The problem is that client can connect to Asterisk server, but when trying to call – there is no voice coming, so I suggest the problem with RTP NATted session. Even if I call presetuped Echo-service to check if it should work at least in client-server flow – there is anyway nothing I can hear.
Tried different directmedia and nat setting – nothing helps.
asterisk is running on a virtual machine after firewall which do nothing then translating all traffic from/to outside. Local IP- 10.0.201.60
sip.conf
[general]
realm=PBX Server
useragent=PBX Server
sdpsession=PBX Server
externaddr=79.x.x.x:5060
externhost=external.host.com
udpbindaddr=10.0.201.60:5060
tlsenable=yes
tlsbindaddr=10.0.201.60:5061
tcpenable=yes
tcpbindaddr=10.0.201.60:5060
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
language=en
context=default
allowoverlap=no
transport=tls,tcp,udp
srvlookup=yes
allowguest=no
alwaysauthreject=yes
limitonpeers=yes
tlscertfile=/usr/asterisk/etc/certs/asterisk.pem
tlscafile=/usr/asterisk/etc/certs/cacert.pem
tlsclientmethod=tlsv12
;tlsciphers=EECDH+AESGCM:EDH+AESGCM:AES256+EECDH:AES256+EDH
tlscipher=ECDH:!3DES:!RC4:!ADH:!AECDH:!NULL:!eNULL
encryption=yes
[clients]
context=clients
type=friend
host=dynamic
qualify=100
;callgroup=1
;pickupgroup=1
call-limit=1
dtmfmode=auto
allow=opus,alaw,ulaw,g729,g723,g722,gsm
100
callerid=100
secret=12345678
directmedia=nonat
nat=force_rport,comedia
101
callerid=“Number 101” <101>
secret=87654321
nat = comedia
extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
[clients]
;Звонок на внутренний номер
exten => _XXX,1,Playback(beep,noanswer)
exten => _XXX,n,Dial(SIP/${EXTEN})
exten => _XXX,n,HangUp()
exten => 1720,1,NoOp(Starting playback service)
exten => 1720,n,Playback(beep) ; Let them know what’s going on
exten => 1720,n,Echo ; Do the echo test
exten => 1720,n,Playback(beep) ; Let them know it’s over
exten => 1720,n,HangUp()
Is there any suggestion what could be wrong? Trying to fight this second day – still no luck.
Thanks!