[asterisk-users] SIP Connection Problems

[Dovid Bender]

I had issues using NAT when having multiple phones as well as single phones 
behind NAT. You can try setting port forwarding on the phones side as well 
as look at a better router. Some routers will make you pull your hair out 
while others will work almost perfectly (this explains my now bald head :) )

Dovid

----- Original Message ----- 
From: "Barry Fawthrop" <barry at ttienterprises.org>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Saturday, August 12, 2006 7:27 AM
Subject: [asterisk-users] SIP Connection Problems


> Hi All
>
> I have a Cisco 7960 which is connected remotely to an Asterisk server.
>
> Both are unfortunately behind NAT.
> The Phone registers and is show in sip show peers, with the correct public 
> ip for the phone and a 100ms qualify time
>
> (1) I can dial the phone from another phone, it will ring but no voice 
> goes through in fact I get this error on * console
>      SIP response 481 "Call Leg/Transaction Does Not Exist
>
> (2) the phone can make calls outbound fine, with voice no problems ?
>
> http://channels.debian.net/paste/3409   holds the SIP debug for the phone 
> extension 650
>
> port forwarding is also set on both sides, and sip.conf has the nat=yes , 
> externip and localnet all set correctly
>
>
> Thank always
>
> Barry
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>