[Asterisk-Users] SIP / GrandStream Configuration

[Michael Koehler]

Sorry, but my * is behind NAT and i have no problems with SIP, and it 
even works with NAT to NAT and without forwarding ports or similar effords.


Michael


Stephen Varga wrote:

>On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote:
>  
>
>>Adam:
>>in reference to my first message, the NAT on the SIP/GS (a D-Link router)
>>has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being
>>forwarded to the Sip/GS.
>>The Asterisk server, also behind another NAT (Linksys), has the same ports
>>opened and forwarded.
>>is it still impossible?
>>URiel
>>    
>>
>
>Nope, it is not currently possible. * behind a NAT for SIP does not work
>because the * real IP address is placed in the SDP information,
>therefore the 'outside' phone can not send the media stream to *. See my
>answers over the last week for the more details and possible work
>arounds.
>
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>  
>
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