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

[Olle E. Johansson]

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.

/O