Goodmorning/evening to everybody, im having quite a weird issue with my PBX, when i make an inbound call (that dials to an extension dialing in the PSTN) no audio is present unless i use the D() option to play a DTMF tone to the calling party, does someone know a possible solution to this? Thanks in advance.
Which channel driver (seems unlikely this would happen on analogue or ISDN, but that still leaves SIP and PJSIP as common ways of connecting to the PSTN, and IAX as a less common way).
If you look at the SIP exchange (see https://wiki.freepbx.org/display/SUP/Providing+Great+Debug), is Asteirsk sending the correct, public, media address in the SDP? Does the router require outgoing traffic to enable inbound RTP? Is the near side of the call reaching an actual human, who could send speech in the same direction as the DTMF?
Which version of FreePBX and which of Asterisk?
Confirm that your firewall is forwarding UDP ports 10000-20000 to the PBX.