[Freeswitch-users] Sip phone lose communication appears registered
steveayre at gmail.com
Tue Sep 2 15:16:08 MSD 2014
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.
On 2 September 2014 01:31, Fernando Flórez <fernando.florez at gmail.com>
> 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.
> Sent from my iPhone
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> Official FreeSWITCH Sites
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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