I originate calls with call-files.
The call file calls a local SIP peer.
After the called person picks up the phone i jump
to an extension in the dialplan to dial out a number specified in the dialplan.
ASTERISK ---> calls local SIP local SIP answers ASTERISK ----> calls external number specified in call file via zap by passing it to an extension in the dialplan that just dials
It works fine. my problem is, that the local SIP user can’t find out if the number was BUSY, CONGESTED or anything else.
Asterisk just hangs up the SIP-Channel without playing the BUSY signal.
Is there a way to make asterisk go on in the dialplan after the failed call, so the local SIP user would know what was going on? I tried to add the “g” option in the dialplan for the Dial command that calls out, but it doesn’t seem to work.