[Freeswitch-users] 30 Second call drop.

[Anthony Minessale]

Hi,

Is it only during voicemal?  If so, Try this:

http://wiki.freeswitch.org/wiki/Variable_record_waste_resources

Otherwise turn on sip trace with "sofia global siptrace on" and "console
loglevel debug" and post it.





On Thu, Jul 18, 2013 at 10:54 AM, Paul <pasha at prosperity4ever.com> wrote:

> 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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> http://www.freeswitchsolutions.com
>
> 
> 
>
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-- 
Anthony Minessale II

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