Thanks, Mark!
Here's a summary of what one needs to do in order to get NAT working with Asterisk. Please note that I have a Cisco ATA-186, and your experience may be slightly different based on the equipment you're using. You'll need to have a CVS updated version of Asterisk as 2003-03-06 ~2:00 PM EST.
NOTE: This currently works for outbound calling only, not inbound. In other words, calls from Asterisk to your NAT-translated device will not make it through.
Configs in Asterisk:
sip.conf:
Add the line "nat=1' to any users/friends/peers that you expect to be coming from behind a NAT device. I have one client behind NAT, and here is what that that peer looks like:
[2410] type=friend username=2410 secret=somepasswordhere host=dynamic context=intern canreinvite=no nat=1
On your Cisco ATA-186:
Set your IP address information as usual (use DHCP, or static, whatever your site requires)
UID0: [your UID]
PWD0: [this UID's password]
UseSIP: 1
SIPRegInterval: 240
GkOrProxy: [ip address of your Asterisk server]
Gateway: [ip address of your Asterisk server]
ConnectMode: 0x00460400
OutBoundProxy: [ip address of your Asterisk server]
The ConnectMode flags are used in v2.14 and v2.15 to "re-register" phones with the correct data. See http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/atarn/186rn214.htm#xtocid17 for details.
That should be all you need to get outbound calls working in their most basic sense.
JT _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
