[asterisk-users] Asterisk calls between 2 private networks

[Kevin Larsen]

For the phone on the public network. you might need to set canreinvite=no. 
My guess is that if you listen really closely you would have about a 
quarter second of audio before it cuts out. Whenever I have had this 
happen it is because the packets didn't know how to reroute from the IP 
address of the Asterisk server to the IP address of the phone. My guess is 
that your network has the proper pathing to send the packets into the 
servers IP address but can't redirect them to the other IP addresses.

If it works, you can leave canreinvite on for phones in the private 
network, but any that will register to the public network should have it 
set to no.

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



From:   Frank <frank at efirehouse.com>
To:     Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users at lists.digium.com>, 
Date:   02/07/2013 08:39 AM
Subject:        [asterisk-users] Asterisk calls between 2 private networks
Sent by:        asterisk-users-bounces at lists.digium.com



My apologies if this topic was already discussed in the past.

Here is my scenario:
Network A - 192.168.1.0
1 Asterisk
1 Digium phone
Router does NAT from the public IP to asterisk, and forward ports 
5060tcp/udp and 10k-20k udp

Network B - 192.168.1.0
1 Digium phone, registering to the public IP of network A


My SIP.CONF has:
nat=yes
localnet=192.168.1.0/255.255.255.0
externaddr=public_ip_of_network_a
directmedia=no



The Digium on network B can register. I can see it when I do "sip show 
peer xxx". When the phones are calling each other, the signaling is 
working. They ring. But when they pick up, there is no audio, in any way.

Has anyone ever worked on the same configuration, and had success ?
If yes, I'd love to hear your story and check your configuration.

Thanks !

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