Remote Extensions one way audio when make calls through SIP Trunk

Hello Everyone ,

I have an issue for remote Extensions when making external calls through Service Provider SIP Trunk as the following :

FreePBX Distro version is : 12.0.65 , Asterisk 11.17.1

The FreePBX have public IP ( Behind NAT ) SIP port 5060 and RTP ports from 1000 to 2000 are opened from the router and the Remote Extensions register successfully on FreePBX Public IP and calling internal extensions successfully and the audio streaming working in both ways .

Also internal Extensions calling outgoing calls through this service provider SIP Trunk very well , the problem appears only when remote Extension make outgoing calls through this SIP Trunk as the audio stream working only in one way.

Kindly find the SIP Trunk Configuration as the following :
host=xx.xx.xx.xx
type=friend
qualify=yes
insecure=very
canreinvite=no
nat=yes
disallow=all
allow=alaw

Asterisk Chan SIP settings as the following :

NAT =yes
IP Configuration = static IP
Override External IP = xx.xx.xx.xx " the local IP address for the FreePBX server however I know I should insert here the Public IP of the FreePBX but when making that the internal and remote Extensions can’t make calls through SIP Trunk "
Local Networks = xx.xx.xx.xx /255.255.255.0 " local subnets for the IP-Phones "

Kindly find Extension settings as the following :
deny=0.0.0.0/0.0.0.0
secret=XXXXXXXX
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=pai
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/150
[email protected]
permit=0.0.0.0/0.0.0.0
callerid=remote test <150>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

Any help please !!!

Could someone help me please ?

The ports should be 10000 to 20000 not 1000 to 2000. Give that a go and reply back :slight_smile:

Sorry I mean the ports opened from 10000 to 20000