[asterisk-users] SIP & NAT

[Eric "ManxPower" Wieling]

According to sip.conf.sample the answer is...well, I guess you can look 
in /path/to/src/asterisk/configs/sip.conf.sample and see for yourself.

Mike Hammett wrote:
> If I have several local networks, can I specify that?
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
> "ManxPower" Wieling
> Sent: Thursday, March 29, 2007 1:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIP & NAT
> 
> Mike Hammett wrote:
>> I hate SIP.  The only reason I'm doing this is that its cheaper than
>> deploying the server to a colo facility.  My provider has given me a
>> non-standard IP block, so I can't do typical routing.
>>
>>  
>>
>> I have Asterisk server <-> MT w\NAT <-> PPPoE <-> MT <-> Provider.
>>
>>  
>>
>> I setup a dst-nat on 5060 to the Asterisk box.
>>
>>  
>>
>> Audio from Asterisk -->  PSTN works great.  Audio Asterisk <-- PSTN does
>> not.
> 
> That would be expected since you did not forward the ports used for RTP. 
>   See /etc/asterisk/rtp.conf  A sample is in the Asterisk source.
> 
> Did you also set localnet= and externip= options in sip.conf [general].
> 
> SIP works just fine with NAT if you have it correctly configured and 
> your server is on a static IP address.