[Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue

[Avi Marcus]

Someone familiar with this might be able to answer right off the bat, but
if you pastebin a SIP trace (and an fs_cli log for completeness) , the
problem should become apparent.

Are you using the internal profile for all the calls? If you use the
external.xml profile, you might need to set your ext-rtp-ip in that file,
too. Seeing a trace will tell (almost) the whole story.

-Avi


On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter at spinato.ca> wrote:

> All,****
>
>   Hopefully someone can assist me - I'll gladly give $50 to the person who
> helps me fix the issue - I have an OpenSIPs server configured as a load
> balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS)
> that receives the call and forwards it to my Freeswitch Server for an IVR.
> When I had the call just routing to the Freeswitch server I got the audio
> working by setting the ext-rtp-ip to the public IP.  Now that I route the
> SIP call through the OpenSIPs server there is no audio - I'm guessing it a
> NAT issue as always.****
>
> ** **
>
> Both the OpenSIPS and FreeSwitch server have an internal private IP -
> OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2****
>
> Both servers also have a public IP that routes to the internal IP OpensIPS
> = 47.1.1.1 and Freeswitch = 47.1.1.2****
>
> ** **
>
> Call gets received by the OpenSIPs via the external IP - routes the call
> to the internal IP on the Freeswitch server which answers the call - but no
> audio - I think the FreeSwitch is trying to route the RTP Audio via its
> internal private IP instead of the public IP of 47.1.1.2.  Not sure if this
> is the real issue or how to configure it route RTP properly  ... all help
> is appreciated.  $50 Paypal to whoever fixes this for me!  Thanks****
>
> ** **
>
> -Peter****
>
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