[Asterisk-Users] SIP gateway question

[Rich Adamson]

Hi Bob,

> >The 1204 then sends "one" more packet to * with both the source and destination
> >ports one digit greater then what was used for the rtp session. I'm assuming
> >that's a bug in their code; anyone seen something like that before?
> >
> That would be RTCP (RTP + 1)
> 
> >3. Has anyone played with this box and found any unusual problems, weird
> >config's, etc?
> >
> I have several of these boxes in use at a few different sites.
> Once installed, I have never gone back in and looked at any of them.
> They just work.
> 
> I have it running in canreinvite mode and all sip phones running p2p.
> The poor * box has really no work to do.

I'm trying to figure out how best to bring pstn calls into * using this
box, and not sure I'm there yet. Since the box doesn't register with *, I'm
using the Redirect method which effectively causes the 1204 to dial x3094.

What I'd like to do is simply drop that incoming call into the ivr menu
directly. Any thoughts on how best to do that?

Rich