[asterisk-users] SIP & NAT

[Mike Hammett]

I have a block of 20 IP addresses, so I can't really carve out /30s and
whatnot to route between locations.

 

Asterisk is the client.  I am doing Interop testing with some vendors before
I ship it out to a colo facility.

 

I have used the NAT setting with Asterisk as the server on the open
Internet.  Would it function similarly with Asterisk as the client?  We are
using IP based authentication.

 

I have more public addresses, but I'm unsure how to route that through so
the Asterisk box can use it.  Perhaps I will look into 1:1 NAT.

 

--Mike

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexander
Lopez
Sent: Thursday, March 29, 2007 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SIP & NAT

 

What do you mean by 'non-standard' IP block?

Is the Asterisk machine behind a NAT, or are only your clients?

Did you look at the nat setting sin sip.conf?

 

Do you have a static public address that can be routed to the Asterisk box?

 

 

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike Hammett
Sent: Thursday, March 29, 2007 11:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP & NAT

 

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.

 

Ideas?

 

 

 

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