[asterisk-users] duration of rfc2833 generated dtmf

[John covici]

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(brent at texascountrytitle.com) 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(brent at texascountrytitle.com) 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(brent at texascountrytitle.com) 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.
 > >  > >
 > >  > >   
 > >  > 
 > >  > 
 > >  > 

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com