I try config pjsip trunk with ip auth but i can call but no sound on other side from extension,
Freepbx version 17
eth0 ip public 103.x.x.x this route via vlan 103
eth1 ip private 10.10.x.x this toute via vlan 10 and is connect to provider sip trunk
domain hallo.x.x
number head +6777xxxxx
extension connect public using domain hallo.x.x, if extension to extension is no problem, but if extension to phone number on phone number side no sound
Set codecs for the trunk to allow only alaw and ulaw. Also confirm that Direct Media is set to No.
If no luck, please report: do incoming calls have two-way audio? If you record an outbound call, does the recording have both sides? Can the remote party hear? Does calling a landline have the same problem?
At the Asterisk command prompt, type pjsip set logger on
Make a failing call, paste the Asterisk log for the call at pastebin.com and post the link here.