<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font size="-1"><font face="Helvetica, Arial, sans-serif">Hi John,<br>
<br>
Try 1.4.2 - there was a bug in earlier versions that produced the
symptoms you describe (<a class="moz-txt-link-freetext" href="http://bugs.digium.com/view.php?id=8848">http://bugs.digium.com/view.php?id=8848</a>, and
various related ones).<br>
<br>
A.<br>
</font></font><br>
John Hughes wrote:
<blockquote cite="mid461E5EA4.4030400@Calva.COM" type="cite">
<pre wrap="">Playing with hints/presence/BLF on asterisk I've made the following
"discoveries".
1. The wiki at <a class="moz-txt-link-freetext" href="http://www.voip-info.org/wiki/view/Asterisk+presence">http://www.voip-info.org/wiki/view/Asterisk+presence</a> says:
"If you add incominglimit=1 to your peer in sip.conf, the SIP
channel will notify you when that extension is busy."
As "incominglimit" is obsolete you can use "call-limit". Also you
don't need to limit it to one, just having a call-limit at all
works. (Tried with call-limit 20).
What is the logic behind the linking of presence to call-limit?
2. A phone is only seen as busy if it's received an incoming call.
Outgoing calls don't change the state.
Why?
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
</body>
</html>