Re: [asterisk-users] codecs/voicemail/DTMF
Sorry, I misread your message as "incoming" and "outgoing" calls. Mr. Jones wrote: Hi Eric, I'm confused on where I would put this? I'm also confused on how this would help with external calls (which we want to be g729) vs internal calls to voicemail (which appear to need to be g711)? Thanks a ton! Brian On 9/19/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: Use type=user for inbound and type=peer for outbound. Have different codec settings for each of them. Mr. Jones wrote: > Hi Folks, > > We're trying to roll Asterisk out to production and are having a few > complications. > > Most specifically we have G711 for our inbound origination, but would > prefer G729 for outbound termination, so far so good - it appears that > dtmfmode=auto works in both cases. > > The area I'm having trouble with is, in order to have g729 on the > outbound I have: > > disallow=all > allow=g729 > allow=ulaw > allow=alaw > > In sip.conf at the [general] level. > > When we call voicemail, or the auto attendant internally touchtones > don't work and we get: > > WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not > supported on codec g729. Use RFC2833 > > I'm just guessing, but I thought "auto" was supposed to negotiate the > DTMF mode. Since it appears that the voicemail can't handle RFC2833, > is there some way to force the codec to resort to G711? > > Thanks! > > Brian > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codecs/voicemail/DTMF
Hi Eric, I'm confused on where I would put this? I'm also confused on how this would help with external calls (which we want to be g729) vs internal calls to voicemail (which appear to need to be g711)? Thanks a ton! Brian On 9/19/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: Use type=user for inbound and type=peer for outbound. Have different codec settings for each of them. Mr. Jones wrote: > Hi Folks, > > We're trying to roll Asterisk out to production and are having a few > complications. > > Most specifically we have G711 for our inbound origination, but would > prefer G729 for outbound termination, so far so good - it appears that > dtmfmode=auto works in both cases. > > The area I'm having trouble with is, in order to have g729 on the > outbound I have: > > disallow=all > allow=g729 > allow=ulaw > allow=alaw > > In sip.conf at the [general] level. > > When we call voicemail, or the auto attendant internally touchtones > don't work and we get: > > WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not > supported on codec g729. Use RFC2833 > > I'm just guessing, but I thought "auto" was supposed to negotiate the > DTMF mode. Since it appears that the voicemail can't handle RFC2833, > is there some way to force the codec to resort to G711? > > Thanks! > > Brian > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codecs/voicemail/DTMF
Use type=user for inbound and type=peer for outbound. Have different codec settings for each of them. Mr. Jones wrote: Hi Folks, We're trying to roll Asterisk out to production and are having a few complications. Most specifically we have G711 for our inbound origination, but would prefer G729 for outbound termination, so far so good - it appears that dtmfmode=auto works in both cases. The area I'm having trouble with is, in order to have g729 on the outbound I have: disallow=all allow=g729 allow=ulaw allow=alaw In sip.conf at the [general] level. When we call voicemail, or the auto attendant internally touchtones don't work and we get: WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 I'm just guessing, but I thought "auto" was supposed to negotiate the DTMF mode. Since it appears that the voicemail can't handle RFC2833, is there some way to force the codec to resort to G711? Thanks! Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] codecs/voicemail/DTMF
Hi Folks, We're trying to roll Asterisk out to production and are having a few complications. Most specifically we have G711 for our inbound origination, but would prefer G729 for outbound termination, so far so good - it appears that dtmfmode=auto works in both cases. The area I'm having trouble with is, in order to have g729 on the outbound I have: disallow=all allow=g729 allow=ulaw allow=alaw In sip.conf at the [general] level. When we call voicemail, or the auto attendant internally touchtones don't work and we get: WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 I'm just guessing, but I thought "auto" was supposed to negotiate the DTMF mode. Since it appears that the voicemail can't handle RFC2833, is there some way to force the codec to resort to G711? Thanks! Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
