[Asterisk-Users] LiveVoip DTMF Issues

[Brian Dingman]

Mark,
I have the same exact settings except I moved to 1.0.5. DTMF
recognition is fundamental to using *. Problems like this shouldn't
happen. As for the LiveVoip DID's, the two of them I have are down and
out. They were rendering fast busy signals - totally different problem
than DTMF, so support pulled them to figure out what was going on.

Just a guess, but maybe the problem lies in the soft switches that
they use or with the CLEC's. I remember an issue with Voicepulse a
while back with answer supervision. This problem was isolated to a
specific CLEC and possibly even a switch. Eventually they fixed the
issue. Not all DID's are created equal. In my case, the DID without
the problem was from Level3 and the toll free one with the issue was
from Qwest. Maybe that tells us something.

My VP Connect DID' s are from ITC (239-580) and Paetec (610-994)... these work.


On Tue, 25 Jan 2005 14:08:17 -0500, Mark Eissler <mark at mixtur.com> wrote:
> What does it say about * and providers? Err, I dunno but the whole
> issue is giving me a splitting headache! Is everyone else using g.711
> too?
> 
> This is my setup:
> 
> -Asterisk 1.0.2
> -IAX (currently set to trunk=no) to VPC
> -codec is g.711
> -tos bits are 0x18 (low delay, high throughput)
> -jitterbuffer=no
> 
> Is my understanding correct that with IAX dtmf is always sent out of
> band regardless of the codec selected?
> 
> Question: Why do you suppose only one line is okay with LiveVoip (with
> regard to DTMF)? It must be something outside of Asterisk that's
> causing the problem. Voicepulse doesn't really get too specific when
> they acknowledge a problem though.
> 
> -mark
>