[Asterisk-Users] asterisk-grandstream call

[Bill Michaelson]

I am trying to muddle my way tthrough getting something - actually 
anything to work - with Asterisk.  I've acquired a Grandstream phone and 
I've got * on a Red Hat 9 box.   I've gotten to a point where I can see 
(via ethereal) that the phone REGISTER's successfully with asterisk, and 
then I try to dial into voicemail.  This is what I observe in the packet 
trace...

GS: INVITE -> *
*: Status 100 (Trying) -> GS
*: Status 200 (OK with session description) -> GS
So far, seems reasonable - but I'm a complete novice with this protocol.

Then I see a huge stream of UDP packets sent by * to the GS on port 
5004, but the GS only replies with an ICMP destination unreachable to 
each packet.  I'm guessing that this is an RTP audio stream, but I don't 
know why the GS is not ready or otherwise unwilling to receive the 
packets.  Examining the GS config, I've confirmed that the "local RTP 
port" is set to 5004.

I have many questions about how this should work, but I'll save some 
bandwidth and leave it to someone here to suggest what should be checked 
next.

Thanks.

--
Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED]
Thanks for putting up with my spam filter!
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