[Freeswitch-users] server to NAT'd server gateways

[Bote Man]

FYI, I have just moved the High Availability and keepalived pages to Confluence.

 

https://freeswitch.org/confluence/display/FREESWITCH/High+Availability

 

Bote

 

 

From: Sergey Safarov
Sent: Wednesday, 25 February, 2015 02:08
Subject: Re: [Freeswitch-users] server to NAT'd server gateways

 

1) Create VPN connection betwen Sip_B (client) and Sip_A (server);

2) On VPN interface configure static IP address;

3) Configure station on Sip_B and gateway on Sib_A;

4) Route call from Sip_A to Sip_B like you has configured routing call  Sip_B to Sip_A;

5) To allow FS listen on VPN IP when VPN connection is down cconfigure sysctl variable "net.ipv4.ip_nonlocal_bind = 1"

https://wiki.freeswitch.org/wiki/Freeswitch_HA

 

On Wed, Feb 25, 2015 at 5:37 AM, Robert Oldham <robert at oldhamtechnology.com> wrote:

I have two FreeSWITCH servers that I need to gateway calls between.
Sip_A is on a public network with a static IP address. Sip_B is on a
private network that is NAT'd. I am trying to pass calls from devices
connected to Sip_B, through Sip_A and on to another server. I am also
trying to pass calls from Sip_A into Sip_B. I don't have access to the
firewall at the location of Sip_B so I am not able to forward standard
ports into Sip_B.

I am fairly new to FreeSWITCH, so I'm not certain how to solve this
problem well. Many of the documents I have read online recommend using
ACLs and port forwards for this. However, the private network does not
have a static IP address that I can give access to and, as mentioned, I
cannot make port forwards. Instead, I have created an extension on Sip_A
and a gateway on Sip_B that registers using the extension on Sip_A. That
works fine for passing calls from Sip_B through Sip_A and on. Sip_B has
been maintaining a registered connection to Sip_A through the NAT'd
connection for several days now without trouble.

Not knowing a better solution, I have been searching for a gateway
configuration on Sip_A that would allow it to route calls back over the
registered extension from Sip_B. I have not found a way to accomplish it
yet. As an alternative, I have also tried without success to create a
passive gateway on Sip_A that the gateway on Sip_B could register with.

Any help is appreciated!

Thanks,
Robert



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