[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: 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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-- 
Andres
Network Admin
http://www.telesip.net
"Providing Wholesale Florida 
SIP/IAX2 Termination for US$0.01/minute"