Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4. Is this new in 1.6?
on Tuesday 04/14/2009 Brent Davidson([email protected]) wrote > To the best of my knowledge, the only way for you to control the > duration sent to the PSTN lines is for you to be directly connected to > the lines so you can set the tone duration in zapata.conf / dahdi.conf > or to use inband signalling. > > One thing you might try is researching the "SipDtmfMode" command. It > allows you to change the DTMF mode on an active channel. A suggestion > might be to set up the dial command with the M() option that point to a > Macro that changes the DTMF to INBAND once you are connected to the > problem number. At least in theory, if your provider is expecting > RFC2833 and they get inband, they should just ignore the inband > signaling and pass it on as part of the audio stream. The only problem > is that this may only work if you use uLaw or aLaw for your codec and I > don't know exactly how to set the tone duration without having a > zapata.conf or dahdi.conf entry. Even with one of those files, I don't > know how Asterisk chooses to do the rfc2833 to inband translation or > where it pulls the toneduration setting from if no PSTN interface is > involved in the call. > > -Brent > > John covici wrote: > > OK, thanks. If I could convince them to use info, would that be > > better as far as the duration is concerned? > > > > > > on Monday 04/13/2009 Brent Davidson([email protected]) wrote > > > John covici wrote: > > > > Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, > > > > however I would like to increase the duration of the tone, its pretty > > > > short and some IVR's are unhappy or don't detect it. I did poke > > > > around, but it looks like when RFC2833 is used, it actually generates > > > > rtp packets of some sort, so I have no idea how to increase that > > > > duration. > > > > > > > > Any assistance would be appreciated. > > > > > > > > > > > > > > If your provider insists on rfc2833, then their servers will be > > > responsible for setting the tone duration sent to PSTN lines. > > > > > > > _______________________________________________ > -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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
