[ASTERISK-65] sig_analog: Allow simple switch to time out to dialplan

← All Issues || New Issue →

Sort Normal || Priority

Details

If the initial dial tone times out, e.g. here: https://github.com/asterisk/asterisk/blob/master/channels/sig_analog.c#L2254 then instead of just disconnecting to reorder, if the "t" extension exists in the channel's context, then execute that extension.

This would allow for implementation of several telephony behaviors, including:

• warm line (Direct Connect Originating with Delay)
• permanent signal intercept

Comments

You must be logged in to leave a comment.