> > Just installed the new 4-port FXO card and moved two pstn lines from > existing x100p cards to ports on the FXO card. All zapata.conf entries > that were functional on the x100p's were copied to the new FXO channels > (including callprogress=no). > > Observations thus far: > 1. asterisk will spontanously decide a pstn call has arrived, and ring > the sip phone designated in the dailplan. Verified callprogress=no > is in place, and monitoring the two pstn lines with an external > analog phone (with line lamps) indicates no one was on the phone > and no ringing actually occurred.
I saw this a few times today on my X100P.. Problem in zaptel code perhaps.. > 2. Incoming CallerID seems to be slightly less reliable on the FXO > compared to the x100p. (Eg, there "seems" to be more cases of the > callerid showing up as "asterisk".) Yes I see errors more here too.. > 3. The echo issues that have been so well documented with the x100p's > seem to be identical on the new FXO card. > 4. Incoming pstn calls that either go to IVR menues or VM do not properly > sense disconnect supervision. Again, monitoring the pstn line via the > LEDs on an analog phone "does" indicate approximately .5 second of > no-battery (disconnect). After the disconnect, * does not release the > pstn line which then causes dial tone from the Central Office to be > recorded in the VM, etc. The CLI does not indicate a hangup until > _after_ the sip phone hangs up. MAJOR Problem, and reported to support. > I worked with Mark today and this problem is now fixed in the CVS. > Additional testing is being conducted. Anyone seeing these same problems? > > Rich > Scott _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
