[Asterisk-Users] SIP-firewall problem?

[Andy Powell]

Paulo,

You are going to have to open some ports for the RTP streams to get through.
IIRC something like ports 16384 to 16391 which will also need to be forwared 
to your PC. I have had reports that just using 8000 and 8001 but I've been
unable to replicate success with these 2 ports.

HTH

Andy

*********** REPLY SEPARATOR  ***********

On 08/05/2003 at 02:35 Paulo H. Mannheimer wrote:

>Hi,
>
>I have asterisk working well on my intranet, using SJPhone to make SIP
>calls. 
>
>Everything works fine except when I try to connect from outside my
>intranet. 
>I've opened the 5060 UDP port on my D-link firewall/router and, although I
>can 
>dial, complete calls and hear the other party, my voice gets lost (the
>other 
>party does not hear me).
>
>I've took a look on the >sip debug output and everything seems fine. Any
>other 
>hint on what am I missing to configure on my firewall?
>
>Best regards,
>
>Paulo Mannheimer
>
>
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