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

[Dave Fullerton]

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