[Oct 22 11:22:32] DEBUG[1852] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Allocating new SIP dialog for 3c267a1d16d4-8nazi7ekl0nk - INVITE (No RTP) [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 22 11:22:32] DEBUG[1852] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Oct 22 11:22:32] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 22 11:22:32] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: timer [Oct 22 11:22:32] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 22 11:22:32] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 22 11:22:32] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 22 11:22:32] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 22 11:22:32] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -from-change- [Oct 22 11:22:32] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: from-change [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcd1b530' [Oct 22 11:22:32] DEBUG[1852] res_rtp_asterisk.c: Allocated port 16280 for RTP instance '0xcd1b530' [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: RTP instance '0xcd1b530' is setup and ready to go [Oct 22 11:22:32] DEBUG[1852] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcd1b530' [Oct 22 11:22:32] VERBOSE[1852] netsock2.c: == Using SIP RTP TOS bits 184 [Oct 22 11:22:32] VERBOSE[1852] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Setting NAT on RTP to Off [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1541909648 1541909648 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33a9f88 [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33a9f88 [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33a9f88 [Oct 22 11:22:32] ERROR[1852] chan_sip.c: No SRTP module loaded, can't setup SRTP session. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:9tkVWbKzoT8SaVR2CRzkG+/mrOudjt/PkX0g77kb... UNSUPPORTED. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33a9f88 [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33a9f88 [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33a9f88 [Oct 22 11:22:32] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:22:32] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Checking SIP call limits for device phone1 [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: *** Our native formats are 0x8 (alaw) [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: This channel will not be able to handle video. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Session-Expires: 3600 [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Refresher: UAS [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Received Min-SE: 90 [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Session timer started: 253 - 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: SIP/phone1-0000000f: New call is still down.... Trying... [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:32] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:22:32] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:22:32] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:22:32] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:22:32] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:22:32] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:22:32] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:22:32] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:22:32] DEBUG[1854] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone1-0000000f ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Erika Musterfrau AccountCode: Exten: 150 Context: Standard Uniqueid: 1287739352.16 [Oct 22 11:22:32] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-0000000f ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1287739352.16 [Oct 22 11:22:32] DEBUG[1945] pbx.c: Launching 'Dial' [Oct 22 11:22:32] VERBOSE[1945] pbx.c: -- Executing [150@Standard:1] Dial("SIP/phone1-0000000f", "SIP/phone2") in new stack [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Allocating new SIP dialog for 64920f474be527424ec9f40044df4734@192.168.10.70 - INVITE (No RTP) [Oct 22 11:22:32] DEBUG[1945] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcd1ec38' [Oct 22 11:22:32] DEBUG[1945] res_rtp_asterisk.c: Allocated port 12712 for RTP instance '0xcd1ec38' [Oct 22 11:22:32] DEBUG[1945] rtp_engine.c: RTP instance '0xcd1ec38' is setup and ready to go [Oct 22 11:22:32] DEBUG[1945] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcd1ec38' [Oct 22 11:22:32] VERBOSE[1945] netsock2.c: == Using SIP RTP TOS bits 184 [Oct 22 11:22:32] VERBOSE[1945] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Setting NAT on RTP to Off [Oct 22 11:22:32] DEBUG[1945] acl.c: For destination '192.168.10.204', our source address is '192.168.10.70'. [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: *** Our native formats are 0x8 (alaw) [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: This channel will not be able to handle video. [Oct 22 11:22:32] DEBUG[1945] rtp_engine.c: Seeded SDP of 'SIP/phone2-00000010' with that of 'SIP/phone1-0000000f' [Oct 22 11:22:32] DEBUG[1945] channel.c: Not copying variable DIALEDTIME. [Oct 22 11:22:32] DEBUG[1945] channel.c: Not copying variable ANSWEREDTIME. [Oct 22 11:22:32] DEBUG[1945] channel.c: Not copying variable DIALEDPEERNAME. [Oct 22 11:22:32] DEBUG[1945] channel.c: Not copying variable DIALEDPEERNUMBER. [Oct 22 11:22:32] DEBUG[1945] channel.c: Not copying variable DIALSTATUS. [Oct 22 11:22:32] DEBUG[1945] channel.c: Not copying variable SIPCALLID. [Oct 22 11:22:32] DEBUG[1945] channel.c: Not copying variable SIPDOMAIN. [Oct 22 11:22:32] DEBUG[1945] channel.c: Not copying variable SIPURI. [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Outgoing Call for phone2 [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Initializing initreq for method INVITE - callid 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:32] VERBOSE[1945] app_dial.c: -- Called phone2 [Oct 22 11:22:32] DEBUG[1854] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-00000010 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 150 CallerIDName: Hans Muster AccountCode: Exten: Context: Standard Uniqueid: 1287739352.17 [Oct 22 11:22:32] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone1-0000000f Destination: SIP/phone2-00000010 CallerIDNum: 100 CallerIDName: Erika Musterfrau UniqueID: 1287739352.16 DestUniqueID: 1287739352.17 Dialstring: phone2 [Oct 22 11:22:32] DEBUG[1854] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone2-00000010 CallerIDNum: 150 CallerIDName: Uniqueid: 1287739352.17 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Oct 22 11:22:32] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:32] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:32] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Oct 22 11:22:32] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '6' [Oct 22 11:22:32] 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 11:22:32] 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 11:22:32] 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 11:22:32] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 1 [Oct 22 11:22:32] 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 11:22:32] 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 11:22:32] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 8 [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Request 102: Found [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:22:32] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:32] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:32] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Oct 22 11:22:32] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '6' [Oct 22 11:22:32] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000010 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 150 CallerIDName: Uniqueid: 1287739352.17 [Oct 22 11:22:32] VERBOSE[1945] app_dial.c: -- SIP/phone2-00000010 is ringing [Oct 22 11:22:32] DEBUG[1945] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-0000000f' with that of 'SIP/phone2-00000010' [Oct 22 11:22:32] DEBUG[1945] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:32] 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 11:22:32] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Request 102: Found [Oct 22 11:22:32] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:22:32] VERBOSE[1945] app_dial.c: -- SIP/phone2-00000010 is ringing [Oct 22 11:22:32] DEBUG[1945] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-0000000f' with that of 'SIP/phone2-00000010' [Oct 22 11:22:33] DEBUG[1945] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:22:33] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.204:64601 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 11:22:33] DEBUG[1945] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Acked pending invite 102 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Stopping retransmission on '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' of Request 102: Match Found [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: SIP response 200 to standard invite [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 192583745 192583746 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:33] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:33] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:33] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:22:33] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:22:33] VERBOSE[1945] app_dial.c: -- SIP/phone2-00000010 answered SIP/phone1-0000000f [Oct 22 11:22:33] DEBUG[1945] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-0000000f' with that of 'SIP/phone2-00000010' [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: SIP answering channel: SIP/phone1-0000000f [Oct 22 11:22:33] DEBUG[1945] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: Setting framing from config on incoming call [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:33] DEBUG[1945] features.c: bridge answer set, chan answer set [Oct 22 11:22:33] DEBUG[1945] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 22 11:22:33] DEBUG[1945] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 22 11:22:33] VERBOSE[1945] rtp_engine.c: -- Remotely bridging SIP/phone1-0000000f and SIP/phone2-00000010 [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: Deferring reinvite on SIP '3c267a1d16d4-8nazi7ekl0nk' - It's audio will be redirected to IP 192.168.10.204:64600 [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: Sending reinvite on SIP '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' - It's audio soon redirected to IP 192.168.10.203:52110 [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: Initializing already initialized SIP dialog 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 (presumably reinvite) [Oct 22 11:22:33] DEBUG[1945] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:33] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000010 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 150 CallerIDName: Uniqueid: 1287739352.17 [Oct 22 11:22:33] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-0000000f ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1287739352.16 [Oct 22 11:22:33] DEBUG[1854] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone2-00000010 Uniqueid: 1287739352.17 AccountCode: OldAccountCode: [Oct 22 11:22:33] DEBUG[1854] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone1-0000000f Channel2: SIP/phone2-00000010 Uniqueid1: 1287739352.16 Uniqueid2: 1287739352.17 CallerID1: 100 CallerID2: 150 [Oct 22 11:22:33] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:33] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:33] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:22:33] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:22:33] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:22:33] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:22:33] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:22:33] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: INVITE [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:33] 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 11:22:33] 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 11:22:33] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 1 [Oct 22 11:22:33] 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 11:22:33] 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 11:22:33] DEBUG[1852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267a1d16d4-8nazi7ekl0nk' of Response 1: Match Found [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Sending pending reinvite on '3c267a1d16d4-8nazi7ekl0nk' [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Initializing already initialized SIP dialog 3c267a1d16d4-8nazi7ekl0nk (presumably reinvite) [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:33] DEBUG[1945] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:22:33] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.203:52111 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 11:22:33] DEBUG[1852] chan_sip.c: Acked pending invite 103 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Stopping retransmission on '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' of Request 103: Match Found [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 192583745 192583747 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:22:33] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:33] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:33] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:33] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:33] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:33] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:22:33] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:22:33] 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 11:22:34] DEBUG[1852] chan_sip.c: Acked pending invite 102 [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267a1d16d4-8nazi7ekl0nk' of Request 102: Match Found [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1541909648 1541909649 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:22:34] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:22:34] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:22:34] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:22:34] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:22:34] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:22:34] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:22:34] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:22:34] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:22:34] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:22:34] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:34] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:34] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:22:34] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:22:34] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:22:34] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:22:34] 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 11:22:35] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:35] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:36] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:36] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:37] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:37] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 22 11:22:38] DEBUG[1852] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer" [Oct 22 11:22:38] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 22 11:22:38] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: timer [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1541909648 1541909650 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33a9f88 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33a9f88 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33a9f88 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33a9f88 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33a9f88 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33a9f88 [Oct 22 11:22:38] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:22:38] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Session-Expires: 1800 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Refresher: UAC [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Received Min-SE: 90 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Restarting session-timers on a refresh - 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Session timer stopped: -1 - 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Session timer started: 261 - 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: SIP/phone1-0000000f: This call is UP.... [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Setting framing from config on incoming call [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: INVITE [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:38] DEBUG[1945] chan_sip.c: Sending reinvite on SIP '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' - It's audio soon redirected to IP 192.168.10.70:5060 [Oct 22 11:22:38] DEBUG[1945] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:22:38] DEBUG[1945] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:22:38] DEBUG[1945] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:22:38] DEBUG[1945] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:22:38] DEBUG[1945] chan_sip.c: Initializing already initialized SIP dialog 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 (presumably reinvite) [Oct 22 11:22:38] DEBUG[1945] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:22:38] VERBOSE[1945] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/phone2-00000010 [Oct 22 11:22:38] DEBUG[1945] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: INVITE [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:38] DEBUG[1854] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/phone2-00000010 UniqueID: 1287739352.17 Class: default [Oct 22 11:22:38] DEBUG[1945] channel.c: Set channel SIP/phone2-00000010 to write format slin [Oct 22 11:22:38] DEBUG[1945] res_musiconhold.c: SIP/phone2-00000010 Opened file 0 '/var/lib/asterisk/moh/iptam/iptam_moh' [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xcd1ec38' [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:22:38] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.204:64601 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 3161587965 FractionLost: 253 PacketsLost: 29372 HighestSequence: 55726 SequenceNumberCycles: 0 IAJitter: 1298271 LastSR: 54874.0268435456 DLSR: 4.7760(sec) RTT: 16(sec) [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Acked pending invite 104 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Stopping retransmission on '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' of Request 104: Match Found [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 192583745 192583748 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:22:38] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:38] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: INVITE [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:38] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:38] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:38] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:22:38] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:38] 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 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267a1d16d4-8nazi7ekl0nk' of Response 2: Match Found [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:38] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:38] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:39] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:39] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:40] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:40] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: REFER [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267a1d16d4-8nazi7ekl0nk' of Request 103: Match Found [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Got 200 OK on NOTIFY for transfer [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: REFER [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 22 11:22:41] DEBUG[1852] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer" [Oct 22 11:22:41] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 22 11:22:41] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: timer [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1541909648 1541909651 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33a9f88 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33a9f88 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33a9f88 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33a9f88 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33a9f88 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33a9f88 [Oct 22 11:22:41] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Session-Expires: 1800 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Refresher: UAC [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Received Min-SE: 90 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Restarting session-timers on a refresh - 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Session timer stopped: -1 - 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Session timer started: 266 - 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: SIP/phone1-0000000f: This call is UP.... [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Setting framing from config on incoming call [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: INVITE [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:41] DEBUG[1945] chan_sip.c: Sending reinvite on SIP '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' - It's audio soon redirected to IP 192.168.10.203:52110 [Oct 22 11:22:41] DEBUG[1945] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:22:41] DEBUG[1945] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:22:41] DEBUG[1945] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:22:41] DEBUG[1945] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:22:41] DEBUG[1945] chan_sip.c: Initializing already initialized SIP dialog 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 (presumably reinvite) [Oct 22 11:22:41] DEBUG[1945] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:22:41] VERBOSE[1945] res_musiconhold.c: -- Stopped music on hold on SIP/phone2-00000010 [Oct 22 11:22:41] DEBUG[1945] channel.c: Set channel SIP/phone2-00000010 to write format alaw [Oct 22 11:22:41] DEBUG[1945] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 22 11:22:41] DEBUG[1945] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: INVITE [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:41] DEBUG[1854] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/phone2-00000010 UniqueID: 1287739352.17 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267a1d16d4-8nazi7ekl0nk' of Response 4: Match Found [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Acked pending invite 105 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Stopping retransmission on '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' of Request 105: Match Found [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 192583745 192583749 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:22:41] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:41] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:41] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:41] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:41] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:41] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:22:41] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:22:41] 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 11:22:42] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:42] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:43] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:43] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:43] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:43] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:43] DEBUG[1854] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 192.168.10.204:64601 OurSSRC: 996383539 SentNTP: 1287739363.3251286016 SentRTP: 20320 SentPackets: 127 SentOctets: 20320 ReportBlock: FractionLost: 256 CumulativeLoss: 1 IAJitter: 0.0000 TheirLastSR: 3596536940 DLSR: 4.9700 (sec) [Oct 22 11:22:44] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: ACK [Oct 22 11:22:44] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:45] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Session timer stopped: -1 - 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Received bye, issuing owner hangup [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267a1d16d4-8nazi7ekl0nk' Method: BYE [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' Method: INVITE [Oct 22 11:22:45] DEBUG[1945] rtp_engine.c: Oooh, got a hangup [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: Sending reinvite on SIP '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' - It's audio soon redirected to IP 192.168.10.70:5060 [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: Initializing already initialized SIP dialog 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 (presumably reinvite) [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:45] DEBUG[1945] channel.c: Returning from native bridge, channels: SIP/phone1-0000000f, SIP/phone2-00000010 [Oct 22 11:22:45] DEBUG[1945] channel.c: Hanging up channel 'SIP/phone2-00000010' [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: Hangup call SIP/phone2-00000010, SIP callid 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:22:45] DEBUG[1945] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:22:45] DEBUG[1945] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Oct 22 11:22:45] DEBUG[1945] pbx.c: Spawn extension (Standard,150,1) exited non-zero on 'SIP/phone1-0000000f' [Oct 22 11:22:45] VERBOSE[1945] pbx.c: == Spawn extension (Standard, 150, 1) exited non-zero on 'SIP/phone1-0000000f' [Oct 22 11:22:45] DEBUG[1945] channel.c: Soft-Hanging up channel 'SIP/phone1-0000000f' [Oct 22 11:22:45] DEBUG[1945] channel.c: Hanging up channel 'SIP/phone1-0000000f' [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: Hangup call SIP/phone1-0000000f, SIP callid 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:22:45] DEBUG[1945] chan_sip.c: Updating call counter for incoming call [Oct 22 11:22:45] DEBUG[1945] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:22:45] DEBUG[1854] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone1-0000000f Channel2: SIP/phone2-00000010 Uniqueid1: 1287739352.16 Uniqueid2: 1287739352.17 CallerID1: 100 CallerID2: 150 [Oct 22 11:22:45] DEBUG[1854] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 100 Destination: 150 DestinationContext: Standard CallerID: "Erika Musterfrau" <100> Channel: SIP/phone1-0000000f DestinationChannel: SIP/phone2-00000010 LastApplication: Dial LastData: SIP/phone2 StartTime: 2010-10-22 11:22:32 AnswerTime: 2010-10-22 11:22:33 EndTime: 2010-10-22 11:22:45 Duration: 13 BillableSeconds: 12 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1287739352.16 UserField: [Oct 22 11:22:45] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-00000010 Uniqueid: 1287739352.17 CallerIDNum: 150 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Oct 22 11:22:45] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone1-0000000f UniqueID: 1287739352.16 DialStatus: ANSWER [Oct 22 11:22:45] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone1-0000000f Uniqueid: 1287739352.16 CallerIDNum: 100 CallerIDName: Erika Musterfrau Cause: 16 Cause-txt: Normal Clearing [Oct 22 11:22:45] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:45] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:45] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:22:45] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:22:45] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:45] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:45] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:22:45] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:22:45] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:22:45] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:22:45] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Oct 22 11:22:45] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '1' [Oct 22 11:22:45] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:22:45] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:22:45] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Oct 22 11:22:45] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '1' [Oct 22 11:22:45] 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 11:22:45] 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 11:22:45] 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 11:22:45] 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 11:22:45] 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 11:22:45] 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 11:22:45] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 0 [Oct 22 11:22:45] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 0 [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Acked pending invite 106 [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Stopping retransmission on '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' of Request 106: Match Found [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: SIP response 200 to standard invite [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 192583745 192583750 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:22:45] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: build_route: Retaining previous route: [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:22:45] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:22:45] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:22:45] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:22:45] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:22:45] 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 11:22:45] DEBUG[1852] chan_sip.c: Stopping retransmission on '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' of Request 107: Match Found [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: Destroying SIP dialog 7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70 [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: ---------- SIP HISTORY for '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: * SIP Call [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 001. NewChan Channel SIP/phone2-00000010 - from 7f0e4e092a31dd1e2981aea650de [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 005. Rx SIP/2.0 / 102 INVITE / 200 Ok [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 006. TxReq ACK / 102 ACK - ACK [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 007. ReInv Re-invite sent [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 008. TxReqRel INVITE / 103 INVITE - INVITE [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 009. Rx SIP/2.0 / 103 INVITE / 200 Ok [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 010. TxReq ACK / 103 ACK - ACK [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 011. ReInv Re-invite sent [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 012. TxReqRel INVITE / 104 INVITE - INVITE [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 013. Rx SIP/2.0 / 104 INVITE / 200 Ok [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 014. TxReq ACK / 104 ACK - ACK [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 015. ReInv Re-invite sent [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 016. TxReqRel INVITE / 105 INVITE - INVITE [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 017. Rx SIP/2.0 / 105 INVITE / 200 Ok [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 018. TxReq ACK / 105 ACK - ACK [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 019. ReInv Re-invite sent [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 020. TxReqRel INVITE / 106 INVITE - INVITE [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 021. Hangup Cause Normal Clearing [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 022. SchedDestroy 32000 ms [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 023. CancelDestroy [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 024. Rx SIP/2.0 / 106 INVITE / 200 Ok [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 025. TxReq ACK / 106 ACK - ACK [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 026. TxReqRel BYE / 107 BYE - BYE [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 027. SchedDestroy 32000 ms [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 028. Rx SIP/2.0 / 107 BYE / 200 OK [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: 029. NeedDestroy Setting needdestroy because received 200 response [Oct 22 11:22:45] DEBUG[1852] chan_sip.c: ---------- END SIP HISTORY for '7f0e4e092a31dd1e2981aea650ded1a1@192.168.10.70' [Oct 22 11:22:45] DEBUG[1852] rtp_engine.c: Destroyed RTP instance '0xcd1ec38' [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: Auto destroying SIP dialog '3c267a1d16d4-8nazi7ekl0nk' [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: Destroying SIP dialog 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: ---------- SIP HISTORY for '3c267a1d16d4-8nazi7ekl0nk' [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: * SIP Call [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 001. Rx INVITE / 1 INVITE / sip:150@192.168.10.70;user=phone [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 002. NewChan Channel SIP/phone1-0000000f - from 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 003. TxResp SIP/2.0 / 1 INVITE - 100 Trying [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 004. TxResp SIP/2.0 / 1 INVITE - 180 Ringing [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 005. ConnectedLine Called party is now Hans Muster <150> [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 006. TxRespRel SIP/2.0 / 1 INVITE - 200 OK [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 007. Rx ACK / 1 ACK / sip:150@192.168.10.70:5060 [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 008. ReInv Re-invite sent [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 009. TxReqRel INVITE / 102 INVITE - INVITE [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 010. Rx SIP/2.0 / 102 INVITE / 200 Ok [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 011. TxReq ACK / 102 ACK - ACK [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 012. Rx INVITE / 2 INVITE / sip:150@192.168.10.70:5060 [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 013. Hold INVITE [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 014. ReInv Re-invite received [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 015. TxResp SIP/2.0 / 2 INVITE - 100 Trying [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 016. TxRespRel SIP/2.0 / 2 INVITE - 200 OK [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 017. Rx ACK / 2 ACK / sip:150@192.168.10.70:5060 [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 018. Rx REFER / 3 REFER / sip:150@192.168.10.70:5060 [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 019. TxResp SIP/2.0 / 3 REFER - 202 Accepted [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 020. Xfer Refer failed. Bad extension. [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 021. TxReqRel NOTIFY / 103 NOTIFY - NOTIFY [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 022. Rx SIP/2.0 / 103 NOTIFY / 200 Ok [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 023. Rx INVITE / 4 INVITE / sip:150@192.168.10.70:5060 [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 024. Unhold INVITE [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 025. ReInv Re-invite received [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 026. TxResp SIP/2.0 / 4 INVITE - 100 Trying [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 027. TxRespRel SIP/2.0 / 4 INVITE - 200 OK [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 028. Rx ACK / 4 ACK / sip:150@192.168.10.70:5060 [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 029. Rx BYE / 5 BYE / sip:150@192.168.10.70:5060 [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 030. RTCPaudio Quality:ssrc=1411485200;themssrc=0;lp=0;rxjitter=0.000000;rxcou [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 031. RTCPaudioJitter Quality:minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0 [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 032. RTCPaudioLoss Quality:minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.00000 [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 033. RTCPaudioRTT Quality:minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrt [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 034. SchedDestroy 32000 ms [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 035. TxResp SIP/2.0 / 5 BYE - 200 OK [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 036. Hangup Cause Normal Clearing [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: 037. AutoDestroy 3c267a1d16d4-8nazi7ekl0nk [Oct 22 11:23:17] DEBUG[1852] chan_sip.c: ---------- END SIP HISTORY for '3c267a1d16d4-8nazi7ekl0nk' [Oct 22 11:23:17] DEBUG[1852] rtp_engine.c: Destroyed RTP instance '0xcd1b530' [Oct 22 11:26:08] DEBUG[1852] acl.c: For destination '192.168.10.203', our source address is '192.168.10.70'. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Allocating new SIP dialog for 3c267af5b7fb-43aus1a7g43d - INVITE (No RTP) [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 22 11:26:08] DEBUG[1852] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Oct 22 11:26:08] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 22 11:26:08] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: timer [Oct 22 11:26:08] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 22 11:26:08] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 22 11:26:08] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 22 11:26:08] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 22 11:26:08] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -from-change- [Oct 22 11:26:08] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: from-change [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcd1b530' [Oct 22 11:26:08] DEBUG[1852] res_rtp_asterisk.c: Allocated port 13666 for RTP instance '0xcd1b530' [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: RTP instance '0xcd1b530' is setup and ready to go [Oct 22 11:26:08] DEBUG[1852] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcd1b530' [Oct 22 11:26:08] VERBOSE[1852] netsock2.c: == Using SIP RTP TOS bits 184 [Oct 22 11:26:08] VERBOSE[1852] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Setting NAT on RTP to Off [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 505349401 505349401 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33a9f88 [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33a9f88 [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33a9f88 [Oct 22 11:26:08] ERROR[1852] chan_sip.c: No SRTP module loaded, can't setup SRTP session. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:7vHkBEu9gM0pjjMGB1SJW0AtdOycVr6Q7CFQKHFn... UNSUPPORTED. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33a9f88 [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33a9f88 [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33a9f88 [Oct 22 11:26:08] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:26:08] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Checking SIP call limits for device phone1 [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: *** Our native formats are 0x8 (alaw) [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: This channel will not be able to handle video. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Session-Expires: 3600 [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Refresher: UAS [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Received Min-SE: 90 [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Session timer started: 274 - 3c267af5b7fb-43aus1a7g43d [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: SIP/phone1-00000011: New call is still down.... Trying... [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:08] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:26:08] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:26:08] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:26:08] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:26:08] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:26:08] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:26:08] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:26:08] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:26:08] DEBUG[1854] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone1-00000011 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 100 CallerIDName: Erika Musterfrau AccountCode: Exten: 150 Context: Standard Uniqueid: 1287739568.18 [Oct 22 11:26:08] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-00000011 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1287739568.18 [Oct 22 11:26:08] DEBUG[1949] pbx.c: Launching 'Dial' [Oct 22 11:26:08] VERBOSE[1949] pbx.c: -- Executing [150@Standard:1] Dial("SIP/phone1-00000011", "SIP/phone2") in new stack [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Allocating new SIP dialog for 573d869742f83fa9741911167b68b4f1@192.168.10.70 - INVITE (No RTP) [Oct 22 11:26:08] DEBUG[1949] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcd1ec38' [Oct 22 11:26:08] DEBUG[1949] res_rtp_asterisk.c: Allocated port 14860 for RTP instance '0xcd1ec38' [Oct 22 11:26:08] DEBUG[1949] rtp_engine.c: RTP instance '0xcd1ec38' is setup and ready to go [Oct 22 11:26:08] DEBUG[1949] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcd1ec38' [Oct 22 11:26:08] VERBOSE[1949] netsock2.c: == Using SIP RTP TOS bits 184 [Oct 22 11:26:08] VERBOSE[1949] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Setting NAT on RTP to Off [Oct 22 11:26:08] DEBUG[1949] acl.c: For destination '192.168.10.204', our source address is '192.168.10.70'. [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: *** Our native formats are 0x8 (alaw) [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: This channel will not be able to handle video. [Oct 22 11:26:08] DEBUG[1949] rtp_engine.c: Seeded SDP of 'SIP/phone2-00000012' with that of 'SIP/phone1-00000011' [Oct 22 11:26:08] DEBUG[1949] channel.c: Not copying variable DIALEDTIME. [Oct 22 11:26:08] DEBUG[1949] channel.c: Not copying variable ANSWEREDTIME. [Oct 22 11:26:08] DEBUG[1949] channel.c: Not copying variable DIALEDPEERNAME. [Oct 22 11:26:08] DEBUG[1949] channel.c: Not copying variable DIALEDPEERNUMBER. [Oct 22 11:26:08] DEBUG[1949] channel.c: Not copying variable DIALSTATUS. [Oct 22 11:26:08] DEBUG[1949] channel.c: Not copying variable SIPCALLID. [Oct 22 11:26:08] DEBUG[1949] channel.c: Not copying variable SIPDOMAIN. [Oct 22 11:26:08] DEBUG[1949] channel.c: Not copying variable SIPURI. [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Outgoing Call for phone2 [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Initializing initreq for method INVITE - callid 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:08] VERBOSE[1949] app_dial.c: -- Called phone2 [Oct 22 11:26:08] DEBUG[1854] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/phone2-00000012 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 150 CallerIDName: Hans Muster AccountCode: Exten: Context: Standard Uniqueid: 1287739568.19 [Oct 22 11:26:08] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone1-00000011 Destination: SIP/phone2-00000012 CallerIDNum: 100 CallerIDName: Erika Musterfrau UniqueID: 1287739568.18 DestUniqueID: 1287739568.19 Dialstring: phone2 [Oct 22 11:26:08] DEBUG[1854] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone2-00000012 CallerIDNum: 150 CallerIDName: Uniqueid: 1287739568.19 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Oct 22 11:26:08] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:08] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:08] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Oct 22 11:26:08] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '6' [Oct 22 11:26:08] 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 11:26:08] 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 11:26:08] 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 11:26:08] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 1 [Oct 22 11:26:08] 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 11:26:08] 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 11:26:08] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 8 [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Request 102: Found [Oct 22 11:26:08] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:26:08] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:08] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:08] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 6 (Ringing) [Oct 22 11:26:08] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '6' [Oct 22 11:26:08] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000012 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 150 CallerIDName: Uniqueid: 1287739568.19 [Oct 22 11:26:08] VERBOSE[1949] app_dial.c: -- SIP/phone2-00000012 is ringing [Oct 22 11:26:08] DEBUG[1949] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000011' with that of 'SIP/phone2-00000012' [Oct 22 11:26:08] DEBUG[1949] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:08] 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 11:26:09] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Request 102: Found [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:26:09] VERBOSE[1949] app_dial.c: -- SIP/phone2-00000012 is ringing [Oct 22 11:26:09] DEBUG[1949] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000011' with that of 'SIP/phone2-00000012' [Oct 22 11:26:09] DEBUG[1949] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:26:09] DEBUG[1949] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:26:09] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.204:49249 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 11:26:09] DEBUG[1852] chan_sip.c: Acked pending invite 102 [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Stopping retransmission on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' of Request 102: Match Found [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: SIP response 200 to standard invite [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1672769336 1672769337 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:09] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:26:09] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:26:09] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:09] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:26:09] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:26:09] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:26:09] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:26:09] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:09] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:09] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:09] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:09] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:09] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:26:09] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:26:09] VERBOSE[1949] app_dial.c: -- SIP/phone2-00000012 answered SIP/phone1-00000011 [Oct 22 11:26:09] DEBUG[1949] rtp_engine.c: Setting early bridge SDP of 'SIP/phone1-00000011' with that of 'SIP/phone2-00000012' [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: SIP answering channel: SIP/phone1-00000011 [Oct 22 11:26:09] DEBUG[1949] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: Setting framing from config on incoming call [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:09] DEBUG[1949] features.c: bridge answer set, chan answer set [Oct 22 11:26:09] DEBUG[1949] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 22 11:26:09] DEBUG[1949] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 22 11:26:09] VERBOSE[1949] rtp_engine.c: -- Remotely bridging SIP/phone1-00000011 and SIP/phone2-00000012 [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: Deferring reinvite on SIP '3c267af5b7fb-43aus1a7g43d' - It's audio will be redirected to IP 192.168.10.204:49248 [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: Sending reinvite on SIP '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' - It's audio soon redirected to IP 192.168.10.203:55034 [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: Initializing already initialized SIP dialog 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 (presumably reinvite) [Oct 22 11:26:09] DEBUG[1949] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:09] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:09] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:09] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:26:09] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:26:09] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:26:09] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:26:09] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:26:09] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:26:09] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone2-00000012 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 150 CallerIDName: Uniqueid: 1287739568.19 [Oct 22 11:26:09] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone1-00000011 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 100 CallerIDName: Erika Musterfrau Uniqueid: 1287739568.18 [Oct 22 11:26:09] DEBUG[1854] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone2-00000012 Uniqueid: 1287739568.19 AccountCode: OldAccountCode: [Oct 22 11:26:09] DEBUG[1854] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone1-00000011 Channel2: SIP/phone2-00000012 Uniqueid1: 1287739568.18 Uniqueid2: 1287739568.19 CallerID1: 100 CallerID2: 150 [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: INVITE [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:09] 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 11:26:09] 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 11:26:09] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 1 [Oct 22 11:26:09] 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 11:26:09] 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 11:26:09] DEBUG[1852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267af5b7fb-43aus1a7g43d' of Response 1: Match Found [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Sending pending reinvite on '3c267af5b7fb-43aus1a7g43d' [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Initializing already initialized SIP dialog 3c267af5b7fb-43aus1a7g43d (presumably reinvite) [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: ACK [Oct 22 11:26:09] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:09] DEBUG[1949] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:26:09] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.203:55035 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 11:26:10] DEBUG[1852] chan_sip.c: Acked pending invite 103 [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Stopping retransmission on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' of Request 103: Match Found [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1672769336 1672769338 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: ACK [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:10] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:10] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:10] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:26:10] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:26:10] 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 11:26:10] DEBUG[1852] chan_sip.c: Acked pending invite 102 [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267af5b7fb-43aus1a7g43d' of Request 102: Match Found [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3c267af5b7fb-43aus1a7g43d [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 505349401 505349402 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:26:10] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:26:10] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1b6dc [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Updating call counter for incoming call [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: ACK [Oct 22 11:26:10] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:10] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:26:10] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:26:10] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 2 (In use) [Oct 22 11:26:10] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '2' [Oct 22 11:26:10] 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 11:26:11] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: ACK [Oct 22 11:26:11] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:12] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: ACK [Oct 22 11:26:12] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: ACK [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 22 11:26:13] DEBUG[1852] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer" [Oct 22 11:26:13] DEBUG[1852] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 22 11:26:13] DEBUG[1852] sip/reqresp_parser.c: Matched SIP option: timer [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 505349401 505349403 IN IP4 192.168.10.203... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.203... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33a9f88 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33a9f88 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33a9f88 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33a9f88 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33a9f88 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33a9f88 [Oct 22 11:26:13] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33a9f88 to 0xcd1b6dc [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:26:13] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Session-Expires: 1800 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Refresher: UAC [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Received Min-SE: 90 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Restarting session-timers on a refresh - 3c267af5b7fb-43aus1a7g43d [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Session timer stopped: -1 - 3c267af5b7fb-43aus1a7g43d [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Session timer started: 282 - 3c267af5b7fb-43aus1a7g43d [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: SIP/phone1-00000011: This call is UP.... [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Setting framing from config on incoming call [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: INVITE [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:13] DEBUG[1949] chan_sip.c: Sending reinvite on SIP '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' - It's audio soon redirected to IP 192.168.10.70:5060 [Oct 22 11:26:13] DEBUG[1949] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:26:13] DEBUG[1949] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:26:13] DEBUG[1949] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:13] DEBUG[1949] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:13] DEBUG[1949] chan_sip.c: Initializing already initialized SIP dialog 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 (presumably reinvite) [Oct 22 11:26:13] DEBUG[1949] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:26:13] VERBOSE[1949] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/phone2-00000012 [Oct 22 11:26:13] DEBUG[1949] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: INVITE [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:13] DEBUG[1854] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/phone2-00000012 UniqueID: 1287739568.19 Class: default [Oct 22 11:26:13] DEBUG[1949] channel.c: Set channel SIP/phone2-00000012 to write format slin [Oct 22 11:26:13] DEBUG[1949] res_musiconhold.c: SIP/phone2-00000012 Opened file 0 '/var/lib/asterisk/moh/iptam/iptam_moh' [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xcd1ec38' [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:26:13] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.204:49249 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 3897688317 FractionLost: 253 PacketsLost: 21224 HighestSequence: 54634 SequenceNumberCycles: 0 IAJitter: 1405711 LastSR: 55090.1342177280 DLSR: 3.4980(sec) RTT: 16(sec) [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Acked pending invite 104 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Stopping retransmission on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' of Request 104: Match Found [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1672769336 1672769339 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:26:13] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:13] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: INVITE [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:13] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:13] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:13] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:26:13] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:26:13] 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 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267af5b7fb-43aus1a7g43d' of Response 2: Match Found [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: ACK [Oct 22 11:26:13] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:13] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: ACK [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: chan1->name: SIP/phone1-00000011 [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:14] DEBUG[1852] channel.c: Planning to masquerade channel SIP/phone2-00000012 into the structure of AsyncGoto/SIP/phone2-00000012 [Oct 22 11:26:14] DEBUG[1852] channel.c: Done planning to masquerade channel SIP/phone2-00000012 into the structure of AsyncGoto/SIP/phone2-00000012 [Oct 22 11:26:14] DEBUG[1852] channel.c: Actually Masquerading SIP/phone2-00000012(6) into the structure of AsyncGoto/SIP/phone2-00000012(6) [Oct 22 11:26:14] DEBUG[1852] channel.c: Set channel SIP/phone2-00000012 to write format slin [Oct 22 11:26:14] DEBUG[1852] channel.c: Putting channel SIP/phone2-00000012 in slin/alaw formats [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: SIP Fixup: New owner for dialogue 35e5cc4c512734b4312a29887660fcd0@192.168.10.70: SIP/phone2-00000012 (Old parent: AsyncGoto/SIP/phone2-00000012) [Oct 22 11:26:14] DEBUG[1852] channel.c: Released clone lock on 'AsyncGoto/SIP/phone2-00000012' [Oct 22 11:26:14] DEBUG[1852] channel.c: Done Masquerading SIP/phone2-00000012 (6) [Oct 22 11:26:14] DEBUG[1852] res_rtp_asterisk.c: Changing ssrc from 1352431174 to 1990453384 due to a source change [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: Blind transfer succeeded. Telling transferer. [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3c267af5b7fb-43aus1a7g43d' Method: REFER [Oct 22 11:26:14] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: AsyncGoto/SIP/phone2-00000012 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: CallerIDName: AccountCode: Exten: Context: Standard Uniqueid: 1287739574.20 [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/phone2-00000012 CloneState: Up Original: AsyncGoto/SIP/phone2-00000012 OriginalState: Up [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone2-00000012 Newname: SIP/phone2-00000012 Uniqueid: 1287739568.19 [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Rename Privilege: call,all Channel: AsyncGoto/SIP/phone2-00000012 Newname: SIP/phone2-00000012 Uniqueid: 1287739574.20 [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/phone2-00000012 Newname: AsyncGoto/SIP/phone2-00000012 Uniqueid: 1287739568.19 [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone2-00000012 CallerIDNum: 150 CallerIDName: Uniqueid: 1287739574.20 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Blind Channel: SIP/phone1-00000011 Uniqueid: 1287739568.18 SIP-Callid: 3c267af5b7fb-43aus1a7g43d TargetChannel: AsyncGoto/SIP/phone2-00000012 TargetUniqueid: 1287739568.19 TransferExten: 180 TransferContext: Standard [Oct 22 11:26:14] DEBUG[1949] rtp_engine.c: Oooh, something is weird, backing out [Oct 22 11:26:14] WARNING[1949] rtp_engine.c: Channel 'AsyncGoto/SIP/phone2-00000012' failed to break RTP bridge [Oct 22 11:26:14] DEBUG[1949] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/phone1-00000011, c1=AsyncGoto/SIP/phone2-00000012, flags: No,No,Yes,Yes [Oct 22 11:26:14] DEBUG[1949] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 22 11:26:14] DEBUG[1949] channel.c: Bridge stops bridging channels SIP/phone1-00000011 and AsyncGoto/SIP/phone2-00000012 [Oct 22 11:26:14] VERBOSE[1949] res_musiconhold.c: -- Stopped music on hold on AsyncGoto/SIP/phone2-00000012 [Oct 22 11:26:14] DEBUG[1949] channel.c: Set channel AsyncGoto/SIP/phone2-00000012 to write format alaw [Oct 22 11:26:14] DEBUG[1949] channel.c: Hanging up zombie 'AsyncGoto/SIP/phone2-00000012' [Oct 22 11:26:14] DEBUG[1949] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Oct 22 11:26:14] DEBUG[1949] pbx.c: Spawn extension (Standard,150,1) exited non-zero on 'SIP/phone1-00000011' [Oct 22 11:26:14] VERBOSE[1949] pbx.c: == Spawn extension (Standard, 150, 1) exited non-zero on 'SIP/phone1-00000011' [Oct 22 11:26:14] DEBUG[1949] channel.c: Soft-Hanging up channel 'SIP/phone1-00000011' [Oct 22 11:26:14] DEBUG[1949] channel.c: Hanging up channel 'SIP/phone1-00000011' [Oct 22 11:26:14] DEBUG[1949] chan_sip.c: Updating call counter for incoming call [Oct 22 11:26:14] DEBUG[1949] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 3c267af5b7fb-43aus1a7g43d. [Oct 22 11:26:14] DEBUG[1949] chan_sip.c: Session timer stopped: -1 - 3c267af5b7fb-43aus1a7g43d [Oct 22 11:26:14] DEBUG[1951] pbx.c: Launching 'Dial' [Oct 22 11:26:14] VERBOSE[1951] pbx.c: -- Executing [180@Standard:1] Dial("SIP/phone2-00000012", "SIP/phone3") in new stack [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: Allocating new SIP dialog for 50354d8947e8640a525cdf593f4f8962@192.168.10.70 - INVITE (No RTP) [Oct 22 11:26:14] DEBUG[1951] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xcd3aa38' [Oct 22 11:26:14] DEBUG[1951] res_rtp_asterisk.c: Allocated port 13696 for RTP instance '0xcd3aa38' [Oct 22 11:26:14] DEBUG[1951] rtp_engine.c: RTP instance '0xcd3aa38' is setup and ready to go [Oct 22 11:26:14] DEBUG[1951] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xcd3aa38' [Oct 22 11:26:14] VERBOSE[1951] netsock2.c: == Using SIP RTP TOS bits 184 [Oct 22 11:26:14] VERBOSE[1951] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: Setting NAT on RTP to Off [Oct 22 11:26:14] DEBUG[1951] acl.c: For destination '192.168.10.206', our source address is '192.168.10.70'. [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.10.70:5060 [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: *** Our native formats are 0x8 (alaw) [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: This channel will not be able to handle video. [Oct 22 11:26:14] DEBUG[1951] rtp_engine.c: Seeded SDP of 'SIP/phone3-00000013' with that of 'SIP/phone2-00000012' [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable DIALEDTIME. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable ANSWEREDTIME. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable DIALEDPEERNAME. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable DIALEDPEERNUMBER. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable DIALSTATUS. [Oct 22 11:26:14] DEBUG[1951] channel.c: Copying soft-transferable variable SIPTRANSFER_REFERER. [Oct 22 11:26:14] DEBUG[1951] channel.c: Copying soft-transferable variable SIPTRANSFER. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable SIPDOMAIN. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable BLINDTRANSFER. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable SIPREFERREDBYHDR. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable SIPREFERRINGCONTEXT. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable BRIDGEPVTCALLID. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable BRIDGEPEER. [Oct 22 11:26:14] DEBUG[1951] channel.c: Not copying variable SIPCALLID. [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: Outgoing Call for phone3 [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: Initializing initreq for method INVITE - callid 3104c26d41301ed744e67b1d11b1d164@192.168.10.70 [Oct 22 11:26:14] DEBUG[1951] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.206:2050 [Oct 22 11:26:14] VERBOSE[1951] app_dial.c: -- Called phone3 [Oct 22 11:26:14] DEBUG[1951] channel.c: Set channel SIP/phone2-00000012 to write format alaw [Oct 22 11:26:14] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone1-00000011 Channel2: AsyncGoto/SIP/phone2-00000012 Uniqueid1: 1287739568.18 Uniqueid2: 1287739568.19 CallerID1: 100 CallerID2: [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 100 Destination: 150 DestinationContext: Standard CallerID: "Erika Musterfrau" <100> Channel: SIP/phone1-00000011 DestinationChannel: SIP/phone2-00000012 LastApplication: Dial LastData: SIP/phone2 StartTime: 2010-10-22 11:26:08 AnswerTime: 2010-10-22 11:26:09 EndTime: 2010-10-22 11:26:14 Duration: 6 BillableSeconds: 5 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1287739568.18 UserField: [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: AsyncGoto/SIP/phone2-00000012 Uniqueid: 1287739568.19 CallerIDNum: CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone1-00000011 UniqueID: 1287739568.18 DialStatus: ANSWER [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone1-00000011 Uniqueid: 1287739568.18 CallerIDNum: 100 CallerIDName: Erika Musterfrau Cause: 16 Cause-txt: Normal Clearing [Oct 22 11:26:14] DEBUG[1854] 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: 1287739574.21 [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/phone2-00000012 Destination: SIP/phone3-00000013 CallerIDNum: 150 CallerIDName: UniqueID: 1287739574.20 DestUniqueID: 1287739574.21 Dialstring: phone3 [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/phone3-00000013 CallerIDNum: 180 CallerIDName: Uniqueid: 1287739574.21 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Oct 22 11:26:14] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for AsyncGoto - SIP/phone2 [Oct 22 11:26:14] DEBUG[1845] devicestate.c: Changing state for AsyncGoto/SIP/phone2 - state 4 (Invalid) [Oct 22 11:26:14] DEBUG[1845] devicestate.c: device 'AsyncGoto/SIP/phone2' state '4' [Oct 22 11:26:14] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:26:14] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:26:14] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Oct 22 11:26:14] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '1' [Oct 22 11:26:14] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone1 [Oct 22 11:26:14] DEBUG[1845] chan_sip.c: Checking device state for peer phone1 [Oct 22 11:26:14] DEBUG[1845] devicestate.c: Changing state for SIP/phone1 - state 1 (Not in use) [Oct 22 11:26:14] DEBUG[1845] devicestate.c: device 'SIP/phone1' state '1' [Oct 22 11:26:14] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Oct 22 11:26:14] DEBUG[1845] chan_sip.c: Checking device state for peer phone3 [Oct 22 11:26:14] DEBUG[1845] devicestate.c: Changing state for SIP/phone3 - state 6 (Ringing) [Oct 22 11:26:14] DEBUG[1845] devicestate.c: device 'SIP/phone3' state '6' [Oct 22 11:26:14] 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 11:26:14] DEBUG[1846] 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. [Oct 22 11:26:14] DEBUG[1872] app_queue.c: Device 'AsyncGoto/SIP/phone2' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Oct 22 11:26:14] 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 11:26:14] 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 11:26:14] DEBUG[1872] 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. [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 100 Context: Standard Hint: SIP/phone1 Status: 0 [Oct 22 11:26:14] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 8 [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Request 102: Found [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] VERBOSE[1951] app_dial.c: -- SIP/phone3-00000013 is ringing [Oct 22 11:26:15] DEBUG[1951] rtp_engine.c: Setting early bridge SDP of 'SIP/phone2-00000012' with that of 'SIP/phone3-00000013' [Oct 22 11:26:15] DEBUG[1951] channel.c: Driver for channel 'SIP/phone2-00000012' does not support indication 3, emulating it [Oct 22 11:26:15] DEBUG[1951] channel.c: Set channel SIP/phone2-00000012 to write format slin [Oct 22 11:26:15] DEBUG[1951] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 22 11:26:15] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Oct 22 11:26:15] DEBUG[1845] chan_sip.c: Checking device state for peer phone3 [Oct 22 11:26:15] DEBUG[1845] devicestate.c: Changing state for SIP/phone3 - state 6 (Ringing) [Oct 22 11:26:15] DEBUG[1845] devicestate.c: device 'SIP/phone3' state '6' [Oct 22 11:26:15] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000013 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 180 CallerIDName: Uniqueid: 1287739574.21 [Oct 22 11:26:15] DEBUG[1872] 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. [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: Difference is 680, ms is 105 [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267af5b7fb-43aus1a7g43d' of Request 103: Match Found [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: Stopping retransmission on '3c267af5b7fb-43aus1a7g43d' of Request 104: Match Found [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3c267af5b7fb-43aus1a7g43d [Oct 22 11:26:15] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1b530' [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: Received bye, no owner, selfdestruct soon. [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.203:2051 [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Request 102: Found [Oct 22 11:26:15] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:26:15] VERBOSE[1951] app_dial.c: -- SIP/phone3-00000013 is ringing [Oct 22 11:26:15] DEBUG[1951] rtp_engine.c: Setting early bridge SDP of 'SIP/phone2-00000012' with that of 'SIP/phone3-00000013' [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:15] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Request 102: Found [Oct 22 11:26:16] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:26:16] VERBOSE[1951] app_dial.c: -- SIP/phone3-00000013 is ringing [Oct 22 11:26:16] DEBUG[1951] rtp_engine.c: Setting early bridge SDP of 'SIP/phone2-00000012' with that of 'SIP/phone3-00000013' [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:16] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:17] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1852] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Request 102: Found [Oct 22 11:26:18] DEBUG[1852] chan_sip.c: SIP response 180 to standard invite [Oct 22 11:26:18] VERBOSE[1951] app_dial.c: -- SIP/phone3-00000013 is ringing [Oct 22 11:26:18] DEBUG[1951] rtp_engine.c: Setting early bridge SDP of 'SIP/phone2-00000012' with that of 'SIP/phone3-00000013' [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1854] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 192.168.10.204:49249 OurSSRC: 1990453384 SentNTP: 1287739578.2777632768 SentRTP: 39880 SentPackets: 245 SentOctets: 39200 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0004 TheirLastSR: 3610600019 DLSR: 4.9730 (sec) [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:26:18] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.204:49249 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 3897688064 FractionLost: 0 PacketsLost: 21224 HighestSequence: 54880 SequenceNumberCycles: 0 IAJitter: 2 LastSR: 55098.2684354560 DLSR: 0.0280(sec) RTT: 3(sec) [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:18] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: No remote address on RTP instance '0xcd3aa38' so dropping frame [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: Got RTCP report of 68 bytes [Oct 22 11:26:19] DEBUG[1854] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 192.168.10.206:49197 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 11:26:19] DEBUG[1951] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd3aa38' [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xcd3aa38' [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Acked pending invite 102 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Stopping retransmission on '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' of Request 102: Match Found [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: SIP response 200 to standard invite [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 442753788 442753789 IN IP4 192.168.10.206... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.206... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd3aa38' [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd3abe4 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd3abe4 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd3abe4 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.206:2050 [Oct 22 11:26:19] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Oct 22 11:26:19] DEBUG[1845] chan_sip.c: Checking device state for peer phone3 [Oct 22 11:26:19] DEBUG[1845] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Oct 22 11:26:19] DEBUG[1845] devicestate.c: device 'SIP/phone3' state '2' [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: Initializing already initialized SIP dialog 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 (presumably reinvite) [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:19] VERBOSE[1951] app_dial.c: -- SIP/phone3-00000013 answered SIP/phone2-00000012 [Oct 22 11:26:19] DEBUG[1951] rtp_engine.c: Setting early bridge SDP of 'SIP/phone2-00000012' with that of 'SIP/phone3-00000013' [Oct 22 11:26:19] DEBUG[1951] channel.c: Set channel SIP/phone2-00000012 to write format alaw [Oct 22 11:26:19] DEBUG[1951] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 22 11:26:19] DEBUG[1951] features.c: bridge answer set, chan answer set [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: Changing ssrc from 1990453384 to 78303737 due to a source change [Oct 22 11:26:19] DEBUG[1951] res_rtp_asterisk.c: Changing ssrc from 1468828581 to 1910880057 due to a source change [Oct 22 11:26:19] VERBOSE[1951] rtp_engine.c: -- Remotely bridging SIP/phone2-00000012 and SIP/phone3-00000013 [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: Deferring reinvite on SIP '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' - It's audio will be redirected to IP 192.168.10.206:49196 [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: Sending reinvite on SIP '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' - It's audio soon redirected to IP 192.168.10.204:49248 [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: Initializing already initialized SIP dialog 3104c26d41301ed744e67b1d11b1d164@192.168.10.70 (presumably reinvite) [Oct 22 11:26:19] DEBUG[1951] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.206:2050 [Oct 22 11:26:19] DEBUG[1854] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/phone3-00000013 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 180 CallerIDName: Uniqueid: 1287739574.21 [Oct 22 11:26:19] DEBUG[1854] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/phone3-00000013 Uniqueid: 1287739574.21 AccountCode: OldAccountCode: [Oct 22 11:26:19] DEBUG[1854] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/phone2-00000012 Channel2: SIP/phone3-00000013 Uniqueid1: 1287739574.20 Uniqueid2: 1287739574.21 CallerID1: 150 CallerID2: 180 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Method: INVITE [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:19] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Oct 22 11:26:19] DEBUG[1845] chan_sip.c: Checking device state for peer phone3 [Oct 22 11:26:19] DEBUG[1845] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Oct 22 11:26:19] DEBUG[1845] devicestate.c: device 'SIP/phone3' state '2' [Oct 22 11:26:19] DEBUG[1846] 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. [Oct 22 11:26:19] DEBUG[1872] 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. [Oct 22 11:26:19] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 1 [Oct 22 11:26:19] DEBUG[1872] 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. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Acked pending invite 105 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Stopping retransmission on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' of Request 105: Match Found [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1672769336 1672769340 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Sending pending reinvite on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Initializing already initialized SIP dialog 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 (presumably reinvite) [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Method: INVITE [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:19] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:19] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:19] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:26:19] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:26:19] 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 11:26:19] DEBUG[1852] chan_sip.c: Acked pending invite 103 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Stopping retransmission on '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' of Request 103: Match Found [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 3104c26d41301ed744e67b1d11b1d164@192.168.10.70 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 442753788 442753790 IN IP4 192.168.10.206... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.206... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd3aa38' [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd3abe4 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd3abe4 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd3abe4 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.206:2050 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Method: INVITE [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:19] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Oct 22 11:26:19] DEBUG[1845] chan_sip.c: Checking device state for peer phone3 [Oct 22 11:26:19] DEBUG[1845] devicestate.c: Changing state for SIP/phone3 - state 2 (In use) [Oct 22 11:26:19] DEBUG[1845] devicestate.c: device 'SIP/phone3' state '2' [Oct 22 11:26:19] DEBUG[1872] 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. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Acked pending invite 106 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Stopping retransmission on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' of Request 106: Match Found [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: SIP response 200 to RE-invite on outgoing call 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1672769336 1672769341 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:26:19] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:19] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: We have an owner, now see if we need to change this call [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Method: INVITE [Oct 22 11:26:19] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:19] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:19] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:19] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 2 (In use) [Oct 22 11:26:19] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '2' [Oct 22 11:26:19] 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 11:26:20] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Method: INVITE [Oct 22 11:26:20] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Method: INVITE [Oct 22 11:26:21] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Method: INVITE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3104c26d41301ed744e67b1d11b1d164@192.168.10.70 [Oct 22 11:26:22] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd3aa38' [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Received bye, issuing owner hangup [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.10.206:2050 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '3104c26d41301ed744e67b1d11b1d164@192.168.10.70' Method: BYE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' Method: INVITE [Oct 22 11:26:22] DEBUG[1951] rtp_engine.c: Oooh, got a hangup [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: Sending reinvite on SIP '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' - It's audio soon redirected to IP 192.168.10.70:5060 [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: -- Done with adding codecs to SDP [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: Initializing already initialized SIP dialog 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 (presumably reinvite) [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:22] DEBUG[1951] channel.c: Returning from native bridge, channels: SIP/phone2-00000012, SIP/phone3-00000013 [Oct 22 11:26:22] DEBUG[1951] channel.c: Hanging up channel 'SIP/phone3-00000013' [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: Hangup call SIP/phone3-00000013, SIP callid 3104c26d41301ed744e67b1d11b1d164@192.168.10.70 [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:22] DEBUG[1951] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd3aa38' [Oct 22 11:26:22] DEBUG[1951] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Oct 22 11:26:22] DEBUG[1951] pbx.c: Spawn extension (Standard,180,1) exited non-zero on 'SIP/phone2-00000012' [Oct 22 11:26:22] VERBOSE[1951] pbx.c: == Spawn extension (Standard, 180, 1) exited non-zero on 'SIP/phone2-00000012' [Oct 22 11:26:22] DEBUG[1951] channel.c: Soft-Hanging up channel 'SIP/phone2-00000012' [Oct 22 11:26:22] DEBUG[1951] channel.c: Hanging up channel 'SIP/phone2-00000012' [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: Hangup call SIP/phone2-00000012, SIP callid 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 [Oct 22 11:26:22] DEBUG[1951] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:22] DEBUG[1951] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:26:22] DEBUG[1854] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/phone2-00000012 Channel2: SIP/phone3-00000013 Uniqueid1: 1287739574.20 Uniqueid2: 1287739574.21 CallerID1: 150 CallerID2: 180 [Oct 22 11:26:22] DEBUG[1854] manager.c: Examining event: Event: Cdr Privilege: cdr,all AccountCode: Source: 150 Destination: 180 DestinationContext: Standard CallerID: 150 Channel: SIP/phone2-00000012 DestinationChannel: SIP/phone3-00000013 LastApplication: Dial LastData: SIP/phone3 StartTime: 2010-10-22 11:26:14 AnswerTime: 2010-10-22 11:26:19 EndTime: 2010-10-22 11:26:22 Duration: 8 BillableSeconds: 3 Disposition: ANSWERED AMAFlags: DOCUMENTATION UniqueID: 1287739568.19 UserField: [Oct 22 11:26:22] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone3-00000013 Uniqueid: 1287739574.21 CallerIDNum: 180 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Oct 22 11:26:22] DEBUG[1854] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/phone2-00000012 UniqueID: 1287739574.20 DialStatus: ANSWER [Oct 22 11:26:22] DEBUG[1854] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/phone2-00000012 Uniqueid: 1287739574.20 CallerIDNum: 150 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Oct 22 11:26:22] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Oct 22 11:26:22] DEBUG[1845] chan_sip.c: Checking device state for peer phone3 [Oct 22 11:26:22] DEBUG[1845] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Oct 22 11:26:22] DEBUG[1845] devicestate.c: device 'SIP/phone3' state '1' [Oct 22 11:26:22] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone3 [Oct 22 11:26:22] DEBUG[1845] chan_sip.c: Checking device state for peer phone3 [Oct 22 11:26:22] DEBUG[1845] devicestate.c: Changing state for SIP/phone3 - state 1 (Not in use) [Oct 22 11:26:22] DEBUG[1845] devicestate.c: device 'SIP/phone3' state '1' [Oct 22 11:26:22] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:22] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:22] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:26:22] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:26:22] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:22] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:22] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:26:22] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:26:22] DEBUG[1846] 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. [Oct 22 11:26:22] 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 11:26:22] DEBUG[1872] 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. [Oct 22 11:26:22] DEBUG[1872] 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. [Oct 22 11:26:22] 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 11:26:22] 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 11:26:22] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 180 Context: Standard Hint: SIP/phone3 Status: 0 [Oct 22 11:26:22] DEBUG[1854] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 150 Context: Standard Hint: SIP/phone2 Status: 0 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Acked pending invite 107 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Stopping retransmission on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' of Request 107: Match Found [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: SIP response 200 to standard invite [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing session-level SDP o=root 1672769336 1672769342 IN IP4 192.168.10.204... UNSUPPORTED. [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED. [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.204... OK. [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Setting payload 8 based on m type on 0xb33aa588 [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Setting payload 0 based on m type on 0xb33aa588 [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Setting payload 101 based on m type on 0xb33aa588 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 pcma/8000... OK. [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 pcmu/8000... OK. [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Incorporating payload 0 on 0xb33aa588 [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Incorporating payload 8 on 0xb33aa588 [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Incorporating payload 101 on 0xb33aa588 [Oct 22 11:26:22] DEBUG[1852] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xcd1ec38' [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Copying payload 0 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Copying payload 8 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Copying payload 101 from 0xb33aa588 to 0xcd1ede4 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Updating call counter for outgoing call [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: build_route: Retaining previous route: [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 192.168.10.204:2050 [Oct 22 11:26:22] DEBUG[1845] devicestate.c: No provider found, checking channel drivers for SIP - phone2 [Oct 22 11:26:22] DEBUG[1845] chan_sip.c: Checking device state for peer phone2 [Oct 22 11:26:22] DEBUG[1845] devicestate.c: Changing state for SIP/phone2 - state 1 (Not in use) [Oct 22 11:26:22] DEBUG[1845] devicestate.c: device 'SIP/phone2' state '1' [Oct 22 11:26:22] 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 11:26:22] DEBUG[1852] chan_sip.c: Stopping retransmission on '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' of Request 108: Match Found [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: Destroying SIP dialog 35e5cc4c512734b4312a29887660fcd0@192.168.10.70 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: ---------- SIP HISTORY for '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: * SIP Call [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 001. NewChan Channel SIP/phone2-00000012 - from 35e5cc4c512734b4312a29887660 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 005. Rx SIP/2.0 / 102 INVITE / 200 Ok [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 006. TxReq ACK / 102 ACK - ACK [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 007. ReInv Re-invite sent [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 008. TxReqRel INVITE / 103 INVITE - INVITE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 009. Rx SIP/2.0 / 103 INVITE / 200 Ok [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 010. TxReq ACK / 103 ACK - ACK [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 011. ReInv Re-invite sent [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 012. TxReqRel INVITE / 104 INVITE - INVITE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 013. Rx SIP/2.0 / 104 INVITE / 200 Ok [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 014. TxReq ACK / 104 ACK - ACK [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 015. Masq Old channel: AsyncGoto/SIP/phone2-00000012 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 016. Masq (cont) ...new owner: SIP/phone2-00000012 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 017. ConnectedLine Calling party is now -> Max Mustermann (via phone1@192.168.10.7 [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 018. TxReqRel INVITE / 105 INVITE - INVITE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 019. Rx SIP/2.0 / 105 INVITE / 200 Ok [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 020. TxReq ACK / 105 ACK - ACK [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 021. ReInv Re-invite sent [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 022. TxReqRel INVITE / 106 INVITE - INVITE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 023. Rx SIP/2.0 / 106 INVITE / 200 Ok [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 024. TxReq ACK / 106 ACK - ACK [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 025. ReInv Re-invite sent [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 026. TxReqRel INVITE / 107 INVITE - INVITE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 027. Hangup Cause Normal Clearing [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 028. SchedDestroy 32000 ms [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 029. CancelDestroy [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 030. Rx SIP/2.0 / 107 INVITE / 200 Ok [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 031. TxReq ACK / 107 ACK - ACK [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 032. TxReqRel BYE / 108 BYE - BYE [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 033. SchedDestroy 32000 ms [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 034. Rx SIP/2.0 / 108 BYE / 200 OK [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: 035. NeedDestroy Setting needdestroy because received 200 response [Oct 22 11:26:22] DEBUG[1852] chan_sip.c: ---------- END SIP HISTORY for '35e5cc4c512734b4312a29887660fcd0@192.168.10.70' [Oct 22 11:26:22] DEBUG[1852] rtp_engine.c: Destroyed RTP instance '0xcd1ec38'