[Asterisk-Users] Asterisk Behind NAT

[Steve Clark]

rudolfl at optusnet.com.au wrote:
> Hi,
> 
> I am working on exact same problem now and open to any suggestions.
> 
> So far I :
> 1. Made my NAT device to forward port 5060 to Asterisk server. 
> 2. Added line 'nat=yes' to the sip.conf for the user that is on outside.
> 
> At the moment, outside phone registers with Asterisk, but I can only place calls in
> one direction and when cal is established, no sound path exist. Asterisk tries to
> talk to the remote phone using its local IP address and this does not work. 
> 
> Let us know if you get anywhere and I will keep you posted too.
> Rudolf
> 
> 
> 
>>sammy ominsky <s at avoidant.org> wrote:
>>
>>Hi all,
>>
>>I've done quite a bit of reading, and I see that it's going to be 
>>difficult, but as a last-ditch effort before implementing a suggestion 
>>I don't like at all, I figured I'd ask...
>>
>>Has anyone successfully put an asterisk box on an internal network 
>>behind a NAT device and been able to connect with SIP from outside?  
>>The real point behind all this is to implement QoS for the voice 
>>traffic, and putting a third box in front of the asterisk and NAT boxes 
>>
>>has been deemed "too expensive".
>>
>>Currently, asterisk has a public IP, as does the NAT box behind which 
>>all the office machines sit.  If it can be done, the NAT box would be 
>>the best place to do the QoS, so why not ask, right?
>>
>>Alternatively, I'm open to any suggestions that would work.  I've been 
>>handed this challenge on my first day on a new job... :/
>>
>>Thanks,
>>
>>---sambo
>>
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> 

I believe you will also have to set externip in sip.conf to you public ip address.

Then you have allow rtp packet thru the fw and have them natted without altering 
the ports. You should then be able to call out and have sound.
When you call in you should get answered but probably wont have sound until the 
inside phone starts sending rtp packets.

HTH,
Steve
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