[Asterisk-Users] This should work right??? Any caveats that youguys know about?

Ashish Raikwar akumar at uf4.net
Tue Aug 2 06:07:24 MST 2005


hi

Solution of your problem is in this article which i am pasting from an
online document....
A SIP phone usually registers with a SIP proxy. This message comes from the
inside of the NAT to the server on the outside. Now, there's an open
connection in the NAT device. As soon as there's no more packets on that
connection, the NAT device cancels the connection and forgets all about it.
The trick is to keep the packets flowing, forcing the NAT device to keep the
connection open.

Some phones send NAT "keep-alive" packets by themselves. X-lite and Sipura
have this feature. If the phone can't do it, configure Asterisk to do it.
Setting "qualify=yes" in the [peer] section for this device, Asterisk starts
sending packets to the device, keeping the NAT connection open. You will
also be able to see some timing for packets between Asterisk and the phone
when you do "sip show peers" at the CLI.

Now, when Asterisk wants to place a call to the phone, the NAT welcomes the
packets and forwards them happily to your phone.

Conclusion: If Asterisk is on a public IP address and your phone is on the
inside of a NAT device, we need to keep the NAT connection open by
frequently sending dummy packets between the devices. This will keep the
connection open for incoming calls.


----- Original Message -----
From: "brent clements" <bcasterisktechlist at gmail.com>
To: <Asterisk-Users at lists.digium.com>
Sent: Tuesday, August 02, 2005 4:03 AM
Subject: [Asterisk-Users] This should work right??? Any caveats that youguys
know about?


Hello, long time lurker, first time writer....


We have the following set up

ITSP
|
|
Internet
|
|
Cisco 2600
|
|
Switch----Asterisk Server running 1.0.9(has public ip)
|
|
Cisco 515e Pix Firewall running Pix OS 5.3(run's a class c 1-to-1 nat and
pat)
|
|
Grandstream GXP-2000(run latest fw from grandstream site 1.0.1.9)


The grandstream registers with the public asterisk server fine. I even
see one of the dynamic nat addresses being assigned.

The Pix Firewall has sip fixed up and all VOIP related ports are wide open.

This is the issue: We can make outgoing calls, but we can't receive
calls when the grandstream is behind the firewall If we move the
grandstream in front of the pix and give it a public ip, everything
works fine.

What is even wierder is the fact that one of our network users who is
behind the pix firewall can use ATT's VOIP service just fine.

Are there any things I should be looking for? In general is the setup
above pretty common? I've looked through the Wiki and searched google
many times but nothing that can give me any pointers.

Thanks!
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