[asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem

[Rainer Piper]

that's funny

I recompiled asterisk without bridge_native_rtp.so
to force asterisk to go to simple_bridge and not to native_bridge...

!!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu




Am 07.05.2014 07:11, schrieb Rainer Piper:
> PS.
>
> if I configure both extension 7000 and 7001 to,
>
> disallow=all
> allow=alaw
> or
> disallow=all
> allow=g722
>
> everything is fine. as long as the allowed codec is equal in both 
> extensions.
>
>
>
> Am 07.05.2014 07:00, schrieb Rainer Piper:
>> Hi!
>>
>> my asterisk-12.2.0 with pjsip-2.2.0  does not translate codecs any 
>> more. I tried every combination. silent on both sides.
>>
>> I compiled pjsip with no resample in pjsip.
>> ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample*  --disable-video --disable-opencore-amr
>> is there a way to force asterisk back to do the codec translation?
>>
>> Attachment:
>> sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to 
>> the B-Leg 7000 NativeFormats: (alaw)
>>
>>
>> -- 
>> *Rainer Piper*
>> Integration engineer
>> Koeslinstr. 56
>> 53123 BONN
>> GERMANY
>>
>> ------------------------------------------------------------------------
>> ------------------------------------------------------------------------
>>
>>
>
>
> -- 
> *Rainer Piper*
> Integration engineer
> Koeslinstr. 56
> 53123 BONN
> GERMANY
> www.soho-piper.de <http://www.soho-piper.de>
> ------------------------------------------------------------------------
>
> ------------------------------------------------------------------------
>
>


-- 
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
www.soho-piper.de <http://www.soho-piper.de>
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