[Asterisk-Dev] Asterisk PSTN gateway billing issue

[C. Maj]

On Tue, 31 May 2005, Noel Sharpe waxed:

> Hi
>
> I am having an issue while trying to use asterisk as a PSTN gateway.
> Let me explain my setup:
>
> I have a SER proxy server that routes calls to different destinations to
> various PSTN gateways.  One provider runs asterisk and is connected to a
> bunch of PSTN lines (actually it's a GSM hive).  The problem is that as
> soon as asterisk tries to dial the destination number using it's ZAP
> interface, it reports back to SER that the call has been answered.  The
> problem is that SER then starts billing the customer, as it believes
> that the final leg of the call has been answered.

8<'s

> Unless I am missing something in the configuration, it looks to me that
> this piece of functionality is missing from *.  If it truly is missing,
> I'd like to have a go at implementing it, and I'd appreciate some
> feedback as to how this should work. BTW, this seems to work correctly
> for another provider who is using a Cisco gateway.

Analog zap channels are marked as answered unless you set
callprogress=yes in zapata.conf, but there's currently only
support for US, Costa Rica, and Brazil tones and patterns in
dsp.c -- I'm assuming you are using UK tones ?  If the cisco
unit is already providing this functionality, then it seems
likely somebody at some time had to configure it to listen
for your tones and detect sufficient non-DTMF line energy
to trigger an answer.

OTOH, if you have a PRI, this is a non-issue -- setting
callprogress=no is correct -- but you've got some other
problem(s).

--Chris


-- 
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May