[Asterisk-Users] Sipura stun settings

[AJ Grinnell]

Well, it worked for 1 call, but now I am back to getting half a ring from
the ATA and then nothing. I am only seeing one rtp packet recieved per call.
Any other ideas?


-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Andres
Sent: Wednesday, May 26, 2004 5:28 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Sipura stun settings


AJ Grinnell wrote:

>I am using sipura spa-1000s and spa-2000s behind a firewall. My asterisk
>server and STUN server are outside the firewall on a public network. I
would
>like the Sipuras to be able to reinvite, so I set canreinvite=yes in my
>sip.conf, and set the STUN server under the SIP tab in the Sipuras.
However,
>I am not able to hear the other caller (the Sipura is not recieving RTP
>packets, it is sending just fine). Am I missing something on the Sipura
>config? I am not sure what all of the VIA options mean, and which ones I
>should use. Cant find any good info out there, can someone hrer help me
out?
>Thank you.
>
>
>
>
>
You need these settings:
Substitute_VIA_Addr               "Yes" ;
STUN_Enable                       "Yes" ;
NAT_Mapping_Enable[1]             "Yes" ;
NAT_Keep_Alive_Enable[1]         "Yes" ;
STUN_Test_Enable             "Yes";

and of course define your STUN Server.

--
Andres
Network Admin
http://www.telesip.net



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