<--- SIP read from UDP:192.168.169.102:5060 ---> INVITE sip:1001@192.168.169.60:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.169.102;branch=z9hG4bK0aa102135062c7541 Max-Forwards: 70 From: "BME" ;tag=a2e37c0386 To: Call-ID: 2c063ea24067f43a CSeq: 11352 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="1000",realm="asterisk",nonce="6c827055",uri="sip:1001@192.168.169.60:5060;user=phone",response="bfd9611073351bc91af88de7d12f7e16",algorithm=MD5 Contact: "BME" ;+sip.instance="" Session-Expires: 900 Supported: path, 100rel, replaces, timer User-Agent: Aastra 6739i/3.2.2.41 Content-Type: application/sdp Content-Length: 301 v=0 o=MxSIP 0 1 IN IP4 192.168.169.102 s=SIP Call c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.102 a=sendrecv <-------------> [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] --- (16 headers 14 lines) --- [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Sending to 192.168.169.102:5060 (NAT) [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Initializing initreq for method INVITE - callid 2c063ea24067f43a [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Using INVITE request as basis request - 2c063ea24067f43a [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Found peer '1000' for '1000' from 192.168.169.102:5060 [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x96d19a8' [Jun 23 16:56:54] DEBUG[1540] res_rtp_asterisk.c: Allocated port 12206 for RTP instance '0x96d19a8' [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: RTP instance '0x96d19a8' is setup and ready to go [Jun 23 16:56:54] DEBUG[1540] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x96d19a8' [Jun 23 16:56:54] VERBOSE[1540] netsock2.c: [Jun 23 16:56:54] == Using SIP RTP TOS bits 184 [Jun 23 16:56:54] VERBOSE[1540] netsock2.c: [Jun 23 16:56:54] == Using SIP RTP CoS mark 5 [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Setting NAT on RTP to On [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.169.102... UNSUPPORTED. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.102... OK. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Found RTP audio format 8 [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5dec924 [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Found RTP audio format 18 [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Setting payload 18 based on m type on 0xb5dec924 [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Found RTP audio format 101 [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5dec924 [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Found audio description format PCMA for ID 8 [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Found audio description format G729 for ID 18 [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Found audio description format telephone-event for ID 101 [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.102... UNSUPPORTED. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5dec924 [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Incorporating payload 18 on 0xb5dec924 [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5dec924 [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:56:54] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x96d19a8' [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Peer audio RTP is at port 192.168.169.102:8000 [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5dec924 to 0x96d1b54 [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Copying payload 18 from 0xb5dec924 to 0x96d1b54 [Jun 23 16:56:54] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5dec924 to 0x96d1b54 [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Checking SIP call limits for device 1000 [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Updating call counter for incoming call [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Call from peer '1000' is 1 out of 2147483647 [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] Looking for 1001 in ClassOfService (domain 192.168.169.60:5060) [Jun 23 16:56:54] DEBUG[1532] chan_sip.c: Checking device state for peer 1000 [Jun 23 16:56:54] DEBUG[1532] devicestate.c: Changing state for SIP/1000 - state 2 (In use) [Jun 23 16:56:54] DEBUG[1532] devicestate.c: device 'SIP/1000' state '2' [Jun 23 16:56:54] DEBUG[1567] app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:54] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1000 [Jun 23 16:56:54] DEBUG[1533] db.c: Unable to find key 'DND1000' in family 'CustomDevstate' [Jun 23 16:56:54] DEBUG[1533] app_queue.c: Extension '1000@ext-local' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 1000 Context: ext-local Hint: SIP/1000&Custom:DND1000 Status: 1 [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/1000-00000014 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 1000 CallerIDName: P1000 AccountCode: Exten: 1001 Context: ClassOfService Uniqueid: 1308841014.20 [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: This channel will not be able to handle video. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: build_route: Contact hop: "BME" ;+sip.instance="" [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] list_route: hop: [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Session-Expires: 900 [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Session timer started: 543 - 2c063ea24067f43a [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: SIP/1000-00000014: New call is still down.... Trying... [Jun 23 16:56:54] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:54] <--- Transmitting (NAT) to 192.168.169.102:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.169.102;branch=z9hG4bK0aa102135062c7541;received=192.168.169.102;rport=5060 From: "BME" ;tag=a2e37c0386 To: Call-ID: 2c063ea24067f43a CSeq: 11352 INVITE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 900;refresher=uas Contact: Content-Length: 0 <------------> [Jun 23 16:56:54] DEBUG[1540] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:56:54] DEBUG[1532] chan_sip.c: Checking device state for peer 1000 [Jun 23 16:56:54] DEBUG[1532] devicestate.c: Changing state for SIP/1000 - state 2 (In use) [Jun 23 16:56:54] DEBUG[1532] devicestate.c: device 'SIP/1000' state '2' [Jun 23 16:56:54] DEBUG[1567] app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:54] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1000 [Jun 23 16:56:54] DEBUG[1533] db.c: Unable to find key 'DND1000' in family 'CustomDevstate' [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/1000-00000014 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 1000 CallerIDName: P1000 Uniqueid: 1308841014.20 [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Verbose' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@ClassOfService:1] Verbose("SIP/1000-00000014", "2|TNE2:Passage COS") in new stack [Jun 23 16:56:54] WARNING[2037] pbx.c: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Verbose(2|TNE2:Passage COS)) [Jun 23 16:56:54] VERBOSE[2037] app_verbose.c: [Jun 23 16:56:54] 2|TNE2:Passage COS [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1000' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@ClassOfService:2] Set("SIP/1000-00000014", "DEVICE=1000") in new stack [Jun 23 16:56:54] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1000' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@ClassOfService:3] Set("SIP/1000-00000014", "COSNUM=1000") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'from-internal' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@ClassOfService:4] Set("SIP/1000-00000014", "AMPUSERCOS=from-internal") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'BME' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@ClassOfService:5] Set("SIP/1000-00000014", "AMPUSERNAME=BME") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@ClassOfService:6] Set("SIP/1000-00000014", "CDR(userfield)=1000") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@ClassOfService:7] Set("SIP/1000-00000014", "CDR(name)=BME") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@ClassOfService:8] GotoIf("SIP/1000-00000014", "0?pasdecos") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Goto' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@ClassOfService:9] Goto("SIP/1000-00000014", "from-internal,1001,1") in new stack [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Goto (from-internal,1001,1) [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@from-internal:1] ExecIf("SIP/1000-00000014", "0?Set(__RINGTIMER=0)") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Macro' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [1001@from-internal:2] Macro("SIP/1000-00000014", "exten-vm,novm,1001,0,0,0") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Macro' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-exten-vm:1] Macro("SIP/1000-00000014", "user-callerid,") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1000' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1000' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:1] Set("SIP/1000-00000014", "AMPUSER=1000") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1000-00000014", "0?report") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1000' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1000-00000014", "1?Set(REALCALLERIDNUM=1000)") in new stack [Jun 23 16:56:54] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1000' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1000' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:4] Set("SIP/1000-00000014", "AMPUSER=1000") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'BME' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:5] Set("SIP/1000-00000014", "AMPUSERCIDNAME=BME") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1000-00000014", "0?report") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1000' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:7] Set("SIP/1000-00000014", "AMPUSERCID=1000") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:8] Set("SIP/1000-00000014", "CALLERID(all)="BME" <1000>") in new stack [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/1000-00000014 CallerIDNum: 1000 CallerIDName: BME Uniqueid: 1308841014.20 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '4' [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1000/concurrency_limit' in family 'AMPUSER' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: AMPUSER/1000/concurrency_limit not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '0' [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1000/concurrency_limit' in family 'AMPUSER' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: AMPUSER/1000/concurrency_limit not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:9] GotoIf("SIP/1000-00000014", "0?limit") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '4' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:10] ExecIf("SIP/1000-00000014", "0?Set(GROUP(concurrency_limit)=1000)") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '4' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:11] ExecIf("SIP/1000-00000014", "0?Set(CHANNEL(language)=)") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1000-00000014", "0?continue") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '-1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '64' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:13] Set("SIP/1000-00000014", "__TTL=64") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1000-00000014", "1?continue") in new stack [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Goto (macro-user-callerid,s,25) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1000' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:25] Set("SIP/1000-00000014", "CALLERID(number)=1000") in new stack [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/1000-00000014 CallerIDNum: 1000 CallerIDName: BME Uniqueid: 1308841014.20 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'BME' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:26] Set("SIP/1000-00000014", "CALLERID(name)=BME") in new stack [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/1000-00000014 CallerIDNum: 1000 CallerIDName: BME Uniqueid: 1308841014.20 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'fr' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'fr' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-user-callerid:27] Set("SIP/1000-00000014", "CHANNEL(language)=fr") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Macro [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-exten-vm:2] Set("SIP/1000-00000014", "RingGroupMethod=none") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-exten-vm:3] Set("SIP/1000-00000014", "__EXTTOCALL=1001") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-exten-vm:4] Set("SIP/1000-00000014", "__PICKUPMARK=1001") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1001' in family 'CFU' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: CFU/1001 not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1001' in family 'CFB' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: CFB/1001 not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '""' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-exten-vm:5] Set("SIP/1000-00000014", "RT=""") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Macro' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-exten-vm:6] Macro("SIP/1000-00000014", "record-enable,1001,IN") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-record-enable:1] GotoIf("SIP/1000-00000014", "1?check") in new stack [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Goto (macro-record-enable,s,4) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-record-enable:4] ExecIf("SIP/1000-00000014", "0?MacroExit()") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-record-enable:5] GotoIf("SIP/1000-00000014", "0?Group:OUT") in new stack [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Goto (macro-record-enable,s,14) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-record-enable:14] GotoIf("SIP/1000-00000014", "1?IN") in new stack [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Goto (macro-record-enable,s,18) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'out=Adhoc|in=Adhoc' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'in=Adhoc' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-record-enable:18] ExecIf("SIP/1000-00000014", "1?MacroExit()") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Macro [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Macro' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-exten-vm:7] Macro("SIP/1000-00000014", "dial-one,"",,1001") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:1] Set("SIP/1000-00000014", "DEXTEN=1001") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:2] Set("SIP/1000-00000014", "DIALSTATUS_CW=") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1001/screen' in family 'AMPUSER' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: AMPUSER/1001/screen not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GosubIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:3] GosubIf("SIP/1000-00000014", "0?screen,1") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GosubIf [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1001/screen' in family 'AMPUSER' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: AMPUSER/1001/screen not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1001' in family 'CF' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: CF/1001 not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GosubIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:4] GosubIf("SIP/1000-00000014", "0?cf,1") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GosubIf [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1001' in family 'CF' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: CF/1001 not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1001' in family 'DND' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: DND/1001 not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:5] GotoIf("SIP/1000-00000014", "1?skip1") in new stack [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Goto (macro-dial-one,s,8) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:8] GotoIf("SIP/1000-00000014", "0?nodial") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:9] GotoIf("SIP/1000-00000014", "0?continue") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'ENABLED' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'ENABLED' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:10] Set("SIP/1000-00000014", "EXTHASCW=ENABLED") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1001' in family 'CFB' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: CFB/1001 not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key '1001' in family 'CFU' [Jun 23 16:56:54] DEBUG[2037] func_db.c: DB: CFU/1001 not found in database. [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:11] GotoIf("SIP/1000-00000014", "0?next1:cwinusebusy") in new stack [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Goto (macro-dial-one,s,23) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:23] GotoIf("SIP/1000-00000014", "1?next3:continue") in new stack [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Goto (macro-dial-one,s,24) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:56:54] DEBUG[2037] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:56:54] DEBUG[2037] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:24] ExecIf("SIP/1000-00000014", "0?Set(DIALSTATUS_CW=BUSY)") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:56:54] DEBUG[2037] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:56:54] DEBUG[2037] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:54] DEBUG[2037] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:25] GotoIf("SIP/1000-00000014", "0?nodial") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GosubIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:26] GosubIf("SIP/1000-00000014", "1?dstring,1:dlocal,1") in new stack [Jun 23 16:56:54] DEBUG[2037] app_stack.c: Channel SIP/1000-00000014 has no datastore, so we're allocating one. [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GosubIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Incrementing gosub_level [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:1] Set("SIP/1000-00000014", "DSTRING=") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1001' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:2] Set("SIP/1000-00000014", "DEVICES=1001") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/1000-00000014", "0?Return()") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/1000-00000014", "0?Set(DEVICES=001)") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:5] Set("SIP/1000-00000014", "LOOPCNT=1") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:6] Set("SIP/1000-00000014", "ITER=1") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1001' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'SIP/1001' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:7] Set("SIP/1000-00000014", "THISDIAL=SIP/1001") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GosubIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/1000-00000014", "1?zap2dahdi,1") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GosubIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Incrementing gosub_level [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/1000-00000014", "0?Return()") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/1000-00000014", "NEWDIAL=") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/1000-00000014", "LOOPCNT2=1") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/1000-00000014", "ITER2=1") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'SIP/1001' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/1000-00000014", "THISPART2=SIP/1001") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/1000-00000014", "0?Set(THISPART2=DAHDI/1001)") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/1000-00000014", "NEWDIAL=SIP/1001&") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '2' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/1000-00000014", "ITER2=2") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/1000-00000014", "0?begin2") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '9' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '8' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/1000-00000014", "THISDIAL=SIP/1001") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Return' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/1000-00000014", "") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Return [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Decrementing gosub_level [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:9] Set("SIP/1000-00000014", "DSTRING=SIP/1001&") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '2' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:10] Set("SIP/1000-00000014", "ITER=2") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/1000-00000014", "0?begin") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '9' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '8' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:12] Set("SIP/1000-00000014", "DSTRING=SIP/1001") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Return' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [dstring@macro-dial-one:13] Return("SIP/1000-00000014", "") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Return [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Decrementing gosub_level [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '8' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:27] GotoIf("SIP/1000-00000014", "0?nodial") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:28] GotoIf("SIP/1000-00000014", "1?skiptrace") in new stack [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Goto (macro-dial-one,s,30) [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] func_strings.c: FUNCTION REGEX ((M[(]auto-blkvm[)]))() [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is '' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:30] Set("SIP/1000-00000014", "D_OPTIONS=") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:31] ExecIf("SIP/1000-00000014", "0?SIPAddHeader(Alert-Info: )") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:32] ExecIf("SIP/1000-00000014", "0?SIPAddHeader()") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'ExecIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:33] ExecIf("SIP/1000-00000014", "0?Set(CHANNEL(musicclass)=)") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: ExecIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GosubIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:34] GosubIf("SIP/1000-00000014", "0?qwait,1") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GosubIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:35] Set("SIP/1000-00000014", "__CWIGNORE=") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:36] Set("SIP/1000-00000014", "__KEEPCID=TRUE") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:37] GotoIf("SIP/1000-00000014", "0?usegoto,1") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'Cedric Autier' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Expression result is '0' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:38] GotoIf("SIP/1000-00000014", "0?godial") in new stack [Jun 23 16:56:54] DEBUG[2037] pbx.c: Not taking any branch [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:56:54] DEBUG[2037] pbx.c: Function result is 'Cedric Autier' [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:39] Set("SIP/1000-00000014", "CONNECTEDLINE(name,i)=Cedric Autier") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:40] Set("SIP/1000-00000014", "CONNECTEDLINE(num)=1001") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Set' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:41] Set("SIP/1000-00000014", "D_OPTIONS=I") in new stack [Jun 23 16:56:54] DEBUG[2037] app_macro.c: Executed application: Set [Jun 23 16:56:54] DEBUG[2037] pbx.c: Launching 'Dial' [Jun 23 16:56:54] VERBOSE[2037] pbx.c: [Jun 23 16:56:54] -- Executing [s@macro-dial-one:42] Dial("SIP/1000-00000014", "SIP/1001,"",I") in new stack [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Allocating new SIP dialog for 150d480045a5781f76fee7d82f4524a2@127.0.1.1:0 - INVITE (No RTP) [Jun 23 16:56:54] DEBUG[2037] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb4b9b4f0' [Jun 23 16:56:54] DEBUG[2037] res_rtp_asterisk.c: Allocated port 15544 for RTP instance '0xb4b9b4f0' [Jun 23 16:56:54] DEBUG[2037] rtp_engine.c: RTP instance '0xb4b9b4f0' is setup and ready to go [Jun 23 16:56:54] DEBUG[2037] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb4b9b4f0' [Jun 23 16:56:54] VERBOSE[2037] netsock2.c: [Jun 23 16:56:54] == Using SIP RTP TOS bits 184 [Jun 23 16:56:54] VERBOSE[2037] netsock2.c: [Jun 23 16:56:54] == Using SIP RTP CoS mark 5 [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Setting NAT on RTP to On [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 23 16:56:54] DEBUG[2037] acl.c: For destination '192.168.169.110', our source address is '192.168.169.60'. [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: This channel will not be able to handle video. [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/1001-00000015 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 1001 CallerIDName: P1001 AccountCode: Exten: Context: ClassOfService Uniqueid: 1308841014.21 [Jun 23 16:56:54] DEBUG[2037] rtp_engine.c: Seeded SDP of 'SIP/1001-00000015' with that of 'SIP/1000-00000014' [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DIALEDTIME. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable ANSWEREDTIME. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DIALEDPEERNAME. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DIALEDPEERNUMBER. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DIALSTATUS. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable MACRO_DEPTH. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable D_OPTIONS. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DB_RESULT. [Jun 23 16:56:54] DEBUG[2037] channel.c: Copying hard-transferable variable KEEPCID. [Jun 23 16:56:54] DEBUG[2037] channel.c: Copying hard-transferable variable CWIGNORE. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable GOSUB_RETVAL. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DSTRING. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable ITER. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable THISDIAL. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable ITER2. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable NEWDIAL. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable THISPART2. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable LOOPCNT2. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable LOOPCNT. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DEVICES. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable EXTHASCW. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DIALSTATUS_CW. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DEXTEN. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable ARG3. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable ARG2. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable ARG1. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable MACRO_PRIORITY. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable MACRO_CONTEXT. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable MACRO_EXTEN. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable RT. [Jun 23 16:56:54] DEBUG[2037] channel.c: Copying hard-transferable variable PICKUPMARK. [Jun 23 16:56:54] DEBUG[2037] channel.c: Copying hard-transferable variable EXTTOCALL. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable RingGroupMethod. [Jun 23 16:56:54] DEBUG[2037] channel.c: Copying hard-transferable variable TTL. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable AMPUSERCID. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable AMPUSERCIDNAME. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable AMPUSER. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable REALCALLERIDNUM. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable ARG5. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable ARG4. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable AMPUSERNAME. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable AMPUSERCOS. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable COSNUM. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable DEVICE. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable SIPCALLID. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable SIPDOMAIN. [Jun 23 16:56:54] DEBUG[2037] channel.c: Not copying variable SIPURI. [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Outgoing Call for 1001 [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Call to peer '1001' is 1 out of 2147483647 [Jun 23 16:56:54] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:56:54] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 6 (Ringing) [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False Text flag: False [Jun 23 16:56:54] DEBUG[1532] devicestate.c: device 'SIP/1001' state '6' [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 23 16:56:54] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:54] Audio is at 5060 [Jun 23 16:56:54] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:54] Adding codec 0x8 (alaw) to SDP [Jun 23 16:56:54] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:54] Adding codec 0x2 (gsm) to SDP [Jun 23 16:56:54] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:54] Adding codec 0x4 (ulaw) to SDP [Jun 23 16:56:54] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 23 16:56:54] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:54] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:54] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:54] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:56:54] DEBUG[1533] app_queue.c: Extension '1001@ext-local' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:56:54] DEBUG[1533] chan_sip.c: Strict routing enforced for session 70a40468b789eb3f [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Initializing initreq for method INVITE - callid 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:56:54] VERBOSE[1533] chan_sip.c: [Jun 23 16:56:54] set_destination: Parsing for address/port to send to [Jun 23 16:56:54] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:54] Reliably Transmitting (NAT) to 192.168.169.110:5060: INVITE sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK2e0b97d3;rport Max-Forwards: 70 From: "BME" ;tag=as343b04db To: Contact: Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 102 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:56:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "BME" Content-Type: application/sdp Content-Length: 285 v=0 o=root 671084360 671084360 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 15544 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:56:54] VERBOSE[1533] chan_sip.c: [Jun 23 16:56:54] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:56:54] DEBUG[2037] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:56:54] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:54] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:54] VERBOSE[1533] chan_sip.c: [Jun 23 16:56:54] Reliably Transmitting (NAT) to 192.168.169.102:5060: NOTIFY sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7c7ea669;rport Max-Forwards: 70 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Contact: Call-ID: 70a40468b789eb3f CSeq: 120 NOTIFY User-Agent: FPBX-2.9.0(1.8.4.2) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 228

early

--- [Jun 23 16:56:54] DEBUG[1533] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:56:54] VERBOSE[1533] chan_sip.c: [Jun 23 16:56:54] == Extension Changed 1001[ext-local] new state Ringing for Notify User 1000 [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 1001 Context: ext-local Hint: SIP/1001&Custom:DND1001 Status: 8 [Jun 23 16:56:54] VERBOSE[2037] app_dial.c: [Jun 23 16:56:54] -- Called 1001 [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/1000-00000014 Destination: SIP/1001-00000015 CallerIDNum: 1000 CallerIDName: BME UniqueID: 1308841014.20 DestUniqueID: 1308841014.21 Dialstring: 1001 [Jun 23 16:56:54] DEBUG[1633] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/1001-00000015 CallerIDNum: 1001 CallerIDName: Uniqueid: 1308841014.21 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 23 16:56:55] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:55] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK2e0b97d3;rport=5060;received=192.168.169.60 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Cedric Autier" ;+sip.instance="" Server: Aastra 6731i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:56:55] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:55] --- (12 headers 0 lines) --- [Jun 23 16:56:55] DEBUG[1540] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '761431ef61781390634985167bd5a036@192.168.169.60:5060' Request 102: Found [Jun 23 16:56:55] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:56:55] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 6 (Ringing) [Jun 23 16:56:55] DEBUG[1532] devicestate.c: device 'SIP/1001' state '6' [Jun 23 16:56:55] DEBUG[1633] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/1001-00000015 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 1001 CallerIDName: Uniqueid: 1308841014.21 [Jun 23 16:56:55] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 23 16:56:55] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:55] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:55] VERBOSE[2037] app_dial.c: [Jun 23 16:56:55] -- SIP/1001-00000015 is ringing [Jun 23 16:56:55] DEBUG[2037] rtp_engine.c: Setting early bridge SDP of 'SIP/1000-00000014' with that of 'SIP/1001-00000015' [Jun 23 16:56:55] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:56:55] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:55] <--- Transmitting (NAT) to 192.168.169.102:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.169.102;branch=z9hG4bK0aa102135062c7541;received=192.168.169.102;rport=5060 From: "BME" ;tag=a2e37c0386 To: ;tag=as22755b3e Call-ID: 2c063ea24067f43a CSeq: 11352 INVITE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 900;refresher=uas Contact: P-Asserted-Identity: "Cedric Autier" Content-Length: 0 <------------> [Jun 23 16:56:55] DEBUG[2037] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:56:55] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:56:55] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:55] Retransmitting #1 (NAT) to 192.168.169.102:5060: NOTIFY sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7c7ea669;rport Max-Forwards: 70 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Contact: Call-ID: 70a40468b789eb3f CSeq: 120 NOTIFY User-Agent: FPBX-2.9.0(1.8.4.2) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 228

early

--- [Jun 23 16:56:55] DEBUG[1540] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:56:55] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:55] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7c7ea669;rport=5060;received=192.168.169.60 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Call-ID: 70a40468b789eb3f CSeq: 120 NOTIFY Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 6739i/3.2.2.41 Supported: path Content-Length: 0 <-------------> [Jun 23 16:56:55] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:55] --- (11 headers 0 lines) --- [Jun 23 16:56:55] DEBUG[1540] chan_sip.c: Acked pending invite 120 [Jun 23 16:56:55] DEBUG[1540] chan_sip.c: Stopping retransmission on '70a40468b789eb3f' of Request 120: Match Found [Jun 23 16:56:55] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:55] SIP Response message for INCOMING dialog NOTIFY arrived [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK2e0b97d3;rport=5060;received=192.168.169.60 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Cedric Autier" ;+sip.instance="" Server: Aastra 6731i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 1 IN IP4 192.168.169.110 s=SIP Call c=IN IP4 192.168.169.110 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.110 a=sendrecv <-------------> [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] --- (14 headers 13 lines) --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Acked pending invite 102 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Stopping retransmission on '761431ef61781390634985167bd5a036@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.110... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found RTP audio format 8 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found RTP audio format 101 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found audio description format PCMA for ID 8 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found audio description format telephone-event for ID 101 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:56:56] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4b9b4f0' [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Peer audio RTP is at port 192.168.169.110:8000 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0xb4b9b69c [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0xb4b9b69c [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: build_route: Contact hop: "Cedric Autier" ;+sip.instance="" [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] list_route: hop: [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Session-Expires: 900 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Refresher: UAS [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Session timer started: 549 - 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Strict routing enforced for session 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: Parsing for address/port to send to [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Transmitting (NAT) to 192.168.169.110:5060: ACK sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK190e6086;rport Max-Forwards: 70 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Contact: Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 102 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:56:56] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:56:56] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 2 (In use) [Jun 23 16:56:56] DEBUG[1532] devicestate.c: device 'SIP/1001' state '2' [Jun 23 16:56:56] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:56] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:56] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:56] DEBUG[1633] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 1001 Context: ext-local Hint: SIP/1001&Custom:DND1001 Status: 1 [Jun 23 16:56:56] DEBUG[1533] app_queue.c: Extension '1001@ext-local' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:56] DEBUG[1533] chan_sip.c: Strict routing enforced for session 70a40468b789eb3f [Jun 23 16:56:56] VERBOSE[1533] chan_sip.c: [Jun 23 16:56:56] set_destination: Parsing for address/port to send to [Jun 23 16:56:56] VERBOSE[1533] chan_sip.c: [Jun 23 16:56:56] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:56:56] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:56] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:56] VERBOSE[1533] chan_sip.c: [Jun 23 16:56:56] Reliably Transmitting (NAT) to 192.168.169.102:5060: NOTIFY sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK08a72347;rport Max-Forwards: 70 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Contact: Call-ID: 70a40468b789eb3f CSeq: 121 NOTIFY User-Agent: FPBX-2.9.0(1.8.4.2) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 209

confirmed

--- [Jun 23 16:56:56] DEBUG[1533] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:56:56] VERBOSE[1533] chan_sip.c: [Jun 23 16:56:56] == Extension Changed 1001[ext-local] new state InUse for Notify User 1000 [Jun 23 16:56:56] VERBOSE[2037] app_dial.c: [Jun 23 16:56:56] -- Connected line update to SIP/1000-00000014 prevented. [Jun 23 16:56:56] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:56:56] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 2 (In use) [Jun 23 16:56:56] DEBUG[1532] devicestate.c: device 'SIP/1001' state '2' [Jun 23 16:56:56] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:56] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:56] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:56] DEBUG[1633] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/1001-00000015 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 1001 CallerIDName: Uniqueid: 1308841014.21 [Jun 23 16:56:56] VERBOSE[2037] app_dial.c: [Jun 23 16:56:56] -- SIP/1001-00000015 answered SIP/1000-00000014 [Jun 23 16:56:56] DEBUG[2037] rtp_engine.c: Setting early bridge SDP of 'SIP/1000-00000014' with that of 'SIP/1001-00000015' [Jun 23 16:56:56] DEBUG[1532] chan_sip.c: Checking device state for peer 1000 [Jun 23 16:56:56] DEBUG[1532] devicestate.c: Changing state for SIP/1000 - state 2 (In use) [Jun 23 16:56:56] DEBUG[1532] devicestate.c: device 'SIP/1000' state '2' [Jun 23 16:56:56] DEBUG[1567] app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:56] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1000 [Jun 23 16:56:56] DEBUG[1533] db.c: Unable to find key 'DND1000' in family 'CustomDevstate' [Jun 23 16:56:56] DEBUG[1633] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/1000-00000014 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 1000 CallerIDName: BME Uniqueid: 1308841014.20 [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: SIP answering channel: SIP/1000-00000014 [Jun 23 16:56:56] DEBUG[2037] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Setting framing from config on incoming call [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Audio is at 5060 [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Adding codec 0x8 (alaw) to SDP [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] <--- Reliably Transmitting (NAT) to 192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.102;branch=z9hG4bK0aa102135062c7541;received=192.168.169.102;rport=5060 From: "BME" ;tag=a2e37c0386 To: ;tag=as22755b3e Call-ID: 2c063ea24067f43a CSeq: 11352 INVITE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 900;refresher=uas Contact: P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 238 v=0 o=root 191191818 191191818 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 12206 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:56:56] DEBUG[1633] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/1001-00000015 Uniqueid: 1308841014.21 AccountCode: OldAccountCode: [Jun 23 16:56:56] DEBUG[1633] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/1000-00000014 Channel2: SIP/1001-00000015 Uniqueid1: 1308841014.20 Uniqueid2: 1308841014.21 CallerID1: 1000 CallerID2: 1001 [Jun 23 16:56:56] DEBUG[2037] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:56:56] DEBUG[2037] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:56:56] VERBOSE[2037] rtp_engine.c: [Jun 23 16:56:56] -- Remotely bridging SIP/1000-00000014 and SIP/1001-00000015 [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Deferring reinvite on SIP '2c063ea24067f43a' - It's audio will be redirected to IP 192.168.169.110:8000 [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Sending reinvite on SIP '761431ef61781390634985167bd5a036@192.168.169.60:5060' - It's audio soon redirected to IP 192.168.169.102:8000 [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Strict routing enforced for session 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] set_destination: Parsing for address/port to send to [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: ** Our native-bridge filtered capablity: 0x8 (alaw) [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Audio is at 5060 [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Adding codec 0x8 (alaw) to SDP [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Initializing already initialized SIP dialog 761431ef61781390634985167bd5a036@192.168.169.60:5060 (presumably reinvite) [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Reliably Transmitting (NAT) to 192.168.169.110:5060: INVITE sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK32c4f3b9;rport Max-Forwards: 70 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Contact: Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 103 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "BME" Content-Type: application/sdp Content-Length: 239 v=0 o=root 671084360 671084361 IN IP4 192.168.169.102 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:56:56] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:56:56] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK08a72347;rport=5060;received=192.168.169.60 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Call-ID: 70a40468b789eb3f CSeq: 121 NOTIFY Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 6739i/3.2.2.41 Supported: path Content-Length: 0 <-------------> [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] --- (11 headers 0 lines) --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Acked pending invite 121 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Stopping retransmission on '70a40468b789eb3f' of Request 121: Match Found [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] SIP Response message for INCOMING dialog NOTIFY arrived [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: INVITE [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] <--- SIP read from UDP:192.168.169.102:5060 ---> ACK sip:1001@192.168.169.60:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.169.102;branch=z9hG4bKb5a591bbf06374fa2 Max-Forwards: 70 From: "BME" ;tag=a2e37c0386 To: ;tag=as22755b3e Call-ID: 2c063ea24067f43a CSeq: 11352 ACK Authorization: Digest username="1000",realm="asterisk",nonce="6c827055",uri="sip:1001@192.168.169.60:5060;user=phone",response="bfd9611073351bc91af88de7d12f7e16",algorithm=MD5 User-Agent: Aastra 6739i/3.2.2.41 Content-Length: 0 <-------------> [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] --- (10 headers 0 lines) --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Stopping retransmission on '2c063ea24067f43a' of Response 11352: Match Found [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Sending pending reinvite on '2c063ea24067f43a' [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Strict routing enforced for session 2c063ea24067f43a [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: Parsing for address/port to send to [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: ** Our native-bridge filtered capablity: 0x8 (alaw) [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Audio is at 5060 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Adding codec 0x8 (alaw) to SDP [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Initializing already initialized SIP dialog 2c063ea24067f43a (presumably reinvite) [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Reliably Transmitting (NAT) to 192.168.169.102:5060: INVITE sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK787c8f5a;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 102 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 239 v=0 o=root 191191818 191191819 IN IP4 192.168.169.110 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.110 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK32c4f3b9;rport=5060;received=192.168.169.60 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Cedric Autier" ;+sip.instance="" Server: Aastra 6731i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 2 IN IP4 192.168.169.110 s=SIP Call c=IN IP4 192.168.169.110 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.110 a=sendrecv <-------------> [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] --- (14 headers 13 lines) --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Acked pending invite 103 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Stopping retransmission on '761431ef61781390634985167bd5a036@192.168.169.60:5060' of Request 103: Match Found [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.110... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found RTP audio format 8 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found RTP audio format 101 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found audio description format PCMA for ID 8 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found audio description format telephone-event for ID 101 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:56:56] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4b9b4f0' [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Peer audio RTP is at port 192.168.169.110:8000 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0xb4b9b69c [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0xb4b9b69c [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:56:56] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:56:56] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 2 (In use) [Jun 23 16:56:56] DEBUG[1532] devicestate.c: device 'SIP/1001' state '2' [Jun 23 16:56:56] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:56] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:56] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Strict routing enforced for session 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: Parsing for address/port to send to [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Transmitting (NAT) to 192.168.169.110:5060: ACK sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK0df3d9f5;rport Max-Forwards: 70 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Contact: Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 103 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Retransmitting #1 (NAT) to 192.168.169.102:5060: INVITE sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK787c8f5a;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 102 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 239 v=0 o=root 191191818 191191819 IN IP4 192.168.169.110 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.110 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK787c8f5a;rport=5060;received=192.168.169.60 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Call-ID: 2c063ea24067f43a CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "BME" ;+sip.instance="" Server: Aastra 6739i/3.2.2.41 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 2 IN IP4 192.168.169.102 s=SIP Call c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.102 a=sendrecv <-------------> [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] --- (14 headers 13 lines) --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Acked pending invite 102 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Stopping retransmission on '2c063ea24067f43a' of Request 102: Match Found [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.169.102... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.102... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found RTP audio format 8 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found RTP audio format 101 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found audio description format PCMA for ID 8 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found audio description format telephone-event for ID 101 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.102... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:56:56] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x96d19a8' [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Peer audio RTP is at port 192.168.169.102:8000 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0x96d1b54 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0x96d1b54 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Updating call counter for incoming call [Jun 23 16:56:56] DEBUG[1532] chan_sip.c: Checking device state for peer 1000 [Jun 23 16:56:56] DEBUG[1532] devicestate.c: Changing state for SIP/1000 - state 2 (In use) [Jun 23 16:56:56] DEBUG[1532] devicestate.c: device 'SIP/1000' state '2' [Jun 23 16:56:56] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1000 [Jun 23 16:56:56] DEBUG[1567] app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:56] DEBUG[1533] db.c: Unable to find key 'DND1000' in family 'CustomDevstate' [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Strict routing enforced for session 2c063ea24067f43a [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: Parsing for address/port to send to [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Transmitting (NAT) to 192.168.169.102:5060: ACK sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK6d9749d8;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 102 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:56:56] DEBUG[2037] rtp_engine.c: Oooh, 'SIP/1000-00000014' changed end address to 192.168.169.102:8000 (format alaw) [Jun 23 16:56:56] DEBUG[2037] rtp_engine.c: Oooh, 'SIP/1000-00000014' was 192.168.169.102:8000/(format unknown) [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Sending reinvite on SIP '761431ef61781390634985167bd5a036@192.168.169.60:5060' - It's audio soon redirected to IP 192.168.169.102:8000 [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Strict routing enforced for session 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] set_destination: Parsing for address/port to send to [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: ** Our native-bridge filtered capablity: 0x8 (alaw) [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Audio is at 5060 [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Adding codec 0x8 (alaw) to SDP [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Initializing already initialized SIP dialog 761431ef61781390634985167bd5a036@192.168.169.60:5060 (presumably reinvite) [Jun 23 16:56:56] VERBOSE[2037] chan_sip.c: [Jun 23 16:56:56] Reliably Transmitting (NAT) to 192.168.169.110:5060: INVITE sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7ed486f4;rport Max-Forwards: 70 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Contact: Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 104 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "BME" Content-Type: application/sdp Content-Length: 239 v=0 o=root 671084360 671084362 IN IP4 192.168.169.102 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:56:56] DEBUG[2037] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7ed486f4;rport=5060;received=192.168.169.60 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Cedric Autier" ;+sip.instance="" Server: Aastra 6731i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 3 IN IP4 192.168.169.110 s=SIP Call c=IN IP4 192.168.169.110 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.110 a=sendrecv <-------------> [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] --- (14 headers 13 lines) --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Acked pending invite 104 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Stopping retransmission on '761431ef61781390634985167bd5a036@192.168.169.60:5060' of Request 104: Match Found [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.110... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found RTP audio format 8 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found RTP audio format 101 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found audio description format PCMA for ID 8 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Found audio description format telephone-event for ID 101 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:56:56] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4b9b4f0' [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Peer audio RTP is at port 192.168.169.110:8000 [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0xb4b9b69c [Jun 23 16:56:56] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0xb4b9b69c [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:56:56] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:56:56] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 2 (In use) [Jun 23 16:56:56] DEBUG[1532] devicestate.c: device 'SIP/1001' state '2' [Jun 23 16:56:56] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:56:56] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:56:56] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Strict routing enforced for session 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: Parsing for address/port to send to [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:56:56] VERBOSE[1540] chan_sip.c: [Jun 23 16:56:56] Transmitting (NAT) to 192.168.169.110:5060: ACK sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK374497a6;rport Max-Forwards: 70 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Contact: Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 104 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:56:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:56:57] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:56:57] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:56:58] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:56:58] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:56:59] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:56:59] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:00] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:00] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:01] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:01] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:02] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:02] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:03] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:03] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:04] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:04] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:05] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:05] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:06] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:06] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:07] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:07] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:08] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:08] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:09] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:09] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:10] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:10] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:11] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:11] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:12] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:12] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:13] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:13] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:14] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:14] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:15] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:15] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:16] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:16] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:17] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:17] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:18] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:18] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:19] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:19] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:20] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:20] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:21] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:21] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:22] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:22] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:23] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:23] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:24] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:24] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:25] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:25] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:26] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:26] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] <--- SIP read from UDP:192.168.169.110:5060 ---> INVITE sip:1000@192.168.169.60:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK192c289c8e2768e4a Max-Forwards: 70 From: ;tag=2053185131 To: "BME" ;tag=as343b04db Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 5623 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Cedric Autier" ;+sip.instance="" Min-SE: 90 Session-Expires: 900;refresher=uac Supported: path, 100rel, replaces, timer User-Agent: Aastra 6731i/3.2.1.43 Content-Type: application/sdp Content-Length: 301 v=0 o=MxSIP 0 4 IN IP4 192.168.169.110 s=SIP Call c=IN IP4 192.168.169.110 t=0 0 m=audio 8000 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.110 a=sendonly <-------------> [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] --- (16 headers 14 lines) --- [Jun 23 16:57:27] DEBUG[1540] sip/reqresp_parser.c: Begin: parsing SIP "Supported: path, 100rel, replaces, timer" [Jun 23 16:57:27] DEBUG[1540] sip/reqresp_parser.c: Found SIP option: -path- [Jun 23 16:57:27] DEBUG[1540] sip/reqresp_parser.c: Matched SIP option: path [Jun 23 16:57:27] DEBUG[1540] sip/reqresp_parser.c: Found SIP option: -100rel- [Jun 23 16:57:27] DEBUG[1540] sip/reqresp_parser.c: Matched SIP option: 100rel [Jun 23 16:57:27] DEBUG[1540] sip/reqresp_parser.c: Found SIP option: -replaces- [Jun 23 16:57:27] DEBUG[1540] sip/reqresp_parser.c: Matched SIP option: replaces [Jun 23 16:57:27] DEBUG[1540] sip/reqresp_parser.c: Found SIP option: -timer- [Jun 23 16:57:27] DEBUG[1540] sip/reqresp_parser.c: Matched SIP option: timer [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Sending to 192.168.169.110:5060 (NAT) [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Initializing initreq for method INVITE - callid 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 4 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.110... OK. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found RTP audio format 8 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5dec924 [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found RTP audio format 18 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Setting payload 18 based on m type on 0xb5dec924 [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found RTP audio format 101 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5dec924 [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found audio description format PCMA for ID 8 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found audio description format G729 for ID 18 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found audio description format telephone-event for ID 101 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5dec924 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Incorporating payload 18 on 0xb5dec924 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5dec924 [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:57:27] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4b9b4f0' [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Peer audio RTP is at port 192.168.169.110:8000 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5dec924 to 0xb4b9b69c [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Copying payload 18 from 0xb5dec924 to 0xb4b9b69c [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5dec924 to 0xb4b9b69c [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:57:27] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4b9b4f0' [Jun 23 16:57:27] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:27] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 8 (On Hold) [Jun 23 16:57:27] DEBUG[1532] devicestate.c: device 'SIP/1001' state '8' [Jun 23 16:57:27] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:27] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:27] DEBUG[1533] app_queue.c: Extension '1001@ext-local' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Jun 23 16:57:27] DEBUG[1533] chan_sip.c: Strict routing enforced for session 70a40468b789eb3f [Jun 23 16:57:27] VERBOSE[1533] chan_sip.c: [Jun 23 16:57:27] set_destination: Parsing for address/port to send to [Jun 23 16:57:27] VERBOSE[1533] chan_sip.c: [Jun 23 16:57:27] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:57:27] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Jun 23 16:57:27] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:27] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:27] VERBOSE[1533] chan_sip.c: [Jun 23 16:57:27] Reliably Transmitting (NAT) to 192.168.169.102:5060: NOTIFY sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK43859e08;rport Max-Forwards: 70 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Contact: Call-ID: 70a40468b789eb3f CSeq: 122 NOTIFY User-Agent: FPBX-2.9.0(1.8.4.2) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 324

confirmed

--- [Jun 23 16:57:27] DEBUG[1533] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:27] VERBOSE[1533] chan_sip.c: [Jun 23 16:57:27] == Extension Changed 1001[ext-local] new state Hold for Notify User 1000 [Jun 23 16:57:27] DEBUG[1633] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 1001 Context: ext-local Hint: SIP/1001&Custom:DND1001 Status: 16 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Got a SIP re-invite for call 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Session-Expires: 900 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Refresher: UAC [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Received Min-SE: 90 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Restarting session-timers on a refresh - 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Session timer stopped: -1 - 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Session timer started: 556 - 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: SIP/1001-00000015: This call is UP.... [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] <--- Transmitting (NAT) to 192.168.169.110:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK192c289c8e2768e4a;received=192.168.169.110;rport=5060 From: ;tag=2053185131 To: "BME" ;tag=as343b04db Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 5623 INVITE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 900;refresher=uac Contact: Content-Length: 0 <------------> [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Setting framing from config on incoming call [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Audio is at 5060 [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] <--- Reliably Transmitting (NAT) to 192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK192c289c8e2768e4a;received=192.168.169.110;rport=5060 From: ;tag=2053185131 To: "BME" ;tag=as343b04db Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 5623 INVITE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 900;refresher=uac Contact: Content-Type: application/sdp Content-Length: 239 v=0 o=root 671084360 671084363 IN IP4 192.168.169.102 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly <------------> [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:27] DEBUG[2037] chan_sip.c: Sending reinvite on SIP '2c063ea24067f43a' - It's audio soon redirected to IP 192.168.169.60:5060 [Jun 23 16:57:27] DEBUG[2037] chan_sip.c: Strict routing enforced for session 2c063ea24067f43a [Jun 23 16:57:27] VERBOSE[2037] chan_sip.c: [Jun 23 16:57:27] set_destination: Parsing for address/port to send to [Jun 23 16:57:27] VERBOSE[2037] chan_sip.c: [Jun 23 16:57:27] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:57:27] DEBUG[2037] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:27] DEBUG[2037] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 23 16:57:27] VERBOSE[2037] chan_sip.c: [Jun 23 16:57:27] Audio is at 5060 [Jun 23 16:57:27] VERBOSE[2037] chan_sip.c: [Jun 23 16:57:27] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:27] VERBOSE[2037] chan_sip.c: [Jun 23 16:57:27] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:27] DEBUG[2037] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:27] DEBUG[2037] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:27] DEBUG[2037] chan_sip.c: Initializing already initialized SIP dialog 2c063ea24067f43a (presumably reinvite) [Jun 23 16:57:27] VERBOSE[2037] chan_sip.c: [Jun 23 16:57:27] Reliably Transmitting (NAT) to 192.168.169.102:5060: INVITE sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK00c08820;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 103 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 238 v=0 o=root 191191818 191191820 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 12206 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:27] DEBUG[2037] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:57:27] VERBOSE[2037] res_musiconhold.c: [Jun 23 16:57:27] -- Started music on hold, class 'default', on SIP/1000-00000014 [Jun 23 16:57:27] DEBUG[2037] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:57:27] DEBUG[1633] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Start Channel: SIP/1000-00000014 UniqueID: 1308841014.20 Class: default [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK43859e08;rport=5060;received=192.168.169.60 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Call-ID: 70a40468b789eb3f CSeq: 122 NOTIFY Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 6739i/3.2.2.41 Supported: path Content-Length: 0 <-------------> [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] --- (11 headers 0 lines) --- [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Acked pending invite 122 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Stopping retransmission on '70a40468b789eb3f' of Request 122: Match Found [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] SIP Response message for INCOMING dialog NOTIFY arrived [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:27] DEBUG[2037] channel.c: Set channel SIP/1000-00000014 to write format slin [Jun 23 16:57:27] DEBUG[2037] res_musiconhold.c: SIP/1000-00000014 Opened file 0 '/var/lib/asterisk/mohmp3/fpm-sunshine' [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x96d19a8' [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] <--- SIP read from UDP:192.168.169.110:5060 ---> ACK sip:1000@192.168.169.60:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK0e6ca67988257e70c Max-Forwards: 70 From: ;tag=2053185131 To: "BME" ;tag=as343b04db Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 5623 ACK User-Agent: Aastra 6731i/3.2.1.43 Content-Length: 0 <-------------> [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] --- (9 headers 0 lines) --- [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Stopping retransmission on '761431ef61781390634985167bd5a036@192.168.169.60:5060' of Response 5623: Match Found [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Retransmitting #1 (NAT) to 192.168.169.102:5060: INVITE sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK00c08820;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 103 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 238 v=0 o=root 191191818 191191820 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 12206 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK00c08820;rport=5060;received=192.168.169.60 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Call-ID: 2c063ea24067f43a CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "BME" ;+sip.instance="" Server: Aastra 6739i/3.2.2.41 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 3 IN IP4 192.168.169.102 s=SIP Call c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.102 a=sendrecv <-------------> [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] --- (14 headers 13 lines) --- [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Acked pending invite 103 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Stopping retransmission on '2c063ea24067f43a' of Request 103: Match Found [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.169.102... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.102... OK. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found RTP audio format 8 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found RTP audio format 101 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found audio description format PCMA for ID 8 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Found audio description format telephone-event for ID 101 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.102... UNSUPPORTED. [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:57:27] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x96d19a8' [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Peer audio RTP is at port 192.168.169.102:8000 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0x96d1b54 [Jun 23 16:57:27] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0x96d1b54 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Updating call counter for incoming call [Jun 23 16:57:27] DEBUG[1532] chan_sip.c: Checking device state for peer 1000 [Jun 23 16:57:27] DEBUG[1532] devicestate.c: Changing state for SIP/1000 - state 2 (In use) [Jun 23 16:57:27] DEBUG[1532] devicestate.c: device 'SIP/1000' state '2' [Jun 23 16:57:27] DEBUG[1567] app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:57:27] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1000 [Jun 23 16:57:27] DEBUG[1533] db.c: Unable to find key 'DND1000' in family 'CustomDevstate' [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Strict routing enforced for session 2c063ea24067f43a [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] set_destination: Parsing for address/port to send to [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:57:27] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:27] Transmitting (NAT) to 192.168.169.102:5060: ACK sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK6a93f242;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 103 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:27] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:28] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:28] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:29] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:29] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] <--- SIP read from UDP:192.168.169.110:5060 ---> INVITE sip:1002@192.168.169.60:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK21da4d0f54992589f Max-Forwards: 70 From: "Cedric Autier" ;tag=1b6ace01bc To: Call-ID: 1c445c5017451751 CSeq: 18897 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Cedric Autier" ;+sip.instance="" Session-Expires: 900 Supported: path, 100rel, replaces, timer User-Agent: Aastra 6731i/3.2.1.43 Content-Type: application/sdp Content-Length: 301 v=0 o=MxSIP 0 1 IN IP4 192.168.169.110 s=SIP Call c=IN IP4 192.168.169.110 t=0 0 m=audio 8002 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8003 IN IP4 192.168.169.110 a=sendrecv <-------------> [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] --- (15 headers 14 lines) --- [Jun 23 16:57:30] DEBUG[1540] acl.c: For destination '192.168.169.110', our source address is '192.168.169.60'. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 1c445c5017451751 - INVITE (No RTP) [Jun 23 16:57:30] DEBUG[1540] sip/reqresp_parser.c: Begin: parsing SIP "Supported: path, 100rel, replaces, timer" [Jun 23 16:57:30] DEBUG[1540] sip/reqresp_parser.c: Found SIP option: -path- [Jun 23 16:57:30] DEBUG[1540] sip/reqresp_parser.c: Matched SIP option: path [Jun 23 16:57:30] DEBUG[1540] sip/reqresp_parser.c: Found SIP option: -100rel- [Jun 23 16:57:30] DEBUG[1540] sip/reqresp_parser.c: Matched SIP option: 100rel [Jun 23 16:57:30] DEBUG[1540] sip/reqresp_parser.c: Found SIP option: -replaces- [Jun 23 16:57:30] DEBUG[1540] sip/reqresp_parser.c: Matched SIP option: replaces [Jun 23 16:57:30] DEBUG[1540] sip/reqresp_parser.c: Found SIP option: -timer- [Jun 23 16:57:30] DEBUG[1540] sip/reqresp_parser.c: Matched SIP option: timer [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Sending to 192.168.169.110:5060 (no NAT) [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Initializing initreq for method INVITE - callid 1c445c5017451751 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Using INVITE request as basis request - 1c445c5017451751 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Found peer '1001' for '1001' from 192.168.169.110:5060 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] <--- Reliably Transmitting (NAT) to 192.168.169.110:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK21da4d0f54992589f;received=192.168.169.110;rport=5060 From: "Cedric Autier" ;tag=1b6ace01bc To: ;tag=as3c8787c3 Call-ID: 1c445c5017451751 CSeq: 18897 INVITE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="00e9145c" Content-Length: 0 <------------> [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Scheduling destruction of SIP dialog '1c445c5017451751' in 6400 ms (Method: INVITE) [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] <--- SIP read from UDP:192.168.169.110:5060 ---> ACK sip:1002@192.168.169.60:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK21da4d0f54992589f Max-Forwards: 70 From: "Cedric Autier" ;tag=1b6ace01bc To: ;tag=as3c8787c3 Call-ID: 1c445c5017451751 CSeq: 18897 ACK User-Agent: Aastra 6731i/3.2.1.43 Content-Length: 0 <-------------> [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] --- (9 headers 0 lines) --- [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Stopping retransmission on '1c445c5017451751' of Response 18897: Match Found [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] <--- SIP read from UDP:192.168.169.110:5060 ---> INVITE sip:1002@192.168.169.60:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK24d8e1d34cd98f20e Max-Forwards: 70 From: "Cedric Autier" ;tag=1b6ace01bc To: Call-ID: 1c445c5017451751 CSeq: 18898 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="1001",realm="asterisk",nonce="00e9145c",uri="sip:1002@192.168.169.60:5060;user=phone",response="6eaebbef77f6177114ad1b15b475a71d",algorithm=MD5 Contact: "Cedric Autier" ;+sip.instance="" Session-Expires: 900 Supported: path, 100rel, replaces, timer User-Agent: Aastra 6731i/3.2.1.43 Content-Type: application/sdp Content-Length: 301 v=0 o=MxSIP 0 1 IN IP4 192.168.169.110 s=SIP Call c=IN IP4 192.168.169.110 t=0 0 m=audio 8002 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8003 IN IP4 192.168.169.110 a=sendrecv <-------------> [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] --- (16 headers 14 lines) --- [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Sending to 192.168.169.110:5060 (NAT) [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Initializing initreq for method INVITE - callid 1c445c5017451751 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Using INVITE request as basis request - 1c445c5017451751 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Found peer '1001' for '1001' from 192.168.169.110:5060 [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x96c7c20' [Jun 23 16:57:30] DEBUG[1540] res_rtp_asterisk.c: Allocated port 10672 for RTP instance '0x96c7c20' [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: RTP instance '0x96c7c20' is setup and ready to go [Jun 23 16:57:30] DEBUG[1540] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x96c7c20' [Jun 23 16:57:30] VERBOSE[1540] netsock2.c: [Jun 23 16:57:30] == Using SIP RTP TOS bits 184 [Jun 23 16:57:30] VERBOSE[1540] netsock2.c: [Jun 23 16:57:30] == Using SIP RTP CoS mark 5 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Setting NAT on RTP to On [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.110... OK. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Found RTP audio format 8 [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5dec924 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Found RTP audio format 18 [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Setting payload 18 based on m type on 0xb5dec924 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Found RTP audio format 101 [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5dec924 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Found audio description format PCMA for ID 8 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Found audio description format G729 for ID 18 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Found audio description format telephone-event for ID 101 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8003 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5dec924 [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Incorporating payload 18 on 0xb5dec924 [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5dec924 [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:57:30] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x96c7c20' [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Peer audio RTP is at port 192.168.169.110:8002 [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5dec924 to 0x96c7dcc [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Copying payload 18 from 0xb5dec924 to 0x96c7dcc [Jun 23 16:57:30] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5dec924 to 0x96c7dcc [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Checking SIP call limits for device 1001 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Updating call counter for incoming call [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Call from peer '1001' is 2 out of 2147483647 [Jun 23 16:57:30] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:30] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 8 (On Hold) [Jun 23 16:57:30] DEBUG[1532] devicestate.c: device 'SIP/1001' state '8' [Jun 23 16:57:30] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Jun 23 16:57:30] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:30] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Looking for 1002 in ClassOfService (domain 192.168.169.60:5060) [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/1001-00000016 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 1001 CallerIDName: P1001 AccountCode: Exten: 1002 Context: ClassOfService Uniqueid: 1308841050.22 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: This channel will not be able to handle video. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: build_route: Contact hop: "Cedric Autier" ;+sip.instance="" [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] list_route: hop: [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header. [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Session-Expires: 900 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Session timer started: 562 - 1c445c5017451751 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: SIP/1001-00000016: New call is still down.... Trying... [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] <--- Transmitting (NAT) to 192.168.169.110:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK24d8e1d34cd98f20e;received=192.168.169.110;rport=5060 From: "Cedric Autier" ;tag=1b6ace01bc To: Call-ID: 1c445c5017451751 CSeq: 18898 INVITE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 900;refresher=uas Contact: Content-Length: 0 <------------> [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:30] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:30] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 8 (On Hold) [Jun 23 16:57:30] DEBUG[1532] devicestate.c: device 'SIP/1001' state '8' [Jun 23 16:57:30] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:30] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:30] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/1001-00000016 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 1001 CallerIDName: P1001 Uniqueid: 1308841050.22 [Jun 23 16:57:30] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Verbose' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@ClassOfService:1] Verbose("SIP/1001-00000016", "2|TNE2:Passage COS") in new stack [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:30] WARNING[2041] pbx.c: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Verbose(2|TNE2:Passage COS)) [Jun 23 16:57:30] VERBOSE[2041] app_verbose.c: [Jun 23 16:57:30] 2|TNE2:Passage COS [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1001' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@ClassOfService:2] Set("SIP/1001-00000016", "DEVICE=1001") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1001' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@ClassOfService:3] Set("SIP/1001-00000016", "COSNUM=1001") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'from-internal' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@ClassOfService:4] Set("SIP/1001-00000016", "AMPUSERCOS=from-internal") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'Cedric Autier' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@ClassOfService:5] Set("SIP/1001-00000016", "AMPUSERNAME=Cedric Autier") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@ClassOfService:6] Set("SIP/1001-00000016", "CDR(userfield)=1001") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@ClassOfService:7] Set("SIP/1001-00000016", "CDR(name)=Cedric Autier") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@ClassOfService:8] GotoIf("SIP/1001-00000016", "0?pasdecos") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Goto' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@ClassOfService:9] Goto("SIP/1001-00000016", "from-internal,1002,1") in new stack [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Goto (from-internal,1002,1) [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@from-internal:1] ExecIf("SIP/1001-00000016", "0?Set(__RINGTIMER=0)") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Macro' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [1002@from-internal:2] Macro("SIP/1001-00000016", "exten-vm,novm,1002,0,0,0") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Macro' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-exten-vm:1] Macro("SIP/1001-00000016", "user-callerid,") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1001' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1001' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:1] Set("SIP/1001-00000016", "AMPUSER=1001") in new stack [Jun 23 16:57:30] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1001-00000016", "0?report") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1001' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1001-00000016", "1?Set(REALCALLERIDNUM=1001)") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1001' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1001' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:4] Set("SIP/1001-00000016", "AMPUSER=1001") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'Cedric Autier' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:5] Set("SIP/1001-00000016", "AMPUSERCIDNAME=Cedric Autier") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1001-00000016", "0?report") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1001' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:7] Set("SIP/1001-00000016", "AMPUSERCID=1001") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:8] Set("SIP/1001-00000016", "CALLERID(all)="Cedric Autier" <1001>") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/1001-00000016 CallerIDNum: 1001 CallerIDName: Cedric Autier Uniqueid: 1308841050.22 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '4' [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1001/concurrency_limit' in family 'AMPUSER' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: AMPUSER/1001/concurrency_limit not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '0' [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1001/concurrency_limit' in family 'AMPUSER' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: AMPUSER/1001/concurrency_limit not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:9] GotoIf("SIP/1001-00000016", "0?limit") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '4' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:10] ExecIf("SIP/1001-00000016", "0?Set(GROUP(concurrency_limit)=1001)") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '4' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:11] ExecIf("SIP/1001-00000016", "0?Set(CHANNEL(language)=)") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1001-00000016", "0?continue") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '-1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '64' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:13] Set("SIP/1001-00000016", "__TTL=64") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1001-00000016", "1?continue") in new stack [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Goto (macro-user-callerid,s,25) [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1001' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:25] Set("SIP/1001-00000016", "CALLERID(number)=1001") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/1001-00000016 CallerIDNum: 1001 CallerIDName: Cedric Autier Uniqueid: 1308841050.22 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'Cedric Autier' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:26] Set("SIP/1001-00000016", "CALLERID(name)=Cedric Autier") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/1001-00000016 CallerIDNum: 1001 CallerIDName: Cedric Autier Uniqueid: 1308841050.22 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'fr' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'fr' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-user-callerid:27] Set("SIP/1001-00000016", "CHANNEL(language)=fr") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Macro [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-exten-vm:2] Set("SIP/1001-00000016", "RingGroupMethod=none") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-exten-vm:3] Set("SIP/1001-00000016", "__EXTTOCALL=1002") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-exten-vm:4] Set("SIP/1001-00000016", "__PICKUPMARK=1002") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1002' in family 'CFU' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: CFU/1002 not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1002' in family 'CFB' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: CFB/1002 not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '""' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-exten-vm:5] Set("SIP/1001-00000016", "RT=""") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Macro' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-exten-vm:6] Macro("SIP/1001-00000016", "record-enable,1002,IN") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-record-enable:1] GotoIf("SIP/1001-00000016", "1?check") in new stack [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Goto (macro-record-enable,s,4) [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-record-enable:4] ExecIf("SIP/1001-00000016", "0?MacroExit()") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-record-enable:5] GotoIf("SIP/1001-00000016", "0?Group:OUT") in new stack [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Goto (macro-record-enable,s,14) [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-record-enable:14] GotoIf("SIP/1001-00000016", "1?IN") in new stack [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Goto (macro-record-enable,s,18) [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'out=Adhoc|in=Adhoc' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'in=Adhoc' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-record-enable:18] ExecIf("SIP/1001-00000016", "1?MacroExit()") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Macro [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Macro' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-exten-vm:7] Macro("SIP/1001-00000016", "dial-one,"",,1002") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:1] Set("SIP/1001-00000016", "DEXTEN=1002") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:2] Set("SIP/1001-00000016", "DIALSTATUS_CW=") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1002/screen' in family 'AMPUSER' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: AMPUSER/1002/screen not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GosubIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:3] GosubIf("SIP/1001-00000016", "0?screen,1") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GosubIf [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1002/screen' in family 'AMPUSER' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: AMPUSER/1002/screen not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1002' in family 'CF' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: CF/1002 not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GosubIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:4] GosubIf("SIP/1001-00000016", "0?cf,1") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GosubIf [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1002' in family 'CF' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: CF/1002 not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1002' in family 'DND' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: DND/1002 not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:5] GotoIf("SIP/1001-00000016", "1?skip1") in new stack [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Goto (macro-dial-one,s,8) [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:8] GotoIf("SIP/1001-00000016", "0?nodial") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:9] GotoIf("SIP/1001-00000016", "0?continue") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'ENABLED' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'ENABLED' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:10] Set("SIP/1001-00000016", "EXTHASCW=ENABLED") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1002' in family 'CFB' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: CFB/1002 not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key '1002' in family 'CFU' [Jun 23 16:57:30] DEBUG[2041] func_db.c: DB: CFU/1002 not found in database. [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:11] GotoIf("SIP/1001-00000016", "0?next1:cwinusebusy") in new stack [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Goto (macro-dial-one,s,23) [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:23] GotoIf("SIP/1001-00000016", "1?next3:continue") in new stack [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Goto (macro-dial-one,s,24) [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:57:30] DEBUG[2041] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:57:30] DEBUG[2041] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:24] ExecIf("SIP/1001-00000016", "0?Set(DIALSTATUS_CW=BUSY)") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:57:30] DEBUG[2041] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:57:30] DEBUG[2041] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:30] DEBUG[2041] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'NOT_INUSE' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:25] GotoIf("SIP/1001-00000016", "0?nodial") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GosubIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:26] GosubIf("SIP/1001-00000016", "1?dstring,1:dlocal,1") in new stack [Jun 23 16:57:30] DEBUG[2041] app_stack.c: Channel SIP/1001-00000016 has no datastore, so we're allocating one. [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GosubIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Incrementing gosub_level [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:1] Set("SIP/1001-00000016", "DSTRING=") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1002' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:2] Set("SIP/1001-00000016", "DEVICES=1002") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/1001-00000016", "0?Return()") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/1001-00000016", "0?Set(DEVICES=002)") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:5] Set("SIP/1001-00000016", "LOOPCNT=1") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:6] Set("SIP/1001-00000016", "ITER=1") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1002' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'SIP/1002' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:7] Set("SIP/1001-00000016", "THISDIAL=SIP/1002") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GosubIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/1001-00000016", "1?zap2dahdi,1") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GosubIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Incrementing gosub_level [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/1001-00000016", "0?Return()") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/1001-00000016", "NEWDIAL=") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/1001-00000016", "LOOPCNT2=1") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/1001-00000016", "ITER2=1") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'SIP/1002' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/1001-00000016", "THISPART2=SIP/1002") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/1001-00000016", "0?Set(THISPART2=DAHDI/1002)") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/1001-00000016", "NEWDIAL=SIP/1002&") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '2' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/1001-00000016", "ITER2=2") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/1001-00000016", "0?begin2") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '9' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '8' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/1001-00000016", "THISDIAL=SIP/1002") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Return' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/1001-00000016", "") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Return [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Decrementing gosub_level [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:9] Set("SIP/1001-00000016", "DSTRING=SIP/1002&") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '2' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:10] Set("SIP/1001-00000016", "ITER=2") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/1001-00000016", "0?begin") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '9' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '8' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:12] Set("SIP/1001-00000016", "DSTRING=SIP/1002") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Return' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [dstring@macro-dial-one:13] Return("SIP/1001-00000016", "") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Return [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Decrementing gosub_level [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '8' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:27] GotoIf("SIP/1001-00000016", "0?nodial") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:28] GotoIf("SIP/1001-00000016", "1?skiptrace") in new stack [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Goto (macro-dial-one,s,30) [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] func_strings.c: FUNCTION REGEX ((M[(]auto-blkvm[)]))() [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is '' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:30] Set("SIP/1001-00000016", "D_OPTIONS=") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:31] ExecIf("SIP/1001-00000016", "0?SIPAddHeader(Alert-Info: )") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:32] ExecIf("SIP/1001-00000016", "0?SIPAddHeader()") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'ExecIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:33] ExecIf("SIP/1001-00000016", "0?Set(CHANNEL(musicclass)=)") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: ExecIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GosubIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:34] GosubIf("SIP/1001-00000016", "0?qwait,1") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GosubIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:35] Set("SIP/1001-00000016", "__CWIGNORE=") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:36] Set("SIP/1001-00000016", "__KEEPCID=TRUE") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:37] GotoIf("SIP/1001-00000016", "0?usegoto,1") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'TCE' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Expression result is '0' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:38] GotoIf("SIP/1001-00000016", "0?godial") in new stack [Jun 23 16:57:30] DEBUG[2041] pbx.c: Not taking any branch [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:30] DEBUG[2041] pbx.c: Function result is 'TCE' [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:39] Set("SIP/1001-00000016", "CONNECTEDLINE(name,i)=TCE") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:40] Set("SIP/1001-00000016", "CONNECTEDLINE(num)=1002") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Set' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:41] Set("SIP/1001-00000016", "D_OPTIONS=I") in new stack [Jun 23 16:57:30] DEBUG[2041] app_macro.c: Executed application: Set [Jun 23 16:57:30] DEBUG[2041] pbx.c: Launching 'Dial' [Jun 23 16:57:30] VERBOSE[2041] pbx.c: [Jun 23 16:57:30] -- Executing [s@macro-dial-one:42] Dial("SIP/1001-00000016", "SIP/1002,"",I") in new stack [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Allocating new SIP dialog for 03033fa33ec266aa0613454e01c0b11c@127.0.1.1:0 - INVITE (No RTP) [Jun 23 16:57:30] DEBUG[2041] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6ed2020' [Jun 23 16:57:30] DEBUG[2041] res_rtp_asterisk.c: Allocated port 15714 for RTP instance '0xb6ed2020' [Jun 23 16:57:30] DEBUG[2041] rtp_engine.c: RTP instance '0xb6ed2020' is setup and ready to go [Jun 23 16:57:30] DEBUG[2041] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6ed2020' [Jun 23 16:57:30] VERBOSE[2041] netsock2.c: [Jun 23 16:57:30] == Using SIP RTP TOS bits 184 [Jun 23 16:57:30] VERBOSE[2041] netsock2.c: [Jun 23 16:57:30] == Using SIP RTP CoS mark 5 [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Setting NAT on RTP to On [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 23 16:57:30] DEBUG[2041] acl.c: For destination '192.168.169.100', our source address is '192.168.169.60'. [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: This channel will not be able to handle video. [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/1002-00000017 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: 1002 CallerIDName: P1002 AccountCode: Exten: Context: ClassOfService Uniqueid: 1308841050.23 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:30] DEBUG[2041] rtp_engine.c: Seeded SDP of 'SIP/1002-00000017' with that of 'SIP/1001-00000016' [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DIALEDTIME. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable ANSWEREDTIME. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DIALEDPEERNAME. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DIALEDPEERNUMBER. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DIALSTATUS. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable MACRO_DEPTH. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable D_OPTIONS. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DB_RESULT. [Jun 23 16:57:30] DEBUG[2041] channel.c: Copying hard-transferable variable KEEPCID. [Jun 23 16:57:30] DEBUG[2041] channel.c: Copying hard-transferable variable CWIGNORE. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable GOSUB_RETVAL. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DSTRING. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable ITER. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable THISDIAL. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable ITER2. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable NEWDIAL. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable THISPART2. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable LOOPCNT2. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable LOOPCNT. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DEVICES. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable EXTHASCW. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DIALSTATUS_CW. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DEXTEN. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable ARG3. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable ARG2. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable ARG1. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable MACRO_PRIORITY. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable MACRO_CONTEXT. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable MACRO_EXTEN. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable RT. [Jun 23 16:57:30] DEBUG[2041] channel.c: Copying hard-transferable variable PICKUPMARK. [Jun 23 16:57:30] DEBUG[2041] channel.c: Copying hard-transferable variable EXTTOCALL. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable RingGroupMethod. [Jun 23 16:57:30] DEBUG[2041] channel.c: Copying hard-transferable variable TTL. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable AMPUSERCID. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable AMPUSERCIDNAME. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable AMPUSER. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable REALCALLERIDNUM. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable ARG5. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable ARG4. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable AMPUSERNAME. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable AMPUSERCOS. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable COSNUM. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable DEVICE. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable SIPCALLID. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable SIPDOMAIN. [Jun 23 16:57:30] DEBUG[2041] channel.c: Not copying variable SIPURI. [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Outgoing Call for 1002 [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Call to peer '1002' is 1 out of 2147483647 [Jun 23 16:57:30] DEBUG[1532] chan_sip.c: Checking device state for peer 1002 [Jun 23 16:57:30] DEBUG[1532] devicestate.c: Changing state for SIP/1002 - state 6 (Ringing) [Jun 23 16:57:30] DEBUG[1532] devicestate.c: device 'SIP/1002' state '6' [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False Text flag: False [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 23 16:57:30] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:30] Audio is at 5060 [Jun 23 16:57:30] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:30] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:30] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:30] Adding codec 0x2 (gsm) to SDP [Jun 23 16:57:30] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:30] Adding codec 0x4 (ulaw) to SDP [Jun 23 16:57:30] DEBUG[1567] app_queue.c: Device 'SIP/1002' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 23 16:57:30] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:30] DEBUG[1533] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:30] DEBUG[1533] app_queue.c: Extension '1002@ext-local' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 1002 Context: ext-local Hint: SIP/1002&Custom:DND1002 Status: 8 [Jun 23 16:57:30] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:30] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Initializing initreq for method INVITE - callid 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:30] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:30] Reliably Transmitting (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK4dc3a374;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 102 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:57:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 283 v=0 o=root 72735859 72735859 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 15714 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/1001-00000016 Destination: SIP/1002-00000017 CallerIDNum: 1001 CallerIDName: Cedric Autier UniqueID: 1308841050.22 DestUniqueID: 1308841050.23 Dialstring: 1002 [Jun 23 16:57:30] VERBOSE[2041] app_dial.c: [Jun 23 16:57:30] -- Called 1002 [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/1002-00000017 CallerIDNum: 1002 CallerIDName: Uniqueid: 1308841050.23 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] Retransmitting #1 (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK4dc3a374;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 102 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:57:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 283 v=0 o=root 72735859 72735859 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 15714 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK4dc3a374;rport=5060;received=192.168.169.60 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "TCE" ;+sip.instance="" Server: Aastra 57i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:57:30] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:30] --- (12 headers 0 lines) --- [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Request 102: Found [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:30] DEBUG[1532] chan_sip.c: Checking device state for peer 1002 [Jun 23 16:57:30] DEBUG[1532] devicestate.c: Changing state for SIP/1002 - state 6 (Ringing) [Jun 23 16:57:30] DEBUG[1532] devicestate.c: device 'SIP/1002' state '6' [Jun 23 16:57:30] DEBUG[1633] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/1002-00000017 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 1002 CallerIDName: Uniqueid: 1308841050.23 [Jun 23 16:57:30] DEBUG[1567] app_queue.c: Device 'SIP/1002' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 23 16:57:30] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:30] DEBUG[1533] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:30] VERBOSE[2041] app_dial.c: [Jun 23 16:57:30] -- SIP/1002-00000017 is ringing [Jun 23 16:57:30] DEBUG[2041] rtp_engine.c: Setting early bridge SDP of 'SIP/1001-00000016' with that of 'SIP/1002-00000017' [Jun 23 16:57:30] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:30] <--- Transmitting (NAT) to 192.168.169.110:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK24d8e1d34cd98f20e;received=192.168.169.110;rport=5060 From: "Cedric Autier" ;tag=1b6ace01bc To: ;tag=as3d35f19b Call-ID: 1c445c5017451751 CSeq: 18898 INVITE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 900;refresher=uas Contact: P-Asserted-Identity: "TCE" Content-Length: 0 <------------> [Jun 23 16:57:30] DEBUG[2041] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:30] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:57:30] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:30] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:31] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Got RTCP report of 84 bytes [Jun 23 16:57:31] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:32] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:32] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK4dc3a374;rport=5060;received=192.168.169.60 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "TCE" ;+sip.instance="" Server: Aastra 57i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 1 IN IP4 192.168.169.100 s=SIP Call c=IN IP4 192.168.169.100 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.100 a=sendrecv <-------------> [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] --- (14 headers 13 lines) --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Acked pending invite 102 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Stopping retransmission on '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.169.100... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.100... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found RTP audio format 8 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found RTP audio format 101 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found audio description format PCMA for ID 8 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found audio description format telephone-event for ID 101 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.100... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:57:33] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6ed2020' [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Peer audio RTP is at port 192.168.169.100:8000 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0xb6ed21cc [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0xb6ed21cc [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: build_route: Contact hop: "TCE" ;+sip.instance="" [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] list_route: hop: [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Session-Expires: 900 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Refresher: UAS [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Session timer started: 567 - 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: Parsing for address/port to send to [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Transmitting (NAT) to 192.168.169.100:5060: ACK sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK34f033b0;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 102 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:33] DEBUG[1532] chan_sip.c: Checking device state for peer 1002 [Jun 23 16:57:33] DEBUG[1532] devicestate.c: Changing state for SIP/1002 - state 2 (In use) [Jun 23 16:57:33] DEBUG[1532] devicestate.c: device 'SIP/1002' state '2' [Jun 23 16:57:33] DEBUG[1567] app_queue.c: Device 'SIP/1002' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:57:33] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:33] DEBUG[1533] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:33] DEBUG[1533] app_queue.c: Extension '1002@ext-local' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:57:33] DEBUG[1633] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 1002 Context: ext-local Hint: SIP/1002&Custom:DND1002 Status: 1 [Jun 23 16:57:33] VERBOSE[2041] app_dial.c: [Jun 23 16:57:33] -- Connected line update to SIP/1001-00000016 prevented. [Jun 23 16:57:33] VERBOSE[2041] app_dial.c: [Jun 23 16:57:33] -- SIP/1002-00000017 answered SIP/1001-00000016 [Jun 23 16:57:33] DEBUG[2041] rtp_engine.c: Setting early bridge SDP of 'SIP/1001-00000016' with that of 'SIP/1002-00000017' [Jun 23 16:57:33] DEBUG[1633] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/1002-00000017 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 1002 CallerIDName: Uniqueid: 1308841050.23 [Jun 23 16:57:33] DEBUG[1633] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/1001-00000016 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 1001 CallerIDName: Cedric Autier Uniqueid: 1308841050.22 [Jun 23 16:57:33] DEBUG[1532] chan_sip.c: Checking device state for peer 1002 [Jun 23 16:57:33] DEBUG[1532] devicestate.c: Changing state for SIP/1002 - state 2 (In use) [Jun 23 16:57:33] DEBUG[1532] devicestate.c: device 'SIP/1002' state '2' [Jun 23 16:57:33] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:33] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 8 (On Hold) [Jun 23 16:57:33] DEBUG[1532] devicestate.c: device 'SIP/1001' state '8' [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: SIP answering channel: SIP/1001-00000016 [Jun 23 16:57:33] DEBUG[2041] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Setting framing from config on incoming call [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] Audio is at 5060 [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] <--- Reliably Transmitting (NAT) to 192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK24d8e1d34cd98f20e;received=192.168.169.110;rport=5060 From: "Cedric Autier" ;tag=1b6ace01bc To: ;tag=as3d35f19b Call-ID: 1c445c5017451751 CSeq: 18898 INVITE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 900;refresher=uas Contact: P-Asserted-Identity: "TCE" Content-Type: application/sdp Content-Length: 240 v=0 o=root 1538385577 1538385577 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 10672 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jun 23 16:57:33] DEBUG[1567] app_queue.c: Device 'SIP/1002' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:57:33] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:33] DEBUG[2041] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:57:33] DEBUG[2041] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:57:33] VERBOSE[2041] rtp_engine.c: [Jun 23 16:57:33] -- Remotely bridging SIP/1001-00000016 and SIP/1002-00000017 [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Deferring reinvite on SIP '1c445c5017451751' - It's audio will be redirected to IP 192.168.169.100:8000 [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Sending reinvite on SIP '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' - It's audio soon redirected to IP 192.168.169.110:8002 [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] set_destination: Parsing for address/port to send to [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: ** Our native-bridge filtered capablity: 0x8 (alaw) [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] Audio is at 5060 [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Initializing already initialized SIP dialog 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 (presumably reinvite) [Jun 23 16:57:33] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:33] Reliably Transmitting (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK215ba30d;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 103 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 237 v=0 o=root 72735859 72735860 IN IP4 192.168.169.110 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.110 t=0 0 m=audio 8002 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: INVITE [Jun 23 16:57:33] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:57:33] DEBUG[1633] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/1002-00000017 Uniqueid: 1308841050.23 AccountCode: OldAccountCode: [Jun 23 16:57:33] DEBUG[1633] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/1001-00000016 Channel2: SIP/1002-00000017 Uniqueid1: 1308841050.22 Uniqueid2: 1308841050.23 CallerID1: 1001 CallerID2: 1002 [Jun 23 16:57:33] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:33] DEBUG[1533] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:33] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:33] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:33] DEBUG[1634] manager.c: Running action 'Getvar' [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] <--- SIP read from UDP:192.168.169.110:5060 ---> ACK sip:1002@192.168.169.60:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK730e095d496e20be3 Max-Forwards: 70 From: "Cedric Autier" ;tag=1b6ace01bc To: ;tag=as3d35f19b Call-ID: 1c445c5017451751 CSeq: 18898 ACK Authorization: Digest username="1001",realm="asterisk",nonce="00e9145c",uri="sip:1002@192.168.169.60:5060;user=phone",response="6eaebbef77f6177114ad1b15b475a71d",algorithm=MD5 User-Agent: Aastra 6731i/3.2.1.43 Content-Length: 0 <-------------> [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] --- (10 headers 0 lines) --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Stopping retransmission on '1c445c5017451751' of Response 18898: Match Found [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Sending pending reinvite on '1c445c5017451751' [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Strict routing enforced for session 1c445c5017451751 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: Parsing for address/port to send to [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: ** Our native-bridge filtered capablity: 0x8 (alaw) [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Audio is at 5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Initializing already initialized SIP dialog 1c445c5017451751 (presumably reinvite) [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Reliably Transmitting (NAT) to 192.168.169.110:5060: INVITE sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK1a0bddbd;rport Max-Forwards: 70 From: ;tag=as3d35f19b To: "Cedric Autier" ;tag=1b6ace01bc Contact: Call-ID: 1c445c5017451751 CSeq: 102 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "TCE" Content-Type: application/sdp Content-Length: 241 v=0 o=root 1538385577 1538385578 IN IP4 192.168.169.100 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.100 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Retransmitting #1 (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK215ba30d;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 103 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 237 v=0 o=root 72735859 72735860 IN IP4 192.168.169.110 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.110 t=0 0 m=audio 8002 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK1a0bddbd;rport=5060;received=192.168.169.60 From: ;tag=as3d35f19b To: "Cedric Autier" ;tag=1b6ace01bc Call-ID: 1c445c5017451751 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Cedric Autier" ;+sip.instance="" Server: Aastra 6731i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 2 IN IP4 192.168.169.110 s=SIP Call c=IN IP4 192.168.169.110 t=0 0 m=audio 8002 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8003 IN IP4 192.168.169.110 a=sendrecv <-------------> [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] --- (14 headers 13 lines) --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Acked pending invite 102 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Stopping retransmission on '1c445c5017451751' of Request 102: Match Found [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.110... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found RTP audio format 8 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found RTP audio format 101 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found audio description format PCMA for ID 8 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found audio description format telephone-event for ID 101 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8003 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:57:33] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x96c7c20' [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Peer audio RTP is at port 192.168.169.110:8002 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0x96c7dcc [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0x96c7dcc [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Updating call counter for incoming call [Jun 23 16:57:33] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:33] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 8 (On Hold) [Jun 23 16:57:33] DEBUG[1532] devicestate.c: device 'SIP/1001' state '8' [Jun 23 16:57:33] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Jun 23 16:57:33] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:33] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Strict routing enforced for session 1c445c5017451751 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: Parsing for address/port to send to [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Transmitting (NAT) to 192.168.169.110:5060: ACK sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK60ca9207;rport Max-Forwards: 70 From: ;tag=as3d35f19b To: "Cedric Autier" ;tag=1b6ace01bc Contact: Call-ID: 1c445c5017451751 CSeq: 102 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:33] DEBUG[2041] rtp_engine.c: Oooh, 'SIP/1001-00000016' changed end address to 192.168.169.110:8002 (format alaw) [Jun 23 16:57:33] DEBUG[2041] rtp_engine.c: Oooh, 'SIP/1001-00000016' was 192.168.169.110:8002/(format unknown) [Jun 23 16:57:33] DEBUG[2041] chan_sip.c: Deferring reinvite on SIP '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' - It's audio will be redirected to IP 192.168.169.110:8002 [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK215ba30d;rport=5060;received=192.168.169.60 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "TCE" ;+sip.instance="" Server: Aastra 57i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 2 IN IP4 192.168.169.100 s=SIP Call c=IN IP4 192.168.169.100 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.100 a=sendrecv <-------------> [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] --- (14 headers 13 lines) --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Acked pending invite 103 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Stopping retransmission on '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' of Request 103: Match Found [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 192.168.169.100... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.100... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found RTP audio format 8 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found RTP audio format 101 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found audio description format PCMA for ID 8 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found audio description format telephone-event for ID 101 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.100... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:57:33] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6ed2020' [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Peer audio RTP is at port 192.168.169.100:8000 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0xb6ed21cc [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0xb6ed21cc [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: Parsing for address/port to send to [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Transmitting (NAT) to 192.168.169.100:5060: ACK sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK4b58c68d;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 103 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Sending pending reinvite on '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: Parsing for address/port to send to [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: ** Our native-bridge filtered capablity: 0x8 (alaw) [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Audio is at 5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Initializing already initialized SIP dialog 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 (presumably reinvite) [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Reliably Transmitting (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK5ecf4a9a;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 104 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 237 v=0 o=root 72735859 72735861 IN IP4 192.168.169.110 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.110 t=0 0 m=audio 8002 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:33] DEBUG[1532] chan_sip.c: Checking device state for peer 1002 [Jun 23 16:57:33] DEBUG[1532] devicestate.c: Changing state for SIP/1002 - state 2 (In use) [Jun 23 16:57:33] DEBUG[1532] devicestate.c: device 'SIP/1002' state '2' [Jun 23 16:57:33] DEBUG[1567] app_queue.c: Device 'SIP/1002' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:57:33] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:33] DEBUG[1533] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Retransmitting #1 (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK5ecf4a9a;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 104 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 237 v=0 o=root 72735859 72735861 IN IP4 192.168.169.110 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.110 t=0 0 m=audio 8002 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK5ecf4a9a;rport=5060;received=192.168.169.60 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "TCE" ;+sip.instance="" Server: Aastra 57i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 3 IN IP4 192.168.169.100 s=SIP Call c=IN IP4 192.168.169.100 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.100 a=sendrecv <-------------> [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] --- (14 headers 13 lines) --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Acked pending invite 104 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Stopping retransmission on '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' of Request 104: Match Found [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.169.100... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.100... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found RTP audio format 8 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found RTP audio format 101 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found audio description format PCMA for ID 8 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Found audio description format telephone-event for ID 101 [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8001 IN IP4 192.168.169.100... UNSUPPORTED. [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:57:33] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6ed2020' [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Peer audio RTP is at port 192.168.169.100:8000 [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0xb6ed21cc [Jun 23 16:57:33] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0xb6ed21cc [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: Parsing for address/port to send to [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:33] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:33] Transmitting (NAT) to 192.168.169.100:5060: ACK sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK6f29a330;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 104 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:33] DEBUG[1532] chan_sip.c: Checking device state for peer 1002 [Jun 23 16:57:33] DEBUG[1532] devicestate.c: Changing state for SIP/1002 - state 2 (In use) [Jun 23 16:57:33] DEBUG[1532] devicestate.c: device 'SIP/1002' state '2' [Jun 23 16:57:33] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:33] DEBUG[1567] app_queue.c: Device 'SIP/1002' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 23 16:57:33] DEBUG[1533] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:33] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:34] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:34] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:34] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:34] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:35] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:35] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:35] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:35] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:36] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: ACK [Jun 23 16:57:36] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:36] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Got RTCP report of 84 bytes [Jun 23 16:57:36] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.110:5060 ---> REFER sip:1000@192.168.169.60:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bKc2b80dcce9b6f0d39 Max-Forwards: 70 From: ;tag=2053185131 To: "BME" ;tag=as343b04db Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 5624 REFER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Cedric Autier" ;+sip.instance="" Refer-To: Referred-By: Supported: path, timer User-Agent: Aastra 6731i/3.2.1.43 Content-Length: 0 <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (15 headers 0 lines) --- [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Call 761431ef61781390634985167bd5a036@192.168.169.60:5060 got a SIP call transfer from caller: (REFER)! [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 1c445c5017451751 F-tag: 1b6ace01bc T-tag: as3d35f19b [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] SIP transfer to extension 1002@from-internal-xfer by 1001@192.168.169.60 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: SIP attended transfer: Transferer channel SIP/1001-00000015, transferee channel SIP/1000-00000014 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- Transmitting (NAT) to 192.168.169.110:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bKc2b80dcce9b6f0d39;received=192.168.169.110;rport=5060 From: ;tag=2053185131 To: "BME" ;tag=as343b04db Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 5624 REFER Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: SIP transfer: Four channels to handle [Jun 23 16:57:37] VERBOSE[1540] res_musiconhold.c: [Jun 23 16:57:37] -- Stopped music on hold on SIP/1000-00000014 [Jun 23 16:57:37] DEBUG[1540] channel.c: Set channel SIP/1000-00000014 to write format alaw [Jun 23 16:57:37] DEBUG[1540] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 23 16:57:37] DEBUG[1540] channel.c: Planning to masquerade channel SIP/1000-00000014 into the structure of SIP/1001-00000016 [Jun 23 16:57:37] DEBUG[1540] channel.c: Done planning to masquerade channel SIP/1000-00000014 into the structure of SIP/1001-00000016 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Strict routing enforced for session 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.110:5060: NOTIFY sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK0cfcec07;rport Max-Forwards: 70 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Contact: Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 105 NOTIFY User-Agent: FPBX-2.9.0(1.8.4.2) Event: refer;id=5624 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 16 SIP/2.0 200 OK --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Sending reinvite on SIP '1c445c5017451751' - It's audio soon redirected to IP 192.168.169.60:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Strict routing enforced for session 1c445c5017451751 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Audio is at 5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Initializing already initialized SIP dialog 1c445c5017451751 (presumably reinvite) [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.110:5060: INVITE sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK4b77f53a;rport Max-Forwards: 70 From: ;tag=as3d35f19b To: "Cedric Autier" ;tag=1b6ace01bc Contact: Call-ID: 1c445c5017451751 CSeq: 103 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 240 v=0 o=root 1538385577 1538385579 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 10672 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: SIP Fixup: New owner for dialogue 1c445c5017451751: SIP/1000-00000014 (Old parent: SIP/1000-00000014) [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Hangup call SIP/1000-00000014, SIP callid 1c445c5017451751 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: update_call_counter(1001) - decrement call limit counter on hangup [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Updating call counter for incoming call [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Call from peer '1001' removed from call limit 2147483647 [Jun 23 16:57:37] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x96c7c20' [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Scheduling destruction of SIP dialog '1c445c5017451751' in 6400 ms (Method: ACK) [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Session timer stopped: -1 - 1c445c5017451751 [Jun 23 16:57:37] DEBUG[1540] channel.c: Putting channel SIP/1000-00000014 in alaw/alaw formats [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: SIP Fixup: New owner for dialogue 2c063ea24067f43a: SIP/1000-00000014 (Old parent: SIP/1001-00000016) [Jun 23 16:57:37] DEBUG[1540] channel.c: Released clone lock on 'SIP/1001-00000016' [Jun 23 16:57:37] DEBUG[1540] channel.c: Done Masquerading SIP/1000-00000014 (6) [Jun 23 16:57:37] DEBUG[1540] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jun 23 16:57:37] DEBUG[1540] res_rtp_asterisk.c: Changing ssrc from 1606514386 to 812976714 due to a source change [Jun 23 16:57:37] DEBUG[1540] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Attended Channel: SIP/1001-00000015 Uniqueid: 1308841014.21 SIP-Callid: 761431ef61781390634985167bd5a036@192.168.169.60:5060 TargetChannel: SIP/1001-00000016 TargetUniqueid: 1308841050.22 [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/1000-00000014 UniqueID: 1308841014.20 [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Masquerade Privilege: call,all Clone: SIP/1000-00000014 CloneState: Up Original: SIP/1001-00000016 OriginalState: Up [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 8 (On Hold) [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/1000-00000014 Newname: SIP/1000-00000014 Uniqueid: 1308841014.20 [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1001' state '8' [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/1001-00000016 Newname: SIP/1000-00000014 Uniqueid: 1308841050.22 [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Rename Privilege: call,all Channel: SIP/1000-00000014 Newname: SIP/1001-00000016 Uniqueid: 1308841014.20 [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/1000-00000014 CallerIDNum: 1000 CallerIDName: BME Uniqueid: 1308841050.22 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: MusicOnHold Privilege: call,all State: Stop Channel: SIP/1000-00000014 UniqueID: 1308841050.22 [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[2041] rtp_engine.c: Oooh, something is weird, backing out [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Sending reinvite on SIP '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' - It's audio soon redirected to IP 192.168.169.60:5060 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Audio is at 5060 [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Initializing already initialized SIP dialog 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 (presumably reinvite) [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK75752858;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 105 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 236 v=0 o=root 72735859 72735862 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 15714 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:37] VERBOSE[2041] rtp_engine.c: [Jun 23 16:57:37] -- Remotely bridging SIP/1000-00000014 and SIP/1002-00000017 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Sending reinvite on SIP '2c063ea24067f43a' - It's audio soon redirected to IP 192.168.169.100:8000 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Strict routing enforced for session 2c063ea24067f43a [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our native-bridge filtered capablity: 0x8 (alaw) [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Audio is at 5060 [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Initializing already initialized SIP dialog 2c063ea24067f43a (presumably reinvite) [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.102:5060: INVITE sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7b03c887;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 104 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 239 v=0 o=root 191191818 191191821 IN IP4 192.168.169.100 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.100 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Deferring reinvite on SIP '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' - It's audio will be redirected to IP 192.168.169.102:8000 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: REFER [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our native-bridge filtered capablity: 0x8 (alaw) [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Audio is at 5060 [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Initializing already initialized SIP dialog 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 (presumably reinvite) [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK64f08575;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 106 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "BME" Content-Type: application/sdp Content-Length: 237 v=0 o=root 72735859 72735863 IN IP4 192.168.169.102 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Strict routing enforced for session 2c063ea24067f43a [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: ** Our native-bridge filtered capablity: 0x8 (alaw) [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Audio is at 5060 [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Adding codec 0x8 (alaw) to SDP [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Adding non-codec 0x1 (telephone-event) to SDP [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: -- Done with adding codecs to SDP [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Initializing already initialized SIP dialog 2c063ea24067f43a (presumably reinvite) [Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.102:5060: INVITE sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK3f57ba26;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 105 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "TCE" Content-Type: application/sdp Content-Length: 239 v=0 o=root 191191818 191191822 IN IP4 192.168.169.100 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.100 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2c063ea24067f43a' Method: ACK [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: REFER [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] DEBUG[2037] rtp_engine.c: Oooh, something is weird, backing out [Jun 23 16:57:37] WARNING[2037] rtp_engine.c: Channel 'SIP/1001-00000016' failed to break RTP bridge [Jun 23 16:57:37] DEBUG[2037] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/1001-00000016, c1=SIP/1001-00000015, flags: Yes,Yes,No,No [Jun 23 16:57:37] DEBUG[2037] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/1001-00000016 Channel2: SIP/1001-00000015 Uniqueid1: 1308841014.20 Uniqueid2: 1308841014.21 CallerID1: 1001 CallerID2: 1001 [Jun 23 16:57:37] DEBUG[2037] channel.c: Bridge stops bridging channels SIP/1001-00000016 and SIP/1001-00000015 [Jun 23 16:57:37] DEBUG[2037] pbx.c: Launching 'Macro' [Jun 23 16:57:37] VERBOSE[2037] pbx.c: [Jun 23 16:57:37] -- Executing [h@macro-dial-one:1] Macro("SIP/1001-00000016", "hangupcall,") in new stack [Jun 23 16:57:37] DEBUG[2037] pbx.c: Expression result is '1' [Jun 23 16:57:37] DEBUG[2037] pbx.c: Launching 'GotoIf' [Jun 23 16:57:37] VERBOSE[2037] pbx.c: [Jun 23 16:57:37] -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1001-00000016", "1?theend") in new stack [Jun 23 16:57:37] VERBOSE[2037] pbx.c: [Jun 23 16:57:37] -- Goto (macro-hangupcall,s,3) [Jun 23 16:57:37] DEBUG[2037] app_macro.c: Executed application: GotoIf [Jun 23 16:57:37] DEBUG[2037] pbx.c: Launching 'Hangup' [Jun 23 16:57:37] VERBOSE[2037] pbx.c: [Jun 23 16:57:37] -- Executing [s@macro-hangupcall:3] Hangup("SIP/1001-00000016", "") in new stack [Jun 23 16:57:37] DEBUG[2037] app_macro.c: Spawn extension (macro-hangupcall,s,3) exited non-zero on 'SIP/1001-00000016' in macro 'hangupcall' [Jun 23 16:57:37] VERBOSE[2037] app_macro.c: [Jun 23 16:57:37] == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/1001-00000016' in macro 'hangupcall' [Jun 23 16:57:37] DEBUG[2037] cdr_mysql.c: Inserting a CDR record. [Jun 23 16:57:37] DEBUG[2037] cdr_mysql.c: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid,userfield) VALUES ('2011-06-23 16:56:54','\"BME\" <1000>','1000','1001','from-internal','SIP/1000-00000014','SIP/1001-00000015','Dial','SIP/1001,\"\",I','43','41','ANSWERED','3','1308841014.20','1000') [Jun 23 16:57:37] DEBUG[2041] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [Jun 23 16:57:37] DEBUG[2041] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [Jun 23 16:57:37] DEBUG[2041] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xb6ed2020' [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK3f57ba26;rport=5060;received=192.168.169.60 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Call-ID: 2c063ea24067f43a CSeq: 105 INVITE Retry-After: 3 Server: Aastra 6739i/3.2.2.41 Content-Length: 0 <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (9 headers 0 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Acked pending invite 105 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '2c063ea24067f43a' of Request 105: Match Found [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] -- Got SIP response 500 "Internal Server Error" back from 192.168.169.102:5060 [Jun 23 16:57:37] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x96d19a8' [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Strict routing enforced for session 2c063ea24067f43a [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Transmitting (NAT) to 192.168.169.102:5060: ACK sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK3f57ba26;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 105 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2c063ea24067f43a [Jun 23 16:57:37] DEBUG[2041] rtp_engine.c: Got a FRAME_CONTROL (8) frame on channel SIP/1000-00000014 [Jun 23 16:57:37] DEBUG[2041] channel.c: Returning from native bridge, channels: SIP/1000-00000014, SIP/1002-00000017 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] DEBUG[2041] pbx.c: Launching 'Macro' [Jun 23 16:57:37] VERBOSE[2041] pbx.c: [Jun 23 16:57:37] -- Executing [h@macro-dial-one:1] Macro("SIP/1000-00000014", "hangupcall,") in new stack [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/1000-00000014 Channel2: SIP/1002-00000017 Uniqueid1: 1308841050.22 Uniqueid2: 1308841050.23 CallerID1: 1000 CallerID2: 1002 [Jun 23 16:57:37] DEBUG[2041] pbx.c: Expression result is '1' [Jun 23 16:57:37] DEBUG[2041] pbx.c: Launching 'GotoIf' [Jun 23 16:57:37] VERBOSE[2041] pbx.c: [Jun 23 16:57:37] -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1000-00000014", "1?theend") in new stack [Jun 23 16:57:37] VERBOSE[2041] pbx.c: [Jun 23 16:57:37] -- Goto (macro-hangupcall,s,3) [Jun 23 16:57:37] DEBUG[2041] app_macro.c: Executed application: GotoIf [Jun 23 16:57:37] DEBUG[2041] pbx.c: Launching 'Hangup' [Jun 23 16:57:37] VERBOSE[2041] pbx.c: [Jun 23 16:57:37] -- Executing [s@macro-hangupcall:3] Hangup("SIP/1000-00000014", "") in new stack [Jun 23 16:57:37] DEBUG[2041] app_macro.c: Spawn extension (macro-hangupcall,s,3) exited non-zero on 'SIP/1000-00000014' in macro 'hangupcall' [Jun 23 16:57:37] VERBOSE[2041] app_macro.c: [Jun 23 16:57:37] == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/1000-00000014' in macro 'hangupcall' [Jun 23 16:57:37] DEBUG[2041] features.c: Spawn extension (macro-dial-one,h,1) exited non-zero on 'SIP/1000-00000014' [Jun 23 16:57:37] VERBOSE[2041] features.c: [Jun 23 16:57:37] == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/1000-00000014' [Jun 23 16:57:37] DEBUG[2041] cdr_mysql.c: Inserting a CDR record. [Jun 23 16:57:37] DEBUG[2041] cdr_mysql.c: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid,userfield) VALUES ('2011-06-23 16:57:30','\"Cedric Autier\" <1001>','1001','1002','from-internal','SIP/1001-00000016','SIP/1002-00000017','Dial','SIP/1002,\"\",I','7','4','ANSWERED','3','1308841050.22','1001') [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7b03c887;rport=5060;received=192.168.169.60 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Call-ID: 2c063ea24067f43a CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "BME" ;+sip.instance="" Server: Aastra 6739i/3.2.2.41 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 4 IN IP4 192.168.169.102 s=SIP Call c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.102 a=sendrecv <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (14 headers 13 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '2c063ea24067f43a' of Request 104: Match Found [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Got response on call that is already terminated: 2c063ea24067f43a (ignoring) [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK0cfcec07;rport=5060;received=192.168.169.60 From: "BME" ;tag=as343b04db To: ;tag=2053185131 Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 105 NOTIFY Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 6731i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (11 headers 0 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '761431ef61781390634985167bd5a036@192.168.169.60:5060' of Request 105: Match Found [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] SIP Response message for INCOMING dialog NOTIFY arrived [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Got 200 OK on NOTIFY for transfer [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.110:5060 ---> BYE sip:1000@192.168.169.60:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK8414cd2cdac18f9b5 Max-Forwards: 70 From: ;tag=2053185131 To: "BME" ;tag=as343b04db Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 5625 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Supported: path, timer User-Agent: Aastra 6731i/3.2.1.43 Content-Length: 0 <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (12 headers 0 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Initializing initreq for method BYE - callid 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Sending to 192.168.169.110:5060 (NAT) [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Setting SIP_ALREADYGONE on dialog 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:37] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4b9b4f0' [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Session timer stopped: -1 - 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Scheduling destruction of SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' in 6400 ms (Method: BYE) [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Received bye, issuing owner hangup [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- Transmitting (NAT) to 192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.110;branch=z9hG4bK8414cd2cdac18f9b5;received=192.168.169.110;rport=5060 From: ;tag=2053185131 To: "BME" ;tag=as343b04db Call-ID: 761431ef61781390634985167bd5a036@192.168.169.60:5060 CSeq: 5625 BYE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK4b77f53a;rport=5060;received=192.168.169.60 From: ;tag=as3d35f19b To: "Cedric Autier" ;tag=1b6ace01bc Call-ID: 1c445c5017451751 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Cedric Autier" ;+sip.instance="" Server: Aastra 6731i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 3 IN IP4 192.168.169.110 s=SIP Call c=IN IP4 192.168.169.110 t=0 0 m=audio 8002 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8003 IN IP4 192.168.169.110 a=sendrecv <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (14 headers 13 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Acked pending invite 103 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '1c445c5017451751' of Request 103: Match Found [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing session-level SDP o=MxSIP 0 3 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.110... OK. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Found RTP audio format 8 [Jun 23 16:57:37] DEBUG[1540] rtp_engine.c: Setting payload 8 based on m type on 0xb5decfb4 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Found RTP audio format 101 [Jun 23 16:57:37] DEBUG[1540] rtp_engine.c: Setting payload 101 based on m type on 0xb5decfb4 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Found audio description format PCMA for ID 8 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Found audio description format telephone-event for ID 101 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=rtcp:8003 IN IP4 192.168.169.110... UNSUPPORTED. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 23 16:57:37] DEBUG[1540] rtp_engine.c: Incorporating payload 8 on 0xb5decfb4 [Jun 23 16:57:37] DEBUG[1540] rtp_engine.c: Incorporating payload 101 on 0xb5decfb4 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 23 16:57:37] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x96c7c20' [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Peer audio RTP is at port 192.168.169.110:8002 [Jun 23 16:57:37] DEBUG[1540] rtp_engine.c: Copying payload 8 from 0xb5decfb4 to 0x96c7dcc [Jun 23 16:57:37] DEBUG[1540] rtp_engine.c: Copying payload 101 from 0xb5decfb4 to 0x96c7dcc [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Updating call counter for incoming call [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: build_route: Retaining previous route: [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Strict routing enforced for session 1c445c5017451751 [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 8 (On Hold) [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1001' state '8' [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Transmitting (NAT) to 192.168.169.110:5060: ACK sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK490f541f;rport Max-Forwards: 70 From: ;tag=as3d35f19b To: "Cedric Autier" ;tag=1b6ace01bc Contact: Call-ID: 1c445c5017451751 CSeq: 103 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Strict routing enforced for session 1c445c5017451751 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.110:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.110:5060: BYE sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK6542cdb4;rport Max-Forwards: 70 From: ;tag=as3d35f19b To: "Cedric Autier" ;tag=1b6ace01bc Call-ID: 1c445c5017451751 CSeq: 104 BYE User-Agent: FPBX-2.9.0(1.8.4.2) Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="192.168.169.60:5060", nonce="", response="4601c522149feb15941be2c7cb2fa016" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '8' (On Hold) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'BYE sip:100' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Scheduling destruction of SIP dialog '1c445c5017451751' in 6400 ms (Method: ACK) [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Retransmitting #1 (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK75752858;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 105 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 236 v=0 o=root 72735859 72735862 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 15714 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Retransmitting #1 (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK64f08575;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 106 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "BME" Content-Type: application/sdp Content-Length: 237 v=0 o=root 72735859 72735863 IN IP4 192.168.169.102 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK6542cdb4;rport=5060;received=192.168.169.60 From: ;tag=as3d35f19b To: "Cedric Autier" ;tag=1b6ace01bc Call-ID: 1c445c5017451751 CSeq: 104 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 6731i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (11 headers 0 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '1c445c5017451751' of Request 104: Match Found [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Destroying SIP dialog 1c445c5017451751 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Really destroying SIP dialog '1c445c5017451751' Method: ACK [Jun 23 16:57:37] DEBUG[1540] rtp_engine.c: Destroyed RTP instance '0x96c7c20' [Jun 23 16:57:37] DEBUG[2037] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid,userfield) VALUES ('"BME" <1000>','1000','1001','from-internal','SIP/1000-00000014','SIP/1001-00000015','Dial','SIP/1001,"",I','2011-06-23 16:56:54','2011-06-23 16:56:56','2011-06-23 16:57:37','43','41','ANSWERED','DOCUMENTATION','1308841014.20','1000') [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK64f08575;rport=5060;received=192.168.169.60 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 106 INVITE Retry-After: 5 Server: Aastra 57i/3.2.1.43 Content-Length: 0 <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (9 headers 0 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Acked pending invite 106 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' of Request 106: Match Found [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] -- Got SIP response 500 "Internal Server Error" back from 192.168.169.100:5060 [Jun 23 16:57:37] DEBUG[1540] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6ed2020' [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Transmitting (NAT) to 192.168.169.100:5060: ACK sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK64f08575;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 106 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Setting SIP_ALREADYGONE on dialog 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK64f08575;rport=5060;received=192.168.169.60 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 106 INVITE Retry-After: 5 Server: Aastra 57i/3.2.1.43 Content-Length: 0 <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (9 headers 0 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' of Request 106: Match Not Found [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Transmitting (NAT) to 192.168.169.100:5060: ACK sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK64f08575;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 106 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Retransmitting #2 (NAT) to 192.168.169.100:5060: INVITE sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK75752858;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 105 INVITE User-Agent: FPBX-2.9.0(1.8.4.2) Require: timer Session-Expires: 900;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) P-Asserted-Identity: "Cedric Autier" Content-Type: application/sdp Content-Length: 236 v=0 o=root 72735859 72735862 IN IP4 192.168.169.60 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.169.60 t=0 0 m=audio 15714 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] DEBUG[2041] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid,userfield) VALUES ('"Cedric Autier" <1001>','1001','1002','from-internal','SIP/1001-00000016','SIP/1002-00000017','Dial','SIP/1002,"",I','2011-06-23 16:57:30','2011-06-23 16:57:33','2011-06-23 16:57:37','7','4','ANSWERED','DOCUMENTATION','1308841050.22','1001') [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK75752858;rport=5060;received=192.168.169.60 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "TCE" ;+sip.instance="" Server: Aastra 57i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 4 IN IP4 192.168.169.100 s=SIP Call c=IN IP4 192.168.169.100 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.100 a=sendrecv <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (14 headers 13 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' of Request 105: Match Found [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Got response on call that is already terminated: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 (ignoring) [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] DEBUG[2037] channel.c: Hanging up channel 'SIP/1001-00000015' [Jun 23 16:57:37] DEBUG[2037] chan_sip.c: update_call_counter(1001) - decrement call limit counter on hangup [Jun 23 16:57:37] DEBUG[2037] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 1 (Not in use) [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1001' state '1' [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:37] DEBUG[1533] app_queue.c: Extension '1001@ext-local' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1533] chan_sip.c: Strict routing enforced for session 70a40468b789eb3f [Jun 23 16:57:37] VERBOSE[1533] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[1533] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 1001 Context: ext-local Hint: SIP/1001&Custom:DND1001 Status: 0 [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:37] VERBOSE[1533] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.102:5060: NOTIFY sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK07d9feb3;rport Max-Forwards: 70 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Contact: Call-ID: 70a40468b789eb3f CSeq: 123 NOTIFY User-Agent: FPBX-2.9.0(1.8.4.2) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 210

terminated

--- [Jun 23 16:57:37] DEBUG[1533] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:37] VERBOSE[1533] chan_sip.c: [Jun 23 16:57:37] == Extension Changed 1001[ext-local] new state Idle for Notify User 1000 [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[2037] chan_sip.c: Call to peer '1001' removed from call limit 2147483647 [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 1 (Not in use) [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1001' state '1' [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/1001-00000015 Uniqueid: 1308841014.21 CallerIDNum: 1001 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 1 (Not in use) [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1001' state '1' [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/1001-00000016 UniqueID: 1308841014.20 DialStatus: ANSWER [Jun 23 16:57:37] DEBUG[2037] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jun 23 16:57:37] DEBUG[2037] app_macro.c: Spawn extension (macro-dial-one,s,42) exited non-zero on 'SIP/1001-00000016' in macro 'dial-one' [Jun 23 16:57:37] VERBOSE[2037] app_macro.c: [Jun 23 16:57:37] == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'SIP/1001-00000016' in macro 'dial-one' [Jun 23 16:57:37] DEBUG[2037] app_macro.c: Spawn extension (macro-exten-vm,s,7) exited non-zero on 'SIP/1001-00000016' in macro 'exten-vm' [Jun 23 16:57:37] VERBOSE[2037] app_macro.c: [Jun 23 16:57:37] == Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/1001-00000016' in macro 'exten-vm' [Jun 23 16:57:37] DEBUG[2037] pbx.c: Spawn extension (from-internal,1001,2) exited non-zero on 'SIP/1001-00000016' [Jun 23 16:57:37] VERBOSE[2037] pbx.c: [Jun 23 16:57:37] == Spawn extension (from-internal, 1001, 2) exited non-zero on 'SIP/1001-00000016' [Jun 23 16:57:37] DEBUG[2037] channel.c: Soft-Hanging up channel 'SIP/1001-00000016' [Jun 23 16:57:37] DEBUG[2037] channel.c: Hanging up zombie 'SIP/1001-00000016' [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/1001-00000016 Uniqueid: 1308841014.20 CallerIDNum: 1001 CallerIDName: Cedric Autier Cause: 16 Cause-txt: Normal Clearing [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1001 [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1001 - state 1 (Not in use) [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1001' state '1' [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1001' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK07d9feb3;rport=5060;received=192.168.169.60 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Call-ID: 70a40468b789eb3f CSeq: 123 NOTIFY Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 6739i/3.2.2.41 Supported: path Content-Length: 0 <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (11 headers 0 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Acked pending invite 123 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '70a40468b789eb3f' of Request 123: Match Found [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] SIP Response message for INCOMING dialog NOTIFY arrived [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:37] DEBUG[2041] channel.c: Hanging up channel 'SIP/1002-00000017' [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Hangup call SIP/1002-00000017, SIP callid 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: update_call_counter(1002) - decrement call limit counter on hangup [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Updating call counter for outgoing call [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Call to peer '1002' removed from call limit 2147483647 [Jun 23 16:57:37] DEBUG[2041] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6ed2020' [Jun 23 16:57:37] DEBUG[2041] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jun 23 16:57:37] DEBUG[2041] app_macro.c: Spawn extension (macro-dial-one,s,42) exited non-zero on 'SIP/1000-00000014' in macro 'dial-one' [Jun 23 16:57:37] VERBOSE[2041] app_macro.c: [Jun 23 16:57:37] == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'SIP/1000-00000014' in macro 'dial-one' [Jun 23 16:57:37] DEBUG[2041] app_macro.c: Spawn extension (macro-exten-vm,s,7) exited non-zero on 'SIP/1000-00000014' in macro 'exten-vm' [Jun 23 16:57:37] VERBOSE[2041] app_macro.c: [Jun 23 16:57:37] == Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/1000-00000014' in macro 'exten-vm' [Jun 23 16:57:37] DEBUG[2041] pbx.c: Spawn extension (from-internal,1002,2) exited non-zero on 'SIP/1000-00000014' [Jun 23 16:57:37] VERBOSE[2041] pbx.c: [Jun 23 16:57:37] == Spawn extension (from-internal, 1002, 2) exited non-zero on 'SIP/1000-00000014' [Jun 23 16:57:37] DEBUG[2041] channel.c: Soft-Hanging up channel 'SIP/1000-00000014' [Jun 23 16:57:37] DEBUG[2041] pbx.c: Launching 'Hangup' [Jun 23 16:57:37] VERBOSE[2041] pbx.c: [Jun 23 16:57:37] -- Executing [h@from-internal:1] Hangup("SIP/1000-00000014", "") in new stack [Jun 23 16:57:37] DEBUG[2041] pbx.c: Spawn extension (from-internal,h,1) exited non-zero on 'SIP/1000-00000014' [Jun 23 16:57:37] VERBOSE[2041] pbx.c: [Jun 23 16:57:37] == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1000-00000014' [Jun 23 16:57:37] DEBUG[2041] channel.c: Hanging up channel 'SIP/1000-00000014' [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Hangup call SIP/1000-00000014, SIP callid 2c063ea24067f43a [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: update_call_counter(1000) - decrement call limit counter on hangup [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Updating call counter for incoming call [Jun 23 16:57:37] DEBUG[2041] chan_sip.c: Call from peer '1000' removed from call limit 2147483647 [Jun 23 16:57:37] DEBUG[2041] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x96d19a8' [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1002 [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1002 - state 1 (Not in use) [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1002' state '1' [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1002 [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1002 - state 1 (Not in use) [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1002' state '1' [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1000 [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1000 - state 1 (Not in use) [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1000' state '1' [Jun 23 16:57:37] DEBUG[1532] chan_sip.c: Checking device state for peer 1000 [Jun 23 16:57:37] DEBUG[1532] devicestate.c: Changing state for SIP/1000 - state 1 (Not in use) [Jun 23 16:57:37] DEBUG[1532] devicestate.c: device 'SIP/1000' state '1' [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/1002-00000017 Uniqueid: 1308841050.23 CallerIDNum: 1002 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/1000-00000014 UniqueID: 1308841050.22 DialStatus: ANSWER [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/1000-00000014 Uniqueid: 1308841050.22 CallerIDNum: 1000 CallerIDName: BME Cause: 16 Cause-txt: Normal Clearing [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1002' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1002' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1000' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1567] app_queue.c: Device 'SIP/1000' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:37] DEBUG[1533] app_queue.c: Extension '1002@ext-local' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1002 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1002' in family 'CustomDevstate' [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1000 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1000' in family 'CustomDevstate' [Jun 23 16:57:37] DEBUG[1533] app_queue.c: Extension '1000@ext-local' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 23 16:57:37] DEBUG[1533] devicestate.c: Checking if I can find provider for "Custom" - number: DND1000 [Jun 23 16:57:37] DEBUG[1533] db.c: Unable to find key 'DND1000' in family 'CustomDevstate' [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 1002 Context: ext-local Hint: SIP/1002&Custom:DND1002 Status: 0 [Jun 23 16:57:37] DEBUG[1633] manager.c: Examining event: Event: ExtensionStatus Privilege: call,all Exten: 1000 Context: ext-local Hint: SIP/1000&Custom:DND1000 Status: 0 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7b03c887;rport=5060;received=192.168.169.60 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Call-ID: 2c063ea24067f43a CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "BME" ;+sip.instance="" Server: Aastra 6739i/3.2.2.41 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 4 IN IP4 192.168.169.102 s=SIP Call c=IN IP4 192.168.169.102 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.102 a=sendrecv <-------------> [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] --- (14 headers 13 lines) --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Stopping retransmission on '2c063ea24067f43a' of Request 104: Match Not Found [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Strict routing enforced for session 2c063ea24067f43a [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: Parsing for address/port to send to [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37] Transmitting (NAT) to 192.168.169.102:5060: ACK sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7c700f2f;rport Max-Forwards: 70 From: ;tag=as22755b3e To: "BME" ;tag=a2e37c0386 Contact: Call-ID: 2c063ea24067f43a CSeq: 104 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:38] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:38] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK75752858;rport=5060;received=192.168.169.60 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "TCE" ;+sip.instance="" Server: Aastra 57i/3.2.1.43 Session-Expires: 900;refresher=uas Supported: path, replaces, timer Content-Type: application/sdp Content-Length: 275 v=0 o=MxSIP 0 4 IN IP4 192.168.169.100 s=SIP Call c=IN IP4 192.168.169.100 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=rtcp:8001 IN IP4 192.168.169.100 a=sendrecv <-------------> [Jun 23 16:57:38] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:38] --- (14 headers 13 lines) --- [Jun 23 16:57:38] DEBUG[1540] chan_sip.c: Stopping retransmission on '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' of Request 105: Match Not Found [Jun 23 16:57:38] DEBUG[1540] chan_sip.c: Strict routing enforced for session 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 [Jun 23 16:57:38] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:38] set_destination: Parsing for address/port to send to [Jun 23 16:57:38] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:38] set_destination: set destination to 192.168.169.100:5060 [Jun 23 16:57:38] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:38] Transmitting (NAT) to 192.168.169.100:5060: ACK sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK3d9d8993;rport Max-Forwards: 70 From: "Cedric Autier" ;tag=as709529d2 To: ;tag=4086919603 Contact: Call-ID: 628b80782bab27c7114ba92c403bb627@192.168.169.60:5060 CSeq: 105 ACK User-Agent: FPBX-2.9.0(1.8.4.2) Content-Length: 0 --- [Jun 23 16:57:38] DEBUG[1540] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:38] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:39] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:40] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:41] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:42] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:43] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:43] DEBUG[1540] chan_sip.c: Auto destroying SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' [Jun 23 16:57:43] DEBUG[1540] chan_sip.c: Destroying SIP dialog 761431ef61781390634985167bd5a036@192.168.169.60:5060 [Jun 23 16:57:43] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:43] Really destroying SIP dialog '761431ef61781390634985167bd5a036@192.168.169.60:5060' Method: BYE [Jun 23 16:57:43] DEBUG[1540] rtp_engine.c: Destroyed RTP instance '0xb4b9b4f0' [Jun 23 16:57:43] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:44] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:45] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:46] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:47] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:48] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:49] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:50] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:51] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:52] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 056bec0f56a37c41580445e220ff425f@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 16:57:53] DEBUG[1540] acl.c: For destination '192.168.169.110', our source address is '192.168.169.60'. [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 1a3ed9ee31c6770106fee3ee717581e1@192.168.169.60:5060 [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] Reliably Transmitting (NAT) to 192.168.169.110:5060: OPTIONS sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK09d57a0d;rport Max-Forwards: 70 From: "Unknown" ;tag=as71840f34 To: Contact: Call-ID: 1a3ed9ee31c6770106fee3ee717581e1@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:57:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK09d57a0d;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as71840f34 To: ;tag=1395383796 Call-ID: 1a3ed9ee31c6770106fee3ee717581e1@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 6731i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] --- (10 headers 0 lines) --- [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Stopping retransmission on '1a3ed9ee31c6770106fee3ee717581e1@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Destroying SIP dialog 1a3ed9ee31c6770106fee3ee717581e1@192.168.169.60:5060 [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] Really destroying SIP dialog '1a3ed9ee31c6770106fee3ee717581e1@192.168.169.60:5060' Method: OPTIONS [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 0182d59c2d01cbf34bba352236fcc5ae@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 16:57:53] DEBUG[1540] acl.c: For destination '192.168.169.102', our source address is '192.168.169.60'. [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 7080a101717efc8d6e52fe4a4083bc5a@192.168.169.60:5060 [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] Reliably Transmitting (NAT) to 192.168.169.102:5060: OPTIONS sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK51bf0564;rport Max-Forwards: 70 From: "Unknown" ;tag=as1ded0a3d To: Contact: Call-ID: 7080a101717efc8d6e52fe4a4083bc5a@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:57:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK51bf0564;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as1ded0a3d To: ;tag=3843227056 Call-ID: 7080a101717efc8d6e52fe4a4083bc5a@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 6739i/3.2.2.41 Supported: path Content-Length: 0 <-------------> [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] --- (10 headers 0 lines) --- [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Stopping retransmission on '7080a101717efc8d6e52fe4a4083bc5a@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Destroying SIP dialog 7080a101717efc8d6e52fe4a4083bc5a@192.168.169.60:5060 [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] Really destroying SIP dialog '7080a101717efc8d6e52fe4a4083bc5a@192.168.169.60:5060' Method: OPTIONS [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 5b13d73c27adc00706338908402f8f35@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 16:57:53] DEBUG[1540] acl.c: For destination '192.168.169.100', our source address is '192.168.169.60'. [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 1f8800f27b2b06e80f026acd06ab2dc9@192.168.169.60:5060 [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] Reliably Transmitting (NAT) to 192.168.169.100:5060: OPTIONS sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK1e52b005;rport Max-Forwards: 70 From: "Unknown" ;tag=as5cdfe0a5 To: Contact: Call-ID: 1f8800f27b2b06e80f026acd06ab2dc9@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:57:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK1e52b005;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as5cdfe0a5 To: ;tag=2465131259 Call-ID: 1f8800f27b2b06e80f026acd06ab2dc9@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] --- (10 headers 0 lines) --- [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Stopping retransmission on '1f8800f27b2b06e80f026acd06ab2dc9@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Destroying SIP dialog 1f8800f27b2b06e80f026acd06ab2dc9@192.168.169.60:5060 [Jun 23 16:57:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:53] Really destroying SIP dialog '1f8800f27b2b06e80f026acd06ab2dc9@192.168.169.60:5060' Method: OPTIONS [Jun 23 16:57:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:54] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:55] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:57] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:58] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:57:59] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:00] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:01] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:02] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:03] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:04] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:05] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:06] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:07] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:08] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:09] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:10] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:11] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:12] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] <--- SIP read from UDP:192.168.169.102:5060 ---> SUBSCRIBE sip:1001@192.168.169.60:5060 SIP/2.0 Accept: application/dialog-info+xml Via: SIP/2.0/UDP 192.168.169.102;branch=z9hG4bK47a103b7fc3d5c6e5 Max-Forwards: 70 From: "BME" ;tag=ef7b15a827 To: ;tag=as1d31e096 Call-ID: 70a40468b789eb3f CSeq: 30673 SUBSCRIBE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="1000",realm="asterisk",nonce="651d0f8e",uri="sip:1001@192.168.169.60:5060",response="851aaf3521cf1f2aaa0761468bd12678",algorithm=MD5 Contact: "BME" ;+sip.instance="" Event: dialog Expires: 900 Supported: path User-Agent: Aastra 6739i/3.2.2.41 Content-Length: 0 <-------------> [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] --- (17 headers 0 lines) --- [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Got a re-subscribe on existing subscription 70a40468b789eb3f [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] Found peer '1000' for '1000' from 192.168.169.102:5060 [Jun 23 16:58:13] NOTICE[1540] chan_sip.c: Correct auth, but based on stale nonce received from '"BME" ;tag=ef7b15a827' [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] <--- Transmitting (NAT) to 192.168.169.102:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.169.102;branch=z9hG4bK47a103b7fc3d5c6e5;received=192.168.169.102;rport=5060 From: "BME" ;tag=ef7b15a827 To: ;tag=as1d31e096 Call-ID: 70a40468b789eb3f CSeq: 30673 SUBSCRIBE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="366d7a87", stale=true Content-Length: 0 <------------> [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] Scheduling destruction of SIP dialog '70a40468b789eb3f' in 6400 ms (Method: SUBSCRIBE) [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] <--- SIP read from UDP:192.168.169.102:5060 ---> SUBSCRIBE sip:1001@192.168.169.60:5060 SIP/2.0 Accept: application/dialog-info+xml Via: SIP/2.0/UDP 192.168.169.102;branch=z9hG4bK498134286c68cf774 Max-Forwards: 70 From: "BME" ;tag=ef7b15a827 To: ;tag=as1d31e096 Call-ID: 70a40468b789eb3f CSeq: 30674 SUBSCRIBE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="1000",realm="asterisk",nonce="366d7a87",uri="sip:1001@192.168.169.60:5060",response="2cfda3149a3f69769f8f61270b7a91a0",algorithm=MD5 Contact: "BME" ;+sip.instance="" Event: dialog Expires: 900 Supported: path User-Agent: Aastra 6739i/3.2.2.41 Content-Length: 0 <-------------> [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] --- (17 headers 0 lines) --- [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Got a re-subscribe on existing subscription 70a40468b789eb3f [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] Found peer '1000' for '1000' from 192.168.169.102:5060 [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] Looking for 1001 in from-internal (domain 192.168.169.60:5060) [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Adding subscription for extension 1001 context from-internal for peer 1000 [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] Scheduling destruction of SIP dialog '70a40468b789eb3f' in 910000 ms (Method: SUBSCRIBE) [Jun 23 16:58:13] DEBUG[1540] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:58:13] DEBUG[1540] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] <--- Transmitting (NAT) to 192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.102;branch=z9hG4bK498134286c68cf774;received=192.168.169.102;rport=5060 From: "BME" ;tag=ef7b15a827 To: ;tag=as1d31e096 Call-ID: 70a40468b789eb3f CSeq: 30674 SUBSCRIBE Server: FPBX-2.9.0(1.8.4.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 900 Contact: ;expires=900 Content-Length: 0 <------------> [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Strict routing enforced for session 70a40468b789eb3f [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] set_destination: Parsing for address/port to send to [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] set_destination: set destination to 192.168.169.102:5060 [Jun 23 16:58:13] DEBUG[1540] devicestate.c: Checking if I can find provider for "Custom" - number: DND1001 [Jun 23 16:58:13] DEBUG[1540] db.c: Unable to find key 'DND1001' in family 'CustomDevstate' [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] Reliably Transmitting (NAT) to 192.168.169.102:5060: NOTIFY sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK2a119d3b;rport Max-Forwards: 70 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Contact: Call-ID: 70a40468b789eb3f CSeq: 124 NOTIFY User-Agent: FPBX-2.9.0(1.8.4.2) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 210

terminated

--- [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK2a119d3b;rport=5060;received=192.168.169.60 From: ;tag=as1d31e096 To: "BME" ;tag=ef7b15a827 Call-ID: 70a40468b789eb3f CSeq: 124 NOTIFY Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 6739i/3.2.2.41 Supported: path Content-Length: 0 <-------------> [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] --- (11 headers 0 lines) --- [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Acked pending invite 124 [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Stopping retransmission on '70a40468b789eb3f' of Request 124: Match Found [Jun 23 16:58:13] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:13] SIP Response message for INCOMING dialog NOTIFY arrived [Jun 23 16:58:13] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:14] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:15] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:16] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:17] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:18] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:19] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:20] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:21] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:22] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:23] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:24] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:25] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:26] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:28] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:29] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:31] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:32] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:34] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:35] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:36] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:38] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:39] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:40] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:41] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:42] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:43] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:44] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:45] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:46] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:47] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:48] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:49] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:50] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:51] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:52] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 753363f276b143433a28d27d0d560244@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 16:58:53] DEBUG[1540] acl.c: For destination '192.168.169.110', our source address is '192.168.169.60'. [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 1ee691f4504b0b515af2ded90dcee5a3@192.168.169.60:5060 [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] Reliably Transmitting (NAT) to 192.168.169.110:5060: OPTIONS sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK4390cbae;rport Max-Forwards: 70 From: "Unknown" ;tag=as25eca427 To: Contact: Call-ID: 1ee691f4504b0b515af2ded90dcee5a3@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:58:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK4390cbae;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as25eca427 To: ;tag=932524632 Call-ID: 1ee691f4504b0b515af2ded90dcee5a3@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 6731i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] --- (10 headers 0 lines) --- [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Stopping retransmission on '1ee691f4504b0b515af2ded90dcee5a3@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Destroying SIP dialog 1ee691f4504b0b515af2ded90dcee5a3@192.168.169.60:5060 [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] Really destroying SIP dialog '1ee691f4504b0b515af2ded90dcee5a3@192.168.169.60:5060' Method: OPTIONS [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 4af85ed7626948610676d6af1da87512@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 16:58:53] DEBUG[1540] acl.c: For destination '192.168.169.102', our source address is '192.168.169.60'. [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 23cd88625defe75329352f9f40a1dd46@192.168.169.60:5060 [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] Reliably Transmitting (NAT) to 192.168.169.102:5060: OPTIONS sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK3f92375a;rport Max-Forwards: 70 From: "Unknown" ;tag=as6df4e5c8 To: Contact: Call-ID: 23cd88625defe75329352f9f40a1dd46@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:58:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK3f92375a;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as6df4e5c8 To: ;tag=634822873 Call-ID: 23cd88625defe75329352f9f40a1dd46@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 6739i/3.2.2.41 Supported: path Content-Length: 0 <-------------> [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] --- (10 headers 0 lines) --- [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Stopping retransmission on '23cd88625defe75329352f9f40a1dd46@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Destroying SIP dialog 23cd88625defe75329352f9f40a1dd46@192.168.169.60:5060 [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] Really destroying SIP dialog '23cd88625defe75329352f9f40a1dd46@192.168.169.60:5060' Method: OPTIONS [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 053e16ed5dc44aa221bf4c4f5b6aab80@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 16:58:53] DEBUG[1540] acl.c: For destination '192.168.169.100', our source address is '192.168.169.60'. [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 5725b20b075ac855397c2e037d937d5c@192.168.169.60:5060 [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] Reliably Transmitting (NAT) to 192.168.169.100:5060: OPTIONS sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK54fe1ba3;rport Max-Forwards: 70 From: "Unknown" ;tag=as757e79a1 To: Contact: Call-ID: 5725b20b075ac855397c2e037d937d5c@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:58:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK54fe1ba3;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as757e79a1 To: ;tag=174049524 Call-ID: 5725b20b075ac855397c2e037d937d5c@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] --- (10 headers 0 lines) --- [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Stopping retransmission on '5725b20b075ac855397c2e037d937d5c@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Destroying SIP dialog 5725b20b075ac855397c2e037d937d5c@192.168.169.60:5060 [Jun 23 16:58:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:58:53] Really destroying SIP dialog '5725b20b075ac855397c2e037d937d5c@192.168.169.60:5060' Method: OPTIONS [Jun 23 16:58:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:54] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:55] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:57] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:58] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:58:59] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:00] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:01] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:02] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:03] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:04] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:05] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:06] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:07] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:08] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:09] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:10] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:11] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:12] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:13] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:14] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:15] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:16] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:17] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:18] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:19] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:20] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:21] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:22] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:23] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:24] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:25] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:26] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:28] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:29] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:31] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:32] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:34] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:35] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:36] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:38] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:39] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:40] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:41] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:42] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:43] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:44] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:45] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:46] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:47] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:48] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:49] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:50] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:51] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:52] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 0fd6c0fd7a23ef43051e1d4e218e6ccc@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 16:59:53] DEBUG[1540] acl.c: For destination '192.168.169.110', our source address is '192.168.169.60'. [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 245065764315c7663e714a2b0d1151bf@192.168.169.60:5060 [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] Reliably Transmitting (NAT) to 192.168.169.110:5060: OPTIONS sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK1e1ba5c6;rport Max-Forwards: 70 From: "Unknown" ;tag=as3e1a51a2 To: Contact: Call-ID: 245065764315c7663e714a2b0d1151bf@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:59:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK1e1ba5c6;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as3e1a51a2 To: ;tag=2640191552 Call-ID: 245065764315c7663e714a2b0d1151bf@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 6731i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] --- (10 headers 0 lines) --- [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Stopping retransmission on '245065764315c7663e714a2b0d1151bf@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Destroying SIP dialog 245065764315c7663e714a2b0d1151bf@192.168.169.60:5060 [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] Really destroying SIP dialog '245065764315c7663e714a2b0d1151bf@192.168.169.60:5060' Method: OPTIONS [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 2a596c9f4f9db55e437d7b6a60b818e8@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 16:59:53] DEBUG[1540] acl.c: For destination '192.168.169.102', our source address is '192.168.169.60'. [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 7c9fe0f240b714ad0a6046aa29cc15f2@192.168.169.60:5060 [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] Reliably Transmitting (NAT) to 192.168.169.102:5060: OPTIONS sip:1000@192.168.169.102:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7b0408e6;rport Max-Forwards: 70 From: "Unknown" ;tag=as0f270086 To: Contact: Call-ID: 7c9fe0f240b714ad0a6046aa29cc15f2@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:59:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.102:5060 [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] <--- SIP read from UDP:192.168.169.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7b0408e6;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as0f270086 To: ;tag=3007406226 Call-ID: 7c9fe0f240b714ad0a6046aa29cc15f2@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 6739i/3.2.2.41 Supported: path Content-Length: 0 <-------------> [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] --- (10 headers 0 lines) --- [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Stopping retransmission on '7c9fe0f240b714ad0a6046aa29cc15f2@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Destroying SIP dialog 7c9fe0f240b714ad0a6046aa29cc15f2@192.168.169.60:5060 [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] Really destroying SIP dialog '7c9fe0f240b714ad0a6046aa29cc15f2@192.168.169.60:5060' Method: OPTIONS [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 6b91ce2278f2ec651c0b0d76637264a1@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 16:59:53] DEBUG[1540] acl.c: For destination '192.168.169.100', our source address is '192.168.169.60'. [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 5e76278405a9854a2a06f2a87b1acec8@192.168.169.60:5060 [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] Reliably Transmitting (NAT) to 192.168.169.100:5060: OPTIONS sip:1002@192.168.169.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK44856327;rport Max-Forwards: 70 From: "Unknown" ;tag=as7ec9efe1 To: Contact: Call-ID: 5e76278405a9854a2a06f2a87b1acec8@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 14:59:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.100:5060 [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] <--- SIP read from UDP:192.168.169.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK44856327;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as7ec9efe1 To: ;tag=450169795 Call-ID: 5e76278405a9854a2a06f2a87b1acec8@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/3.2.1.43 Supported: path Content-Length: 0 <-------------> [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] --- (10 headers 0 lines) --- [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Stopping retransmission on '5e76278405a9854a2a06f2a87b1acec8@192.168.169.60:5060' of Request 102: Match Found [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Destroying SIP dialog 5e76278405a9854a2a06f2a87b1acec8@192.168.169.60:5060 [Jun 23 16:59:53] VERBOSE[1540] chan_sip.c: [Jun 23 16:59:53] Really destroying SIP dialog '5e76278405a9854a2a06f2a87b1acec8@192.168.169.60:5060' Method: OPTIONS [Jun 23 16:59:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:54] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:55] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:56] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:57] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:58] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 16:59:59] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:00] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:01] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:02] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:03] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:04] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:05] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:06] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:07] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:08] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:09] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:10] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:11] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:12] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:13] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:14] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:15] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:16] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:17] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:18] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:19] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:20] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:21] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:22] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:23] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:24] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:25] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:26] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:27] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:28] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:28] VERBOSE[1976] asterisk.c: [Jun 23 17:00:28] -- Remote UNIX connection disconnected [Jun 23 17:00:29] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:30] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:31] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:32] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:33] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:34] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:35] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:36] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:37] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:38] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:39] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:40] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:41] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:42] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:43] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:44] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:45] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:46] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:47] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:48] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:49] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:50] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:51] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:52] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:53] DEBUG[1540] chan_sip.c: Allocating new SIP dialog for 78a6490e40a7aabf401889262116d89d@127.0.1.1:0 - OPTIONS (No RTP) [Jun 23 17:00:53] DEBUG[1540] acl.c: For destination '192.168.169.110', our source address is '192.168.169.60'. [Jun 23 17:00:53] DEBUG[1540] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.169.60:5060 [Jun 23 17:00:53] DEBUG[1540] chan_sip.c: Initializing initreq for method OPTIONS - callid 1368480f616cb8f918925dba4acb1131@192.168.169.60:5060 [Jun 23 17:00:53] VERBOSE[1540] chan_sip.c: [Jun 23 17:00:53] Reliably Transmitting (NAT) to 192.168.169.110:5060: OPTIONS sip:1001@192.168.169.110:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK6116f767;rport Max-Forwards: 70 From: "Unknown" ;tag=as163feb60 To: Contact: Call-ID: 1368480f616cb8f918925dba4acb1131@192.168.169.60:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.4.2) Date: Thu, 23 Jun 2011 15:00:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 23 17:00:53] DEBUG[1540] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.169.110:5060 [Jun 23 17:00:53] DEBUG[1540] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '628b80782bab27c7114ba92c403bb627@192.168.169.60:5060' Method: INVITE [Jun 23 17:00:53] VERBOSE[1540] chan_sip.c: [Jun 23 17:00:53] <--- SIP read from UDP:192.168.169.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK6116f767;rport=5060;received=192.168.169.60 From: "Unknown" ;tag=as163feb60 To: ;tag=1050679096 Call-ID: 1368480f616cb8f918925dba4acb1131@192.168.169.60:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 6731i/3.2.1.43 Supported: path