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

[Carlos Alperin]

I downloaded this two days ago from digium ftp.

It reports to be 1.4.0-beta2.

I also have the g726nonstandard=yes on the sip.conf file.

Now, do I need to modify or not the rtp.c that is on /main directory?

I already checked, I don't have audio on g726, with both ports on g729, and
with both ports on ilbc.

I have audio with one port in g729 and the other on ulaw, or both on ulaw.

I don't see any rtp traffic on the previous cases. 

Thanks,

Carlos Alperin

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, December 06, 2006 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FW: G.726 on Asterisk 1.4.0

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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