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

[Olivier CALVANO]

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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>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
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