Начало кусок плана:
[incoming]
exten=> 79221882778,1,Wait(1)
exten=> 79221882778,n,PlayBack(/home/music/output)
exten=> 79221882778,n,Dial(SIP/5001,10)
exten=> 79221882778,n,Voicemail(5001@default,u)
exten=> 79221882778,n,Hangup()
Конец кусок плана:
Без проблем проходит соединение с Голосовым ящиком. Но когда пытаются оставить сообщение появляется следующая ошибка:
[Jan 29 12:35:05] WARNING[13719]: app.c:855 __ast_play_and_record: No audio available on SIP/multifon-0000001b??
Если соединиться с внутреннего экстеншна и надиктовать, то такой ошибки не происходит. В чем может быть проблема?
Если поднять трубку не дождавшись голосовой почты, то абонента слышно.
Да конечно прописан, я и могу оставить сообщение когда звоню с другого внутреннего экстеншна, а вот когда звонок проходит из вне а точнее от SIP мегафона, появляется следующая ошибка. При этом уведомление на e-mail приходит, но wav файл пустой.
Тоже самое, только сообщение изменилось.
-- Executing [79221882778@incoming:1] Wait("SIP/multifon-00000027", "15") in new stack
-- Executing [79221882778@incoming:2] Playback("SIP/multifon-00000027", "/home/music/output") in new stack
-- <SIP/multifon-00000027> Playing '/home/music/output.ulaw' (language 'en')
-- Executing [79221882778@incoming:3] Dial("SIP/multifon-00000027", "SIP/5001,10") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/5001
-- SIP/5001-00000028 is ringing
-- Nobody picked up in 10000 ms
-- Executing [79221882778@incoming:4] VoiceMail("SIP/multifon-00000027", "5001@default") in new stack
-- <SIP/multifon-00000027> Playing 'vm-intro.slin' (language 'en')
-- <SIP/multifon-00000027> Playing 'beep.slin' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/5001/tmp/M6ZXNl format: wav49, 0x7f630401a7d8
-- x=1, open writing: /var/spool/asterisk/voicemail/default/5001/tmp/M6ZXNl format: gsm, 0x7f6304033a48
-- x=2, open writing: /var/spool/asterisk/voicemail/default/5001/tmp/M6ZXNl format: wav, 0x7f6304012de8
[Jan 29 14:18:15] WARNING[13812]: app.c:855 __ast_play_and_record: No audio available on SIP/multifon-00000027??
-- User hung up
== Spawn extension (incoming, 79221882778, 4) exited non-zero on 'SIP/multifon-00000027'
Та же самая ошибка [Jan 29 14:18:15] WARNING[13812]: app.c:855 __ast_play_and_record: No audio available on SIP/multifon-00000027??
То же самое поведение, уведомление на e-mail пришло:
Dear ivan:
Just wanted to let you know you were just left a 0:00 long message (number 10)
in mailbox 5001 from 73433734397, on Tuesday, January 29, 2013 at 02:18:15 PM so you might
want to check it when you get a chance. Thanks!
--Asterisk
-------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
Wav в приложении 60Кб пустой.
Может попробовать с кодеками поиграть? Сейчас разрешены alaw ulaw gsm
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [79221882778@incoming:1] Wait("SIP/multifon-0000002c", "15") in new stack
-- Executing [79221882778@incoming:2] Playback("SIP/multifon-0000002c", "/home/music/output") in new stack
-- <SIP/multifon-0000002c> Playing '/home/music/output.ulaw' (language 'en')
-- Executing [79221882778@incoming:3] Dial("SIP/multifon-0000002c", "SIP/5001,10") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/5001
-- SIP/5001-0000002d is ringing
-- Got SIP response 486 "Busy Here" back from 192.168.0.41:5061
-- SIP/5001-0000002d is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [79221882778@incoming:4] VoiceMail("SIP/multifon-0000002c", "5001@default,u") in new stack
-- <SIP/multifon-0000002c> Playing 'vm-theperson.slin' (language 'en')
-- <SIP/multifon-0000002c> Playing 'digits/5.slin' (language 'en')
-- <SIP/multifon-0000002c> Playing 'digits/0.slin' (language 'en')
-- <SIP/multifon-0000002c> Playing 'digits/0.slin' (language 'en')
-- <SIP/multifon-0000002c> Playing 'digits/1.slin' (language 'en')
-- <SIP/multifon-0000002c> Playing 'vm-isunavail.slin' (language 'en')
-- <SIP/multifon-0000002c> Playing 'vm-intro.slin' (language 'en')
-- <SIP/multifon-0000002c> Playing 'beep.slin' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/5001/tmp/VjxJMp format: wav, 0x7f6304012de8
[Jan 29 14:54:07] WARNING[13845]: app.c:855 __ast_play_and_record: No audio available on SIP/multifon-0000002c??
-- User hung up
== Parsing '/var/spool/asterisk/voicemail/default/5001/INBOX/msg0011.txt': == Found
== Spawn extension (incoming, 79221882778, 4) exited non-zero on 'SIP/multifon-0000002c'
*CLI> sip set debug peer multifon
при переходе на VoiceMail asterisk должен отправить SIP 200 Ok
если не отправляет попробовать поставить Answer перед VoiceMail
если не поможет собрать из сорцов current
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.229:5060;received=18.18.22.14;branch=z9hG4bK1aedad40;rport=5060
From: sip:79221882778-dcfceti8137e4@10.190.35.4:5060;tag=as3371061c
To: <sip:73433734397@10.190.35.17>;tag=95ffcd055e0f78f7d5d397020e89288d4ec819d1
Call-ID: 0203205590814000000149AB@SFESIP2-id1-ext
CSeq: 102 BYE
Content-Length: 0
Что то ответило 200, после ошибки.
При Answer тоже самое.
== Using SIP RTP CoS mark 5
-- Executing [79221882778@incoming:1] Answer("SIP/multifon-00000000", "") in new stack
-- Executing [79221882778@incoming:2] Playback("SIP/multifon-00000000", "/home/music/output") in new stack
-- <SIP/multifon-00000000> Playing '/home/music/output.ulaw' (language 'en')
-- Executing [79221882778@incoming:3] Dial("SIP/multifon-00000000", "SIP/5001,10") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/5001
-- SIP/5001-00000001 is ringing
-- Got SIP response 486 "Busy Here" back from 192.168.0.41:5061
-- SIP/5001-00000001 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [79221882778@incoming:4] VoiceMail("SIP/multifon-00000000", "5001@default,u") in new stack
-- <SIP/multifon-00000000> Playing 'vm-theperson.slin' (language 'en')
-- <SIP/multifon-00000000> Playing 'digits/5.slin' (language 'en')
-- <SIP/multifon-00000000> Playing 'digits/0.slin' (language 'en')
-- <SIP/multifon-00000000> Playing 'digits/0.slin' (language 'en')
-- <SIP/multifon-00000000> Playing 'digits/1.slin' (language 'en')
-- <SIP/multifon-00000000> Playing 'vm-isunavail.slin' (language 'en')
-- <SIP/multifon-00000000> Playing 'vm-intro.slin' (language 'en')
-- <SIP/multifon-00000000> Playing 'beep.slin' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/5001/tmp/vrFVR6 format: wav, 0x7fc7d80188f8
[Jan 29 15:17:09] WARNING[1445]: app.c:855 __ast_play_and_record: No audio available on SIP/multifon-00000000??
-- User hung up
== Parsing '/var/spool/asterisk/voicemail/default/5001/INBOX/msg0015.txt': == Found
== Spawn extension (incoming, 79221882778, 4) exited non-zero on 'SIP/multifon-00000000'