what is the lowest value I can use for sip-force-expires ?
Yes Asterisk config using the qualify=yes.I will modify the Registration time out by 50sec and see how it is behaving.ThanksLloyd
On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <david.ponzone@gmail.com> wrote:On most low-end routers, the NAT table will expire UDP translations after 60 sec.Did you configure your phones to send a NAT keep-alive every X seconds, with X < 60 ?You can also use sip-force-expires on the FS side.In your Asterisk config, do you use qualify=yes ?David Ponzone Direction Techniqueemail: david.ponzone@ipeva.frtel: 01 74 03 18 97gsm: 06 66 98 76 34Service Client IPevatel: 0811 46 26 26Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 08/04/2010 à 23:00, Aloysius Lloyd a écrit :Hi All,I am having lots of problem with NAT and FreeSWITCH.FreeeSWITCH running Public IP.Phones Connecting to FreeeSWITCH through NAT. Cable Provider Router/Modem : SMC Gateway.The Phone Losing the Registrations. The Environment working for Asterisk without any problem.
Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make FreeSWITCH work.How to solve this problem.
Thanks,
Lloyd_______________________________________________
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