[asterisk-users] Can't get G.729 to work...

[Danny Nicholas]

You should only need a reboot for DAHDI changes (not always then...)

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ben Schorr
Sent: Tuesday, December 15, 2009 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...

Ahhh...yes, I think that may have been it.  I moved G.729 to the top of
that list (just below disallow) and now I have a "restart when
convenient" pending.  Is that sufficient or do I have to actually reboot
the server for the change to take effect?

Best wishes and aloha, 

Ben M. Schorr
Chief Executive Officer
______________________________________________
Roland Schorr & Tower
www.rolandschorr.com
bens at rolandschorr.com

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Jeff LaCoursiere
> Sent: Tuesday, December 15, 2009 8:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Can't get G.729 to work...
> 
> 
> On Tue, 15 Dec 2009, Ben Schorr wrote:
> 
> > Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones.
> >
> >
> >
> > I've got G.729 loaded in the modules on the Asterisk server and on
the
> > Polycom phones I've set G.729 to be the first preference of codec,
but
> > still when I go SIP SHOW CHANNELS during active calls it still shows
> > "(ULAW)" (G.711) as the codec in use.
> >
> >
> >
> > I'm a newbie at Asterisk, can anybody suggest what I might be
> > overlooking?
> >
> 
> In the sip.conf entry for your peer you need to specify the codec
negotiation
> order.  Though you put g.729 first on the phone, asterisk probably has
ulaw
> first, and is taking precedence.  In the sip.conf entry put this:
> 
> disallow=all
> allow=g729
> 
> j
> 
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