[asterisk-dev] [svn-commits] mjordan: testsuite/asterisk/trunk r3711 - /asterisk/trunk/tests/channels/SIP/...

[Matthew Jordan]

On 04/16/2013 09:38 AM, Olle E. Johansson wrote:
> 
> 16 apr 2013 kl. 16:33 skrev SVN commits to the Digium repositories <svn-commits at lists.digium.com>:
> 
>> Asterisk 12 uses DTMFBegin/DTMFEnd. In this case, we only really care
>> about the DTMFEnd events - so subscribe for those and treat them as
>> if they were the DTMF event in previous versions of Asterisk
> 
> Please notice that this is not the solution we will have. I have a large patch that adds DTMFcontinue in order to
> properly handle DTMF. I don't know if that's relevant here, but we need to move that patch forward to fix a lot
> of DTMF issues.
> 
> In this code we have 
> 
> DTMF Begin
> DTMF continue - repeated for as long as we get DTMF in
> DTMF End
> 
> The DTMF continue has a duration data field, that is needed to handle playout on the other side of the bridge.
> 
> Just to update you in case it matters for this code.
> 

Luckily, this is just AMI's representation of DTMF. It's just responding
to whatever the core (over the Stasis-Core message bus) tells it to do -
which, even with a DTMF continue, would still include a Begin/End. So
this should be okay with any changes we make to DTMF.

While the core would make use of a DTMF continue, I'm not sure if
sending a DTMF continue makes a lot of sense for external applications.
Right now, of course, that isn't really even an option, but it could
certainly be added if there was a need for it.


-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org