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

Tue Dec 15 15:22:51 CST 2009

O.K., so for now (as a test) I just commented out the "allow=ULAW" line
in the SIP.conf (actually it's sip_general_additional.conf on this
FreePBX box) and that does seem to be forcing all traffic to G.729.

I think ultimately I'd like to let the local users use ULAW because it
seems to sound better and just force the 8 remote users to use G.729,
but for now I can live with this while I figure out how to do that.

Thanks for all of your help - and I welcome any additional pointers
you'd like to offer.

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 Ben Schorr
> Sent: Tuesday, December 15, 2009 11:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Can't get G.729 to work...
> 
> O.K., I think I'm catching on.  I only have a single SIP.CONF file
that ALL of the
> extensions are using so I'm gathering that I need to set up a separate
> SIP.CONF file (or perhaps just an included file) for the 8 users at
the remote
> office which ONLY Allows the G.729.
> 
> So now I'm figuring out how to do that.
> 
> 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 Dave Fullerton
> > Sent: Tuesday, December 15, 2009 11:05 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Can't get G.729 to work...
> >
> > That's a bit misleading. Yes calls that travel over a PRI will be
> using ulaw, but
> > only over the PRI leg of the call. The SIP leg can still be using
> G.729 with
> > asterisk transcoding between the two legs.
> >
> > Ben, You haven't shown us the contents of your sip.conf file for the
> peers
> > you are working on but I have a guess as to what is going on. In one
> of your
> > previous messages you state: "I moved G.729 to the top of that list
> (just
> > below disallow)" I'm guessing your list looks something like this:
> >
> > disallow=all
> > allow=g729
> > allow=ulaw
> > allow={maybe something else}
> >
> > This will be fine for all the phones in the office but the remote
> phones need
> > to ONLY have disallow=all and allow=g729 in their entries in
sip.conf
> as Jeff's
> > reply stated. By having the allow=ulaw entry in there you are giving
> asterisk
> > permission to allow any call that is already in the ulaw format
(calls
> from the
> > PRI) to remain in that format when contacting your remote phones. If
> you're
> > still stick post your sip.conf (with the passwords removed) and we
can
> help
> > you out.
> >
> > -Dave
> >
> >
> > Danny Nicholas wrote:
> > > 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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