You hit the nail on the head when you said " , just a few more weeks of testing like we have had for the last month"
Some of the code you see makes you think the "testing" was slightly more than a clean compile :) Matt Florell wrote: > On 3/19/08, Benny Amorsen <[EMAIL PROTECTED]> wrote: > >> "Matt Florell" <[EMAIL PROTECTED]> writes: >> >> > But seriously, several of my clients use SIP exclusively, passing tens >> > of thousand of calls a day on Asterisk 1.2.X with no issues. I have >> > noticed that the load is slightly lower for SIP-only in 1.4, but I >> > have not noticed any stability issues revolving around SIP on 1.2.X. >> >> >> No hung calls? Our 1.2.x customer PBX's are drowning in "channel.c: >> Avoided deadlock for '0x91dbee8', 9 retries!". Of course you can just >> ignore the hung calls if you want, but they mess up hint state and >> prevent graceful restarts. 1.4.x fixes it. >> > > I will say that we did notice some SIP issues with older 1.2 releases, > but on the current 1.2.24+ releases we really haven't had many > problems, and we do not have hung channels. I should mention that most > of these installations have all phones on a LAN and almost none of the > calls are native SIP-bridged since they go through meetme rooms which > might account for why we do not see problems like this. > > As for 1.4.X we are moving closer to putting a live production machine > on it, just a few more weeks of testing like we have had for the last > month, and I should be convinced of it's stability. > > MATT--- > > _______________________________________________ > -- 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 > > > _______________________________________________ -- 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
