[Asterisk-Users] SIP gateway question

[Bob Knight]

Rich Adamson wrote:

>
>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.

-- 
Bob Knight
[-w] the work option
bk at minusw.com
925-449-9163