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: chris at bayareadigital.us
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: chris at bayareadigital.us > > _______________________________________________ > -- 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
since you're using ulaw try setting dtmfmode = inband if this doesnt work try = auto -Jon ----- Original Message ----- From: "Christopher Gray" <chris at bayareadigital.us> To: "Asterisk Users Listserve" <asterisk-users at lists.digium.com> Sent: Friday, January 23, 2009 8:13 PM Subject: [asterisk-users] Passing DTMF> 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: chris at bayareadigital.us > > _______________________________________________ > -- 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
I have had this same problem using via:talk. Even though they tell me they have hundreds of people using Asterisk with their service that have no problems we cannot make it work. I have also had reponses confirming that in this email list. So don't wast your time with via:talk. 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: chris at bayareadigital.us > > _______________________________________________ > -- 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 > > >-- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465