I am making outgoing calls from Asterisk (SIP) to the PSTN using FXO ports of the GXW4104.
The SIP call starts immediately instead of starting when the B party (on PSTN side) answers. As a result, the SIP call includes the ringing duration.
In addition, the SIP call succeeds and gives non zero duration event if the call is rejected e.g. after 15 sec ringing.
Is there any config option to establish the SIP call when the PSTN user answers ?
(NB: could not find any infoon this).