История изменений - Asterisk: Вопросы и Ответы

Помогите разобраться с IVR Asterisk

Есть ivr настроенный на донабор в группы нажмите 1,2,3, сейчас надо добавить возможность донабора внутреннего номера сотрудника, привожу сам ivr ниже:

exten => s,1,NoOp(Start Promt) exten => s,n,Answer() exten => s,n,Background(/usr/share/asterisk/sounds/IVR/privetstvie) exten => s,n(4),Background(/usr/share/asterisk/sounds/IVR/zakazy) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/brak) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/docs) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/transport) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/callrec) exten => s,n(9),WaitExten(5) exten => s,n,Hangup() ;exten => s,n,GoTo(dialivr,0,1) exten => i,1,GoTo(invalidnumber,s,1) exten => 1,1,GoTo(dialivr,1,1) exten => 2,1,GoTo(dialivr,2,1) exten => 3,1,GoTo(dialivr,3,1) exten => 4,1,GoTo(dialivr,4,1) exten => _XXXX,1,Dial(SIP/${EXTEN}) дописал но не работает.

[dial_ivr]

exten => 1,1,NoOp(Call to group phones 5008-5011) exten => 1,n,NoOp(the retry count) exten => 1,n,Set(COUNT1=$[${COUNT1}-1]) exten => 1,n,GoTo(dialivr,1100,1) exten => 1100,1,Macro(recording) exten => 1100,n,Queue(1100,n,tT,r) exten => 1100,n,GotoIf($[${COUNT1} > 0]?:exit) exten => 1100,n,Playback(/usr/share/asterisk/sounds/IVR-GMT/busy) exten => 1100,n,GoTo(dialivr,1,1) exten => 1100,n(exit),Hangup()

и так далее до кнопки 4. Когда набираю четырехзначный номер сотрудника 5036 выдает в cli следующее:

Goto (IVR,s,1) -- Executing [s@IVR:1] NoOp("SIP/SIP-TRUNK-0001f90b", "Start Promt") in new stack -- Executing [s@IVR:2] Answer("SIP/SIP-TRUNK-0001f90b", "") in new stack

0x7f1ba40d4110 -- Probation passed - setting RTP source address to 85.90.97.5:10062 -- Executing [s@IVR:3] BackGround("SIP/SIP-TRUNK-0001f90b", "/usr/share/asterisk/sounds/IVR/privetstvie") in new stack -- <sip sip-trunk-0001f90b=""> Playing '/usr/share/asterisk/sounds/IVR-GMT/privetstvie.ulaw' (language 'en') == CDR updated on SIP/SIP-TRUNK-0001f90b -- Executing [5500@IVR:1] Dial("SIP/SIP-TRUNK-0001f90b", "SIP/5500") in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/SIP-TRUNK-0001f90b' status is 'CHANUNAVAIL' == Spawn extension (CSS-MOBILE, 5004, 3) exited non-zero on 'SIP/SIP-TRUNK-0001f904' == MixMonitor close filestream (mixed)

Подскажите в чем проблема и куда копать, уже всю голову сломал

Помогите разобраться с IVR Asterisk

Есть ivr настроенный на донабор в группы нажмите 1,2,3, сейчас надо добавить возможность донабора внутреннего номера сотрудника, привожу сам ivr ниже:

exten => s,1,NoOp(Start Promt) exten => s,n,Answer() exten => s,n,Background(/usr/share/asterisk/sounds/IVR/privetstvie) exten => s,n(4),Background(/usr/share/asterisk/sounds/IVR/zakazy) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/brak) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/docs) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/transport) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/callrec) exten => s,n(9),WaitExten(5) exten => s,n,Hangup() s,n,Answer()

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/privetstvie)

exten => s,n(4),Background(/usr/share/asterisk/sounds/IVR/zakazy)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/brak)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/docs)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/transport)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/callrec)

exten => s,n(9),WaitExten(5)

exten => s,n,Hangup()

;exten => ~~s,n,GoTo(dialivr,0,1) exten => i,1,GoTo(invalidnumber,s,1) exten => 1,1,GoTo(dialivr,1,1) exten => 2,1,GoTo(dialivr,2,1) exten => 3,1,GoTo(dialivr,3,1) exten => 4,1,GoTo(dialivr,4,1)~~ s,n,GoTo(dial_ivr,0,1)

exten => i,1,GoTo(invalid_number,s,1)

exten => 1,1,GoTo(dial_ivr,1,1)

exten => 2,1,GoTo(dial_ivr,2,1)

exten => 3,1,GoTo(dial_ivr,3,1)

exten => 4,1,GoTo(dial_ivr,4,1)

exten => _XXXX,1,Dial(SIP/${EXTEN}) дописал но не работает.

[dial_ivr]

exten => 1,1,NoOp(Call to group phones ~~5008-5011)~~ 5008-5011)

exten => 1,n,NoOp(the retry ~~count) exten => 1,n,Set(COUNT1=$[${COUNT1}-1]) exten => 1,n,GoTo(dialivr,1100,1) exten => 1100,1,Macro(recording) exten => 1100,n,Queue(1100,n,tT,r)~~ count)

exten => 1,n,Set(COUNT1=$[${COUNT1}-1])

exten => 1,n,GoTo(dial_ivr,1100,1)

exten => 1100,1,Macro(recording)

exten => 1100,n,Queue(1100,n,tT,r)

exten => 1100,n,GotoIf($[${COUNT1} > ~~0]?:exit) exten => 1100,n,Playback(/usr/share/asterisk/sounds/IVR-GMT/busy) exten => 1100,n,GoTo(dial~~ivr,1,1) 0]?:exit)

exten => 1100,n,Playback(/usr/share/asterisk/sounds/IVR-GMT/busy)

exten => 1100,n,GoTo(dial_ivr,1,1)

exten => 1100,n(exit),Hangup()

и так далее до кнопки 4. Когда набираю четырехзначный номер сотрудника 5036 выдает в cli следующее:

Goto (IVR,s,1) -- Executing [s@IVR:1] NoOp("SIP/SIP-TRUNK-0001f90b", "Start Promt") in new ~~stack~~ stack

-- Executing [s@IVR:2] Answer("SIP/SIP-TRUNK-0001f90b", "") in new stack
  stack

   > 0x7f1ba40d4110 -- Probation passed - setting RTP source address to 85.90.97.5:10062
  -- Executing [s@IVR:3] BackGround("SIP/SIP-TRUNK-0001f90b", "/usr/share/asterisk/sounds/IVR/privetstvie") in new stack
      -- <sip sip-trunk-0001f90b=""> 
-- <SIP/SIP-TRUNK-0001f90b> Playing '/usr/share/asterisk/sounds/IVR-GMT/privetstvie.ulaw' (language 'en')

== CDR updated on SIP/SIP-TRUNK-0001f90b -- Executing [5500@IVR:1] Dial("SIP/SIP-TRUNK-0001f90b", "SIP/5500") in new stack == Everyone is busy/congested at this time ~~(1:0/0/1)~~ (1:0/0/1)

-- Auto fallthrough, channel 'SIP/SIP-TRUNK-0001f90b' status is 'CHANUNAVAIL'

received on SIP/SIP-TRUNK-0001f90b

== Spawn extension (CSS-MOBILE, 5004, 3) exited non-zero on ~~'SIP/SIP-TRUNK-0001f904'~~ 'SIP/SIP-TRUNK-0001f904'

== MixMonitor close filestream (mixed)

Подскажите в чем проблема и куда копать, уже всю голову сломал

Помогите разобраться с IVR Asterisk

Есть ivr настроенный на донабор в группы нажмите 1,2,3, сейчас надо добавить возможность донабора внутреннего номера сотрудника, привожу сам ivr ниже:

exten => s,1,NoOp(Start Promt) exten => s,n,Answer()

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/privetstvie)

exten => s,n(4),Background(/usr/share/asterisk/sounds/IVR/zakazy)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/brak)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/docs)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/transport)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/callrec)

exten => s,n(9),WaitExten(5)

exten => s,n,Hangup()

;exten => s,n,GoTo(dial_ivr,0,1)

exten => i,1,GoTo(invalid_number,s,1)

exten => 1,1,GoTo(dial_ivr,1,1)

exten => 2,1,GoTo(dial_ivr,2,1)

exten => 3,1,GoTo(dial_ivr,3,1)

exten => 4,1,GoTo(dial_ivr,4,1)

exten => _XXXX,1,Dial(SIP/${EXTEN}) дописал но не работает.

[dial_ivr]

exten => 1,1,NoOp(Call to group phones 5008-5011)

exten => 1,n,NoOp(the retry count)

exten => 1,n,Set(COUNT1=$[${COUNT1}-1])

exten => 1,n,GoTo(dial_ivr,1100,1)

exten => 1100,1,Macro(recording)

exten => 1100,n,Queue(1100,n,tT,r)

exten => 1100,n,GotoIf($[${COUNT1} > 0]?:exit)

exten => ~~1100,n,Playback(/usr/share/asterisk/sounds/IVR-GMT/busy)~~1100,n,Playback(/usr/share/asterisk/sounds/IVR/busy)

exten => 1100,n,GoTo(dial_ivr,1,1)

exten => 1100,n(exit),Hangup()

и так далее до кнопки 4. Когда набираю четырехзначный номер сотрудника 5036 выдает в cli следующее:

Goto (IVR,s,1) -- Executing [s@IVR:1] NoOp("SIP/SIP-TRUNK-0001f90b", "Start Promt") in new stack

-- Executing [s@IVR:2] Answer("SIP/SIP-TRUNK-0001f90b", "") in new stack

   > 0x7f1ba40d4110 -- Probation passed - setting RTP source address to 85.90.97.5:10062

-- Executing [s@IVR:3] BackGround("SIP/SIP-TRUNK-0001f90b", "/usr/share/asterisk/sounds/IVR/privetstvie") in new stack

-- <SIP/SIP-TRUNK-0001f90b> Playing '/usr/share/asterisk/sounds/IVR-GMT/privetstvie.ulaw' '/usr/share/asterisk/sounds/IVR/privetstvie.ulaw' (language 'en')

== CDR updated on SIP/SIP-TRUNK-0001f90b -- Executing [5500@IVR:1] Dial("SIP/SIP-TRUNK-0001f90b", "SIP/5500") in new stack == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/SIP-TRUNK-0001f90b' status is 'CHANUNAVAIL'

received on SIP/SIP-TRUNK-0001f90b

== Spawn extension (CSS-MOBILE, 5004, 3) exited non-zero on 'SIP/SIP-TRUNK-0001f904'

== MixMonitor close filestream (mixed)

Подскажите в чем проблема и куда копать, уже всю голову сломал

Помогите разобраться с IVR Asterisk

Есть ivr настроенный на донабор в группы нажмите 1,2,3, сейчас надо добавить возможность донабора внутреннего номера сотрудника, привожу сам ivr ниже:

exten => s,1,NoOp(Start Promt) exten => s,n,Answer()

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/privetstvie)

exten => s,n(4),Background(/usr/share/asterisk/sounds/IVR/zakazy)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/brak)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/docs)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/transport)

exten => s,n,Background(/usr/share/asterisk/sounds/IVR/callrec)

exten => s,n(9),WaitExten(5)

exten => s,n,Hangup()

;exten => s,n,GoTo(dial_ivr,0,1)

exten => i,1,GoTo(invalid_number,s,1)

exten => 1,1,GoTo(dial_ivr,1,1)

exten => 2,1,GoTo(dial_ivr,2,1)

exten => 3,1,GoTo(dial_ivr,3,1)

exten => 4,1,GoTo(dial_ivr,4,1)

exten => _XXXX,1,Dial(SIP/${EXTEN}) дописал но не работает.

[dial_ivr]

exten => 1,1,NoOp(Call to group phones 5008-5011)

exten => 1,n,NoOp(the retry count)

exten => 1,n,Set(COUNT1=$[${COUNT1}-1])

exten => 1,n,GoTo(dial_ivr,1100,1)

exten => 1100,1,Macro(recording)

exten => 1100,n,Queue(1100,n,tT,r)

exten => 1100,n,GotoIf($[${COUNT1} > 0]?:exit)

exten => 1100,n,Playback(/usr/share/asterisk/sounds/IVR/busy)

exten => 1100,n,GoTo(dial_ivr,1,1)

exten => 1100,n(exit),Hangup()

и так далее до кнопки 4. Когда набираю четырехзначный номер сотрудника 5036 выдает в cli следующее:

Goto (IVR,s,1) -- Executing [s@IVR:1] NoOp("SIP/SIP-TRUNK-0001f90b", "Start Promt") in new stack

--  Executing [s@IVR:2] Answer("SIP/SIP-TRUNK-0001f90b", "") in new stack

   > 0x7f1ba40d4110 -- Probation passed - setting RTP source address to 85.90.97.5:10062

-- Executing [s@IVR:3] BackGround("SIP/SIP-TRUNK-0001f90b", "/usr/share/asterisk/sounds/IVR/privetstvie") in new stack

-- <SIP/SIP-TRUNK-0001f90b> Playing '/usr/share/asterisk/sounds/IVR/privetstvie.ulaw' (language 'en')

== CDR updated on SIP/SIP-TRUNK-0001f90b -- Executing [5500@IVR:1] Dial("SIP/SIP-TRUNK-0001f90b", "SIP/5500") in new stack == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/SIP-TRUNK-0001f90b' status is 'CHANUNAVAIL'

received on SIP/SIP-TRUNK-0001f90b

== Spawn extension (CSS-MOBILE, 5004, 3) exited non-zero on 'SIP/SIP-TRUNK-0001f904'

== MixMonitor close filestream (mixed)

Подскажите в чем проблема и куда копать, уже всю голову сломал

Помогите разобраться с IVR Asterisk

Есть ivr настроенный на донабор в группы нажмите 1,2,3, сейчас надо добавить возможность донабора внутреннего номера сотрудника, привожу сам ivr ниже:

exten => s,1,NoOp(Start Promt) exten => s,n,Answer() exten => s,n,Background(/usr/share/asterisk/sounds/IVR/privetstvie) exten => s,n(4),Background(/usr/share/asterisk/sounds/IVR/zakazy) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/brak) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/docs) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/transport) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/callrec) exten => s,n(9),WaitExten(5) exten => s,n,Hangup() s,n,Answer() exten => s,n,Background(/usr/share/asterisk/sounds/IVR/privetstvie) exten => s,n(4),Background(/usr/share/asterisk/sounds/IVR/zakazy) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/brak) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/docs) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/transport) exten => s,n,Background(/usr/share/asterisk/sounds/IVR/callrec) exten => s,n(9),WaitExten(5) exten => s,n,Hangup() ;exten => s,n,GoTo(dial_ivr,0,1) exten => i,1,GoTo(invalid_number,s,1) exten => 1,1,GoTo(dial_ivr,1,1) exten => 2,1,GoTo(dial_ivr,2,1) exten => 3,1,GoTo(dial_ivr,3,1) exten => 4,1,GoTo(dial_ivr,4,1) s,n,GoTo(dial_ivr,0,1) exten => i,1,GoTo(invalid_number,s,1) exten => 1,1,GoTo(dial_ivr,1,1) exten => 2,1,GoTo(dial_ivr,2,1) exten => 3,1,GoTo(dial_ivr,3,1) exten => 4,1,GoTo(dial_ivr,4,1) exten => _XXXX,1,Dial(SIP/${EXTEN}) дописал но не работает. [dial_ivr] работает. [dial_ivr] exten => 1,1,NoOp(Call to group phones 5008-5011) 5008-5011) exten => 1,n,NoOp(the retry count) exten => 1,n,Set(COUNT1=$[${COUNT1}-1]) exten => 1,n,GoTo(dial_ivr,1100,1) exten => 1100,1,Macro(recording) exten => 1100,n,Queue(1100,n,tT,r) count) exten => 1,n,Set(COUNT1=$[${COUNT1}-1]) exten => 1,n,GoTo(dial_ivr,1100,1) exten => 1100,1,Macro(recording) exten => 1100,n,Queue(1100,n,tT,r) exten => 1100,n,GotoIf($[${COUNT1} > 0]?:exit) exten => 1100,n,Playback(/usr/share/asterisk/sounds/IVR/busy) exten => 1100,n,GoTo(dial_ivr,1,1) exten => 1100,n(exit),Hangup()0]?:exit) exten => 1100,n,Playback(/usr/share/asterisk/sounds/IVR/busy) exten => 1100,n,GoTo(dial_ivr,1,1) exten => 1100,n(exit),Hangup()

и так далее до кнопки 4. Когда набираю четырехзначный номер сотрудника 5036 выдает в cli следующее:

Goto (IVR,s,1) -- Executing [s@IVR:1] NoOp("SIP/SIP-TRUNK-0001f90b", "Start Promt") in new stack stack Executing [s@IVR:2] Answer("SIP/SIP-TRUNK-0001f90b", "") in new stack > 0x7f1ba40d4110 -- Probation passed - setting RTP source address to 85.90.97.5:10062 Executing [s@IVR:3] BackGround("SIP/SIP-TRUNK-0001f90b", "/usr/share/asterisk/sounds/IVR/privetstvie") in new stack -- <SIP/SIP-TRUNK-0001f90b> Playing '/usr/share/asterisk/sounds/IVR/privetstvie.ulaw' (language 'en') [Aug 17 08:37:04] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '5' received on SIP/SIP-TRUNK-0001f90b, duration 250 ms [Aug 17 08:37:04] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '5' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:04] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '5' received on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:04] DTMF[4436][C-0000c5be]: channel.c:4174 __ast_read: DTMF begin ignored '5' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '5' received on SIP/SIP-TRUNK-0001f90b, duration 459 ms [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '5' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '0' received on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4174 __ast_read: DTMF begin ignored '0' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '0' received on SIP/SIP-TRUNK-0001f90b, duration 250 ms [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '0' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '0' received on SIP/SIP-TRUNK-0001f90b, duration 25 ms [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '0' on SIP/SIP-TRUNK-0001f90b CDR updated on SIP/SIP-TRUNK-0001f90b -- Executing [5500@IVR:1] Dial("SIP/SIP-TRUNK-0001f90b", "SIP/5500") in new stack [Aug 17 08:37:05] WARNING[4436][C-0000c5be]: chan_sip.c:6160 create_addr: Purely numeric hostname (5500), and not a peer--rejecting! [Aug 17 08:37:05] WARNING[4436][C-0000c5be]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) Everyone is busy/congested at this time (1:0/0/1) (1:0/0/1) Auto fallthrough, channel 'SIP/SIP-TRUNK-0001f90b' status is 'CHANUNAVAIL' received on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '0' received on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4181 __ast_read: DTMF begin passthrough '0' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '0' received on SIP/SIP-TRUNK-0001f90b, duration 114 ms [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4125 __ast_read: DTMF end accepted with begin '0' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '0' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '3' received on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4181 __ast_read: DTMF begin passthrough '3' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '3' received on SIP/SIP-TRUNK-0001f90b, duration 250 ms [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4125 __ast_read: DTMF end accepted with begin '3' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4140 __ast_read: DTMF end '3' detected to have actual duration 45 on the wire, emulation will be triggered on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4147 __ast_read: DTMF end '3' has duration 45 but want minimum 80, emulating on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '3' received on SIP/SIP-TRUNK-0001f90b, duration 25 ms [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4248 __ast_read: DTMF end emulation of '3' queued on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '3' received on SIP/SIP-TRUNK-0001f90b, duration 25 ms [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4111 __ast_read: DTMF begin emulation of '3' with duration 80 queued on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '3' received on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4174 __ast_read: DTMF begin ignored '3' on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4248 __ast_read: DTMF end emulation of '3' queued on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '3' received on SIP/SIP-TRUNK-0001f90b, duration 216 ms [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4111 __ast_read: DTMF begin emulation of '3' with duration 216 queued on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4248 __ast_read: DTMF end emulation of '3' queued on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '6' received on SIP/SIP-TRUNK-0001f90b, duration 250 ms [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4111 __ast_read: DTMF begin emulation of '6' with duration 250 queued on SIP/SIP-TRUNK-0001f90b [Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4248 __ast_read: DTMF end emulation of '6' queued on SIP/SIP-TRUNK-0001f90b == Spawn extension (CSS-MOBILE, 5004, 3) exited non-zero on 'SIP/SIP-TRUNK-0001f904' 'SIP/SIP-TRUNK-0001f904' == MixMonitor close filestream (mixed)(mixed)

Подскажите в чем проблема и куда копать, уже всю голову сломал

Помогите разобраться с IVR Asterisk

Есть ivr настроенный на донабор в группы нажмите 1,2,3, сейчас надо добавить возможность донабора внутреннего номера сотрудника, привожу сам ivr ниже:

exten => s,1,NoOp(Start Promt)
exten => s,n,Answer()
exten => s,n,Background(/usr/share/asterisk/sounds/IVR/privetstvie)
exten => s,n(4),Background(/usr/share/asterisk/sounds/IVR/zakazy)
exten => s,n,Background(/usr/share/asterisk/sounds/IVR/brak)
exten => s,n,Background(/usr/share/asterisk/sounds/IVR/docs)
exten => s,n,Background(/usr/share/asterisk/sounds/IVR/transport)
exten => s,n,Background(/usr/share/asterisk/sounds/IVR/callrec)
exten => s,n(9),WaitExten(5)
exten => s,n,Hangup()
;exten => s,n,GoTo(dial_ivr,0,1)
exten => i,1,GoTo(invalid_number,s,1)
exten => 1,1,GoTo(dial_ivr,1,1)
exten => 2,1,GoTo(dial_ivr,2,1)
exten => 3,1,GoTo(dial_ivr,3,1)
exten => 4,1,GoTo(dial_ivr,4,1)
exten => _XXXX,1,Dial(SIP/${EXTEN}) дописал но не работает.

[dial_ivr]
exten => 1,1,NoOp(Call to group phones 5008-5011)
exten => 1,n,NoOp(the retry count)
exten => 1,n,Set(COUNT1=$[${COUNT1}-1])
exten => 1,n,GoTo(dial_ivr,1100,1)
exten => 1100,1,Macro(recording)
exten => 1100,n,Queue(1100,n,tT,r)
exten => 1100,n,GotoIf($[${COUNT1} > 0]?:exit)
exten => 1100,n,Playback(/usr/share/asterisk/sounds/IVR/busy)
exten => 1100,n,GoTo(dial_ivr,1,1)
exten => 1100,n(exit),Hangup()

и так далее до кнопки 4. Когда набираю четырехзначный номер сотрудника 5036 выдает в cli следующее:

Goto (IVR,s,1)
-- Executing [s@IVR:1] NoOp("SIP/SIP-TRUNK-0001f90b", "Start Promt") in new stack
Executing [s@IVR:2] Answer("SIP/SIP-TRUNK-0001f90b", "") in new stack
> 0x7f1ba40d4110 -- Probation passed - setting RTP source address to 85.90.97.5:10062
Executing [s@IVR:3] BackGround("SIP/SIP-TRUNK-0001f90b", "/usr/share/asterisk/sounds/IVR/privetstvie") in new stack
-- <SIP/SIP-TRUNK-0001f90b> Playing '/usr/share/asterisk/sounds/IVR/privetstvie.ulaw' (language 'en')
[Aug 17 08:37:04] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '5' received on SIP/SIP-TRUNK-0001f90b, duration 250 ms
[Aug 17 08:37:04] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '5' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:04] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '5' received on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:04] DTMF[4436][C-0000c5be]: channel.c:4174 __ast_read: DTMF begin ignored '5' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '5' received on SIP/SIP-TRUNK-0001f90b, duration 459 ms
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '5' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '0' received on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4174 __ast_read: DTMF begin ignored '0' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '0' received on SIP/SIP-TRUNK-0001f90b, duration 250 ms
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '0' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '0' received on SIP/SIP-TRUNK-0001f90b, duration 25 ms
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '0' on SIP/SIP-TRUNK-0001f90b
CDR updated on SIP/SIP-TRUNK-0001f90b
-- Executing [5500@IVR:1] Dial("SIP/SIP-TRUNK-0001f90b", "SIP/5500") in new stack
[Aug 17 08:37:05] WARNING[4436][C-0000c5be]: chan_sip.c:6160 create_addr: Purely numeric hostname (5500), and not a peer--rejecting!
[Aug 17 08:37:05] WARNING[4436][C-0000c5be]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
   Everyone is busy/congested at this time (1:0/0/1)
    Auto fallthrough, channel 'SIP/SIP-TRUNK-0001f90b' status is 'CHANUNAVAIL'
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '0' 
received on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4181 __ast_read: DTMF begin passthrough '0' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '0' received on SIP/SIP-TRUNK-0001f90b, duration 114 ms
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4125 __ast_read: DTMF end accepted with begin '0' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:05] DTMF[4436][C-0000c5be]: channel.c:4154 __ast_read: DTMF end passthrough '0' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '3' received on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4181 __ast_read: DTMF begin passthrough '3' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '3' received on SIP/SIP-TRUNK-0001f90b, duration 250 ms
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4125 __ast_read: DTMF end accepted with begin '3' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4140 __ast_read: DTMF end '3' detected to have actual duration 45 on the wire, emulation will be triggered on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4147 __ast_read: DTMF end '3' has duration 45 but want minimum 80, emulating on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '3' received on SIP/SIP-TRUNK-0001f90b, duration 25 ms
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4248 __ast_read: DTMF end emulation of '3' queued on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '3' received on SIP/SIP-TRUNK-0001f90b, duration 25 ms
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4111 __ast_read: DTMF begin emulation of '3' with duration 80 queued on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4170 __ast_read: DTMF begin '3' received on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4174 __ast_read: DTMF begin ignored '3' on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4248 __ast_read: DTMF end emulation of '3' queued on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '3' received on SIP/SIP-TRUNK-0001f90b, duration 216 ms
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4111 __ast_read: DTMF begin emulation of '3' with duration 216 queued on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4248 __ast_read: DTMF end emulation of '3' queued on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4084 __ast_read: DTMF end '6' received on SIP/SIP-TRUNK-0001f90b, duration 250 ms
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4111 __ast_read: DTMF begin emulation of '6' with duration 250 queued on SIP/SIP-TRUNK-0001f90b
[Aug 17 08:37:06] DTMF[4436][C-0000c5be]: channel.c:4248 __ast_read: DTMF end emulation of '6' queued on SIP/SIP-TRUNK-0001f90b
  == Spawn extension (CSS-MOBILE, 5004, 3) exited non-zero on 'SIP/SIP-TRUNK-0001f904'
  == MixMonitor close filestream (mixed)

Подскажите в чем проблема и куда копать, уже всю голову ~~сломал~~сломал.

Проверил Logger, DTMF debug вроде включен console => notice,warning,error,dtmf . При звонке и донаборе в cli пишет так

<sip sip-trunk-0001fdd2=""> Playing '/usr/share/asterisk/sounds/IVR/brak.ulaw' (language 'en')

-- Timeout on SIP/SIP-TRUNK-0001fdd0, continuing...

-- Executing [s@IVR:10] Hangup("SIP/SIP-TRUNK-0001fdd0", "") in new stack

== Spawn extension (IVR, s, 10) exited non-zero on 'SIP/SIP-TRUNK-0001fdd0'

-- Executing [s@IVR:6] BackGround("SIP/SIP-TRUNK-0001fdd2", "/usr/share/asterisk/sounds/IVR-GMT/docs") in new stack

-- <SIP/SIP-TRUNK-0001fdd2> Playing '/usr/share/asterisk/sounds/IVR/docs.ulaw' (language 'en')

== CDR updated on SIP/SIP-TRUNK-0001fdd1

-- Executing [3333@IVR:1] Dial("SIP/SIP-TRUNK-0001fdd1", "SIP/3333") in new stack

== Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/SIP-TRUNK-0001fdd1' status is 'CHANUNAVAIL'

В настройках транка не нашел указания о использовании DTMF, настройки sip trunk type=peer host=xx.xx.xx.xxx context=INCOMING-SIP-TRUNK nat=no insecure = invite,port nat = no canreinvite = no qualify=yes allow=alaw allow=ulaw

Может стоит прописать DTMF?

История изменений [назад]

Помогите разобраться с IVR Asterisk

Помогите разобраться с IVR Asterisk

Помогите разобраться с IVR Asterisk

Помогите разобраться с IVR Asterisk

Помогите разобраться с IVR Asterisk

Помогите разобраться с IVR Asterisk