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

[Ben Schorr]

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
> 
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users