<font face="arial" size="2">you can also try using<br /><br />
busydetect=yes<br />
busycount=4<br /><br />
in your zapata.conf<br /><br />
Hopefuly you won't start getting sudden hang ups, due to false positives and it will be helpful enough.<br /><br />
Alyed </font>
<br />
<br />
<font face="Tahoma, Arial, Sans-Serif" size="2">
<hr align="center" size="2" width="100%" />Return-Path: <asterisk-users-bounces@lists.digium.com> Tue Oct 17 14:30:11 2006<br />Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;<br /> Tue, 17 Oct 2006 14:30:11 -0700<br />Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])<br /> by lists.digium.com (Postfix) with ESMTP id C3E341FCA42;<br /></font>
<br />
<br />On 2006-10-17 09:00:51 -0700, Bjoern Metzdorf <br /><bm @turtle-entertainment.de=""> said:<br /><br />>> I run into that from time to time for this business account we have <br />>> where channels were staying open for a long time so I made a script run <br />>> from cron to hang up any extension over X amount of time:<br />>> <br />>> /usr/sbin/asterisk -rx "show channels concise" |awk -F : '($11 > 5400) <br />>> {print "/usr/sbin/asterisk -rx \"soft hangup " $1 "\""} '|sh<br />>> <br />>> This looks at any calls over 90 minutes then hangs it up. You can <br />>> modify it for your issue say something like:<br />>> <br />>> /usr/sbin/asterisk -rx "show channels concise" |awk -F : <br />>> '/YOUR_X_SIPURA_NUMBER/'|awk -F : '($11 > 5400) {print <br />>> "/usr/sbin/asterisk -rx \"soft hangup " $1 "\""} '|sh<br />>> <br />>> Not practical though for saving money... If someone is on for say 1 <br />>> minute and there is an issue with the channel not hanging up, 5399 <br />>> minutes would still be billed.<br />> <br />> What version are you using?<br />> <br />> I never had these issues with asterisk 1.0.x in 15 months. That leads <br />> me to a problematic 1.2.x or to faulty bristuff-patches.<br />> <br />> I will upgrade asterisk asap to latest 1.2.x and add an absolute <br />> timeout to those destinations.<br />> <br />> But: Are we the only ones experiencing this?<br /><br />That really doesn't sound at all the same to me as what he is <br />describing? In his case (ie not hung up calls) if you are using SIP <br />handsets, then the rtptimeout setting can cut the calls off when there <br />is no audio data flowing.<br /><br />Good Luck,<br />Marty<br /><br /><br />_______________________________________________<br />--Bandwidth and Colocation provided by Easynews.com --<br /><br />asterisk-users mailing list<br />To UNSUBSCRIBE or update options visit:<br /> http://lists.digium.com/mailman/listinfo/asterisk-users<br /><br /></bm>