[Freeswitch-users] Sip phone lose communication appears registered

[Fernando Flórez]

It worked! Thanks! I went a little further and enabled this for all registered clients and not only nat’d.

Thanks,

On Sep 2, 2014, at 6:16 AM, Steven Ayre <steveayre at gmail.com> wrote:

> I assume your clients are behind NAT? UDP is connectionless so the port mappings that are opened timeout after a period of inactivity. It could be that this is happening so they cannot receive calls from FreeSWITCH.
> 
> Try setting http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping or configuring your SIP client to send keepalive packets.
> 
> Steve
> 
> 
> On 2 September 2014 01:31, Fernando Flórez <fernando.florez at gmail.com> wrote:
> Hello,
> 
> Just setup an FS test server on amazon ec2. Everything works flawlessly but after a while sip phone arent reacheable even as they seem registered on FS and are able to do outbound calling.
> 
> Any idea of what could be happening? If i restart the sip phones every starts working again for a litrle while.
> 
> Thanks,
> 
> Sent from my iPhone
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