<div class="gmail_quote">2009/4/15 John covici <span dir="ltr"><<a href="mailto:covici@ccs.covici.com">covici@ccs.covici.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Its not there and the link you gave me says its for sip originating<br>
rather than calls to a sip channel.<br>
<div><div></div><div class="h5"><br>
on Tuesday 04/14/2009 Brent Davidson(<a href="mailto:brent@texascountrytitle.com">brent@texascountrytitle.com</a>) wrote<br>
> It's been around awhile. I've used it in 1.4 Check out this link for<br>
> basic info: <a href="http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode" target="_blank">http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode</a><br>
><br>
> John covici wrote:<br>
> > Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.<br>
> > Is this new in 1.6?<br>
> ><br>
> > on Tuesday 04/14/2009 Brent Davidson(<a href="mailto:brent@texascountrytitle.com">brent@texascountrytitle.com</a>) wrote<br>
> > > One thing you might try is researching the "SipDtmfMode" command. It<br>
> > > allows you to change the DTMF mode on an active channel. A suggestion<br>
> > > might be to set up the dial command with the M() option that point to a<br>
> > > Macro that changes the DTMF to INBAND once you are connected to the<br>
> > > problem number. At least in theory, if your provider is expecting<br>
> > > RFC2833 and they get inband, they should just ignore the inband<br>
> > > signaling and pass it on as part of the audio stream. The only problem<br>
> > > is that this may only work if you use uLaw or aLaw for your codec and I<br>
> > > don't know exactly how to set the tone duration without having a<br>
> > > zapata.conf or dahdi.conf entry. Even with one of those files, I don't<br>
> > > know how Asterisk chooses to do the rfc2833 to inband translation or<br>
> > > where it pulls the toneduration setting from if no PSTN interface is<br>
> > > involved in the call.<br>
</div></div><br></blockquote></div><br>It's been there from at least 1.0...<br>
<br>
But, you are correct, it's for use on incoming SIP calls rather than outgoing SIP calls...<br>
<br>
d<br>