[asterisk-dev] meetme and DTMF going from Zap to SIP-IAX2

Matt Florell astmattf at gmail.com
Fri Aug 10 08:05:07 CDT 2007


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.


On 8/6/07, Russell Bryant <russell at digium.com> 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.
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