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

[Mark Murawski]

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
>>>
>>> --