[Freeswitch-users] Sip phone lose communication appears registered

[Steven Ayre]

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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