Hi Everyone,
I am facing a dilemma here. If I port forward 1-2 to my first
Asterisk server which sets behind pfSense v1.2.3 then I have two way audio.
If I remove it I don't have any audio but call establishes.
Now, I have a second server, so I am stuck with what to do on the NAT. I
On Fri, Jan 14, 2011 at 1:55 PM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am facing a dilemma here. If I port forward 1-2 to my first
Asterisk server which sets behind pfSense v1.2.3 then I have two way audio.
If I remove it I don't have any audio but call establishes.
On Fri, Jan 14, 2011 at 11:55 AM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am facing a dilemma here. If I port forward 1-2 to my first
Asterisk server which sets behind pfSense v1.2.3 then I have two way audio.
If I remove it I don't have any audio but call establishes.
Now,
On Fri, Jan 14, 2011 at 3:34 PM, David Burgess apt@gmail.com wrote:
On Fri, Jan 14, 2011 at 11:55 AM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am facing a dilemma here. If I port forward 1-2 to my first
Asterisk server which sets behind pfSense v1.2.3 then I have two way
I have not worked with Asterisk or SIP at all, but it sounds like what you need
is a combination of sipproxd to get past the NAT issues and some sort of load
balancer like SER or Ultra Monkey to round-robin (or whatever) the two Asterisk
servers. So that you'd wind up with:
asterisk1 -*
doing that SIP will broke
On 11-01-14 04:18 PM, Jason C. Taylor wrote:
I have not worked with Asterisk or SIP at all, but it sounds like what you need
is a combination of sipproxd to get past the NAT issues and some sort of load
balancer like SER or Ultra Monkey to round-robin (or whatever)
Simple solution is to limit the RTP port start and end in each Asterisk
server and use those ports with NON STATIC port setup in outbound NAT and
all should be fine. Thanks for the suggestions.
On Fri, Jan 14, 2011 at 4:21 PM, Francois-Alexandre St-Onge Aubut
fst-o...@idsmicronet.com wrote:
