[asterisk-users] spa3k dtmf problem asterisk 1.2.x

[Doug Crompton]

Anyone that uses the spa3k with Asterisk knows about the dtmf issues of
not being able to get tones properly to an IVR after call completion. You
can make it work by eliminating ALL special keys - transfer, etc. in the
dial and using inband signaling. This has been beat to death over the last
year.

My question is that there were patches to rtp.c that were an attempt to
correct. I tried a few to no avail. Does anyone have a patch that works?
I am currently using 1.2.13

My understand from googling this is that the problem is both a Sipura and
Asterisk problem, although more of the blame is put on Asterisk.

Also the rtp in 1.4 has been completely reworked. Has anyone tested this
with the spa3k? Unfortunately 1.4 is a significant change that involves a
great deal of time to test and is not at all like doing an upgrade within
1.2. So I am not inclined to go that route yet unless it fixes this
problem.

Doug