asterisk users - Apr 2008 - Polycom RTP port range

We've just upgraded to asterisk 1.4 and we have changed the way we
handle our calls a bit. This seems to be giving us a bit of an issue.

We now allow the phones to reinvite. In the rtp.conf file, i've set the
range from 10000-20000. However, when the phones begin talking to one
another, they start talking on ports in the 2000 range. I'm assuming
they don't use the rtp.conf settings then.

If not, is there a way to configure this range on the phones? (They are
polycom 501s).

Any help would be much appreciated.
Rob

On Fri, Apr 18, 2008 at 1:17 PM, Rob Schall <rschall at callone.net>
wrote:
>
> If not, is there a way to configure this range on the phones? (They are
> polycom 501s).


You're right, how could the phone read your asterisk rtp.conf setup?

Download the SIP Administrator Guide for the 501 from the Polycom website,
then check the TCP/IP config section starting on page A-50. On page A-54
you'll find specifics on the ".mediaPortRangeStart" parameter,
which is what
you're after. Modify its settings in the sip.cfg file your phone loads from
the tftp server, and you should be good to go.

(Chapter and pagenumbers are from the Admin Guide v. 2.2.0)
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