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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 b...@rolandschorr.com > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@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 > > b...@rolandschorr.com > > > > > >> -----Original Message----- > >> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > >> boun...@lists.digium.com] On Behalf Of j...@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 > > > > _______________________________________________ > > -- 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 > > > > _______________________________________________ > -- 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 _______________________________________________ -- 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 _______________________________________________ -- 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
