[Asterisk-Users] Zaptel - incoming delay

[Kevin Walsh]

George Gardiner [asterisk at georgej.demon.co.uk] wrote:
> I'm in the UK using BT for two incoming lines, one on Wildcard TDM400P
> and the other on  Wildcard X100P.  I also have a SIP connection to
> voiptalk.org. 
> 
> Incoming calls via SIP/broadband ring on extensions immediately. 
> However, an incoming call via PSTN is displayed on the CLI as an incoming
> call but extensions only ring after about 3 rings. 
> 
> I am assuming this delay is to enable Caller ID detection.  Caller ID
> doesn't work with the Digium hardware in the UK I believe, so it isn't
> much help to me. 
> 
> How do I get rid of this delay?
> 
There's a patch to get Caller*ID working for UK (BT) phone lines.
You'll find it here:

    http://bugs.digium.com/bug_view_page.php?bug_id=0001719

The chan_zap part of the Asterisk patch will probably not apply
properly any longer.  If this is still true then you can use my
update, which you can find here:

    http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg04797.html

The UK (BT) Caller*ID correctly comes in before the phone start ringing,
whereas the North American (Bellcore) ID comes in 20 minutes into the
call.  Well, it's not that bad, but it explains why your phone has to
ring several times before the Caller*ID detection kicks in.

You probably have the following line in your zapata.conf file:

    usecallerid = yes

Once the above patches have been applied, and everything has been
rebuilt and re-installed, you can change that line to the following:

    usecallerid = uk

Asterisk will now answer on the first ring.  The Zaptel hardware needs
to listen to the ring tone to detect the ring, so you still have to
wait.

I'm assuming that you want Caller*ID support.  If not then simply
set the following and you should get an answer on the first ring.
It's been a while since I tried that, but it should work.

    usecallerid = no

Note that the above patches are for the latest CVS version of Asterisk
and the Zaptel drivers.  I can't say what'll happen if you try to apply
them against an ancient Asterisk/Zaptel version.

By the way, TelAppliant will switch your account from SIP to IAX2 if
you ask.  Just something to keep in mind for the future.

I hope this helps.

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