[asterisk-users] Asterisk 1.6 => No sound/voice when i redirect the call

[Olivier CALVANO]

Hi

very thanks, that's work

bye
olivier

2011/4/3 Mark Murawski <markm-lists at intellasoft.net>:
> I gave you the syntax in ael format, if you want to use extensions.conf
> you'll have to use the syntax that's applicable, which is:
>
> [start-audio]
> exten => s,1,Playback(silence/1)
>
>
> On 04/03/11 14:14, Olivier CALVANO wrote:
>>
>> Hi Mark
>>
>> Thanks for your answer, but i am new in asterisk ;=) the "context
>> start-audio ..."
>> i put it into the extension.conf ?
>>
>> because i have a error:
>>
>> [Apr  3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
>> '=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
>> [Apr  3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
>> '=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
>> [Apr  3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
>> ==!!== Unknown directive: s at line 135 -- IGNORING!!!
>>
>> thanks for your help
>>
>> olivier
>>
>>
>>
>>
>> 2011/4/3 Mark Murawski<markm-lists at intellasoft.net>:
>>>
>>> In that situation, I've had to do a pickup macro that kind of "primes"
>>> the
>>> audio.
>>>
>>> Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
>>>
>>> context start-audio {
>>>  s =>  {
>>>    Playback(silence/1);
>>>  }
>>> }
>>>
>>> The above might help... What it does is plays an audio track on the
>>> callee's
>>> channel (SIP/MyOperator-xxxx) before bridging the audio.
>>>
>>>
>>> On 04/03/11 12:01, Olivier CALVANO wrote:
>>>>
>>>> Hi
>>>>
>>>> i use this into my extension :
>>>>
>>>>
>>>>         exten =>    _00339xxxxxxxx,1,Set(foo=${SIP_HEADER(To)})
>>>>         exten =>    _00339xxxxxxxx,2,Set(cut1=${CUT(foo,:,2)})
>>>>         exten =>    _00339xxxxxxxx,3,Set(CLI=${CUT(cut1,>,1)})
>>>>         exten =>    _00339xxxxxxxx,4,Set(toexten=${CUT(CLI,@,1)})
>>>>         exten =>    _00339xxxxxxxx,5,Noop(ORIGINAL NUMBER : [ ${toexten}
>>>> ])
>>>>         exten =>    _00339xxxxxxxx,6,AGI(Ddi-Network.agi,${toexten})
>>>>         exten =>
>>>>  _00339xxxxxxxx,7,Set(CALLERPRES()=prohib_not_screened)
>>>>         exten =>
>>>>  _00339xxxxxxxx,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
>>>>         exten =>    _00339xxxxxxxx,9,Hangup
>>>>
>>>>
>>>> and i have in sip.conf:
>>>>
>>>>
>>>> [MyOperator]
>>>> type=peer
>>>> host=host-of-my-operator
>>>> qualify=yes
>>>> dtmf=rfc2833
>>>> nat=no
>>>> canreinvite=no
>>>> canredirect=yes
>>>> insecure=port,invite
>>>> dtmfmode=rfc2833
>>>> disallow=all
>>>> allow=g729
>>>> allow=alaw
>>>> allow=g723
>>>> defaultuser=0033xxxxxx
>>>> secret=xxxxx
>>>>
>>>>
>>>>
>>>> When i call directly from [MyOperator], no probleme i have sound/Voice
>>>> but when a customer call to the "00339xxx..", the call are correct,
>>>> asterisk
>>>> call to my standard "SIP/MyOperator/${NUMAPPEL}" but no sound/voice
>>>> (i receive the call without problems, only sound off)
>>>>
>>>> anyone have a idea of this problems ?
>>>>
>>>> bye
>>>> Olivier
>>>>
>>>> --
>
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