[asterisk-dev] DTMF queuing
jtodd at digium.com
Tue Jan 27 02:56:27 CST 2009
On Jan 26, 2009, at 7:38 PM, James Lamanna wrote:
>> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote:
>>> Is it just me, or does DTMF queuing not work properly?
>>> I'm consistently faced with the issue that users (and myself) will
>>> dial digits quickly and all I get in the logs are:
>>> end 'X' put into dtmf queue on SIP/xxxxxxxxxx
>> What version are you talking about? If it's not 1.4.23, please try
>> that, as there are some related fixes in that version.
> Sorry, I neglected to mention this is on 18.104.22.168.
> I will try and test 1.4.23 and see if things are better.
> In the meantime, I'll report my findings to see if you guys can
> better explain
> to me what is going on.
> The best DTMF combination (between phone and asterisk) I have found
> sip.conf - dtmfmode=info
> Phone (SPA962) - DTMF Mode = Auto
> This works very well for outbound SIP and Zap trunks and on both ulaw
> and g726 codecs.
> However, this does NOT work for any prompt that is internal to
> asterisk that
> needs to detect DTMF (Voicemail, Authenticate, etc..).
> The only way for these prompts to work is to explicitly put
> in the dial plan. Of course, this breaks when the codec is g726. Why
> do these prompts not work with this setup?
> I've also noticed that when in this mode, nothing is put into the
> dtmf log.
> Does that mean that the phone and asterisk have negotiated inband
> (though if this was the case why would it work with g726..)?
> (please CC me directly since I'm on digest mode at the moment).
Actually, I'm a little surprised you get DTMF working at all in this
Setting dtmfmode=info means that Asterisk will be looking for SIP INFO
messages that contain DTMF events. Have you watched the SIP channel
debug during DTMF events, or set up a tshark or other interceptor to
watch port 5060 as you send DTMF? Perhaps you've got a few things
mucking up the works there. What does RFC2833 get you if you set all
the gear to that?
Try setting everything to RFC2833 and try again. I'd also suggest
follow-up messages go to asterisk-users and not to this list, as this
is not sounding particularly like a question for the -dev list where
core Asterisk code is involved, and you'll probably get more answers
over on -users.
John Todd email:jtodd at digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
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