[Asterisk-Users] asterisk-grandstream call

[Glenn Dalgliesh]

Please include your sip.conf and extension.conf files. Hard to say what is
wrong without seeing the configuration

----- Original Message ----- 
From: "Bill Michaelson" <bill at cosi.com>
To: <asterisk-users at lists.digium.com>
Sent: Monday, February 09, 2004 3:15 PM
Subject: [Asterisk-Users] asterisk-grandstream call


> 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 - bill at cosi.com
> Thanks for putting up with my spam filter!
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>