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

Tue Dec 15 14:45:59 CST 2009

Well, I know I still have a LOT to learn about Asterisk but...how will
they get their incoming phone calls from their DIDs (which the TelCo
sends to their PRI) if I move the remote office onto a SIP provider?

The PRI doesn't seem to cause any problem for the majority of the users
(at the home site) it's just the 8 users at the remote site who are
complaining of 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 10:31 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Can't get G.729 to work...
> 
> 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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