asterisk users - Mar 2003 - NAT, SIP and ATA-186

(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@lists.digium.com
>From: John Todd <jtodd@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

Hi all,
Some problems again.
I am testing a setup like this:

phone-usb-*SIP-eth-SIP*SIP-NAT-Internet-SIP*
i.e. just trying to call outside my network.
When I reach asterisk on the enge of the internal net everything works fine
but when I make further connection to * in the internet I get connected and
seems that DTMF
messeges are passing correctly so basically I can go around menus but I
DON'T HEARanything.
I only know that I am connected by looking at * console.

Any hints?

Thanks in advance.

Best regards,
Krzysztof Bujak



==========================================================================Ta
wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system
antywirusowy na serwerze IT Form.