[Asterisk-Users] Call Waiting on SIP phones

Paul Liew pliew at atp.org.au
Mon Oct 20 19:52:53 MST 2003


----- Original Message ----- 
From: "John Todd" <jtodd at loligo.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, October 21, 2003 10:29 AM
Subject: Re: [Asterisk-Users] Call Waiting on SIP phones


>
> Paul -
>    A few questions and comments:
>
>   1) So, does this also make "incominglimit" and "outgoinglimit" work
> as expected?  The current method doesn't do quite what the average
> user thinks it would do.

Yes it does John - however, I've made incominglimit and outgoinglimit to
imply the same, so we only need incominglimit=1.

>
>   2) Your patch may be relevant to:
> http://bugs.digium.com/bug_view_page.php?bug_id=0000329  - I didn't
> realize that "outgoinglimit=1" would only work for the first call,
> but fail subsequently.  With "type=peer", the "incoming=" and
> "outgoing=" modifiers really don't work at all for me, which makes
> them almost completely useless.  If you have some method to fix that:
> great!

The work "once" applies to incoming (never tested for outgoing) - Its now
fixed for both - so let me know how you go.

>
>   3) If there is an existing bugtracker item, the source code diff's
> are best appended to that particular bug.  If there isn't one open,
> go ahead and open one.  Sending diff's to the mailing list is getting
> less common (and less desirable) as time goes on, since we have the
> bugtracker now.
>
I've added to the bugtracker -
http://bugs.digium.com/bug_view_page.php?bug_id=0000408

I've made it new report as it also fixes the incoming call waiting which
"329" didn't refer to.

Also thanks for your comments and also from Tilghman.

Paul




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