Re: [Asterisk-Users] Sipura stun settings

[Andres]

AJ Grinnell wrote:
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?
 

Does it work ok if canreinvite=no ??  To be honest I have not 
experimented much with canreinvite.  All our Sipura subs register with 
SER and not Asterisk. STUN works fine for us with the settings I sent you.

-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andres
Sent: Wednesday, May 26, 2004 5:28 PM
To: [EMAIL PROTECTED]
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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--
Andres
Network Admin
http://www.telesip.net
"Providing Wholesale Florida 
SIP/IAX2 Termination for US$0.01/minute"

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