> Caveats: There's still a few issues to clear up, namely 'pedantic'
> dialog lookups need to be addressed, and there's an occasional timing
> issue between the rtp stuff, and dialog destruction. It happens about
> as often as asterisk crashes in trunk from a bad peer pointer, btw,
> both problems
> Now.. the system that is sending the SIP packets to the 1.4 box is an
> Asterisk 1.2 box. What I assume is happening is that the 1.2 system is
> sending the old style rapid fire DTMF events, and the Asterisk 1.4 box
> is processing them in the way that a sane RFC-2833 gateway would and
> starting
Hello,
I'm running into an issue that should be simple to correct, but I
must be overlooking something in my troubleshooting. The situation is that I
have an Asterisk 1.4 server sitting in front of a Shoretel PBX acting as a
Media Gateway to convert SIP to PRI. I take SIP channels into the
> > Why not use SIP? They even suggest it when setting up your account.
> > NAT issues?
>
> Two reasons: first, I do indeed have NAT all over the place, although I
> think Luigi's recent work may make that a non-issue.
>
> The bigger reason, though, is that I want to support IAX, which I
> belie
> > SVN-branch-1.2-r51359 as of this morning on the box. I do not know
> > what the previous build was, but it couldn't be more than a
> month old.
> > I may be able to get that data if it is critical.
>
> It's critical. There doesn't appear to be any commit within
> the last month which alte
Hello,
We began experiencing an issue yesterday whereby one of our TDM
gateways, which handles IAX2 <-> Zap handoffs was dropping calls. The box in
question has 18 IAX2 peers defined, never has more than about 30 concurrent
calls and runs under what I consider a very light load. It does ver
