[asterisk-dev] DTMF queuing

[James Lamanna]

> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote:
>
>> Hi,
>> 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
>> etc...
>
>
>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 1.4.18.1.
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 is:

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 SIPDTMFMode(inband)
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..)?

Thanks.
(please CC me directly since I'm on digest mode at the moment).

> --
> Russell Bryant
> Digium, Inc. | Senior Software Engineer, Open Source Team Lead
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org

-- James