[asterisk-users] sip codec convertion on the fly

mike mike at thundersystems.it
Wed Jul 5 18:49:34 MST 2006


really is that easy ?
i thought this will refuse any codec that isn't gsm
instead, it forces the convertion ... amazing !
thank you very much for your help !

regards
.mike


On Wed, 2006-07-05 at 14:58 -0400, Raymond McKay wrote:
> > sip phone (ulaw) -> asterisk -> internet <- asterisk <- sip phone (ulaw)
> >
> > it is possible to force the two asterisk to convert the codec from ulaw
> > to, say, gsm ?
> > i mean, without touching the two sip phones
> 
> 
> Of course.
> 
> On the trunk between the two Asterisk servers, just add
> 
> disallow=all
> allow=gsm
> 
> to the trunk config.  From that point on, only GSM will transverse the trunk 
> and the two asterisk boxes will transcode.  Remember though, transcoding 
> takes processor power so if you have more than one phone on each end, you 
> are going to eat up processor power quick.
> 
> Raymond McKay
> President
> RAYNET Technologies LLC
> http://www.raynettech.com
> (860) 693-2226 x 31
> Toll Free (877) 693-2226 
> 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list