Hello, Just wanted to follow up on this string, After several issues with 1.4.9(IAX out of threads at 8 IAX channels) I upgraded to 1.4.10 as soon as it was released which was better, but still had issues with consistency of the "show channels concise" output(channels with wrong data, duplicates-triplicates-quadruplicates-then no results at all) and the load of 1.4.10 was almost double that of 1.2.23 at the same call load on the same hardware when going above 70 channels.
In the end I downgraded back to 1.2.23 and did a hack of routing the SIP channel to a stand-alone machine with a quad T1 card and two T1 ports crossed-over to each other. It took a couple hours to throw that together, and it is an ugly solution, but it works. The calls go out of the meetme on server A, over SIP trunk to server B where they loop out of one T1 to the other and then to an AGI script where the original server-A T1-connected user's tones are detected. I plan on working on trying to backport the SLA features within meetme to the 1.2.X tree and will post on the tracker if I am successful. MATT--- On 8/6/07, Russell Bryant <[EMAIL PROTECTED]> wrote: > Matt Florell wrote: > > I looked at the code and it is pretty drastic how much has changed in > > app_meetme.c since the 1.2.X tree. I tried the "F" flag with 1.4.9 and > > it works great with every combination of channels I could throw at it > > passing DTMF through. I am now going to start testing 1.4.9 under some > > load to see if I can use it in production. > > That's great news that it works well for you. Thanks for letting me know. > > -- > Russell Bryant > Software Engineer > Digium, Inc. > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
