Brian Roy wrote: > On 3/3/06, Gary Richardson <[EMAIL PROTECTED]> wrote: >> I'm running 1.2.4 and just about every call is cut short. I'm using Cisco >> IP phones as end points. All the outbound calls are routed via SIP through a >> PRI line attached to a Cisco 2811.. >> > > > I'm running 1.2.1 and most of mine get cut short too. I posted this on the > list a few months ago and nobody had any suggestions. BJ said I should > probably post a bug on it but I haven't had time to continue to troubleshoot > it. I will go to 1.2.4 (now 5 probably) and see if mine goes away. I've been > watching change logs and hadn't seen anything surrounding mixmonitor so I've > let it go. > > Please continue to update us if anyone gets some resolution. I'm glad to > know there are lots of us experiencing this. That should be the catalyst to > get it fixed.
The only catalyst to getting it fixed will be if someone posts a bug entry with full details on bugs.digium.com If you do, post again here with the ID and discussion and testing can continue there. -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
