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

[Ben Schorr]

O.K., thanks, I'm catching on (slowly).  Waiting for the next call to
see if the SIP.CONF change did the trick.

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 Danny Nicholas
> Sent: Tuesday, December 15, 2009 9:15 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Can't get G.729 to work...
> 
> 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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