[Asterisk-Users] Using call.sample on Zap hardware - Answering problem

[Julian J. M.]

On Sun, 27 Mar 2005 12:29:55 -0500, Patrick Healy
<pjhealy at healyville.com> wrote:
> I've got a X100P connected to a POTS line and am using it to call out to
> play a recorded message.  I drop a copy of sample.call into
> /var/spool/asterisk/outgoing and Asterisk picks up the line and initiates
> the call.  The problem is that the recorded message starts immediately and
> doesn't wait for the called party to pick up the phone.  When I try this
> same process with a SIP extension, the process works like a champ, it just
> fails on the Zap interface.

This is normal, on analog Zap channels, asterisk doesn't know when the
called party picked up the phone, unless your Telephone Company
provides you with such information, usually via polarity switchs.

You can enable debug on module wctdm (debug=yes), and watch
/var/log/messages. Check if it detects polarity reversal when the
called party picks up the phone, and when it hangs up. If it does, you
could use answeronpolarityswitch=yes, and hanguponpolarityswitch=yes
in zapata.conf.

I've recently submited a patch to the bugtracker
(http://bugs.digium.com/bug_view_page.php?bug_id=0003874), that fixes
some problems with this approach (at least in Spain).

Julian J. M.