBOSE[1175] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.202:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKd8a467852cdb2855a.8e1c59d6df2de8a8f;received=192.168.10.202 From: ;tag=3092808172 To: "Hans Muster" ;tag=as1a46d292 Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 CSeq: 20119 BYE Server: Asterisk PBX 1.8.0-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 1 [110]: Via: SIP/2.0/UDP 192.168.10.202:5060;branch=z9hG4bKd8a467852cdb2855a.8e1c59d6df2de8a8f;received=192.168.10.202 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 2 [ 67]: From: ;tag=3092808172 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 3 [ 56]: To: "Hans Muster" ;tag=as1a46d292 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 4 [ 55]: Call-ID: 43691a3867af054e1f075d71580a7d02@192.168.10.70 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 5 [ 15]: CSeq: 20119 BYE [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 6 [ 28]: Server: Asterisk PBX 1.8.0-1 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Header 10 [ 0]: [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.202:5060 [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:56] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:57:57] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:57] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:57:58] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:58] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:57:59] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:57:59] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:58:00] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:58:00] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:58:01] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:58:01] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:58:02] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:58:02] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:58:03] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:58:03] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:58:04] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:58:04] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: Auto destroying SIP dialog 'bddd1f752853e839' [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: Destroying SIP dialog bddd1f752853e839 [Nov 22 11:58:05] VERBOSE[1175] chan_sip.c: Really destroying SIP dialog 'bddd1f752853e839' Method: REGISTER [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: ---------- SIP HISTORY for 'bddd1f752853e839' [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: * SIP Call [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: 001. Rx REGISTER / 9010 REGISTER / sip:192.168.10.70 [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: 002. TxResp SIP/2.0 / 9010 REGISTER - 200 OK [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: 003. RegRequest Succeeded : Account [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: 004. SchedDestroy 32000 ms [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: 005. AutoDestroy bddd1f752853e839 [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: ---------- END SIP HISTORY for 'bddd1f752853e839' [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:58:05] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:58:06] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:58:06] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:58:07] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:58:07] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE [Nov 22 11:58:08] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '8bd610aef94fb8be' Method: ACK [Nov 22 11:58:08] DEBUG[1175] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '43691a3867af054e1f075d71580a7d02@192.168.10.70' Method: BYE