asterisk.ru

Добрый день.Сразу скажу, что делаю на еластиксе.
Хочу настроить IVR, голос записал, меню сделал, перевод входящего на ивр сделал. При звонке идет как будто хорошо, ивр трубку берет, аудио проигрывает, но нажатие кнопок ни к чему не приводит. Завел специально для обработки неверного направления один ивр с соотв аудио и по таймауту на второй ивр с соотв аудио. Так вот не работают ни корректные нажатия, типа 1 - на номер N, ни некорректные своей подсказки по неверному направлению и по таймауту я тоже не услышал. Т.е. перевод из первого ивр не производится никак.
Подскажите, пожалуйста, какой уровень ошибок поставить -rvvv или vvvv и sip set debug on и я выложу по возможности краткий лог, или есть другие варианты отладки?
Спасибо!

Google => "как включить дебаг DTMF" site:.asterisk.ru

спасибо не знал что за аббревиатура, включил, теперь вижу нажатия при внутренних звонках звонках между экстеншнами записи типа:

Код: Выделить всё

[Kvoip*CLI> [H[2Jvoip*CLI> 
[0K[2015-11-18 15:13:53] [1;32mDEBUG[0m[8397][C-0000001c]: [1;37mres_rtp_asterisk.c[0m:[1;37m3453[0m [1;37mcreate_dtmf_frame[0m: Creating BEGIN DTMF Frame: 48 (0), at 192.168.0.121:4020

[Kvoip*CLI> 
[0K[2015-11-18 15:13:53] [1;32mDTMF[0m[8397][C-0000001c]: [1;37mchannel.c[0m:[1;37m4214[0m [1;37m__ast_read[0m: DTMF begin '0' received on SIP/100-00000022

[Kvoip*CLI> 
[0K[2015-11-18 15:13:53] [1;32mDTMF[0m[8397][C-0000001c]: [1;37mchannel.c[0m:[1;37m4225[0m [1;37m__ast_read[0m: DTMF begin passthrough '0' on SIP/100-00000022

[Kvoip*CLI> 
[0K[2015-11-18 15:13:53] [1;32mDEBUG[0m[8397][C-0000001c]: [1;37mchannel.c[0m:[1;37m7703[0m [1;37mast_generic_bridge[0m: Got DTMF begin on channel (SIP/100-00000022)

[Kvoip*CLI> 
[0K[2015-11-18 15:13:53] [1;32mDEBUG[0m[8397][C-0000001c]: [1;37mchannel.c[0m:[1;37m8128[0m [1;37mast_channel_bridge[0m: Bridge stops bridging channels SIP/100-00000022 and SIP/102-00000023

[Kvoip*CLI> 
[0K[2015-11-18 15:13:53] [1;32mDEBUG[0m[8397][C-0000001c]: [1;37mres_rtp_asterisk.c[0m:[1;37m3453[0m [1;37mcreate_dtmf_frame[0m: Creating END DTMF Frame: 48 (0), at 192.168.0.121:4020
[2015-11-18 15:13:53] [1;32mDTMF[0m[8397][C-0000001c]: [1;37mchannel.c[0m:[1;37m4128[0m [1;37m__ast_read[0m: DTMF end '0' received on SIP/100-00000022, duration 200 ms

[Kvoip*CLI> 
[0K[2015-11-18 15:13:53] [1;32mDTMF[0m[8397][C-0000001c]: [1;37mchannel.c[0m:[1;37m4169[0m [1;37m__ast_read[0m: DTMF end accepted with begin '0' on SIP/100-00000022
[2015-11-18 15:13:53] [1;32mDTMF[0m[8397][C-0000001c]: [1;37mchannel.c[0m:[1;37m4198[0m [1;37m__ast_read[0m: DTMF end passthrough '0' on SIP/100-00000022

но при входящем звонке я сколько ни нажимаю на кнопки - ничего в консоли нет. может что типа слушателя нажатий не активировано на голосовом меню?

1. Вывод инфы в CLI при входящем звонке
2. Настройка транка из которого прилетает входящий.

P.S. И не превращайте ЕГО величество ФОРУМ в чат.

2.

Код: Выделить всё

[dialog]
type=friend
host=91.227.140.72
username=099060
secret=*********
nat=yes
directmedia=no
qualify=no
fromuser=099060
dtmfmode=rfc2833
insecure=port,invite
context=from-trunk-sip-dialog

1.

Код: Выделить всё

[2015-11-18 16:16:29] DEBUG[6828] chan_sip.c: Allocating new SIP dialog for 76c57ec031402391115add4d1d9cf7c0@192.168.0.70:5060 - OPTIONS (No RTP)
[2015-11-18 16:16:29] DEBUG[6828] chan_sip.c: SIP call-id changed from '76c57ec031402391115add4d1d9cf7c0@192.168.0.70:5060' to '3a56299b5914b74f40f50c4d556c6939@192.168.0.70:5060'
[2015-11-18 16:16:29] DEBUG[6828] chan_sip.c: Initializing initreq for method OPTIONS - callid 3a56299b5914b74f40f50c4d556c6939@192.168.0.70:5060
[2015-11-18 16:16:29] DEBUG[6828] chan_sip.c: Stopping retransmission on '3a56299b5914b74f40f50c4d556c6939@192.168.0.70:5060' of Request 102: Match Found
[2015-11-18 16:16:37] DEBUG[6828] chan_sip.c: Target address 91.227.140.72:5060 is not local, substituting externaddr
[2015-11-18 16:16:37] DEBUG[6828] chan_sip.c: Allocating new SIP dialog for 200107008bb11310a8c17845c4f02832@siprise-brn - INVITE (No RTP)
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xaef88d8'
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Allocated port 19896 for RTP instance '0xaef88d8'
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] rtp_engine.c: RTP instance '0xaef88d8' is setup and ready to go
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xaed76f8'
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Allocated port 16542 for RTP instance '0xaed76f8'
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] rtp_engine.c: RTP instance '0xaed76f8' is setup and ready to go
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xaed76f8'
[2015-11-18 16:16:37] VERBOSE[6828][C-0000001d] netsock2.c:   == Using SIP VIDEO TOS bits 136
[2015-11-18 16:16:37] VERBOSE[6828][C-0000001d] netsock2.c:   == Using SIP VIDEO CoS mark 6
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xaef88d8'
[2015-11-18 16:16:37] VERBOSE[6828][C-0000001d] netsock2.c:   == Using SIP RTP TOS bits 184
[2015-11-18 16:16:37] VERBOSE[6828][C-0000001d] netsock2.c:   == Using SIP RTP CoS mark 5
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] chan_sip.c: Setting NAT on RTP to On
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] chan_sip.c: Setting NAT on VRTP to On
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] rtp_engine.c: Setting payload 18 based on m type on 0x2b73b3df7530
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] rtp_engine.c: Setting payload 8 based on m type on 0x2b73b3df7530
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] rtp_engine.c: Setting payload 0 based on m type on 0x2b73b3df7530
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] rtp_engine.c: Setting payload 101 based on m type on 0x2b73b3df7530
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaef88d8'
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xaef88d8'
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaed76f8'
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] chan_sip.c: Checking SIP call limits for device 099060
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [099060@from-trunk-sip-dialog:1] Set("SIP/dialog-00000024", "GROUP()=OUT_2") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Goto'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [099060@from-trunk-sip-dialog:2] Goto("SIP/dialog-00000024", "from-trunk,099060,1") in new stack
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Goto (from-trunk,099060,1)
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'NoOp'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [099060@from-trunk:1] NoOp("SIP/dialog-00000024", "Catch-All DID Match - Found 099060 - You probably want a DID for this.") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [099060@from-trunk:2] Set("SIP/dialog-00000024", "__FROM_DID=099060") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Goto'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [099060@from-trunk:3] Goto("SIP/dialog-00000024", "ext-did,s,1") in new stack
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Goto (ext-did,s,1)
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Expression result is '0'
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'ExecIf'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ext-did:1] ExecIf("SIP/dialog-00000024", "0?Set(__FROM_DID=s)") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Gosub'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ext-did:2] Gosub("SIP/dialog-00000024", "app-blacklist-check,s,1()") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] app_stack.c: Channel SIP/dialog-00000024 has no datastore, so we're allocating one.
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] db.c: Unable to find key '555146' in family 'blacklist'
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] db.c: Unable to find key 'Baltijskaya, 77, Sultanova Elena Vladimirovna' in family 'blacklist'
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Expression result is '0'
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'GotoIf'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@app-blacklist-check:1] GotoIf("SIP/dialog-00000024", "0?blacklisted") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Not taking any branch
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@app-blacklist-check:2] Set("SIP/dialog-00000024", "CALLED_BLACKLIST=1") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Return'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@app-blacklist-check:3] Return("SIP/dialog-00000024", "") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ext-did:3] Set("SIP/dialog-00000024", "CDR(did)=099060") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Expression result is '0'
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'ExecIf'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ext-did:4] ExecIf("SIP/dialog-00000024", "0 ?Set(CALLERID(name)=555146)") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ext-did:5] Set("SIP/dialog-00000024", "CHANNEL(musicclass)=default") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ext-did:6] Set("SIP/dialog-00000024", "__MOHCLASS=default") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ext-did:7] Set("SIP/dialog-00000024", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ext-did:8] Set("SIP/dialog-00000024", "CALLERPRES()=allowed_not_screened") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Goto'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ext-did:9] Goto("SIP/dialog-00000024", "ivr-4,s,1") in new stack
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Goto (ivr-4,s,1)
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:1] Set("SIP/dialog-00000024", "TIMEOUT_LOOPCOUNT=0") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:2] Set("SIP/dialog-00000024", "INVALID_LOOPCOUNT=0") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:3] Set("SIP/dialog-00000024", "_IVR_CONTEXT_ivr-4=") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:4] Set("SIP/dialog-00000024", "_IVR_CONTEXT=ivr-4") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:5] Set("SIP/dialog-00000024", "__IVR_RETVM=") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Expression result is '0'
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'GotoIf'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:6] GotoIf("SIP/dialog-00000024", "0?skip") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Not taking any branch
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] pbx.c: Launching 'Answer'
[2015-11-18 16:16:37] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:7] Answer("SIP/dialog-00000024", "") in new stack
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] chan_sip.c: SIP answering channel: SIP/dialog-00000024
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] chan_sip.c: Setting framing from config on incoming call
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] chan_sip.c: ** Our capability: (ulaw|alaw|g729) Video flag: True Text flag: True
[2015-11-18 16:16:37] DEBUG[8498][C-0000001d] chan_sip.c: ** Our prefcodec: (nothing) 
[2015-11-18 16:16:37] DEBUG[6828][C-0000001d] chan_sip.c: Stopping retransmission on '200107008bb11310a8c17845c4f02832@siprise-brn' of Response 1: Match Found
[2015-11-18 16:16:38] DEBUG[8498][C-0000001d] pbx.c: Launching 'Wait'
[2015-11-18 16:16:38] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:8] Wait("SIP/dialog-00000024", "1") in new stack
[2015-11-18 16:16:38] DEBUG[8498][C-0000001d] channel.c: Set channel SIP/dialog-00000024 to write format slin
[2015-11-18 16:16:38] DEBUG[8498][C-0000001d] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2015-11-18 16:16:38] DEBUG[8498][C-0000001d] channel.c: Started silence generator on 'SIP/dialog-00000024'
[2015-11-18 16:16:38] DEBUG[8498][C-0000001d] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
[2015-11-18 16:16:38] DEBUG[8498][C-0000001d] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
[2015-11-18 16:16:38] DEBUG[8498][C-0000001d] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xaef88d8'
[2015-11-18 16:16:39] DEBUG[8498][C-0000001d] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2015-11-18 16:16:39] DEBUG[8498][C-0000001d] channel.c: Stopped silence generator on 'SIP/dialog-00000024'
[2015-11-18 16:16:39] DEBUG[8498][C-0000001d] channel.c: Set channel SIP/dialog-00000024 to write format ulaw
[2015-11-18 16:16:39] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:39] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:9] Set("SIP/dialog-00000024", "IVR_MSG=custom/testIVRaudio") in new stack
[2015-11-18 16:16:39] DEBUG[8498][C-0000001d] pbx.c: Launching 'Set'
[2015-11-18 16:16:39] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:10] Set("SIP/dialog-00000024", "TIMEOUT(digit)=3") in new stack
[2015-11-18 16:16:39] VERBOSE[8498][C-0000001d] func_timeout.c:     -- Digit timeout set to 3.000
[2015-11-18 16:16:39] DEBUG[8498][C-0000001d] pbx.c: Expression result is '1'
[2015-11-18 16:16:39] DEBUG[8498][C-0000001d] pbx.c: Launching 'ExecIf'
[2015-11-18 16:16:39] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:11] ExecIf("SIP/dialog-00000024", "1?Background(custom/testIVRaudio)") in new stack
[2015-11-18 16:16:39] DEBUG[8498][C-0000001d] channel.c: Set channel SIP/dialog-00000024 to write format slin
[2015-11-18 16:16:39] DEBUG[8498][C-0000001d] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2015-11-18 16:16:39] VERBOSE[8498][C-0000001d] file.c:     -- <SIP/dialog-00000024> Playing 'custom/testIVRaudio.slin' (language 'en')
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Allocating new SIP dialog for 1ba8320b430cabe220b9e34d08938638@192.168.0.70 - REGISTER (No RTP)
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Target address 91.227.140.72:5060 is not local, substituting externaddr
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Scheduled a registration timeout for siprise.dialog-it.ru id  #3125 
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Initializing initreq for method REGISTER - callid 1ba8320b430cabe220b9e34d08938638@192.168.0.70
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Stopping retransmission on '1ba8320b430cabe220b9e34d08938638@192.168.0.70' of Request 222: Match Found
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Registration successful
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Cancelling timeout 3125
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Allocating new SIP dialog for 503cbbf663d95c883b017d520f550a14@192.168.0.70:5060 - OPTIONS (No RTP)
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: SIP call-id changed from '503cbbf663d95c883b017d520f550a14@192.168.0.70:5060' to '01cafd404963f08277104ea82ae77b21@192.168.0.70:5060'
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Initializing initreq for method OPTIONS - callid 01cafd404963f08277104ea82ae77b21@192.168.0.70:5060
[2015-11-18 16:16:40] DEBUG[6828] chan_sip.c: Stopping retransmission on '01cafd404963f08277104ea82ae77b21@192.168.0.70:5060' of Request 102: Match Found
[2015-11-18 16:16:49] DEBUG[8498][C-0000001d] channel.c: Scheduling timer at (138 requested / 138 actual) timer ticks per second
[2015-11-18 16:16:49] DEBUG[8498][C-0000001d] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2015-11-18 16:16:49] DEBUG[8498][C-0000001d] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2015-11-18 16:16:49] DEBUG[8498][C-0000001d] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2015-11-18 16:16:49] DEBUG[8498][C-0000001d] channel.c: Set channel SIP/dialog-00000024 to write format ulaw
[2015-11-18 16:16:49] DEBUG[8498][C-0000001d] pbx.c: Launching 'WaitExten'
[2015-11-18 16:16:49] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:12] WaitExten("SIP/dialog-00000024", "30,") in new stack
[2015-11-18 16:16:53] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaef88d8'
[2015-11-18 16:16:53] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaed76f8'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] pbx.c: Spawn extension (ivr-4,s,12) exited non-zero on 'SIP/dialog-00000024'
[2015-11-18 16:16:53] VERBOSE[8498][C-0000001d] pbx.c:   == Spawn extension (ivr-4, s, 12) exited non-zero on 'SIP/dialog-00000024'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] channel.c: Soft-Hanging up channel 'SIP/dialog-00000024'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] channel.c: Soft-Hanging up channel 'SIP/dialog-00000024'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] pbx.c: Launching 'Hangup'
[2015-11-18 16:16:53] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [h@ivr-4:1] Hangup("SIP/dialog-00000024", "") in new stack
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] channel.c: Soft-Hanging up channel 'SIP/dialog-00000024'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] pbx.c: Spawn extension (ivr-4,h,1) exited non-zero on 'SIP/dialog-00000024'
[2015-11-18 16:16:53] VERBOSE[8498][C-0000001d] pbx.c:   == Spawn extension (ivr-4, h, 1) exited non-zero on 'SIP/dialog-00000024'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] channel.c: Hanging up channel 'SIP/dialog-00000024'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] chan_sip.c: Hangup call SIP/dialog-00000024, SIP callid 200107008bb11310a8c17845c4f02832@siprise-brn
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaef88d8'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaed76f8'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] cdr_mysql.c: Inserting a CDR record.
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] cdr_mysql.c: SQL command as follows: INSERT INTO cdr (`calldate`,`clid`,`src`,`dst`,`dcontext`,`channel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`uniqueid`) VALUES ('2015-11-18 16:16:37','\"Baltijskaya, 77, Sultanova Elena Vladimirovna\" <555146>','555146','s','ivr-4','SIP/dialog-00000024','WaitExten','30,','16','16','ANSWERED','3','1447841797.37')
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test,src,dst) VALUES ('2015-11-18 16:16:37','"Baltijskaya, 77, Sultanova Elena Vladimirovna" <555146>','ivr-4','SIP/dialog-00000024','','WaitExten','30,','16','16','ANSWERED','DOCUMENTATION','','1447841797.37','','','555146','s')

вроде никакой не слил конфиденциальной инфы.
ps, уточните, пожалуйста, что я делаю не так?

через sip show settings показывает
relax DTMF No
конфиги находятся такие:

не знаю в какой файл надо это написать чтобы включить, этот параметр фигурирует в темах с такой проблемой.

Поиграйтесь с параметром dtmfmode=rfc2833 - там разные варианты.

Попробовал все варианты, после смены конфига делал:
core reload - перечитать все конфиги
core restart now - перезагрузить Asterisk немедленно
абсолютно ничего не изменилось, но зато увидел такие строки с ошибкой

как мне кажется эта функция наверное расположена в скрипте который работает уже после DTMF, т.е. это плохо, но проблема не в этом.

Судя по логу DTMF там ловится, я увидел - был нажат 0. Задействовали ли Вы 0 в схеме нажатий на вэб интерфейсе?

В следующем логе видно, что ждёт нажатий, но самих тонов нету, уже выключили дебаг DTMF что ли?

Код: Выделить всё

[2015-11-18 16:16:49] DEBUG[8498][C-0000001d] channel.c: Set channel SIP/dialog-00000024 to write format ulaw
[2015-11-18 16:16:49] DEBUG[8498][C-0000001d] pbx.c: Launching 'WaitExten'
[2015-11-18 16:16:49] VERBOSE[8498][C-0000001d] pbx.c:     -- Executing [s@ivr-4:12] WaitExten("SIP/dialog-00000024", "30,") in new stack
[2015-11-18 16:16:53] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaef88d8'
[2015-11-18 16:16:53] DEBUG[6828][C-0000001d] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaed76f8'
[2015-11-18 16:16:53] DEBUG[8498][C-0000001d] pbx.c: Spawn extension (ivr-4,s,12) exited non-zero on 'SIP/dialog-00000024'
[2015-11-18 16:16:53] VERBOSE[8498][C-0000001d] pbx.c:   == Spawn extension (ivr-4, s, 12) exited non-zero on 'SIP/dialog-00000024'

не дождался и hangup.

да, 0 задействован. Да когда звониил пробовал нажимать и 0 и 1 и другие цифры.
DTMF дебаг не отключал, но я почему то правда теперь его не вижу там где раньше видел - при внутренних звонках.

asterisk.ru

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