[Asterisk-Users] No rering on misoperation on SIP ATA

[Johann Steinwendtner]

Hello !

Following scenario:
Party A: SIP Analog Terminal Adapter Grandstream HT486 (analog phone)
Party B: any other external PSTN set
Asterisk 1.0.9

Party A calls external party. Call is established. Party A presses the
flash key and goes on hook.
The external Party still gets Music on Hold. No disconnection.

I would have expected that Party A would rering.
Is this a problem of the Grandstream Adapter or is this a problem of
Asterisk ?

Hans