From what I have read most dtmf problems are the phones them selfs. I use a Grandstream HandyTone 286 ATA. It has known dtmf isues. However I have had good luck with setting both the ATA and asterisk to dtmf mode rfc2833. However I would get the occasional "dtmf talk off" problem where people's voices would generate a dtmf tone. A know problem with most ATA's.
To experiment I set the ATA to use inband dtmf and I left asterisk set to rfc2833. Before this when I would call a POTS line and press a button on the asterisk phone I would just hear a slight blip of dtmf on the POTS phone. Now since changing the ATA to inband and leaving asterisk at rfc2833, the dtmf going through on the POTS phone is a long tone. I am guessing that since asterisk is only set to use rfc2833 in my conf, that the inband dtmf is passing straight through and not getting regenerated. I cannot confirm yet if it has fixed my dtmf talk off problems, but I have not had any problems navigating through company ivr's (of course I didn't before either.) Sam Christopher Gray wrote: > Hello: > > I need to be able to reliably send out touchtone to any calling party who > comes > into my pbx. The standard things to help with this have been done as far as > I > know: > > 1. dtmfmode is rfc2833. > > 2. The phones themselves are set to rfc2833. > > 3. allow=ulaw > > 4. On internal calls between extensions, touchtone works fine. > > Also, I have reviewed sip.conf with my carriers. > > Now for the question: does anybody know of a carrier that can reliably allow > an > extension in my pbx to send touchtone to a calling party? > > I have tried Vitelity and VoicePulse. Neither can do this, and VoicePulse > indicates they know it's a problem and will fix it at some unknown time in > the > future. > > For the curious, here is the reason for the need. My wife, who works as a > translator, will use this extension to receive calls from companies needing > translation. When she receives such a call, step 1 for her is to enter an > employee id code. At the end of the call, she must enter an additional code > to > receive an ending time. > > Vitelity can't do this at all. VoicePulse works about 75% of the time which > is > not acceptable. > > Thanks for any advice. > > Chris > > > > > > ---------------------------------------- > Christopher Gray, President > Bay Area Digital > > Promoting good health with innovative technology > > 870 Market Street, #653 > San Francisco, CA 94102 > Phone: (415) 217-6667 > fax: (415) 962-2520 > Email: [email protected] > > _______________________________________________ > -- 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
