[2015-07-17 15:21:47] DEBUG[1781] chan_sip.c: Allocating new SIP dialog for SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL - INVITE (No RTP) [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel" [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] sip/reqresp_parser.c: Found SIP option: -100rel- [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] sip/reqresp_parser.c: Matched SIP option: 100rel [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '109.235.35.252' into... [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: ...host '109.235.35.252' and port ''. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: Splitting 'pocos.nl:5060' into... [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: ...host 'pocos.nl' and port ''. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4d0bcd8138' [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Allocated port 18188 for RTP instance '0x7f4d0bcd8138' [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '109.235.32.49' into... [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: ...host '109.235.32.49' and port ''. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] rtp_engine.c: RTP instance '0x7f4d0bcd8138' is setup and ready to go [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4d0bcd8138' [2015-07-17 15:21:47] VERBOSE[1781][C-0000ed11] netsock2.c: == Using SIP RTP CoS mark 5 [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP o=- 14123045 14123045 IN IP4 109.235.35.252... OK. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '91.236.19.12' into... [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: ...host '91.236.19.12' and port ''. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP c=IN IP4 91.236.19.12... OK. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] rtp_engine.c: Setting payload 8 based on m type on 0x7f4ce6279c90 [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] rtp_engine.c: Setting payload 96 based on m type on 0x7f4ce6279c90 [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '109.235.35.252' into... [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: ...host '109.235.35.252' and port ''. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 109.235.35.252... OK. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP b=RR:0... UNSUPPORTED OR FAILED. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP b=RS:0... UNSUPPORTED OR FAILED. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=maxptime:40... UNSUPPORTED OR FAILED. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=nortpproxy:yes... UNSUPPORTED OR FAILED. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4d0bcd8138' [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] rtp_engine.c: Copying payload 8 from 0x7f4ce6279c90 to 0x7f4d0bcd8300 [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] rtp_engine.c: Copying payload 96 from 0x7f4ce6279c90 to 0x7f4d0bcd8300 [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4d0bcd8138' [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: We're settling with these formats: (alaw) [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Checking SIP call limits for device [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Updating call counter for incoming call [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: Splitting 'mo-pbx.pocos.nl:5060' into... [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: ...host 'mo-pbx.pocos.nl' and port ''. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: Splitting 'pocos.nl:5060' into... [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] netsock2.c: ...host 'pocos.nl' and port ''. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: *** Our native formats are (alaw) [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: *** Joint capabilities are (alaw) [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: *** Our capabilities are (alaw) [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: This channel will not be able to handle video. [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: build_route: Record-Route hop: [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: SIP/sbc2momt-0002191e: New call is still down.... Trying... [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 109.235.35.252:5060 [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'EXTEN' is '00735232329' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'SIPCALLID' is 'SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo:1] NoOp("SIP/sbc2momt-0002191e", "MO: 00735232329 SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'DumpChan' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo:2] DumpChan("SIP/sbc2momt-0002191e", "3") in new stack [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] app_dumpchan.c: -- [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'Set' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo:3] Set("SIP/sbc2momt-0002191e", "TRANSFERERNAME=") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'EXTEN' is '00735232329' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'AGI' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo:4] AGI("SIP/sbc2momt-0002191e", "/etc/asterisk/agi/mo,00735232329") in new stack [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] res_agi.c: -- Launched AGI Script /etc/asterisk/agi/mo [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'SIPTRANSFER' is NULL [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'SIPREFERREDBYHDR' is NULL [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] res_agi.c: /etc/asterisk/agi/mo,00735232329: +31658061038 [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] res_agi.c: -- AGI Script Executing Application: (Goto) Options: (mo-pbx,00735232329,1) [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Goto (mo-pbx,00735232329,1) [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] res_agi.c: -- AGI Script /etc/asterisk/agi/mo completed, returning 0 [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo-pbx:1] NoOp("SIP/sbc2momt-0002191e", "MO-PBX") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'Macro' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo-pbx:2] Macro("SIP/sbc2momt-0002191e", "setinternalaccountcode") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Function CDR(accountcode) result is '5707' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:1] NoOp("SIP/sbc2momt-0002191e", "5707") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] app_macro.c: Executed application: NoOp [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Function CDR(accountcode) result is '5707' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Expression result is '0' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'GotoIf' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:2] GotoIf("SIP/sbc2momt-0002191e", "0?exit") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Not taking any branch [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] app_macro.c: Executed application: GotoIf [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Function CDR(accountcode) result is '5707' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'Set' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:3] Set("SIP/sbc2momt-0002191e", "CDR(accountcode)=internal-5707") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] app_macro.c: Executed application: Set [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'MacroExit' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:4] MacroExit("SIP/sbc2momt-0002191e", "") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'SIPCALLID' is 'SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'Set' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo-pbx:3] Set("SIP/sbc2momt-0002191e", "CDR(userfield)=SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL:FOP") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'DumpChan' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo-pbx:4] DumpChan("SIP/sbc2momt-0002191e", "3") in new stack [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] app_dumpchan.c: -- [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'TRANSFERERNAME' is '' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Function LEN() result is '0' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Expression result is '0' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'EXTEN' is '00735232329' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'GotoIf' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo-pbx:5] GotoIf("SIP/sbc2momt-0002191e", "0?mo,00735232329,1") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Not taking any branch [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'ResetCDR' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo-pbx:6] ResetCDR("SIP/sbc2momt-0002191e", "") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'mo_sipurl' is '00735232329:password::401:udp@78.109.2.69:5060' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Result of 'mo_dialoptions' is 'M(setdtmftx^info)' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] pbx.c: Launching 'Dial' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [00735232329@mo-pbx:7] Dial("SIP/sbc2momt-0002191e", "SIP/00735232329:password::401:udp@78.109.2.69:5060,120,TM(setdtmftx^info)") in new stack [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Allocating new SIP dialog for 109875574faba4db3f96155a5e38cf6e@mobile.pocos.nl - INVITE (No RTP) [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4cf001fbf8' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Allocated port 18198 for RTP instance '0x7f4cf001fbf8' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] netsock2.c: Splitting '109.235.32.49' into... [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] netsock2.c: ...host '109.235.32.49' and port ''. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] rtp_engine.c: RTP instance '0x7f4cf001fbf8' is setup and ready to go [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4cf001fbf8' [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] netsock2.c: == Using SIP RTP CoS mark 5 [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] acl.c: For destination '78.109.2.69', our source address is '109.235.32.49'. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 109.235.32.49:5060 [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: SIP call-id changed from '109875574faba4db3f96155a5e38cf6e@mobile.pocos.nl' to '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: *** Our native formats are (alaw) [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: *** Joint capabilities are (alaw) [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: *** Our capabilities are (alaw) [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: This channel will not be able to handle video. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel_internal_api.c: Channel Call ID changing from [C-0000ed11] to [C-0000ed11] [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_dialoptions from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sipurl from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_context from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_dialtimeout from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mt_dialtimeout from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_is_pgsmonly from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_dtmftx from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_voicemail from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sip_exten from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sip_passwd from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sip_authname from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sip_host from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_clip from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_accountcode from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_forcedonpbx from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_msisdn from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_id from SIP/sbc2momt-0002191e to SIP/78.109.2.69:5060-0002191f. [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Outgoing Call for 00735232329 [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Updating call counter for outgoing call [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: ** Our capability: (alaw) Video flag: False Text flag: False [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: ** Our prefcodec: (alaw) [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: -- Done with adding codecs to SDP [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Initializing initreq for method INVITE - callid 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl [2015-07-17 15:21:47] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:21:47] VERBOSE[47965][C-0000ed11] app_dial.c: -- Called SIP/00735232329:password::401:udp@78.109.2.69:5060 [2015-07-17 15:21:47] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' Request 102: Found [2015-07-17 15:21:47] DEBUG[1781][C-0000ed11] chan_sip.c: SIP response 100 to standard invite [2015-07-17 15:21:49] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag 8a2f3c70 [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' Request 102: Found [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: SIP response 183 to standard invite [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: build_route: Contact hop: [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP o=- 97034665 1 IN IP4 78.109.2.69... OK. [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP s=Swyx LinkMgr... UNSUPPORTED OR FAILED. [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '78.109.2.69' into... [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port ''. [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP c=IN IP4 78.109.2.69... OK. [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] rtp_engine.c: Setting payload 8 based on m type on 0x7f4ce6278cf0 [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4cf001fbf8' [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] rtp_engine.c: Copying payload 8 from 0x7f4ce6278cf0 to 0x7f4cf001fdc0 [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4cf001fbf8' [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: We're settling with these formats: (alaw) [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: We have an owner, now see if we need to change this call [2015-07-17 15:21:49] VERBOSE[47965][C-0000ed11] app_dial.c: -- SIP/78.109.2.69:5060-0002191f is making progress passing it to SIP/sbc2momt-0002191e [2015-07-17 15:21:49] DEBUG[47965][C-0000ed11] chan_sip.c: Setting framing from config on incoming call [2015-07-17 15:21:49] DEBUG[47965][C-0000ed11] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [2015-07-17 15:21:49] DEBUG[47965][C-0000ed11] chan_sip.c: ** Our prefcodec: (nothing) [2015-07-17 15:21:49] DEBUG[47965][C-0000ed11] chan_sip.c: -- Done with adding codecs to SDP [2015-07-17 15:21:49] DEBUG[47965][C-0000ed11] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [2015-07-17 15:21:49] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 109.235.35.252:5060 [2015-07-17 15:21:49] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag 8a2f3c70 [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' Request 102: Found [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: SIP response 180 to standard invite [2015-07-17 15:21:49] DEBUG[1781][C-0000ed11] chan_sip.c: build_route: Contact hop: [2015-07-17 15:21:49] VERBOSE[47965][C-0000ed11] app_dial.c: -- SIP/78.109.2.69:5060-0002191f is ringing [2015-07-17 15:21:49] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 109.235.35.252:5060 [2015-07-17 15:21:51] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag 8a2f3c70 [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: Acked pending invite 102 [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' of Request 102: Match Found [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: SIP response 200 to standard invite [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: Call 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl responded to our reinvite without changing SDP version; ignoring SDP. [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: Updating call counter for outgoing call [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: build_route: Contact hop: [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: Trying to put 'ACK sip:78.' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:21:51] VERBOSE[47965][C-0000ed11] app_dial.c: -- SIP/78.109.2.69:5060-0002191f answered SIP/sbc2momt-0002191e [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] app.c: SIP/78.109.2.69:5060-0002191f Original location: mo-pbx,00735232329,1 [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:21:51] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [s@macro-setdtmftx:1] NoOp("SIP/78.109.2.69:5060-0002191f", "") in new stack [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] app_macro.c: Executed application: NoOp [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] pbx.c: Result of 'ARG1' is 'info' [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] pbx.c: Launching 'SIPDtmfMode' [2015-07-17 15:21:51] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [s@macro-setdtmftx:2] SIPDtmfMode("SIP/78.109.2.69:5060-0002191f", "info") in new stack [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] app_macro.c: Executed application: SIPDtmfMode [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] pbx.c: Launching 'MacroExit' [2015-07-17 15:21:51] VERBOSE[47965][C-0000ed11] pbx.c: -- Executing [s@macro-setdtmftx:3] MacroExit("SIP/78.109.2.69:5060-0002191f", "") in new stack [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] app.c: Macro exited with status 0 [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] app.c: SIP/78.109.2.69:5060-0002191f Ending location: mo-pbx,00735232329,1 [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] chan_sip.c: SIP answering channel: SIP/sbc2momt-0002191e [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] chan_sip.c: Setting framing from config on incoming call [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] chan_sip.c: ** Our prefcodec: (nothing) [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] chan_sip.c: -- Done with adding codecs to SDP [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 109.235.35.252:5060 [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] features.c: bridge answer set, chan answer set [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] features.c: Removing dialed interfaces datastore on SIP/78.109.2.69:5060-0002191f since we're bridging [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] channel.c: setting peeraccount to internal-5707 for SIP/78.109.2.69:5060-0002191f from data on channel SIP/sbc2momt-0002191e [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] channel.c: setting peeraccount to internal-5707 for SIP/sbc2momt-0002191e from data on channel SIP/78.109.2.69:5060-0002191f [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:51] DEBUG[1781] chan_sip.c: = Looking for Call ID: SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL (Checking From) --From tag SDq0j4701-0836716337 --To-tag as40d2e6d4 [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2015-07-17 15:21:51] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on 'SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL' of Response 14417: Match Found [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: 0x7f4d0bcdc680 -- Probation learning mode pass with source address 109.235.35.252:60712 [2015-07-17 15:21:51] VERBOSE[47965][C-0000ed11] res_rtp_asterisk.c: > 0x7f4d0bcdc680 -- Probation passed - setting RTP source address to 109.235.35.252:60712 [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f4cf001fbf8' [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: 0x7f4cf006d5f0 -- Probation learning mode pass with source address 78.109.2.69:55096 [2015-07-17 15:21:51] VERBOSE[47965][C-0000ed11] res_rtp_asterisk.c: > 0x7f4cf006d5f0 -- Probation passed - setting RTP source address to 78.109.2.69:55096 [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [2015-07-17 15:21:51] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 35 (#), at 109.235.35.252:60712 [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] channel.c: Got DTMF begin on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] features.c: Not passing DTMF through, since it may be a feature code [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 35 (#), at 109.235.35.252:60712 [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] channel.c: Got DTMF end on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] features.c: Feature interpret: chan=SIP/sbc2momt-0002191e, peer=SIP/78.109.2.69:5060-0002191f, code=#, sense=1, features=2, dynamic=# [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] features.c: Set feature timer to 1500 ms [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:55] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] features.c: Timed out for feature! [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'INFO sip:78' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:21:57] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag 8a2f3c70 [2015-07-17 15:21:57] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' of Request 103: Match Found [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 49 (1), at 109.235.35.252:60712 [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] channel.c: Got DTMF begin on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] features.c: Passing DTMF through, since it is not a feature code [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 49 (1), at 109.235.35.252:60712 [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] channel.c: Got DTMF end on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'INFO sip:78' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:57] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:58] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag 8a2f3c70 [2015-07-17 15:21:58] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' of Request 104: Match Found [2015-07-17 15:21:59] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Got RTCP report of 52 bytes [2015-07-17 15:21:59] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 52 (4), at 109.235.35.252:60712 [2015-07-17 15:21:59] DEBUG[47965][C-0000ed11] channel.c: Got DTMF begin on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:21:59] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:59] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:59] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:21:59] DEBUG[47965][C-0000ed11] features.c: Passing DTMF through, since it is not a feature code [2015-07-17 15:21:59] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:21:59] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 52 (4), at 109.235.35.252:60712 [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] channel.c: Got DTMF end on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'INFO sip:78' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag 8a2f3c70 [2015-07-17 15:22:00] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' of Request 105: Match Found [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 109.235.35.252:60712 [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] channel.c: Got DTMF begin on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] features.c: Passing DTMF through, since it is not a feature code [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 109.235.35.252:60712 [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] channel.c: Got DTMF end on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'INFO sip:78' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag 8a2f3c70 [2015-07-17 15:22:00] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' of Request 106: Match Found [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 50 (2), at 109.235.35.252:60712 [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] channel.c: Got DTMF begin on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] features.c: Passing DTMF through, since it is not a feature code [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:00] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 50 (2), at 109.235.35.252:60712 [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] channel.c: Got DTMF end on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'INFO sip:78' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag 8a2f3c70 [2015-07-17 15:22:01] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' of Request 107: Match Found [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 35 (#), at 109.235.35.252:60712 [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] channel.c: Got DTMF begin on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] features.c: Not passing DTMF through, since it may be a feature code [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 35 (#), at 109.235.35.252:60712 [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] channel.c: Got DTMF end on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] features.c: Feature interpret: chan=SIP/sbc2momt-0002191e, peer=SIP/78.109.2.69:5060-0002191f, code=#, sense=1, features=2, dynamic=# [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] features.c: Set feature timer to 1500 ms [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:01] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:03] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:03] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:03] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:03] DEBUG[47965][C-0000ed11] features.c: Timed out for feature! [2015-07-17 15:22:03] DEBUG[47965][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:03] DEBUG[47965][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:03] DEBUG[47965][C-0000ed11] chan_sip.c: Trying to put 'INFO sip:78' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:03] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking To) --From tag as32ab4836 --To-tag 8a2f3c70 [2015-07-17 15:22:03] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on '6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl' of Request 108: Match Found [2015-07-17 15:22:03] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:03] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:04] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Got RTCP report of 52 bytes [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 35 (#), at 109.235.35.252:60712 [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Got DTMF begin on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] features.c: Not passing DTMF through, since it may be a feature code [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 35 (#), at 109.235.35.252:60712 [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Got DTMF end on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] features.c: Feature interpret: chan=SIP/sbc2momt-0002191e, peer=SIP/78.109.2.69:5060-0002191f, code=#, sense=1, features=2, dynamic=# [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] features.c: Set feature timer to 1500 ms [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 49 (1), at 109.235.35.252:60712 [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Got DTMF begin on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] features.c: Not passing DTMF through, since it may be a feature code [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 49 (1), at 109.235.35.252:60712 [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Got DTMF end on channel (SIP/sbc2momt-0002191e) [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] features.c: Feature interpret: chan=SIP/sbc2momt-0002191e, peer=SIP/78.109.2.69:5060-0002191f, code=#1, sense=1, features=2, dynamic=# [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] features.c: Feature detected: fname=Attended Transfer sname=atxfer exten=#1 [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] features.c: Executing Attended Transfer SIP/sbc2momt-0002191e, SIP/78.109.2.69:5060-0002191f (sense=1) [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:05] VERBOSE[47965][C-0000ed11] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Set channel SIP/sbc2momt-0002191e to write format gsm [2015-07-17 15:22:05] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-07-17 15:22:05] VERBOSE[47965][C-0000ed11] file.c: -- Playing 'pbx-transfer.gsm' (language 'en') [2015-07-17 15:22:05] DEBUG[3145][C-0000ed11] res_musiconhold.c: SIP/78.109.2.69:5060-0002191f Opened file 0 '/var/lib/asterisk/moh/onholdtone-8kHz' [2015-07-17 15:22:05] DEBUG[3145][C-0000ed11] res_rtp_asterisk.c: Difference is 2200, ms is 295 [2015-07-17 15:22:06] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:06] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:06] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:06] DEBUG[47965][C-0000ed11] channel.c: Set channel SIP/sbc2momt-0002191e to write format alaw [2015-07-17 15:22:06] DEBUG[47965][C-0000ed11] channel.c: Set channel SIP/sbc2momt-0002191e to write format slin [2015-07-17 15:22:06] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-07-17 15:22:07] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 52 (4), at 109.235.35.252:60712 [2015-07-17 15:22:07] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 52 (4), at 109.235.35.252:60712 [2015-07-17 15:22:07] DEBUG[47965][C-0000ed11] channel.c: Set channel SIP/sbc2momt-0002191e to write format alaw [2015-07-17 15:22:07] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:08] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 109.235.35.252:60712 [2015-07-17 15:22:08] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 109.235.35.252:60712 [2015-07-17 15:22:08] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 50 (2), at 109.235.35.252:60712 [2015-07-17 15:22:08] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 50 (2), at 109.235.35.252:60712 [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 35 (#), at 109.235.35.252:60712 [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Creating END DTMF Frame: 35 (#), at 109.235.35.252:60712 [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] features.c: Checking if 402@mo-pbx is a parking exten [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel_internal_api.c: Channel Call ID changing from [C-0000ed11] to [C-0000ed11] [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_dialoptions from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sipurl from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_context from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_dialtimeout from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mt_dialtimeout from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_is_pgsmonly from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_dtmftx from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_voicemail from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sip_exten from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sip_passwd from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sip_authname from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_sip_host from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_clip from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_accountcode from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_forcedonpbx from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_msisdn from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Inheriting variable mo_id from SIP/sbc2momt-0002191e to Local/402@mo-pbx-00000021;1. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:1] NoOp("Local/402@mo-pbx-00000021;2", "MO-PBX") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'Macro' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:2] Macro("Local/402@mo-pbx-00000021;2", "setinternalaccountcode") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Function CDR(accountcode) result is '(null)' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:1] NoOp("Local/402@mo-pbx-00000021;2", "") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] app_macro.c: Executed application: NoOp [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Function CDR(accountcode) result is '(null)' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Expression result is '0' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'GotoIf' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:2] GotoIf("Local/402@mo-pbx-00000021;2", "0?exit") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Not taking any branch [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] app_macro.c: Executed application: GotoIf [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Function CDR(accountcode) result is '(null)' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'Set' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:3] Set("Local/402@mo-pbx-00000021;2", "CDR(accountcode)=internal-") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] app_macro.c: Executed application: Set [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'MacroExit' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:4] MacroExit("Local/402@mo-pbx-00000021;2", "") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'SIPCALLID' is NULL [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'Set' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:3] Set("Local/402@mo-pbx-00000021;2", "CDR(userfield)=:FOP") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'DumpChan' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:4] DumpChan("Local/402@mo-pbx-00000021;2", "3") in new stack [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] app_dumpchan.c: -- [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'TRANSFERERNAME' is 'SIP/sbc2momt-0002191e' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Function LEN(SIP/sbc2momt-0002191e) result is '21' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Expression result is '1' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'EXTEN' is '402' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'GotoIf' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:5] GotoIf("Local/402@mo-pbx-00000021;2", "1?mo,402,1") in new stack [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Goto (mo,402,1) [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'EXTEN' is '402' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'SIPCALLID' is NULL [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo:1] NoOp("Local/402@mo-pbx-00000021;2", "MO: 402 ") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'DumpChan' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo:2] DumpChan("Local/402@mo-pbx-00000021;2", "3") in new stack [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] app_dumpchan.c: -- [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'Set' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo:3] Set("Local/402@mo-pbx-00000021;2", "TRANSFERERNAME=") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'EXTEN' is '402' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'AGI' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo:4] AGI("Local/402@mo-pbx-00000021;2", "/etc/asterisk/agi/mo,402") in new stack [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] res_agi.c: -- Launched AGI Script /etc/asterisk/agi/mo [2015-07-17 15:22:09] WARNING[48031][C-0000ed11] chan_sip.c: This function can only be used on SIP channels. [2015-07-17 15:22:09] WARNING[48031][C-0000ed11] chan_sip.c: This function can only be used on SIP channels. [2015-07-17 15:22:09] WARNING[48031][C-0000ed11] chan_sip.c: This function can only be used on SIP channels. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'SIPTRANSFER' is NULL [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'SIPREFERREDBYHDR' is NULL [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'mo_msisdn' is '+31658061038' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] res_agi.c: /etc/asterisk/agi/mo,402: +31658061038 [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] res_agi.c: -- AGI Script Executing Application: (Goto) Options: (mo-pbx,402,1) [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Goto (mo-pbx,402,1) [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] res_agi.c: -- AGI Script /etc/asterisk/agi/mo completed, returning 0 [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:1] NoOp("Local/402@mo-pbx-00000021;2", "MO-PBX") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'Macro' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:2] Macro("Local/402@mo-pbx-00000021;2", "setinternalaccountcode") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Function CDR(accountcode) result is '5707' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:1] NoOp("Local/402@mo-pbx-00000021;2", "5707") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] app_macro.c: Executed application: NoOp [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Function CDR(accountcode) result is '5707' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Expression result is '0' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'GotoIf' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:2] GotoIf("Local/402@mo-pbx-00000021;2", "0?exit") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Not taking any branch [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] app_macro.c: Executed application: GotoIf [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Function CDR(accountcode) result is '5707' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'Set' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:3] Set("Local/402@mo-pbx-00000021;2", "CDR(accountcode)=internal-5707") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] app_macro.c: Executed application: Set [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'MacroExit' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setinternalaccountcode:4] MacroExit("Local/402@mo-pbx-00000021;2", "") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'SIPCALLID' is NULL [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'Set' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:3] Set("Local/402@mo-pbx-00000021;2", "CDR(userfield)=:FOP") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'DumpChan' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:4] DumpChan("Local/402@mo-pbx-00000021;2", "3") in new stack [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] app_dumpchan.c: -- [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'TRANSFERERNAME' is '' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Function LEN() result is '0' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Expression result is '0' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'EXTEN' is '402' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'GotoIf' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:5] GotoIf("Local/402@mo-pbx-00000021;2", "0?mo,402,1") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Not taking any branch [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'ResetCDR' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:6] ResetCDR("Local/402@mo-pbx-00000021;2", "") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'mo_sipurl' is '402:password::401:udp@78.109.2.69:5060' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Result of 'mo_dialoptions' is 'M(setdtmftx^info)' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] pbx.c: Launching 'Dial' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [402@mo-pbx:7] Dial("Local/402@mo-pbx-00000021;2", "SIP/402:password::401:udp@78.109.2.69:5060,120,TM(setdtmftx^info)") in new stack [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Allocating new SIP dialog for 78aa315728ace240259b603d59980256@mobile.pocos.nl - INVITE (No RTP) [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4cf01aede8' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Allocated port 16822 for RTP instance '0x7f4cf01aede8' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] netsock2.c: Splitting '109.235.32.49' into... [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] netsock2.c: ...host '109.235.32.49' and port ''. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] rtp_engine.c: RTP instance '0x7f4cf01aede8' is setup and ready to go [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4cf01aede8' [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] netsock2.c: == Using SIP RTP CoS mark 5 [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] acl.c: For destination '78.109.2.69', our source address is '109.235.32.49'. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 109.235.32.49:5060 [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: SIP call-id changed from '78aa315728ace240259b603d59980256@mobile.pocos.nl' to '598a9242767a3b1912f983f156460f0e@mobile.pocos.nl' [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: *** Our native formats are (alaw) [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: *** Joint capabilities are (alaw) [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: *** Our capabilities are (alaw) [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: This channel will not be able to handle video. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel_internal_api.c: Channel Call ID changing from [C-0000ed11] to [C-0000ed11] [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_dialoptions from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_sipurl from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_dialtimeout from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mt_dialtimeout from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_is_pgsmonly from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_dtmftx from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_voicemail from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_sip_exten from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_sip_passwd from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_sip_authname from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_sip_host from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_clip from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_accountcode from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_forcedonpbx from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_msisdn from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_id from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] channel.c: Inheriting variable mo_context from Local/402@mo-pbx-00000021;2 to SIP/78.109.2.69:5060-00021924. [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Outgoing Call for 402 [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Updating call counter for outgoing call [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: ** Our capability: (alaw) Video flag: False Text flag: False [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: ** Our prefcodec: (alaw) [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: -- Done with adding codecs to SDP [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Initializing initreq for method INVITE - callid 598a9242767a3b1912f983f156460f0e@mobile.pocos.nl [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] app_dial.c: -- Called SIP/402:password::401:udp@78.109.2.69:5060 [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[1781] chan_sip.c: = Looking for Call ID: 598a9242767a3b1912f983f156460f0e@mobile.pocos.nl (Checking To) --From tag as0adf2c67 --To-tag [2015-07-17 15:22:09] DEBUG[1781][C-0000ed11] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '598a9242767a3b1912f983f156460f0e@mobile.pocos.nl' Request 102: Found [2015-07-17 15:22:09] DEBUG[1781][C-0000ed11] chan_sip.c: SIP response 100 to standard invite [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[1781] chan_sip.c: = Looking for Call ID: OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE. (Checking From) --From tag 3b5d2056 --To-tag [2015-07-17 15:22:09] DEBUG[1781] acl.c: For destination '78.109.2.69', our source address is '109.235.32.49'. [2015-07-17 15:22:09] DEBUG[1781] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 109.235.32.49:5060 [2015-07-17 15:22:09] DEBUG[1781] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:09] DEBUG[1781] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:09] DEBUG[1781] chan_sip.c: Allocating new SIP dialog for OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE. - INVITE (No RTP) [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer" [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] sip/reqresp_parser.c: Found SIP option: -timer- [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] sip/reqresp_parser.c: Matched SIP option: timer [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: Splitting 'PANTEL-1020' into... [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: ...host 'PANTEL-1020' and port ''. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4d0aafda38' [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] res_rtp_asterisk.c: Allocated port 19240 for RTP instance '0x7f4d0aafda38' [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: Splitting '109.235.32.49' into... [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: ...host '109.235.32.49' and port ''. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] rtp_engine.c: RTP instance '0x7f4d0aafda38' is setup and ready to go [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4d0aafda38' [2015-07-17 15:22:09] VERBOSE[1781][C-0000ed14] netsock2.c: == Using SIP RTP CoS mark 5 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP o=- 97054680 1 IN IP4 109.235.32.49... OK. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP s=Swyx IpPbxSrv... UNSUPPORTED OR FAILED. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: Splitting '109.235.32.49' into... [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: ...host '109.235.32.49' and port ''. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP c=IN IP4 109.235.32.49... OK. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] rtp_engine.c: Setting payload 8 based on m type on 0x7f4ce6279c90 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] rtp_engine.c: Setting payload 101 based on m type on 0x7f4ce6279c90 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4d0aafda38' [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] rtp_engine.c: Copying payload 8 from 0x7f4ce6279c90 to 0x7f4d0aafdc00 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] rtp_engine.c: Copying payload 101 from 0x7f4ce6279c90 to 0x7f4d0aafdc00 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4d0aafda38' [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: We're settling with these formats: (alaw) [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Checking SIP call limits for device [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Updating call counter for incoming call [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: Splitting '109.235.32.49:5060' into... [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: ...host '109.235.32.49' and port ''. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: Splitting 'PANTEL-1020' into... [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] netsock2.c: ...host 'PANTEL-1020' and port ''. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Incoming INVITE with 'timer' option supported [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: INVITE also has "Session-Expires" header. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Session-Expires: 90 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Refresher: UAC [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: INVITE also has "Min-SE" header. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Received Min-SE: 90 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: *** Our native formats are (alaw) [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: *** Joint capabilities are (alaw) [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: *** Our capabilities are (alaw) [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: This channel will not be able to handle video. [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: build_route: Contact hop: [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: SIP/PANTEL-1020-00021925: New call is still down.... Trying... [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'EXTEN' is '402' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [402@forcedonpbx:1] NoOp("SIP/PANTEL-1020-00021925", "ForceOnPBX: 402") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'DumpChan' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [402@forcedonpbx:2] DumpChan("SIP/PANTEL-1020-00021925", "3") in new stack [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] app_dumpchan.c: -- [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'AGI' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [402@forcedonpbx:3] AGI("SIP/PANTEL-1020-00021925", "/etc/asterisk/agi/forcedonpbx") in new stack [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] res_agi.c: -- Launched AGI Script /etc/asterisk/agi/forcedonpbx [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Got RTCP report of 52 bytes [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'SIPTRANSFER' is NULL [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'mo_msisdn' is NULL [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'mo_forcedonpbx' is NULL [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] res_agi.c: -- AGI Script Executing Application: (SIPDtmfMode) Options: (info) [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] res_agi.c: -- AGI Script Executing Application: (Goto) Options: (mt,+31658061024,1) [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Goto (mt,+31658061024,1) [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] res_agi.c: -- AGI Script /etc/asterisk/agi/forcedonpbx completed, returning 0 [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:1] NoOp("SIP/PANTEL-1020-00021925", "MT: outbound") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'Macro' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:2] Macro("SIP/PANTEL-1020-00021925", "setinternalaccountcode") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Function CDR(accountcode) result is '(null)' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [s@macro-setinternalaccountcode:1] NoOp("SIP/PANTEL-1020-00021925", "") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] app_macro.c: Executed application: NoOp [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Function CDR(accountcode) result is '(null)' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Expression result is '0' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'GotoIf' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [s@macro-setinternalaccountcode:2] GotoIf("SIP/PANTEL-1020-00021925", "0?exit") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Not taking any branch [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] app_macro.c: Executed application: GotoIf [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Function CDR(accountcode) result is '(null)' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'Set' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [s@macro-setinternalaccountcode:3] Set("SIP/PANTEL-1020-00021925", "CDR(accountcode)=internal-") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] app_macro.c: Executed application: Set [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'MacroExit' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [s@macro-setinternalaccountcode:4] MacroExit("SIP/PANTEL-1020-00021925", "") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'Set' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:3] Set("SIP/PANTEL-1020-00021925", "TIMEOUT(absolute)=14400") in new stack [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] func_timeout.c: -- Channel will hangup at 2015-07-17 19:22:09.725 CEST. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'EXTEN' is '+31658061024' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'mt_dialtimeout' is NULL [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'mt_dialoptions' is 't' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'Dial' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:4] Dial("SIP/PANTEL-1020-00021925", "SIP/+31658061024@pgsmpocosnl,,rt") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Allocating new SIP dialog for 578289fe30b1c5ee7323cfd0131477f2@mobile.pocos.nl - INVITE (No RTP) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4cf0119528' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Allocated port 18032 for RTP instance '0x7f4cf0119528' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] netsock2.c: Splitting '109.235.32.49' into... [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] netsock2.c: ...host '109.235.32.49' and port ''. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] rtp_engine.c: RTP instance '0x7f4cf0119528' is setup and ready to go [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4cf0119528' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] netsock2.c: == Using SIP RTP CoS mark 5 [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] acl.c: For destination '109.235.38.156', our source address is '109.235.32.49'. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 109.235.32.49:5060 [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: SIP call-id changed from '578289fe30b1c5ee7323cfd0131477f2@mobile.pocos.nl' to '2ca6636e14e8a9ce34491ee860c090e0@mobile.pocos.nl' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** Our native formats are (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** Joint capabilities are (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** Our capabilities are (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: This channel will not be able to handle video. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel_internal_api.c: Channel Call ID changing from [C-0000ed14] to [C-0000ed14] [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mt_dialoptions from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_is_pgsmonly from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_dtmftx from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_voicemail from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_sip_exten from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_sip_passwd from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_sip_authname from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_sip_host from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_clip from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_accountcode from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_forcedonpbx from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_msisdn from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_id from SIP/PANTEL-1020-00021925 to SIP/pgsmpocosnl-00021926. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Outgoing Call for +31658061024 [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Updating call counter for outgoing call [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: ** Our capability: (alaw) Video flag: False Text flag: False [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: ** Our prefcodec: (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: -- Done with adding codecs to SDP [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Initializing initreq for method INVITE - callid 2ca6636e14e8a9ce34491ee860c090e0@mobile.pocos.nl [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 109.235.38.156:5080 [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] app_dial.c: -- Called SIP/+31658061024@pgsmpocosnl [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[1781] chan_sip.c: = Looking for Call ID: 598a9242767a3b1912f983f156460f0e@mobile.pocos.nl (Checking To) --From tag as0adf2c67 --To-tag ec3d4c36 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed11] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '598a9242767a3b1912f983f156460f0e@mobile.pocos.nl' Request 102: Found [2015-07-17 15:22:09] DEBUG[1781][C-0000ed11] chan_sip.c: SIP response 180 to standard invite [2015-07-17 15:22:09] DEBUG[1781][C-0000ed11] chan_sip.c: build_route: Contact hop: [2015-07-17 15:22:09] VERBOSE[48031][C-0000ed11] app_dial.c: -- SIP/78.109.2.69:5060-00021924 is ringing [2015-07-17 15:22:09] VERBOSE[47965][C-0000ed11] features.c: -- Local/402@mo-pbx-00000021;1 is ringing [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Driver for channel 'SIP/sbc2momt-0002191e' does not support indication 3, emulating it [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Set channel SIP/sbc2momt-0002191e to write format slin [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Difference is 15808, ms is 1996 [2015-07-17 15:22:09] DEBUG[1781] chan_sip.c: = Looking for Call ID: 2ca6636e14e8a9ce34491ee860c090e0@mobile.pocos.nl (Checking To) --From tag as5a633904 --To-tag 2eb48052 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Acked pending invite 102 [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Stopping retransmission on '2ca6636e14e8a9ce34491ee860c090e0@mobile.pocos.nl' of Request 102: Match Found [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: SIP response 404 to standard invite [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: Trying to put 'ACK sip:+31' onto UDP socket destined for 109.235.38.156:5080 [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Hanging up channel 'SIP/pgsmpocosnl-00021926' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Hangup call SIP/pgsmpocosnl-00021926, SIP callid 2ca6636e14e8a9ce34491ee860c090e0@mobile.pocos.nl [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Updating call counter for outgoing call [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4cf0119528' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'HANGUPCAUSE' is '1' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Expression result is '1' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'GotoIf' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:5] GotoIf("SIP/PANTEL-1020-00021925", "1?pm") in new stack [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Goto (mt,+31658061024,10) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'EXTEN' is '+31658061024' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:10] NoOp("SIP/PANTEL-1020-00021925", "Trying +31658061024 at Private Mobility") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'Set' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:11] Set("SIP/PANTEL-1020-00021925", "fopornotfop=NOTFOP") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'mo_forcedonpbx' is 'yes' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Expression result is '1' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'SIPCALLID' is 'OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE.' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'ExecIf' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:12] ExecIf("SIP/PANTEL-1020-00021925", "1?Set(CDR(userfield)=OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE.:FOP)") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'DIALSTATUS' is 'CHANUNAVAIL' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'HANGUPCAUSE' is '1' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'fopornotfop' is 'NOTFOP' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'mo_forcedonpbx' is 'yes' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'NoOp' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:13] NoOp("SIP/PANTEL-1020-00021925", "PREDIALSTATUS: CHANUNAVAIL HANGUPCAUSE: 1 FOP:NOTFOP yes") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'Set' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:14] Set("SIP/PANTEL-1020-00021925", "TIMEOUT(absolute)=0") in new stack [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] func_timeout.c: -- Channel hangup cancelled. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'EXTEN' is '+31658061024' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'mt_dialtimeout' is NULL [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Result of 'mt_dialoptions' is 't' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] pbx.c: Launching 'Dial' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] pbx.c: -- Executing [+31658061024@mt:15] Dial("SIP/PANTEL-1020-00021925", "SIP/+31658061024@sbc1momt,,t") in new stack [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Allocating new SIP dialog for 4f3ec6e41c7d4eb06982be0b06e2b41e@mobile.pocos.nl - INVITE (No RTP) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f4cf00f8248' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Allocated port 14036 for RTP instance '0x7f4cf00f8248' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] netsock2.c: Splitting '109.235.32.49' into... [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] netsock2.c: ...host '109.235.32.49' and port ''. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] rtp_engine.c: RTP instance '0x7f4cf00f8248' is setup and ready to go [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f4cf00f8248' [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] netsock2.c: == Using SIP RTP CoS mark 5 [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] acl.c: For destination '109.235.34.116', our source address is '109.235.32.49'. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 109.235.32.49:5060 [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Setting NAT on RTP to On [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: SIP call-id changed from '4f3ec6e41c7d4eb06982be0b06e2b41e@mobile.pocos.nl' to '081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl' [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** Our native formats are (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** Joint capabilities are (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** Our capabilities are (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: This channel will not be able to handle video. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel_internal_api.c: Channel Call ID changing from [C-0000ed14] to [C-0000ed14] [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mt_dialoptions from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_is_pgsmonly from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_dtmftx from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_voicemail from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_sip_exten from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_sip_passwd from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_sip_authname from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_sip_host from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_clip from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_accountcode from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_forcedonpbx from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_msisdn from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] channel.c: Inheriting variable mo_id from SIP/PANTEL-1020-00021925 to SIP/sbc1momt-00021927. [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Outgoing Call for +31658061024 [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Updating call counter for outgoing call [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: ** Our capability: (alaw) Video flag: False Text flag: False [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: ** Our prefcodec: (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: -- Done with adding codecs to SDP [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Initializing initreq for method INVITE - callid 081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl [2015-07-17 15:22:09] DEBUG[48034][C-0000ed14] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 109.235.34.116:5060 [2015-07-17 15:22:09] VERBOSE[48034][C-0000ed14] app_dial.c: -- Called SIP/+31658061024@sbc1momt [2015-07-17 15:22:09] DEBUG[1781] chan_sip.c: = Looking for Call ID: 081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl (Checking To) --From tag as122dcd12 --To-tag [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl' Request 102: Found [2015-07-17 15:22:09] DEBUG[1781][C-0000ed14] chan_sip.c: SIP response 100 to standard invite [2015-07-17 15:22:09] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame ...repeated 172 time... [2015-07-17 15:22:13] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:13] DEBUG[1781] chan_sip.c: = Looking for Call ID: 081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl (Checking To) --From tag as122dcd12 --To-tag SD82un599-1578563008 [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl' Request 102: Found [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: SIP response 183 to standard invite [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: build_route: Record-Route hop: [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP o=- 15573613 15573613 IN IP4 109.235.34.116... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] netsock2.c: Splitting '91.236.19.4' into... [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] netsock2.c: ...host '91.236.19.4' and port ''. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP c=IN IP4 91.236.19.4... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP a=sendrecv... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] rtp_engine.c: Setting payload 8 based on m type on 0x7f4ce6278cf0 [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] rtp_engine.c: Setting payload 101 based on m type on 0x7f4ce6278cf0 [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] netsock2.c: Splitting '109.235.34.116' into... [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] netsock2.c: ...host '109.235.34.116' and port ''. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 109.235.34.116... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP b=RR:0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP b=RS:0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP a=maxptime:40... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: Processing media-level (audio) SDP a=nortpproxy:yes... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4cf00f8248' [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] rtp_engine.c: Copying payload 8 from 0x7f4ce6278cf0 to 0x7f4cf00f8410 [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] rtp_engine.c: Copying payload 101 from 0x7f4ce6278cf0 to 0x7f4cf00f8410 [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4cf00f8248' [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: We're settling with these formats: (alaw) [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: We have an owner, now see if we need to change this call [2015-07-17 15:22:13] VERBOSE[48034][C-0000ed14] app_dial.c: -- SIP/sbc1momt-00021927 is making progress passing it to SIP/PANTEL-1020-00021925 [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] rtp_engine.c: Setting early bridge SDP of 'SIP/PANTEL-1020-00021925' with that of 'SIP/sbc1momt-00021927' [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] chan_sip.c: Setting framing from config on incoming call [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] chan_sip.c: ** Our prefcodec: (nothing) [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] chan_sip.c: -- Done with adding codecs to SDP [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:13] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: No remote address on RTP instance '0x7f4cf01aede8' so dropping frame [2015-07-17 15:22:13] DEBUG[1781] chan_sip.c: = Looking for Call ID: 598a9242767a3b1912f983f156460f0e@mobile.pocos.nl (Checking To) --From tag as0adf2c67 --To-tag ec3d4c36 [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '598a9242767a3b1912f983f156460f0e@mobile.pocos.nl' Request 102: Found [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: SIP response 183 to standard invite [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: build_route: Contact hop: [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP o=pocosmobile 798518551 798518551 IN IP4 109.235.32.49... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP s=Pocos Mobile... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '109.235.32.49' into... [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] netsock2.c: ...host '109.235.32.49' and port ''. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP c=IN IP4 109.235.32.49... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] rtp_engine.c: Setting payload 8 based on m type on 0x7f4ce6278cf0 [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4cf01aede8' [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] rtp_engine.c: Copying payload 8 from 0x7f4ce6278cf0 to 0x7f4cf01aefb0 [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4cf01aede8' [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: We're settling with these formats: (alaw) [2015-07-17 15:22:13] DEBUG[1781][C-0000ed11] chan_sip.c: We have an owner, now see if we need to change this call [2015-07-17 15:22:13] VERBOSE[48031][C-0000ed11] app_dial.c: -- SIP/78.109.2.69:5060-00021924 is making progress passing it to Local/402@mo-pbx-00000021;2 [2015-07-17 15:22:13] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [2015-07-17 15:22:13] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [2015-07-17 15:22:13] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f4cf01aede8' [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: 0x7f4d0ad55610 -- Probation learning mode pass with source address 109.235.32.49:16822 [2015-07-17 15:22:13] VERBOSE[48034][C-0000ed14] res_rtp_asterisk.c: > 0x7f4d0ad55610 -- Probation passed - setting RTP source address to 109.235.32.49:16822 [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f4cf00f8248' [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl' Request 102: Found [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: SIP response 183 to standard invite [2015-07-17 15:22:13] DEBUG[1781][C-0000ed14] chan_sip.c: build_route: Record-Route hop: [2015-07-17 15:22:13] VERBOSE[48034][C-0000ed14] app_dial.c: -- SIP/sbc1momt-00021927 is ringing [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] rtp_engine.c: Setting early bridge SDP of 'SIP/PANTEL-1020-00021925' with that of 'SIP/sbc1momt-00021927' [2015-07-17 15:22:13] DEBUG[48034][C-0000ed14] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:14] DEBUG[1781] chan_sip.c: = Looking for Call ID: 081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl (Checking To) --From tag as122dcd12 --To-tag SD82un599-1578563008 [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl' Request 102: Found [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: SIP response 180 to standard invite [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: build_route: Record-Route hop: [2015-07-17 15:22:14] VERBOSE[48034][C-0000ed14] app_dial.c: -- SIP/sbc1momt-00021927 is ringing [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] rtp_engine.c: Setting early bridge SDP of 'SIP/PANTEL-1020-00021925' with that of 'SIP/sbc1momt-00021927' [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: 0x7f4cf00fc790 -- Probation learning mode pass with source address 109.235.34.116:37700 [2015-07-17 15:22:14] VERBOSE[48034][C-0000ed14] res_rtp_asterisk.c: > 0x7f4cf00fc790 -- Probation passed - setting RTP source address to 109.235.34.116:37700 [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160 [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: 0x7f4cf01b3330 -- Probation learning mode pass with source address 109.235.32.49:19240 [2015-07-17 15:22:14] VERBOSE[48031][C-0000ed11] res_rtp_asterisk.c: > 0x7f4cf01b3330 -- Probation passed - setting RTP source address to 109.235.32.49:19240 [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] channel.c: Set channel SIP/sbc2momt-0002191e to write format alaw [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Got RTCP report of 52 bytes [2015-07-17 15:22:14] DEBUG[1781] chan_sip.c: = Looking for Call ID: 081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl (Checking To) --From tag as122dcd12 --To-tag SD82un599-1578563008 [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: Acked pending invite 102 [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: Stopping retransmission on '081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl' of Request 102: Match Found [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: SIP response 200 to standard invite [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: Call 081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl responded to our reinvite without changing SDP version; ignoring SDP. [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: Updating call counter for outgoing call [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: build_route: Record-Route hop: [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] netsock2.c: Splitting '109.235.34.116' into... [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] netsock2.c: ...host '109.235.34.116' and port ''. [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: Trying to put 'ACK sip:91.' onto UDP socket destined for 109.235.34.116:5060 [2015-07-17 15:22:14] VERBOSE[48034][C-0000ed14] app_dial.c: -- SIP/sbc1momt-00021927 answered SIP/PANTEL-1020-00021925 [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] chan_sip.c: SIP answering channel: SIP/PANTEL-1020-00021925 [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] chan_sip.c: Setting framing from config on incoming call [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] chan_sip.c: ** Our prefcodec: (nothing) [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] chan_sip.c: -- Done with adding codecs to SDP [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] chan_sip.c: Session timer started: 21040718 - OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE. 60000ms [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] features.c: bridge answer set, chan answer set [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] features.c: Removing dialed interfaces datastore on SIP/sbc1momt-00021927 since we're bridging [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] channel.c: setting peeraccount to internal- for SIP/sbc1momt-00021927 from data on channel SIP/PANTEL-1020-00021925 [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] channel.c: setting peeraccount to internal- for SIP/PANTEL-1020-00021925 from data on channel SIP/sbc1momt-00021927 [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2015-07-17 15:22:14] DEBUG[1781][C-0000ed14] chan_sip.c: Stopping retransmission on 'OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE.' of Response 1: Match Found [2015-07-17 15:22:14] DEBUG[1781] chan_sip.c: = Looking for Call ID: 598a9242767a3b1912f983f156460f0e@mobile.pocos.nl (Checking To) --From tag as0adf2c67 --To-tag ec3d4c36 [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] chan_sip.c: Acked pending invite 102 [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on '598a9242767a3b1912f983f156460f0e@mobile.pocos.nl' of Request 102: Match Found [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] chan_sip.c: SIP response 200 to standard invite [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] chan_sip.c: Call 598a9242767a3b1912f983f156460f0e@mobile.pocos.nl responded to our reinvite without changing SDP version; ignoring SDP. [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] chan_sip.c: Updating call counter for outgoing call [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] chan_sip.c: build_route: Contact hop: [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:14] DEBUG[1781][C-0000ed11] chan_sip.c: Trying to put 'ACK sip:78.' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:14] VERBOSE[48031][C-0000ed11] app_dial.c: -- SIP/78.109.2.69:5060-00021924 answered Local/402@mo-pbx-00000021;2 [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] app.c: SIP/78.109.2.69:5060-00021924 Original location: mo-pbx,402,1 [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] pbx.c: Launching 'NoOp' [2015-07-17 15:22:14] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setdtmftx:1] NoOp("SIP/78.109.2.69:5060-00021924", "") in new stack [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] app_macro.c: Executed application: NoOp [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] pbx.c: Result of 'ARG1' is 'info' [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] pbx.c: Launching 'SIPDtmfMode' [2015-07-17 15:22:14] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setdtmftx:2] SIPDtmfMode("SIP/78.109.2.69:5060-00021924", "info") in new stack [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] app_macro.c: Executed application: SIPDtmfMode [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] pbx.c: Launching 'MacroExit' [2015-07-17 15:22:14] VERBOSE[48031][C-0000ed11] pbx.c: -- Executing [s@macro-setdtmftx:3] MacroExit("SIP/78.109.2.69:5060-00021924", "") in new stack [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] app.c: Macro exited with status 0 [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] app.c: SIP/78.109.2.69:5060-00021924 Ending location: mo-pbx,402,1 [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] features.c: bridge answer set, chan answer set [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] features.c: Removing dialed interfaces datastore on SIP/78.109.2.69:5060-00021924 since we're bridging [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] channel.c: setting peeraccount to internal-5707 for SIP/78.109.2.69:5060-00021924 from data on channel Local/402@mo-pbx-00000021;2 [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] channel.c: setting peeraccount to internal-5707 for Local/402@mo-pbx-00000021;2 from data on channel SIP/78.109.2.69:5060-00021924 [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] features.c: Dial party C result: newchan:1, outstate:4 [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] features.c: Actually doing an attended transfer. [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] features.c: bridge answer set, chan answer set [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] channel.c: setting peeraccount to internal-5707 for Local/402@mo-pbx-00000021;1 from data on channel SIP/sbc2momt-0002191e [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] channel.c: Got a FRAME_CONTROL (-1) frame on channel Local/402@mo-pbx-00000021;1 [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and Local/402@mo-pbx-00000021;1 [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:14] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:18] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-07-17 15:22:18] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-07-17 15:22:19] DEBUG[1781] chan_sip.c: = Looking for Call ID: SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL (Checking From) --From tag SDq0j4701-0836716337 --To-tag as40d2e6d4 [2015-07-17 15:22:19] DEBUG[1781][C-0000ed11] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [2015-07-17 15:22:19] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '109.235.35.252' into... [2015-07-17 15:22:19] DEBUG[1781][C-0000ed11] netsock2.c: ...host '109.235.35.252' and port ''. [2015-07-17 15:22:19] DEBUG[1781][C-0000ed11] chan_sip.c: Setting SIP_ALREADYGONE on dialog SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL [2015-07-17 15:22:19] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4d0bcd8138' [2015-07-17 15:22:19] DEBUG[1781][C-0000ed11] chan_sip.c: Received bye, issuing owner hangup [2015-07-17 15:22:19] DEBUG[1781][C-0000ed11] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 109.235.35.252:5060 [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Didn't get a frame from channel: SIP/sbc2momt-0002191e [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Bridge stops bridging channels SIP/sbc2momt-0002191e and Local/402@mo-pbx-00000021;1 [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:19] VERBOSE[47965][C-0000ed11] res_musiconhold.c: -- Stopped music on hold on SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:19] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Got RTCP report of 52 bytes [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Planning to masquerade channel SIP/78.109.2.69:5060-0002191f into the structure of Transfered/SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Done planning to masquerade channel SIP/78.109.2.69:5060-0002191f into the structure of Transfered/SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Actually Masquerading SIP/78.109.2.69:5060-0002191f(6) into the structure of Transfered/SIP/78.109.2.69:5060-0002191f(6) [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Putting channel SIP/78.109.2.69:5060-0002191f in alaw/alaw formats [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] chan_sip.c: SIP Fixup: New owner for dialogue 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl: SIP/78.109.2.69:5060-0002191f (Old parent: Transfered/SIP/78.109.2.69:5060-0002191f) [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Driver for channel 'SIP/78.109.2.69:5060-0002191f' does not support indication 3, emulating it [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Set channel SIP/78.109.2.69:5060-0002191f to write format slin [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Done Masquerading SIP/78.109.2.69:5060-0002191f (6) [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Changing ssrc from 1534351286 to 1480679605 due to a source change [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Set channel Local/402@mo-pbx-00000021;1 to write format gsm [2015-07-17 15:22:19] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-07-17 15:22:19] VERBOSE[47965][C-0000ed11] file.c: -- Playing 'beep.gsm' (language 'en') [2015-07-17 15:22:19] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Difference is 1568, ms is 216 [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] channel.c: Set channel Local/402@mo-pbx-00000021;1 to write format alaw [2015-07-17 15:22:20] DEBUG[48061][C-0000ed11] features.c: bridge answer set, chan answer set [2015-07-17 15:22:20] DEBUG[48061][C-0000ed11] channel.c: setting peeraccount to internal-5707 for SIP/78.109.2.69:5060-0002191f from data on channel Local/402@mo-pbx-00000021;1 [2015-07-17 15:22:20] DEBUG[48061][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:20] DEBUG[48061][C-0000ed11] channel.c: Set channel SIP/78.109.2.69:5060-0002191f to write format alaw [2015-07-17 15:22:20] DEBUG[48061][C-0000ed11] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-07-17 15:22:20] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] channel.c: Hanging up channel 'Transfered/SIP/78.109.2.69:5060-0002191f' [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] app_dial.c: Exiting with DIALSTATUS=ANSWER. [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] pbx.c: Spawn extension (mo-pbx,00735232329,7) exited non-zero on 'SIP/sbc2momt-0002191e' [2015-07-17 15:22:20] VERBOSE[47965][C-0000ed11] pbx.c: == Spawn extension (mo-pbx, 00735232329, 7) exited non-zero on 'SIP/sbc2momt-0002191e' [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] channel.c: Soft-Hanging up channel 'SIP/sbc2momt-0002191e' [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] channel.c: Hanging up channel 'SIP/sbc2momt-0002191e' [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] chan_sip.c: Hangup call SIP/sbc2momt-0002191e, SIP callid SDq0j4701-2395957efedf9410d32973e14bcd9007500@VODAFONE.NL [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] chan_sip.c: Updating call counter for incoming call [2015-07-17 15:22:20] DEBUG[47965][C-0000ed11] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4d0bcd8138' [2015-07-17 15:22:23] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-07-17 15:22:23] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-07-17 15:22:24] DEBUG[48061][C-0000ed11] res_rtp_asterisk.c: Got RTCP report of 52 bytes [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: = Looking for Call ID: 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl (Checking From) --From tag 8a2f3c70 --To-tag as32ab4836 [2015-07-17 15:22:24] DEBUG[1781][C-0000ed11] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [2015-07-17 15:22:24] DEBUG[1781][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:24] DEBUG[1781][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:24] DEBUG[1781][C-0000ed11] chan_sip.c: Setting SIP_ALREADYGONE on dialog 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl [2015-07-17 15:22:24] DEBUG[1781][C-0000ed11] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4cf001fbf8' [2015-07-17 15:22:24] DEBUG[1781][C-0000ed11] chan_sip.c: Received bye, issuing owner hangup [2015-07-17 15:22:24] DEBUG[1781][C-0000ed11] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:24] DEBUG[48061][C-0000ed11] channel.c: Didn't get a frame from channel: SIP/78.109.2.69:5060-0002191f [2015-07-17 15:22:24] DEBUG[48061][C-0000ed11] channel.c: Bridge stops bridging channels SIP/78.109.2.69:5060-0002191f and Local/402@mo-pbx-00000021;1 [2015-07-17 15:22:24] DEBUG[48061][C-0000ed11] channel.c: Hanging up channel 'Local/402@mo-pbx-00000021;1' [2015-07-17 15:22:24] DEBUG[48061][C-0000ed11] channel.c: Hanging up channel 'SIP/78.109.2.69:5060-0002191f' [2015-07-17 15:22:24] DEBUG[48061][C-0000ed11] chan_sip.c: Hangup call SIP/78.109.2.69:5060-0002191f, SIP callid 6a334ce70d80f94d79351b67684a79c6@mobile.pocos.nl [2015-07-17 15:22:24] DEBUG[48061][C-0000ed11] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4cf001fbf8' [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] channel.c: Didn't get a frame from channel: Local/402@mo-pbx-00000021;2 [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] channel.c: Bridge stops bridging channels Local/402@mo-pbx-00000021;2 and SIP/78.109.2.69:5060-00021924 [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] channel.c: Hanging up channel 'SIP/78.109.2.69:5060-00021924' [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] chan_sip.c: Hangup call SIP/78.109.2.69:5060-00021924, SIP callid 598a9242767a3b1912f983f156460f0e@mobile.pocos.nl [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4cf01aede8' [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] chan_sip.c: Trying to put 'BYE sip:78.' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: = Looking for Call ID: 598a9242767a3b1912f983f156460f0e@mobile.pocos.nl (Checking To) --From tag as0adf2c67 --To-tag ec3d4c36 [2015-07-17 15:22:24] DEBUG[1781][C-0000ed11] chan_sip.c: Stopping retransmission on '598a9242767a3b1912f983f156460f0e@mobile.pocos.nl' of Request 103: Match Found [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: Destroying SIP dialog 598a9242767a3b1912f983f156460f0e@mobile.pocos.nl [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: * SIP Call [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 001. NewChan Channel SIP/78.109.2.69:5060-00021924 - from 598a9242767a3b1912 [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 100 Trying [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 180 Ringing [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 005. Rx SIP/2.0 / 102 INVITE / 183 Session Progress [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 006. Rx SIP/2.0 / 102 INVITE / 200 OK [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 007. TxReq ACK / 102 ACK - ACK [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 008. Hangup Cause Normal Clearing [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 009. SchedDestroy 32000 ms [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 010. RTCPaudio Quality:ssrc=23502946;themssrc=1569887450;lp=0;rxjitter=0.00000 [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 011. TxReqRel BYE / 103 BYE - BYE [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 012. Rx SIP/2.0 / 103 BYE / 200 OK [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 013. NeedDestroy Setting needdestroy because received 200 response [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: [2015-07-17 15:22:24] DEBUG[1781] rtp_engine.c: Destroyed RTP instance '0x7f4cf01aede8' [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: = Looking for Call ID: OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE. (Checking From) --From tag 3b5d2056 --To-tag as1d9ba6e3 [2015-07-17 15:22:24] DEBUG[1781][C-0000ed14] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [2015-07-17 15:22:24] DEBUG[1781][C-0000ed14] netsock2.c: Splitting '78.109.2.69:5060' into... [2015-07-17 15:22:24] DEBUG[1781][C-0000ed14] netsock2.c: ...host '78.109.2.69' and port '5060'. [2015-07-17 15:22:24] DEBUG[1781][C-0000ed14] chan_sip.c: Setting SIP_ALREADYGONE on dialog OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE. [2015-07-17 15:22:24] DEBUG[1781][C-0000ed14] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4d0aafda38' [2015-07-17 15:22:24] DEBUG[1781][C-0000ed14] chan_sip.c: Session timer stopped: 21040718 - OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE. [2015-07-17 15:22:24] DEBUG[1781][C-0000ed14] chan_sip.c: Received bye, issuing owner hangup [2015-07-17 15:22:24] DEBUG[1781][C-0000ed14] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 78.109.2.69:5060 [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] channel.c: Didn't get a frame from channel: SIP/PANTEL-1020-00021925 [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] channel.c: Bridge stops bridging channels SIP/PANTEL-1020-00021925 and SIP/sbc1momt-00021927 [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] app_dial.c: Exiting with DIALSTATUS=ANSWER. [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] pbx.c: Spawn extension (mo-pbx,402,7) exited non-zero on 'Local/402@mo-pbx-00000021;2' [2015-07-17 15:22:24] VERBOSE[48031][C-0000ed11] pbx.c: == Spawn extension (mo-pbx, 402, 7) exited non-zero on 'Local/402@mo-pbx-00000021;2' [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] channel.c: Soft-Hanging up channel 'Local/402@mo-pbx-00000021;2' [2015-07-17 15:22:24] DEBUG[48031][C-0000ed11] channel.c: Hanging up channel 'Local/402@mo-pbx-00000021;2' [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] channel.c: Hanging up channel 'SIP/sbc1momt-00021927' [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] chan_sip.c: Hangup call SIP/sbc1momt-00021927, SIP callid 081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] chan_sip.c: Updating call counter for outgoing call [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4cf00f8248' [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] netsock2.c: Splitting '109.235.34.116' into... [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] netsock2.c: ...host '109.235.34.116' and port ''. [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] chan_sip.c: Trying to put 'BYE sip:91.' onto UDP socket destined for 109.235.34.116:5060 [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: = Looking for Call ID: 081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl (Checking To) --From tag as122dcd12 --To-tag SD82un599-1578563008 [2015-07-17 15:22:24] DEBUG[1781][C-0000ed14] chan_sip.c: Stopping retransmission on '081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl' of Request 103: Match Found [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: Destroying SIP dialog 081922c51de400b36fdc442e7dc31c18@mobile.pocos.nl [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: * SIP Call [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 001. NewChan Channel SIP/sbc1momt-00021927 - from 081922c51de400b36fdc442e7d [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - INVITE [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 100 trying -- your call is important to [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 004. Rx SIP/2.0 / 102 INVITE / 183 Session Progress [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 005. Rx SIP/2.0 / 102 INVITE / 183 Session Progress [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 006. Rx SIP/2.0 / 102 INVITE / 180 Ringing [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 007. Rx SIP/2.0 / 102 INVITE / 200 OK [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 008. TxReq ACK / 102 ACK - ACK [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 009. Hangup Cause Normal Clearing [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 010. SchedDestroy 19200 ms [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 011. RTCPaudio Quality:ssrc=1672737727;themssrc=2778307470;lp=0;rxjitter=0.000 [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 012. TxReqRel BYE / 103 BYE - BYE [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 013. Rx SIP/2.0 / 103 BYE / 200 OK [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: 014. NeedDestroy Setting needdestroy because received 200 response [2015-07-17 15:22:24] DEBUG[1781] chan_sip.c: [2015-07-17 15:22:24] DEBUG[1781] rtp_engine.c: Destroyed RTP instance '0x7f4cf00f8248' [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] app_dial.c: Exiting with DIALSTATUS=ANSWER. [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] pbx.c: Spawn extension (mt,+31658061024,15) exited non-zero on 'SIP/PANTEL-1020-00021925' [2015-07-17 15:22:24] VERBOSE[48034][C-0000ed14] pbx.c: == Spawn extension (mt, +31658061024, 15) exited non-zero on 'SIP/PANTEL-1020-00021925' [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] channel.c: Soft-Hanging up channel 'SIP/PANTEL-1020-00021925' [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] channel.c: Hanging up channel 'SIP/PANTEL-1020-00021925' [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] chan_sip.c: Hangup call SIP/PANTEL-1020-00021925, SIP callid OGMyYTBlMzkwNWFkYjAwNzU2ZTM4NzMyYjg3M2Q4ZGE. [2015-07-17 15:22:24] DEBUG[48034][C-0000ed14] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4d0aafda38'