[Freeswitch-users] 30 Second call drop.

Mick Stevens mickstevens at yahoo.com
Thu Jul 18 18:37:17 MSD 2013

Hi Paul,

In my experience there 's only one thing worse than trying to solve your own NAT/double-NAT/ALG issues, and that's trying to solve somebody else's....

But, in all seriousness, I use a similar setup to you, multiple FS instances sitting behind Kamailio as proxy/load balancer and have had no NAT issues after reading "FreeSWITCH 1.2, Chapter 12 - Handling NAT".

Alas, particularly without any protocol traces, there are no quick answers here, but like any good diagnostic process/engineer I would suggest you change one parameter at a time & retest, recording the results & moving forward/back a step until you find a configuration that works for you.

Seriously, download the book, worth the cover price alone for the NAT chapter & you'll not only be able to solve your problem but will have a greater understanding of the issue(s) and how to plan to avoid them in the future.

Rgds, Mick

 From: Paul <pasha at prosperity4ever.com>
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, 18 July 2013, 16:54
Subject: [Freeswitch-users] 30 Second call drop.

Hi guys,

After hours of googling and trying every different config options I need
some help!

My scenario is as follows:

Remote Network 1: (10.8.0.X/32 via openVPN tun)
Remote Network 2: (10.9.0.X/32 via openVPN tun)

Main Network (where Freeswitch resides):

My sip gateway/proxy: (kamailio)

I can fully use the VPN from both networks (ssh, web gui control, etc)

phones can register, however here are my problems:

When calling incoming DID none of the extensions ever get ring, it skips
to voicemail... when leaving a message voicemail cuts of after 30-31
seconds exactly. I am wondering if freeswitch isn't detecting my client
sending audio and interprets it as silence and therefore hangs up. The
voicemail message itsel does contain everything said right up to the hang

Sometimes I can call between extensions if they are on the same side of
the VPN subnet (say 101 calling 101 and they both are coming from via
If 100 is on 10.8.0/24 and 101 is on freeswitch just says user
is unavailable.

I'm suspecting this is a NAT issue, all the reading I have done talks
about "external sip and rtp ip" in my case my trunk hooks up from private
ip to private ip, it's the kamailio that's supposed to do the NAT (which I
have verified works fine, I can place outgoing calls through it from other
PBXs no problem with two way audio and working as it should be).

I'm not sure if there is a way to enable nat or not with freeswitch. One
thing I have done (as I've found this worked for some people) is set all
references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to (my
freeswitch eth0). There are no other adapters on the box.

I am hoping that as this is my first time setting up freeswitch that one
of you freeswitch experts can point me in the right direction :)

Thanks in advance guys!

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