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 b...@rolandschorr.com > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@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 > >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 11:47 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 > >>>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 _______________________________________________ -- 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