[Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue

[Peter Spinato]

I changed the external profile to port 5060 - I can see it using that
profile and it still sends out the local IP at the RTP IP . even though
ext-rtp-ip is set to the public one .?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Saturday, December 17, 2011 10:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue

 

Yes, send it to the freeswitch server port  5080

Or if you don't need the internal profile, remove it and set external to use
port 5060.


-Avi

 

On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <peter at spinato.ca> wrote:

The calls go to internal profile - I guess case the OpenSIPs server is local
lan - is there a way to force that calls to external profile as I find calls
that hit that profile load the ext-rtp-ip ip and work.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Saturday, December 17, 2011 11:07 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue

 

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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Checked by AVG - www.avg.com
Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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Checked by AVG - www.avg.com
Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11

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