[Asterisk-Users] Call Waiting on SIP phones

[John Todd]

>Sorry, to repost - but I left a "/*" comment - here it is again
>
>Paul
>[code block removed]

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.

  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!

  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.

JT