[asterisk-users] Problem with DTMF dialing

[Andrew Joakimsen]

On Feb 12, 2008 10:40 AM, Ian <asterisk at iancoetzee.za.net> wrote:
>
>  Hi all,
>
>  its been quite a busy few day with pc's packing up etc, I recompile my
> whole asterisk today using zaptel 1.4.7.1 and now the problem is
> miraculously fixed, I will be sending this report to Digium bugs as well.
>
>  Just a quick heads up for the order in which I had to recompile in order
> for this to work
>
>
> Recompile Zaptel
> Restart Asterisk, asterisk doesn't pick up the zap channels
> Recompile Libpri
> Retart Asterisk, still no zap channels
> Doing the thing I was hoping to skip, Recompile Asterisk
> Everything in working order Did I miss something for me to have to only
> recompile zaptel, or is that the way of doing things?
>
>  Thank you all for your support
>
>  Please scroll down to see the answers to my own stupid questions :-)
>

Asterisk depends on Zaptel (well chan_zap and the respective codecs
do) so always make sure to install first LibPRI, then Zaptel then
Asterisk

FWIW in the wav recording you sent there is alot of static. I am
playing back with amaroK 1.4.7 of openSuSE.


On Feb 12, 2008 11:50 AM, Andres Jimenez <gandresin at gmail.com> wrote:
> I am having similar problems running the same versions of Asterisk,
> libpri  & zaptel.
> The Asterisk bug (http://bugs.digium.com/view.php?id=11855) was
> supossed to be related to FXO only, but I am having issues with a PRI
> line and Digium's TE120P.
>
> Do you guys think it can be the same issue?
>

Maybe it is related but with PRI Asterisk does not generate any tone
it sends a signal regarding your keypress. If you are using SIP phones
make sure the dtmfmode in use is RFC2833.