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 !
Kevin Larsen
2013-Feb-07 14:47 UTC
[asterisk-users] Asterisk calls between 2 private networks
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 ! -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130207/1b9754ee/attachment.htm>
A J Stiles
2013-Feb-07 15:46 UTC
[asterisk-users] Asterisk calls between 2 private networks
On Thursday 07 February 2013, Frank wrote:> 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=noMy (lazy) solution to this problem was to throw hardware at it ..... Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- AJS Answers come *after* questions.
Hans Witvliet
2013-Feb-08 07:38 UTC
[asterisk-users] Asterisk calls between 2 private networks
-----Original Message----- From: Carlos Alvarez <carlos at televolve.com> Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Asterisk calls between 2 private networks Date: Thu, 7 Feb 2013 10:36:36 -0700 On Thu, Feb 7, 2013 at 10:26 AM, Frank <frank at efirehouse.com> wrote: AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. Adding more points of failure and more devices to maintain without any real benefit is always the wrong thing to do. IAX is also flaky as hell. -- _____________________________________________________________________ Carlos, with regards to your comment about IAX, where can i find your bug-report?