[Freeswitch-users] 30 Second call drop.

[Paul]

Hi Mick,

I hear ya on trying to figure out someone else's NAT issues ... I have
this weird feeling that it's not NAT though...

Only reason I threw my problem out there is just in case someone sees the
post and says "Oh yeah, been there, here's what worked for me".

I did buy the book on your recommendation so I will be reading through
chapter 12 tonight, see if I can figure this out.

Thanks for your input!

Paul

On Thu, July 18, 2013 7:37 am, Mick Stevens wrote:
> 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: 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!
>
>
>
>
> _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org http://www.freeswitchsolutions.com
>
>
> 
> 
>
>
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