[Oct 22 11:09:04] VERBOSE[1874] config.c: == Parsing '/etc/asterisk/logger.conf': [Oct 22 11:09:04] DEBUG[1874] config.c: Parsing /etc/asterisk/logger.conf [Oct 22 11:09:04] VERBOSE[1874] config.c: == Found [Oct 22 11:09:04] VERBOSE[1874] logger.c: Asterisk Queue Logger restarted [Oct 22 11:09:08] VERBOSE[1852] chan_sip.c: Reloading SIP [Oct 22 11:09:08] VERBOSE[1852] config.c: == Parsing '/etc/asterisk/sip.conf': [Oct 22 11:09:08] DEBUG[1852] config.c: Parsing /etc/asterisk/sip.conf [Oct 22 11:09:08] VERBOSE[1852] config.c: == Found [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: --------------- SIP reload started [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: --------------- Done destroying registry list [Oct 22 11:09:08] DEBUG[1852] config.c: extract addr from 0.0.0.0 gives 0.0.0.0:0(0) [Oct 22 11:09:08] VERBOSE[1852] netsock2.c: == Using SIP TOS bits 96 [Oct 22 11:09:08] VERBOSE[1852] netsock2.c: == Using SIP CoS mark 4 [Oct 22 11:09:08] DEBUG[1852] tcptls.c: Nothing changed in SIP TCP server [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: SIP TCP server started [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: SIP Seeding peer from astdb: 'phone1' at phone1@192.168.10.203 for 3600 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Allocating new SIP dialog for 448042b17b91e13e6e960efc626f57d9@192.168.10.70 - NOTIFY (No RTP) [Oct 22 11:09:08] DEBUG[1852] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Initializing initreq for method NOTIFY - callid 461db49054eaf2221789536f49a184ac@192.168.10.70 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: SIP Seeding peer from astdb: 'phone2' at phone2@192.168.10.204 for 3600 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Allocating new SIP dialog for 172633db53479f3b4e611d35741a5e1a@192.168.10.70 - NOTIFY (No RTP) [Oct 22 11:09:08] DEBUG[1852] acl.c: For destination '192.168.10.204', our source address is '192.168.10.70'. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Initializing initreq for method NOTIFY - callid 1d03858063ae01f91faf965233d97c22@192.168.10.70 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: SIP Seeding peer from astdb: 'phone3' at phone3@192.168.10.206 for 3600 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Allocating new SIP dialog for 5ad885ba71f3c871606726e765fc1bfe@192.168.10.70 - NOTIFY (No RTP) [Oct 22 11:09:08] DEBUG[1852] acl.c: For destination '192.168.10.206', our source address is '192.168.10.70'. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Initializing initreq for method NOTIFY - callid 6211fcee4a637a9a7dd9614f54698107@192.168.10.70 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.206:2050 [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: SIP reload_config done...Runtime= 0 sec [Oct 22 11:09:08] DEBUG[1852] sched.c: Asterisk Schedule Dump (9 in Q, 226 Total, 4 Cache, 13 high-water) [Oct 22 11:09:08] DEBUG[1852] sched.c: ============================================================= [Oct 22 11:09:08] DEBUG[1852] sched.c: |ID Callback Data Time (sec:ms) | [Oct 22 11:09:08] DEBUG[1852] sched.c: +-----+-----------------+-----------------+-----------------+ [Oct 22 11:09:08] DEBUG[1852] sched.c: |0220 | 0xb3a82a80 | 0xcd2c438 | 000000 : 490818 | [Oct 22 11:09:08] DEBUG[1852] sched.c: |0226 | 0xb3a82a80 | 0xcd203d0 | 000000 : 495115 | [Oct 22 11:09:08] DEBUG[1852] sched.c: |0223 | 0xb3a82a80 | 0xcd2d488 | 000000 : 494032 | [Oct 22 11:09:08] DEBUG[1852] sched.c: |0219 | 0xb3a82530 | 0xcd0d908 | 000031 : 990749 | [Oct 22 11:09:08] DEBUG[1852] sched.c: |0221 | 0xb3a26490 | 0x98ee3f8 | 003609 : 993328 | [Oct 22 11:09:08] DEBUG[1852] sched.c: |0218 | 0xb3a26490 | 0x98ec1a8 | 003609 : 990383 | [Oct 22 11:09:08] DEBUG[1852] sched.c: |0224 | 0xb3a26490 | 0x98ef700 | 003609 : 994772 | [Oct 22 11:09:08] DEBUG[1852] sched.c: |0225 | 0xb3a82530 | 0xcd20ce8 | 000031 : 995091 | [Oct 22 11:09:08] DEBUG[1852] sched.c: |0222 | 0xb3a82530 | 0xcd1bd20 | 000031 : 993769 | [Oct 22 11:09:08] DEBUG[1852] sched.c: ============================================================= [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: --------------- Done destroying pruned peers [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: do_reload finished. peer poke/prune reg contact time = 0 sec. [Oct 22 11:09:08] DEBUG[1852] chan_sip.c: --------------- SIP reload done [Oct 22 11:09:08] DEBUG[1854] manager.c: Examining event: Event: ChannelReload Privilege: system,all ChannelType: SIP ReloadReason: CLIRELOAD (Channel module reload by CLI command) Registry_Count: 0 Peer_Count: 3 [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: Stopping retransmission on '461db49054eaf2221789536f49a184ac@192.168.10.70' of Request 102: Match Found [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: Got 200 accepted on NOTIFY [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: Destroying SIP dialog 461db49054eaf2221789536f49a184ac@192.168.10.70 [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: ---------- SIP HISTORY for '461db49054eaf2221789536f49a184ac@192.168.10.70' [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: * SIP Call [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 001. SchedDestroy 32000 ms [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 002. TxReqRel NOTIFY / 102 NOTIFY - NOTIFY [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 003. Rx SIP/2.0 / 102 NOTIFY / 200 Ok [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 004. NeedDestroy Setting needdestroy because received 200 response [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: ---------- END SIP HISTORY for '461db49054eaf2221789536f49a184ac@192.168.10.70' [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: Stopping retransmission on '6211fcee4a637a9a7dd9614f54698107@192.168.10.70' of Request 102: Match Found [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: Got 200 accepted on NOTIFY [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: Destroying SIP dialog 6211fcee4a637a9a7dd9614f54698107@192.168.10.70 [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: ---------- SIP HISTORY for '6211fcee4a637a9a7dd9614f54698107@192.168.10.70' [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: * SIP Call [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 001. SchedDestroy 32000 ms [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 002. TxReqRel NOTIFY / 102 NOTIFY - NOTIFY [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 003. Rx SIP/2.0 / 102 NOTIFY / 200 Ok [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 004. NeedDestroy Setting needdestroy because received 200 response [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: ---------- END SIP HISTORY for '6211fcee4a637a9a7dd9614f54698107@192.168.10.70' [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: Stopping retransmission on '1d03858063ae01f91faf965233d97c22@192.168.10.70' of Request 102: Match Found [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: Got 200 accepted on NOTIFY [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: Destroying SIP dialog 1d03858063ae01f91faf965233d97c22@192.168.10.70 [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: ---------- SIP HISTORY for '1d03858063ae01f91faf965233d97c22@192.168.10.70' [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: * SIP Call [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 001. SchedDestroy 32000 ms [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 002. TxReqRel NOTIFY / 102 NOTIFY - NOTIFY [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 003. Rx SIP/2.0 / 102 NOTIFY / 200 Ok [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: 004. NeedDestroy Setting needdestroy because received 200 response [Oct 22 11:09:09] DEBUG[1852] chan_sip.c: ---------- END SIP HISTORY for '1d03858063ae01f91faf965233d97c22@192.168.10.70' [Oct 22 11:09:19] DEBUG[1852] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Allocating new SIP dialog for 3c267702d060-9et3ydbvv5tx - INVITE (No RTP) [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 22 11:09:19] DEBUG[1852] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Oct 22 11:09:19] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 22 11:09:19] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: timer [Oct 22 11:09:19] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 22 11:09:19] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 22 11:09:19] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 22 11:09:19] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 22 11:09:19] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -from-change- [Oct 22 11:09:19] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: from-change [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcd1b530' [Oct 22 11:09:19] DEBUG[1852] res_rtp_asterisk.c: Allocated port 17264 for RTP instance '0xcd1b530' [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: RTP instance '0xcd1b530' is setup and ready to go [Oct 22 11:09:19] DEBUG[1852] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcd1b530' [Oct 22 11:09:19] VERBOSE[1852] netsock2.c: == Using SIP RTP TOS bits 184 [Oct 22 11:09:19] VERBOSE[1852] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Setting NAT on RTP to Off [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1799815998 1799815998 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33a9f88 [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33a9f88 [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33a9f88 [Oct 22 11:09:19] ERROR[1852] chan_sip.c: No SRTP module loaded, can't setup SRTP session. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Y14eeL5VxUMtgW/Gm8mvofrm32tA6H4nptmOyWY8... UNSUPPORTED. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33a9f88 [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33a9f88 [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33a9f88 [Oct 22 11:09:19] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:09:19] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Checking SIP call limits for device phone1 [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: *** Our native formats are 0x8 (alaw) [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: This channel will not be able to handle video. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Session-Expires: 3600 [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Refresher: UAS [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Received Min-SE: 90 [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Session timer started: 230 - 3c267702d060-9et3ydbvv5tx [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: SIP/phone1-0000000d: New call is still down.... Trying... [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:19] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:09:19] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:09:19] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:09:19] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:09:19] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:09:19] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:09:19] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:09:19] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:09:19] DEBUG[1854] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone1-0000000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Erika Musterfrau AccountCode: Exten: 150 Context: Standard Uniqueid: 1287738559.14 [Oct 22 11:09:19] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-0000000d ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1287738559.14 [Oct 22 11:09:19] DEBUG[1927] pbx.c: Launching 'Dial' [Oct 22 11:09:19] VERBOSE[1927] pbx.c: -- Executing [150@Standard:1] Dial("SIP/phone1-0000000d", "SIP/phone2") in new stack [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Allocating new SIP dialog for 387b453f12d5e10b5c58684b692fbeb1@192.168.10.70 - INVITE (No RTP) [Oct 22 11:09:19] DEBUG[1927] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcd20ce8' [Oct 22 11:09:19] DEBUG[1927] res_rtp_asterisk.c: Allocated port 16194 for RTP instance '0xcd20ce8' [Oct 22 11:09:19] DEBUG[1927] rtp_engine.c: RTP instance '0xcd20ce8' is setup and ready to go [Oct 22 11:09:19] DEBUG[1927] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcd20ce8' [Oct 22 11:09:19] VERBOSE[1927] netsock2.c: == Using SIP RTP TOS bits 184 [Oct 22 11:09:19] VERBOSE[1927] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Setting NAT on RTP to Off [Oct 22 11:09:19] DEBUG[1927] acl.c: For destination '192.168.10.204', our source address is '192.168.10.70'. [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: *** Our native formats are 0x8 (alaw) [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: This channel will not be able to handle video. [Oct 22 11:09:19] DEBUG[1927] rtp_engine.c: Seeded SDP of 'SIP/phone2-0000000e' with that of 'SIP/phone1-0000000d' [Oct 22 11:09:19] DEBUG[1927] channel.c: Not copying variable DIALEDTIME. [Oct 22 11:09:19] DEBUG[1927] channel.c: Not copying variable ANSWEREDTIME. [Oct 22 11:09:19] DEBUG[1927] channel.c: Not copying variable DIALEDPEERNAME. [Oct 22 11:09:19] DEBUG[1927] channel.c: Not copying variable DIALEDPEERNUMBER. [Oct 22 11:09:19] DEBUG[1927] channel.c: Not copying variable DIALSTATUS. [Oct 22 11:09:19] DEBUG[1927] channel.c: Not copying variable SIPCALLID. [Oct 22 11:09:19] DEBUG[1927] channel.c: Not copying variable SIPDOMAIN. [Oct 22 11:09:19] DEBUG[1927] channel.c: Not copying variable SIPURI. [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Outgoing Call for phone2 [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Initializing initreq for method INVITE - callid 0642fc08466be56c26b98a5562bb4321@192.168.10.70 [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:19] VERBOSE[1927] app_dial.c: -- Called phone2 [Oct 22 11:09:19] DEBUG[1854] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-0000000e ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 150 CallerIDName: Hans Muster AccountCode: Exten: Context: Standard Uniqueid: 1287738559.15 [Oct 22 11:09:19] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone1-0000000d Destination: SIP/phone2-0000000e CallerIDNum: 100 CallerIDName: Erika Musterfrau UniqueID: 1287738559.14 DestUniqueID: 1287738559.15 Dialstring: phone2 [Oct 22 11:09:19] DEBUG[1854] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone2-0000000e CallerIDNum: 150 CallerIDName: Uniqueid: 1287738559.15 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Oct 22 11:09:19] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:09:19] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:09:19] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Oct 22 11:09:19] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '6' [Oct 22 11:09:19] DEBUG[1846] 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. [Oct 22 11:09:19] DEBUG[1872] 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. [Oct 22 11:09:19] DEBUG[1872] 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. [Oct 22 11:09:19] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 1 [Oct 22 11:09:19] DEBUG[1846] 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. [Oct 22 11:09:19] DEBUG[1872] 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. [Oct 22 11:09:19] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 8 [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Request 102: Found [Oct 22 11:09:19] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:09:19] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:09:19] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:09:19] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Oct 22 11:09:19] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '6' [Oct 22 11:09:19] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-0000000e ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 150 CallerIDName: Uniqueid: 1287738559.15 [Oct 22 11:09:19] VERBOSE[1927] app_dial.c: -- SIP/phone2-0000000e is ringing [Oct 22 11:09:19] DEBUG[1927] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-0000000d' with that of 'SIP/phone2-0000000e' [Oct 22 11:09:19] DEBUG[1927] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:19] DEBUG[1872] 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. [Oct 22 11:09:20] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Request 102: Found [Oct 22 11:09:20] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:09:20] VERBOSE[1927] app_dial.c: -- SIP/phone2-0000000e is ringing [Oct 22 11:09:20] DEBUG[1927] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-0000000d' with that of 'SIP/phone2-0000000e' [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Request 102: Found [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:09:21] VERBOSE[1927] app_dial.c: -- SIP/phone2-0000000e is ringing [Oct 22 11:09:21] DEBUG[1927] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-0000000d' with that of 'SIP/phone2-0000000e' [Oct 22 11:09:21] DEBUG[1927] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:09:21] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.204:50541 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 16777216 FractionLost: 0 PacketsLost: 1 HighestSequence: 0 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Oct 22 11:09:21] DEBUG[1927] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd20ce8' [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Acked pending invite 102 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Stopping retransmission on '0642fc08466be56c26b98a5562bb4321@192.168.10.70' of Request 102: Match Found [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: SIP response 200 to standard invite [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1514094839 1514094840 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd20ce8' [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd20e94 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd20e94 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd20e94 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:21] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:09:21] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:09:21] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:09:21] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:09:21] VERBOSE[1927] app_dial.c: -- SIP/phone2-0000000e answered SIP/phone1-0000000d [Oct 22 11:09:21] DEBUG[1927] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-0000000d' with that of 'SIP/phone2-0000000e' [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: SIP answering channel: SIP/phone1-0000000d [Oct 22 11:09:21] DEBUG[1927] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: Setting framing from config on incoming call [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:21] DEBUG[1927] features.c: bridge answer set, chan answer set [Oct 22 11:09:21] DEBUG[1927] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 22 11:09:21] DEBUG[1927] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 22 11:09:21] VERBOSE[1927] rtp_engine.c: -- Remotely bridging SIP/phone1-0000000d and SIP/phone2-0000000e [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: Deferring reinvite on SIP '3c267702d060-9et3ydbvv5tx' - It's audio will be redirected to IP 192.168.10.204:50540 [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: Sending reinvite on SIP '0642fc08466be56c26b98a5562bb4321@192.168.10.70' - It's audio soon redirected to IP 192.168.10.203:56506 [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: Initializing already initialized SIP dialog 0642fc08466be56c26b98a5562bb4321@192.168.10.70 (presumably reinvite) [Oct 22 11:09:21] DEBUG[1927] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:21] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-0000000e ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 150 CallerIDName: Uniqueid: 1287738559.15 [Oct 22 11:09:21] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-0000000d ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1287738559.14 [Oct 22 11:09:21] DEBUG[1854] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone2-0000000e Uniqueid: 1287738559.15 AccountCode: OldAccountCode: [Oct 22 11:09:21] DEBUG[1854] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone1-0000000d Channel2: SIP/phone2-0000000e Uniqueid1: 1287738559.14 Uniqueid2: 1287738559.15 CallerID1: 100 CallerID2: 150 [Oct 22 11:09:21] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:09:21] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:09:21] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:09:21] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:09:21] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:09:21] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:09:21] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:09:21] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: INVITE [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: INVITE [Oct 22 11:09:21] DEBUG[1846] 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. [Oct 22 11:09:21] DEBUG[1872] 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. [Oct 22 11:09:21] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 1 [Oct 22 11:09:21] DEBUG[1872] 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. [Oct 22 11:09:21] DEBUG[1872] 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. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267702d060-9et3ydbvv5tx' of Response 1: Match Found [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Sending pending reinvite on '3c267702d060-9et3ydbvv5tx' [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Initializing already initialized SIP dialog 3c267702d060-9et3ydbvv5tx (presumably reinvite) [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: INVITE [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:21] DEBUG[1927] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:09:21] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.203:56507 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 16777216 FractionLost: 0 PacketsLost: 1 HighestSequence: 0 SequenceNumberCycles: 0 IAJitter: 0 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Acked pending invite 103 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Stopping retransmission on '0642fc08466be56c26b98a5562bb4321@192.168.10.70' of Request 103: Match Found [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 0642fc08466be56c26b98a5562bb4321@192.168.10.70 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1514094839 1514094841 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd20ce8' [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd20e94 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd20e94 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd20e94 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: INVITE [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:21] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:09:21] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:09:21] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:09:21] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:09:21] DEBUG[1872] 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. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Acked pending invite 102 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267702d060-9et3ydbvv5tx' of Request 102: Match Found [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3c267702d060-9et3ydbvv5tx [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1799815998 1799815999 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:09:21] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:09:21] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: INVITE [Oct 22 11:09:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:21] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:09:21] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:09:21] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:09:21] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:09:21] DEBUG[1872] 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. [Oct 22 11:09:22] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: INVITE [Oct 22 11:09:22] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 22 11:09:23] DEBUG[1852] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Oct 22 11:09:23] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 22 11:09:23] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: timer [Oct 22 11:09:23] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 22 11:09:23] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 22 11:09:23] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 22 11:09:23] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 22 11:09:23] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -from-change- [Oct 22 11:09:23] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: from-change [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1514094839 1514094842 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33a9f88 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33a9f88 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33a9f88 [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33a9f88 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33a9f88 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33a9f88 [Oct 22 11:09:23] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd20ce8' [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33a9f88 to 0xcd20e94 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33a9f88 to 0xcd20e94 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33a9f88 to 0xcd20e94 [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:09:23] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd20ce8' [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Session-Expires: 3600 [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Refresher: UAS [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Received Min-SE: 90 [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: SIP/phone2-0000000e: This call is UP.... [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Setting framing from config on incoming call [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: INVITE [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:23] DEBUG[1927] chan_sip.c: Sending reinvite on SIP '3c267702d060-9et3ydbvv5tx' - It's audio soon redirected to IP 192.168.10.70:5060 [Oct 22 11:09:23] DEBUG[1927] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:09:23] DEBUG[1927] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:09:23] DEBUG[1927] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:09:23] DEBUG[1927] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:09:23] DEBUG[1927] chan_sip.c: Initializing already initialized SIP dialog 3c267702d060-9et3ydbvv5tx (presumably reinvite) [Oct 22 11:09:23] DEBUG[1927] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:09:23] VERBOSE[1927] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/phone1-0000000d [Oct 22 11:09:23] DEBUG[1927] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: INVITE [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:23] DEBUG[1854] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/phone1-0000000d UniqueID: 1287738559.14 Class: default [Oct 22 11:09:23] DEBUG[1927] channel.c: Set channel SIP/phone1-0000000d to write format slin [Oct 22 11:09:23] DEBUG[1927] res_musiconhold.c: SIP/phone1-0000000d Opened file 0 '/var/lib/asterisk/moh/iptam/iptam_moh' [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xcd1b530' [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:09:23] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.203:56507 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 3600810239 FractionLost: 255 PacketsLost: 41174 HighestSequence: 17371 SequenceNumberCycles: 0 IAJitter: 894792 LastSR: 54081.2952790016 DLSR: 1.5200(sec) RTT: 504(sec) [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Acked pending invite 103 [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267702d060-9et3ydbvv5tx' of Request 103: Match Found [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3c267702d060-9et3ydbvv5tx [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1799815998 1799816000 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:09:23] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:09:23] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: INVITE [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:23] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:09:23] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:09:23] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:09:23] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:09:23] DEBUG[1872] 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. [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Stopping retransmission on '0642fc08466be56c26b98a5562bb4321@192.168.10.70' of Response 1: Match Found [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: ACK [Oct 22 11:09:23] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:23] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: ACK [Oct 22 11:09:24] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:24] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: ACK [Oct 22 11:09:25] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:25] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: ACK [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: chan1->name: SIP/phone2-0000000e [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:26] DEBUG[1852] channel.c: Soft-Hanging up channel 'SIP/phone1-0000000d' [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Blind transfer succeeded. Telling transferer. [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' Method: REFER [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267702d060-9et3ydbvv5tx' Method: ACK [Oct 22 11:09:26] DEBUG[1854] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Blind Channel: SIP/phone2-0000000e Uniqueid: 1287738559.15 SIP-Callid: 0642fc08466be56c26b98a5562bb4321@192.168.10.70 TargetChannel: SIP/phone1-0000000d TargetUniqueid: 1287738559.14 TransferExten: 180 TransferContext: Standard [Oct 22 11:09:26] VERBOSE[1927] res_musiconhold.c: -- Stopped music on hold on SIP/phone1-0000000d [Oct 22 11:09:26] DEBUG[1927] channel.c: Set channel SIP/phone1-0000000d to write format alaw [Oct 22 11:09:26] DEBUG[1927] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 22 11:09:26] DEBUG[1927] rtp_engine.c: Oooh, got a hangup [Oct 22 11:09:26] DEBUG[1927] channel.c: Returning from native bridge, channels: SIP/phone1-0000000d, SIP/phone2-0000000e [Oct 22 11:09:26] DEBUG[1927] channel.c: Hanging up channel 'SIP/phone2-0000000e' [Oct 22 11:09:26] DEBUG[1927] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:09:26] DEBUG[1927] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 0642fc08466be56c26b98a5562bb4321@192.168.10.70. [Oct 22 11:09:26] DEBUG[1927] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Oct 22 11:09:26] DEBUG[1927] pbx.c: Spawn extension (Standard,180,1) exited non-zero on 'SIP/phone1-0000000d' [Oct 22 11:09:26] VERBOSE[1927] pbx.c: == Spawn extension (Standard, 180, 1) exited non-zero on 'SIP/phone1-0000000d' [Oct 22 11:09:26] DEBUG[1927] channel.c: Soft-Hanging up channel 'SIP/phone1-0000000d' [Oct 22 11:09:26] DEBUG[1927] channel.c: Hanging up channel 'SIP/phone1-0000000d' [Oct 22 11:09:26] DEBUG[1927] chan_sip.c: Hangup call SIP/phone1-0000000d, SIP callid 3c267702d060-9et3ydbvv5tx [Oct 22 11:09:26] DEBUG[1927] chan_sip.c: Updating call counter for incoming call [Oct 22 11:09:26] DEBUG[1927] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:09:26] DEBUG[1927] chan_sip.c: Session timer stopped: -1 - 3c267702d060-9et3ydbvv5tx [Oct 22 11:09:26] DEBUG[1927] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:26] DEBUG[1854] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone1-0000000d Channel2: SIP/phone2-0000000e Uniqueid1: 1287738559.14 Uniqueid2: 1287738559.15 CallerID1: 100 CallerID2: 150 [Oct 22 11:09:26] DEBUG[1854] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 100 Destination: 150 DestinationContext: Standard CallerID: "Erika Musterfrau" <100> Channel: SIP/phone1-0000000d DestinationChannel: SIP/phone2-0000000e LastApplication: Dial LastData: SIP/phone2 StartTime: 2010-10-22 11:09:19 AnswerTime: 2010-10-22 11:09:21 EndTime: 2010-10-22 11:09:26 Duration: 7 BillableSeconds: 5 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1287738559.14 UserField: [Oct 22 11:09:26] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-0000000e Uniqueid: 1287738559.15 CallerIDNum: 150 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Oct 22 11:09:26] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone1-0000000d UniqueID: 1287738559.14 DialStatus: ANSWER [Oct 22 11:09:26] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone1-0000000d Uniqueid: 1287738559.14 CallerIDNum: 100 CallerIDName: Erika Musterfrau Cause: 16 Cause-txt: Normal Clearing [Oct 22 11:09:26] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:09:26] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:09:26] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:09:26] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:09:26] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:09:26] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:09:26] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:09:26] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:09:26] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:09:26] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:09:26] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Oct 22 11:09:26] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '1' [Oct 22 11:09:26] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:09:26] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:09:26] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Oct 22 11:09:26] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '1' [Oct 22 11:09:26] DEBUG[1846] 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. [Oct 22 11:09:26] DEBUG[1846] 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. [Oct 22 11:09:26] DEBUG[1872] 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. [Oct 22 11:09:26] DEBUG[1872] 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. [Oct 22 11:09:26] DEBUG[1872] 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. [Oct 22 11:09:26] DEBUG[1872] 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. [Oct 22 11:09:26] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 0 [Oct 22 11:09:26] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 0 [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267702d060-9et3ydbvv5tx' of Request 104: Match Found [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Destroying SIP dialog 3c267702d060-9et3ydbvv5tx [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: ---------- SIP HISTORY for '3c267702d060-9et3ydbvv5tx' [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: * SIP Call [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 001. Rx INVITE / 1 INVITE / sip:150@192.168.10.70;user=phone [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 002. NewChan Channel SIP/phone1-0000000d - from 3c267702d060-9et3ydbvv5tx [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 003. TxResp SIP/2.0 / 1 INVITE - 100 Trying [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 004. TxResp SIP/2.0 / 1 INVITE - 180 Ringing [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 005. ConnectedLine Called party is now Hans Muster <150> [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 006. TxRespRel SIP/2.0 / 1 INVITE - 200 OK [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 007. Rx ACK / 1 ACK / sip:150@192.168.10.70:5060 [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 008. ReInv Re-invite sent [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 009. TxReqRel INVITE / 102 INVITE - INVITE [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 010. Rx SIP/2.0 / 102 INVITE / 200 Ok [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 011. TxReq ACK / 102 ACK - ACK [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 012. ReInv Re-invite sent [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 013. TxReqRel INVITE / 103 INVITE - INVITE [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 014. Rx SIP/2.0 / 103 INVITE / 200 Ok [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 015. TxReq ACK / 103 ACK - ACK [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 016. Hangup Cause Normal Clearing [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 017. SchedDestroy 32000 ms [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 018. RTCPaudio Quality:ssrc=86636644;themssrc=0;lp=0;rxjitter=0.000000;rxcount [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 019. TxReqRel BYE / 104 BYE - BYE [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 020. Rx SIP/2.0 / 104 BYE / 200 OK [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: 021. NeedDestroy Setting needdestroy because received 200 response [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: ---------- END SIP HISTORY for '3c267702d060-9et3ydbvv5tx' [Oct 22 11:09:26] DEBUG[1852] rtp_engine.c: Destroyed RTP instance '0xcd1b530' [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Stopping retransmission on '0642fc08466be56c26b98a5562bb4321@192.168.10.70' of Request 104: Match Found [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Stopping retransmission on '0642fc08466be56c26b98a5562bb4321@192.168.10.70' of Request 105: Match Found [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0642fc08466be56c26b98a5562bb4321@192.168.10.70 [Oct 22 11:09:26] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd20ce8' [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Received bye, no owner, selfdestruct soon. [Oct 22 11:09:26] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:44] DEBUG[1852] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Oct 22 11:09:44] DEBUG[1852] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:09:44] DEBUG[1852] chan_sip.c: Allocating new SIP dialog for 3c267015c127-g1d1291ldshy - REGISTER (No RTP) [Oct 22 11:09:44] DEBUG[1852] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 22 11:09:44] DEBUG[1852] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 22 11:09:44] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:09:44] DEBUG[1854] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone1 PeerStatus: Registered Address: 192.168.10.203:2051 [Oct 22 11:09:44] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:09:44] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:09:44] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Oct 22 11:09:44] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '1' [Oct 22 11:09:44] DEBUG[1872] 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. [Oct 22 11:09:56] DEBUG[1852] acl.c: For destination '192.168.10.204', our source address is '192.168.10.70'. [Oct 22 11:09:56] DEBUG[1852] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:09:56] DEBUG[1852] chan_sip.c: Allocating new SIP dialog for 3c267015e83a-xl6vwqz6alxx - REGISTER (No RTP) [Oct 22 11:09:56] DEBUG[1852] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 22 11:09:56] DEBUG[1852] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 22 11:09:56] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:09:56] DEBUG[1854] manager.c: Examining event: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/phone2 PeerStatus: Registered Address: 192.168.10.204:2050 [Oct 22 11:09:56] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:09:56] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:09:56] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:09:56] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:09:56] DEBUG[1872] 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. [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: Auto destroying SIP dialog '0642fc08466be56c26b98a5562bb4321@192.168.10.70' [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: Destroying SIP dialog 0642fc08466be56c26b98a5562bb4321@192.168.10.70 [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: This call did not properly clean up call limits. Call ID 0642fc08466be56c26b98a5562bb4321@192.168.10.70 [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: ---------- SIP HISTORY for '0642fc08466be56c26b98a5562bb4321@192.168.10.70' [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: * SIP Call [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 001. NewChan Channel SIP/phone2-0000000e - from 0642fc08466be56c26b98a5562bb [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 005. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 006. Rx SIP/2.0 / 102 INVITE / 200 Ok [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 007. TxReq ACK / 102 ACK - ACK [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 008. ReInv Re-invite sent [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 009. TxReqRel INVITE / 103 INVITE - INVITE [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 010. Rx SIP/2.0 / 103 INVITE / 200 Ok [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 011. TxReq ACK / 103 ACK - ACK [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 012. Rx INVITE / 1 INVITE / sip:100@192.168.10.70:5060 [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 013. Hold INVITE [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 014. ReInv Re-invite received [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 015. TxResp SIP/2.0 / 1 INVITE - 100 Trying [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 016. TxRespRel SIP/2.0 / 1 INVITE - 200 OK [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 017. Rx ACK / 1 ACK / sip:100@192.168.10.70:5060 [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 018. Rx REFER / 2 REFER / sip:100@192.168.10.70:5060 [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 019. TxResp SIP/2.0 / 2 REFER - 202 Accepted [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 020. TxReqRel NOTIFY / 104 NOTIFY - NOTIFY [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 021. TxReqRel NOTIFY / 105 NOTIFY - NOTIFY [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 022. Xfer Refer succeeded. [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 023. SchedDestroy 32000 ms [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 024. Rx SIP/2.0 / 104 NOTIFY / 200 Ok [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 025. Rx SIP/2.0 / 105 NOTIFY / 200 Ok [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 026. Rx BYE / 3 BYE / sip:100@192.168.10.70:5060 [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 027. RTCPaudio Quality:ssrc=295137925;themssrc=0;lp=0;rxjitter=0.000000;rxcoun [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 028. RTCPaudioJitter Quality:minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0 [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 029. RTCPaudioLoss Quality:minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.00000 [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 030. RTCPaudioRTT Quality:minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrt [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 031. CancelDestroy [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 032. SchedDestroy 32000 ms [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 033. TxResp SIP/2.0 / 3 BYE - 200 OK [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: 034. AutoDestroy 0642fc08466be56c26b98a5562bb4321@192.168.10.70 [Oct 22 11:09:58] DEBUG[1852] chan_sip.c: ---------- END SIP HISTORY for '0642fc08466be56c26b98a5562bb4321@192.168.10.70' [Oct 22 11:09:58] DEBUG[1852] rtp_engine.c: Destroyed RTP instance '0xcd20ce8' [Oct 22 11:09:58] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:09:58] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:09:58] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:09:58] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:09:59] DEBUG[1872] 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.