On my SIP carrier, I register to a proxy "sipconnect.dal0.cbeyond.net" which ends up being 192.168.22.212 (They supply a T1 bundle)
#sip show peers Name/username Host Dyn Nat ACL Port Status <snip> Generic-8174691929/817469 192.168.22.212 N 5060 OK (41 ms) Yesterday, they had a problem with their primary server and reverted to a backup server for about 5 minutes. As chance would have it, I received a call to one of my DIDs just before and just after the switch. As you can see below, the first call was on their primary server and the "Found peer" finds the Generic-8174691929 peer I have set up. Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.22.212 : 5060 (NAT) Found peer 'Generic-8174691929' <<<<<<<<<<<<<<<<<<<< Found RTP audio format 0 Found RTP audio format 100 However, just after they changed to the backup service, I received the call below. Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.25.212 : 5060 (NAT) Found no matching peer or user for '192.168.25.212:5060' <<<<<<<<<<<< Found RTP audio format 0 Found RTP audio format 100 Since it was a different IP address, it found no matching peer and failed to find a valid context to send the call to. How should this be addressed in Asterisk to allow for such an incident? Bill _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
