2009/4/15 John covici <[email protected]> > Its not there and the link you gave me says its for sip originating > rather than calls to a sip channel. > > on Tuesday 04/14/2009 Brent Davidson([email protected]) wrote > > It's been around awhile. I've used it in 1.4 Check out this link for > > basic info: > http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode > > > > John covici wrote: > > > 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 > > > > 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. > > It's been there from at least 1.0...
But, you are correct, it's for use on incoming SIP calls rather than outgoing SIP calls... d
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