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

Tue Dec 15 14:31:21 CST 2009

Why not restrict these 8 users to a SIP provider like (but not)
bandwidth.com?  By eliminating the PRI element, you should completely
resolve the problem.

-----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 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...

Oh, dear.  So my users with "less-than-ideal" bandwidth are stuck with
drop-outs and poor sound quality because they can't use the reduced
bandwidth codec for those calls?  :-(

They've been complaining that they often end up on a call where one or
both parties are "cutting in and out".  Unfortunately it's only this one
remote site, with about 8 users, who connect across a VPN to the site
where the server is.  We've tried increasing their bandwidth and
tweaking the QOS settings on their firewalls but so far we haven't been
able to solve it.  I was hoping that switching to a lower bandwidth
CODEC would give them the call reliability they need.

If not then I guess I'm back to the drawing board, with increasingly
impatient users, trying to troubleshoot their call quality issues.

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 10:19 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Can't get G.729 to work...
> 
> IMO you can only use the G.729 on a SIP call.  If the call falls onto
the PRI
> framework, ulaw will be forced.
> 
> -----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 2:11 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Can't get G.729 to work...
> 
> Sorry, I think I may have misspoke...
> 
> What I'm hoping for is that all of the connections between my phones
(or at
> least a particular group of them) and my Asterisk server will use
G.729.
> Currently it seems like it usually is, but not always, and I haven't
figured out
> the pattern.
> 
> All of our calls fall into two categories:
> 
> Internal calls - one extension to another within our single Asterisk
server org.
> External calls - To/From one of our extensions out thru the PRI line
to our
> carrier (Hawaiian Tel) to phone numbers out in the world.
> 
> For some reason it appears that inbound calls from out in the world
are going
> to our phones using ULAW, but outbound calls to the world are using
G.729.
> 
> That's progress but...how can I get my Asterisk server to use G.729 to
pass
> those incoming calls to my phones?
> 
> 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 9:54 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:
> >
> > > O.K., interestingly enough when I call our extensions from my
mobile
> > > phone it still seems to be using ULAW, but when they dial out it
> seems
> > > to be using G.729 now.
> > >
> > > Is there something in Dahdi that I need to configure so that
inbound
> > > calls (from the PRI on a Digium TE205) use G.729 to get to the
> phones
> > > too?
> >
> > A Dahdi channel over a PRI will always be ulaw - that is the
encoding
> on the
> > PRI (at least in the US).  If your phones are using G.729 then
> transcoding will
> > be taking place within asterisk for the bridge between the channels.
> >
> > My guess is you are looking at the PRI channel.  There should be
> another
> > channel for the phone.  That should always be G.729 now.
> >
> > Cheers,
> >
> > j
> >
> > >
> > > 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 at jeff.net
> > >> Sent: Tuesday, December 15, 2009 9:13 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:
> > >>
> > >>> 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?
> > >>
> > >> Just do a "sip reload" at the asterisk CLI prompt and you will be
> > >> good
> > > to go.  It
> > >> won't cutoff any calls in progress.  Then reboot your phone.
> > >>
> > >> Cheers,
> > >>
> > >> j
> > >>
> > >> _______________________________________________
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