I could not find a SIP ALG Setting. I setup the sip-force-expires and ping for the user directory. Only one time registering then lost the connection. Here is the sofia profile internal status Call-ID: 7307ef8fa6044407 User: 202@abc.com Contact: "Mike Derouin" <sip:202@A.B.C.D:5060;transport=udp;srcadr= 192.168.1.51:5060> Agent: Aastra 9143i/2.5.2.30 Status: Registered(UDP)(unknown) EXP(2010-04-09 15:31:21) Host: TestSrv IP: A.B.C.D Port: 5060 Auth-User: 202 Auth-Realm: abc.com MWI-Account: 202@abc.com Please let me know how to fix this issue. Thanks Lloyd On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <david.ponzone@gmail.com>wrote:
I dont think there is a lowest value, but 30 seconds is reasonable in most cases. You can also add a ping parameter with value 30, in the user config with the variables. The result is a SIP OPTIONS sent to the phone every X sec. That's quite the equivalent of qualify=yes in Asterisk.
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Le 09/04/2010 à 03:31, Aloysius Lloyd a écrit :
what is the lowest value I can use for sip-force-expires ?
Thanks Lloyd
On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <lloyd.aloysius@gmail.com>wrote:
Yes Asterisk config using the qualify=yes.
I will modify the Registration time out by 50sec and see how it is behaving.
Thanks Lloyd
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 Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
*Ce 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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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