On Fri, Nov 19, 2010 at 8:23 AM, Benoit Chabrier <[email protected]> wrote: > Thanks Alejandro, you were right it was just a NAT problem ! i add a > stun server in the phone configuration and it works :) >
Cool. Also Asterisk SIP conf file has some NAT settings as well that you can play with and perhaps do away with the stun server config in the phone. Here is a great article that explains in detail the issues with SIP and NAT: http://www.voipuser.org/forum_topic_7295.html > 2010/11/19, Alejandro Imass <[email protected]>: >> On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier <[email protected]> wrote: >>> Hello, >>> >>> I have a Sip phone (Siemens C470IP) which works perfectly with -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
