[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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