On May 22, 2009, at 3:05 PM, Martin wrote: >> Yes, this would be why I said that it is Asterisk's fault and >> provided possible >> workarounds. >> >> Thank you for your helpful and constructive criticism. > LOL yes you could expect now everyone to be critical about something > like this. > Asterisk has been around for quite some time now (6+ years) and this > sounds like a pretty basic problem > that could cause a lot of failed calls with some SIP MTAs. > > I would expect this kind of problem from an Asterisk version before > 1.0.0 > > Martin
Things that find their way to the issue tracker tend to get repaired if there is a reproduce-able problem. If you could include the documentation you've made in a new issue/bug report, that would help this get nailed down. If nobody has reported this as a bug before, then chances are it hasn't been fixed. Additionally, as Asterisk is a community effort it may be possible for you to create a quick patch which if applied would solve the problem for everyone, and including that in the issue report would help to quickly resolve the matter. JT -- John Todd email:[email protected] Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
