[Asterisk-Users] NAT, SIP and ATA-186
John Todd
jtodd at loligo.com
Sun Mar 9 16:19:15 MST 2003
(note: this has only been tested with my Cisco ATA-186 v2.15 and SIP)
Update: inbound calls now work, again thanks to Mark's banging on the
code. I am able to receive calls on my ATA-186 with the settings
below, behind an Apple Airport NAT/PAT translator, as of CVS updates
late afternoon yesterday (2002-03-08). No special changes or holes
were created on the Airport to allow for the translation tricks.
I'm interested to hear if anyone else has success with SIP behind
NAT/PAT using similar or hopefully different SIP hardware.
JT
>Date: Thu, 6 Mar 2003 13:29:20 -0800
>To: asterisk-users at lists.digium.com
>From: John Todd <jtodd at loligo.com>
>Subject: NAT working outbound with Asterisk and ATA-186 phones
>
>
>
>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
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