Im using freeswitch with dozen aastra phones (6757i and 6739i) for about a year. They work perfectly, all the time. If you're behind NAT, it would also be possible that the NAT device is the source of your calls dropping. Erik Van: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Namens Aloysius Lloyd Verzonden: donderdag 10 februari 2011 15:25 Aan: FreeSWITCH Users Help Onderwerp: Re: [Freeswitch-users] Aastra phone registration lost Tim, Can you share your success stories FreeSWITCH and Aastra. Aastra Phones Behind the NAT? In my case Aastra phones registration not a problem. But calls drooped every 60 sec ... in the same environment Linksys and Polycom works perfectly. How stable the Aastra phones with FreeSWITCH system. TIA Lloyd On Thu, Feb 10, 2011 at 6:23 AM, David Ponzone <david.ponzone@ipeva.fr<mailto:david.ponzone@ipeva.fr>> wrote: Philippe, can you confirm there is no NAT Keepalive feature in the Aastra ? That's quite weird. David Ponzone Direction Technique email: david.ponzone@ipeva.fr<mailto: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<http://www.ipeva.fr> - www.ipeva-studio.com<http://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 10/02/2011 à 11:39, Philippe Sultan a écrit : Tim, That helped a lot, thanks! More comments inline. On Thu, Feb 10, 2011 at 3:18 AM, Tim St. Pierre <fs-list@communicatefreely.net<mailto:fs-list@communicatefreely.net>> wrote: I'm wondering about these values: sip registration retry timer: 10 sip registration renewal timer: 15 sip registration timeout retry timer: 10 sip registration period: 60 sip rport: 1 They seem a bit short, especially on a LAN. Do you really need your phones to re-register every 45 seconds? Indeed, I don't. But need a mechanism to keep NAT sessions alive in case phones are being deployed behind NAT boxes. It seems like transporting SIP over TCP from the phone, plus using TCP keepalives will solve that issue though. I also have Thomson ST2030 phones, and they send SIP OPTIONS packets every 50 seconds for that purpose (all handled by the SER proxy). Try setting sip registration period to 600 That's 10 minutes. sip registration renewal timer can be a bit longer - 30 seconds maybe, or 60 if you are at all worried about reliability. Here is what I put in the aastra.cfg file from your suggestions : sip registration period: 3600 sip registration renewal timer: 3565 And that does the trick, that is, the phone registers every 45 seconds. I'll try to update those values to have a larger registration period, rely on TCP keepalives in order to maintain NAT sessions, and have all that NAT handling in SER. Here is the topology : Aastra ---(sip/tcp)--- SER ---(sip/udp)--- FS ST2030 ---(sip/udp)--- SER ---(sip/udp)--- FS Thanks again! Philippe _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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