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. 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".) 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. Additional testing is being conducted. Anyone seeing these same problems? Rich _______________________________________________ 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
