[Asterisk-Users] SIP gateway question

[Rich Adamson]

Just received a Mediatrix 1204 fxo sip gateway and playing with the initial
config's, etc. It's working, but have a ways to go before it could be
considered usable. The box was not designed to "register" like sip phones do.
The incoming pstn line is an ordinary 2-wire analog US pots line, and I'm
using canreinvite=no to forcably keep * in the middle for now.

Questions:

1. The 1204 answers incoming pstn calls correctly, cycles through the invite/
trying/ringing (I have * config'ed to simply ring an internal sip phone for
testing purposes), and I answer the call just fine from the sip phone. When
I hang up the sip phone, * sends a Bye and the 1204 says OK. 

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?

2. The 1204 seems to be set to a 30 millisecond sampling rate while all other
sip phones, etc, are set to 20. Anyone have any thoughts as to whether that
would cause a problem later, or should I change that to 20 milliseconds for
consistency?

3. Has anyone played with this box and found any unusual problems, weird
config's, etc?

The box is essentially in a test/eval mode, anticipating using it to replace
a couple of x100p's.

Rich