[Asterisk-Dev] rtp.c Issue

[Brian Rathman]

Unfortunately, my Asterisk server was not passing DTMF tones to the
satisfaction of any IVR that I call into, so I began digging around and
found the implementation of RFC2833 in the ast_rtp_senddigit function in
rtp.c. When I adjusted the duration value in this function, the IVRs I was
calling into finally began to recognize the DTMF tones I was attempting to
pass. The tones still sound extremely short on the called party end, but
apparently they are long enough.

Unfortunately, there is one other problem that I can not seem to fix and I
was looking for some support from someone who knows the code better. Now
whenever I dial digits on my phone while the call is up, if I dial them in
succession to quickly, they begin to overwrite each other. I don't know if
this was happening before I updated the duration value because I was not
getting any response.

Has anyone experienced similar issues with dtmf relay before and are there
any plans to update rtp.c to account for these issues?

FYI I am using a Cisco 5300 and SNOM 200 as the two SIP endpoints. I am also
using GSM as the codec.

Thanks,
Brian