[asterisk-users] FW: G.726 on Asterisk 1.4.0

Joshua Colp jcolp at digium.com
Wed Dec 6 10:52:46 MST 2006


Carlos Alperin wrote:
>  Ok,
> 
> With everything restore on rtp.c, still I have no audio however the call is
> not destroyed immediately as before.
> 
> I'm going to put a second Granstream box, and findout if between two boxes
> this happen too.
> 
> I cannot believe that we cannot do 2 g726 on the same box at one time.
> 
> Carlos
> 

Make sure you are using the latest 1.4 branch, I already fixed a G726-32 
related bug in there and you must have the g726nonstandard set to yes in 
sip.conf I do believe.

-- 
Joshua Colp
Software Developer
Digium, Inc.


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