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

Tue Dec 15 16:08:29 CST 2009

O.K., I restored the Allow=ulaw in the sip_general_additional.conf file,
then I found the individual extension settings in the
sip_additional.conf file and I added 

disallow=all
allow=g729

to each of the extensions at the remote site.  Then I did a SIP RELOAD.
So we'll see how that goes.

Thanks again for the assist - this has been quite an education.

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 11:47 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Can't get G.729 to work...
> 
> I don't know how FreePBX works, but with vanilla Asterisk you would do
> something like this with your sip.conf:
> 
> [general]
> disallow=all
> allow=ulaw
> allow=g729
> 
> [localA]
> callerid=Local phone A <100>
> username=localA
> secret=blahblah1
> 
> [localB]
> callerid=Local phone B <101>
> username=localB
> secret=blah1blah
> 
> [remoteA]
> callerid=Remote phone A <102>
> disallow=all
> allow=g729
> username=remoteA
> secret=123456
> 
> [remoteB]
> callerid=Remote phone B <103>
> disallow=all
> allow=g729
> username=remoteB
> secret=654321
> 
> You can do this using templates as well, but this will make it easier
to
> understand. See the disallow/allow lines on the remote peers? Those
> override the settings in the general portion of your sip.conf. With
these
> settings the local phones will use ulaw by default and allow g729 when
> needed.
> 
> This will do what you want for the most part. Local phones will use
ulaw for all
> calls between themselves and calls in and out of the PRI. Calls from a
remote
> phone to a local phone will use g.729 end to end. Calls from a local
phone to a
> remote phone will use ulaw between the local phone and asterisk and
g.729
> between asterisk and the remote phone (this is a limitation of
asterisk's
> codec negotiation). Calls from remote phones will use g.729 all the
time.
> 
> I'm sure there is a way to do this through the freepbx gui, but like I
said, I
> have no experience with freepbx.
> 
> -Dave
> 
> 
> 
> Ben Schorr wrote:
> > 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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