James Lamanna schrieb: > If you've used Linksys phones against recent Asterisk 1.4.x you may > have noticed > that they may drop registration for a quick bit and then go back to being ok > if your phone is behind NAT. > If you turn Asterisk's sip debug information on, you'll probably find > errors like these in your logs: > > NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce > received from '"999911" <sip:[email protected]>' > > I believe I have determined that this is caused by a bug in the > Linksys firmware that is related to the NAT Keep-Alive packets. > Because recent Asterisk 1.4.x's do not establish a SIP dialog for > NOTIFY requests, the "489 Bad Event" > replies were going back to the wrong address if your phone was behind NAT. > This lack of reply would cause the next REGISTER message to use the > same nonce as the previous REGISTER, > resulting in the "stale nonce" errors and temporarily dropping > registration. I've also seen the lack of response to > the NAT keep-alive cause the phone to stop being able to register > completely as well. > > Below I've posted a patch that responds with a 200 OK to these > keep-alive requests, and I believe > also solves the temporary loss of registration problem, though more > testing in different environments > for those who experience this problem would be greatly appreciated. > > The patch is against 1.4.32. > > -- James > Hello,
you also just could set the NAT KEEP ALIVE MESSAGE on Ext 1 from $NOTIFY to $OPTIONS and make this extension in your default context: exten => s,1,hangup and you also would get a 200 ok for the keep alive package. IMHO a stale nonce would only occur when a user tries to register faster than 3600s cause of the register timeout used in asterisk. Maybe you should also try to set a higher register timeout on your phones. but i dont have an 1.4 system running, only around 2k of linksys phones on a 1.2.40 and 300 on 1.6.1.18 and i dont see this problem there. best regards. steve -- _____________________________________________________________________ -- 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
