[Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working

[David Hajek]

I have canreinvite=no already, below is my sip.conf entry.

[1360]
username=1360
callerid=Phone 1 <1360>
secret=mysec1
host=dynamic
auth=md5
qualify=1000
dtmfmode=rfc2833
context=from-sip-unrestricted
mailbox=1360
type=friend
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g726
nat=yes
canreinvite=no

-
David Hajek
http://hajek.net/blog

Rich Adamson wrote:

>>I'm trying to configure Sipura 2000 (behind NAT) which connects to 
>>Asterisk (public IP, no NAT) and having interesting results. When Sipura 
>>is behind Linux/NAT firewall it works great and no special NAT settings 
>>on Sipura are necessary. The issue I'm having is when Sipura is behind 
>>Linksys broadband NAT router. Sipura gets registered with Asterisk just 
>>fine, but I can't hear the other party (to be more precise I can hear 
>>first two secs then nothing). So it must be the incoming RTP is blocked 
>>on Linksys. Here I think STUN server enters the game and give some help?
>>
>>I have installed Vovida STUN server and point Sipura to use it. But no 
>>luck, I still can't hear the other party. I've ended up with having 
>>Linksys to forward all ports to my Sipura (DMZ host) which works.
>>
>>What is interesting is that when I'm using Vonage service (Cisco ATA) it 
>>works just fine without touching the Linksys. How come they can get 
>>through it?
>>
>>Any hints?
>>    
>>
>
>Add canreinvite=no to the sipura def's in sip.conf
>
>
>_______________________________________________
>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
>  
>