[asterisk-users] Asterisk calls between 2 private networks
kevin.larsen at pioneerballoon.com
Thu Feb 7 08:47:48 CST 2013
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 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:
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.
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