[Freeswitch-users] 30 Second call drop.

[Paul]

Hi guys,

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

My scenario is as follows:

Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun)
Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)

Main Network (where Freeswitch resides): 10.0.0.0/24

My sip gateway/proxy: 10.0.0.40 (kamailio)
Freeswitch: 10.0.0.34

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
up.

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
192.168.1.0/24 via 10.8.0.0/24)
If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 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 10.0.0.34 (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!