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

[Ben Schorr]

Really?  I didn't see them in the web interface; which is why I turned
to editing the files.  I'll check the web interface again, perhaps I
simply missed them.

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 Andres
> Sent: Wednesday, December 16, 2009 5:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Can't get G.729 to work...
> 
> Ben Schorr wrote:
> 
> >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
> >
> >
> I would not go editing the individual files if you are using FreePBX.
> As soon as you make a change in the web interface it will override any
> manual changes you made.
> 
> Simply do it in the web interface for each extension.  You do have a
> parameter called allow and another called disallow in the web
interface when
> editing the extension (its under device options).  Use them.  For
multiple
> entries just separate them with a comma.
> 
> Andres
> http://www.neuroredes.com
> 
> >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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> >>
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> >>
> >>
> >
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> >
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> >
> >
> >
> 
> 
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