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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
