Hi Lloyd, as you are using Aastra phone 9143i behind NAT, plesse enable "rport" under "Network" settings of Aastra phone,and restart the phone, most likely you should be able to bet back to work unless your SMC Gateway really sucks. Please advise if this helps. Chris On Fri, Apr 9, 2010 at 3:40 PM, Aloysius Lloyd <lloyd.aloysius@gmail.com>wrote:
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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