O.K., so for now (as a test) I just commented out the "allow=ULAW" line in the SIP.conf (actually it's sip_general_additional.conf on this FreePBX box) and that does seem to be forcing all traffic to G.729.
I think ultimately I'd like to let the local users use ULAW because it seems to sound better and just force the 8 remote users to use G.729, but for now I can live with this while I figure out how to do that. Thanks for all of your help - and I welcome any additional pointers you'd like to offer. 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 Ben Schorr > Sent: Tuesday, December 15, 2009 11:16 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Can't get G.729 to work... > > 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 > b...@rolandschorr.com > > > > -----Original Message----- > > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > > boun...@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-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