[Asterisk-Users] Problem with SIP clients

[Ricardo Peironcely]

Has you redirected all the RTP ports? You must redirect the SIP and the 
RTP streams. Take a look to the rtp.conf file of  your asterisk 
installation to configure the RTP ports that you want to use.

Best regards.
Rpr

Alex Piqueras escribió:

> Hi, I have my asterisk server inside a NAT.
> When i connect a softphone SIP inside my net, all go well. Ok.
> But when I try connect a SIP softphone client outside my NAT, I get:
> NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex 
> <sip:phone2 at 192.168.1.21>' failed for '83.41.119.25'
>
> Can someone help me with this?
>
> PD: Sorry for my english
>
>
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