Hi! > > PBX*CLI> sip show subscriptions > > Peer User Call ID Extension Last state > > Type Mailbox 192.168.13.114 222 3c26707958d > > ...@default Idle dialog-info+xml <none> 1 active SIP > > subscription > > The phone is behind natted router on a private IP, the PBX is on a > public IP; could this private IP in the subscriptions be the problem?
Not sure, haven't seen that before. Anyone? Do add a "SIP SET DEBUG IP 192.168.13.114" to the game and see what is happening. What does the result of "route -n" look like? What does "SIP SHOW PEER 223" say concerning "Status" and "Addr->IP", can 223 be called? In general: You will want to accompany nat=yes with canreinvite=nonat (or canreinvite=yes). You could also turn on STUN in the snom and then switch to nat=no in Asterisk. Philipp -- _____________________________________________________________________ -- 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
