[Freeswitch-users] 30 Second call drop.

[Tim Meade]

I have found in the past that a 30 second call drop is normally a firewall issue.

If you have bypass media on:

<action application="set" data="bypass_media=true"/>

Then at almost exactly 30 seconds FS will take you out of the media path.  If your 2 endpoints cannot see each other due to a firewall (or nat) then the media will drop.

In our case it's been firewall settings every time.

Hope that helps.

Tim

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul
Sent: Thursday, July 18, 2013 11:54 AM
To: freeswitch-users at lists.freeswitch.org
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: 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!



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