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
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
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
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
believe
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 very
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 alters the
