I have a FreePBX 18.104.22.168 and asterisk 16.15.0, I have pjsip extensions and a sip trunk, randomly when I call, I have only one way audio, they hear me but I don’t hear anything.
I thought it was a nat problem and I have configured the trunk as follows.
type = peer
secret = ECOxxxxxxxxx
qualify = yes
progressinbound = yes
nat = force_rport, comedy
insecure = port, invite
host = x.x.x.x
fromuser = 9xxxxxxxx
fromdomain = x.x.x.x
externrefresh = 10
disallow = all
defaultuser = xxxxxxxx
allow = alaw & ulaw & gsm
In asterisk sip settings → nat settings I have configured the subnet in which the terminals are and the public IP of the connection
in audio codecs they are labeled ulaw, alaw, gsm, g726, g722
The ports for rtp are 10000-20000
I have tried with the rtp set debug on in one of the calls if audio in one sense but I do not see anything that clarifies it, it seems that there are connections from the pbx to the trunk operator, any ideas? What do you suggest, what can I try to find the fault?
There is a tp-link wr-841N router at the output and it does not have the rtp ports open to the private IP of the switchboard. would it be convenient to open them?