Well, first thing, what is the signalling at /etc/asterisk/zapata.conf set on ?
Secondly, do you have the following lines:
busydetect=yes
hanguponpolarityswitch=yes

Also make sure you set /etc/zaptel.conf to work in israel zone. It can
help detecting the right signals.

It will also be best if you could send us both files, it might help us
to see other issues that can effect this, that I do not recall at this
time.


On Dec 26, 2007 6:15 PM, Shachar Shemesh <[EMAIL PROTECTED]> wrote:
> Hi all,
>
>
> I thought that perhaps this problem will go away if I just upgrade to
> the latest Asterisk, but it did not.
>
> Zaptel 1.4.7.1
>
> libpri 1.4.3
>
> Asterisk 1.4.16.2
>
>
> Zaptel.conf marks all FXO lines as "FXSKS".
>
>
> There are several problems, but I haven't had a chance to test them all
> since I upgraded, so I'll start with the one most burning.
>
>
> The problem is that when an incoming call hangs up, the FXO line does
> not get "disconnected". If the called arrived at a human, this is not a
> big deal. When the human (usually on a FXS line) hangs up, Asterisk also
> hangs up the FXO side. If the call gets routed to voice mail and the
> person leaves a message, this is also typically not so bad. The voice
> mail auto-terminates after five seconds of silence.
>
>
> The real problem is if the person has hung up during the office wide
> greeting. In such a case, by the time the timeout kicks in and the
> phones in the office actually ring, anyone picking up gets to hear a
> "busy" signal. Worse, if no one is at the office, and the call gets
> redirected to voice mail, the voice mail is going to record the entire
> "busy" signal, until it turns into that reminder that the phone is off
> the hook (faster and louder), and only once the Bezeq system gives up
> and starts producing silence does the voicemail hang up and free the
> line. The result is a series of approx. 4:30 seconds messages in my
> voice mail.
>
>
> If anyone has any idea how to solve this problem, it would be much
> appreciated.
>
>
> The other two problems (unconfirmed with latest versions) are:
>
> 1. Asterisk does not receive the caller ID from Bezeq.
>
> 2. I cannot define a call group for outgoing calls (Zap/g1).
>
>
> The hardware are a couple Digium TDM cards bought from Atellis. If you
> tell me the cards are defective, I will simply replace them under warranty.
>
>
> Thanks
>
> Shachar



Ido
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