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 <[email protected]> To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]>, Date: 02/07/2013 08:39 AM Subject: [asterisk-users] Asterisk calls between 2 private networks Sent by: [email protected] 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 ! -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
