About this case: the old SIP channel behaves correctly. On Sun, May 17, 2020 at 2:44 AM Saint Michael <vene...@gmail.com> wrote:
> My phone is located behind a NAT, 172.16.0.0/21. > Asterisk 16 is on a public IP. > PJSIP has the config below: > force_rport=yes > direct_media=yes > disable_direct_media_on_nat = yes > direct_media_method=invite > > But when I send a call I see the RTP being sent to my private address, vs > the public IP. This only happens when Asterisk has dialed the call to > another carrier. If instead of Dial I choose Answer() and MusicOnHold, then > the RTP gets shipped to the right address. > This is a sample of the erroneous behavior: > Got RTP packet from XX.XX.XX.XX:17510 (type 00, seq 024786, ts 017440, > len 000160) > Sent RTP packet to 172.16.7.254:50798 (type 00, seq 010736, ts > 017440, len 000160) > > 172.16.7.254 is my private address. > What am I missing? Should I open a bug? > Asterisk should never, ever send RTP to a private address when Asterisk > itself is on a public IP. > Before you ask, the dialplan is 3 lines, > '_X.' => 1. NoOP() > 2. Dial(PJSIP/${EXTEN}@carrier) > 3. Hangup() > > > > > > > > > > > >
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