[Asterisk-Dev] SIP / H.323 Gateway with Asterisk

[Frits Wiersma]

I had to go back to an older version of the H.323 from nufone, somewhere 
around 05/10/04, to make it work again with our gatekeeper. I am still trying 
to find out what the problem was. 

It might have to to with the change in:

H323Connection::AnswerCallResponse MyH323Connection::OnAnswerCall
{
        /* The call will be answered later with "AnsweringCall()" function.
         */
        return H323Connection::AnswerCallAlertWithMedia;
}


On Tuesday 07 September 2004 15:36, Aaron S. Joyner wrote:

> Aaron S. Joyner wrote:
> > We are using SIP phones to make calls through Asterisk, out an H.323
> > connection.  Depending on the end point, some times these calls do not
> > pass audio.  The call setup appears normal, everything about the
> > negotiation appears to be correct, but no audio is passed out of
> > asterisk, in either direction.  From a packet dump we can see that
> > audio is coming in from both sides, but no audio is being sent out
> > from Asterisk.
>
> <lots of detail snipped, see previous post>
>
> I'm posting again because we have the better part of a resolution.
> There were two problems at work.  The reason we couldn't talk to the
> remote asterisk box was a codec issue, which was easily resolved.  The
> reason we couldn't talk to the remote Cisco box was less
> straight-forward.  We tried it with the oh323 driver, and it worked
> fine.  But it still does not work with the nufone asterisk/channels/h323
> driver.  We are going to work on it some more today in hopes to get it
> working with the nufone driver - as I think it's simply a minor
> configuration issue.  If I come up with any big revelations, I'll be
> sure to post them back here.  But at the worst, for now, it's working
> with the oh323 channel driver.