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

[Mark Murawski]

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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