You sure about that? Having a config that looks like this: port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=g726 context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown pedantic=no progressinband=no
And then a user that looks like this: [570601] username=570601 accountcode=75415 type=friend secret=6edfa qualify=yes port=5060 pickupgroup= nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all allow=g729 context=from-internal canreinvite=no callgroup= callerid="Test VoIP Accounts" <570601> Seemed to lock EVERYONE to using g729!!! On 7/27/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote: > in ur sip.conf under the device definition you can set it > > for example device name is asterisk is pap2 > > [pap2] > username=pap2 > secret=blabla > type=friend > disallow=all > allow=g729 > > Then asterisk will only use g729 for incoming as well as outgoing calls on > this device . > > > On 27/07/07, Matt <[EMAIL PROTECTED]> wrote: > > Right.. what I'm asking is: > > > > If I set my PAP2T to use G723 or G729.... outgoing calls from that > > device go in that format. > > However, incoming calls to the device from asterisk are running at > > G711u. The PBX will access any format G711u, G723, G729 or GSM. > > What do I need to do to make asterisk use the same codec back to the > > ATA as it is using to the PBX? > > > > On 7/27/07, dave cantera <[EMAIL PROTECTED]> wrote: > > > > > > baji, mhoppes, > > > remember, if you have Only the g729 codec allowed or if this is the > only > > > allow= entry in the sip.conf file, callers requesting any other codec > will > > > be rejected.... > > > daveC > > > > > > > > > Baji Panchumarti wrote: > > > On 7/27/07, Matt <[EMAIL PROTECTED]> wrote: > > > > > > > > > Can someone comfirm my logic here? > > > > > > If I want a phone to use G729.... I can set it to use G729... do I > > > also need to set it in Asterisk? I'm thinking no... as long as > > > asterisk WILL do G729... if that's all the device accepts it should go > > > to that codec, yes? > > > > > > (based on my understanding, take it for what it is worth) > > > > > > if allow=all or allow=g729 is in your > > > asterisk configuration (sip.conf / iax.conf ) then asterisk will > > > stream packets in g729 (assuming you have any licesnses > > > needed in place). > > > > > > -baji. > > > > > > -- > > > > > > _______________________________________________ > > > --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 > > > > > > > > > > > > > > > > > > -- > > > My wife's sister is in California. > > > I should buy her a Videophone2008! > > > > > > Truly, The Next Best Thing to Being There! > > > -- > > > > > > WorldWideVideoPhones.com > > > 856.380.0894 > > > > > > > > > > > > > > > _______________________________________________ > > > --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