[Solved]Can't hear each other on call

Hello everyone,

I’m having a problem with FreePBX when I call or anyone call me I can’t hear them and they can’t hear me too.

Someone told me that I have problem with RTP ports I’m using the ports from 10000-20000 how should I check them from FreePBX are they open ?

Internal calls does work fine?
You have to check on your router if you are forwarding 10000-20000 to the PBX.

@PitzKey,

Yes internal calls works fine.

this is the topology

P.S. Phone modem it’s connected directly to eth0 not over the switch.

In that thread you said that you can’t ping your gateway, is that resolved already?

Under SIP Settings have you added 192.168.15.0/24?

@PitzKey
Yes that threat has been resolved because I have route 192.168.15.70 to 192.168.47.1 so routed host to gateway and it’s working okay but now I have this second problem that when I call or when I recieve the call we can’t hear each other what should I do about this ?

Can you answer this?

Sorry,

You mean to Settings - Asterisk SIP Settings ?

Local Networks : 192.168.47.0/24, 192.168.88.0/24 and 192.168.15.0/24

Can you post your Trunk configuration?

Also, please provide a call trace.

Outgoing:

host=192.168.15.70
type=friend
trustrpid=no
sendrpid=yes&pai
qualify=yes
nat=yes
insecure=invite,port
dtmfmode=rfc2833
disallow=all
context=from-pstn
canreinvite=no
amaflags=default
allow=ulaw&alaw

Incoming:

host=192.168.15.70
type=friend
trustrpid=no
sendrpid=yes&pai
qualify=yes
nat=yes
insecure=invite,port
dtmfmode=rfc2833
disallow=all
context=from-pstn
canreinvite=no
amaflags=default
allow=ulaw&alaw&g726&g722

How to trace call ?

We told you here to not set nat=yes. Why did you change it?

Sorry the NAT it’s yes nat=yes ?

Please read carefully what @BlazeStudios posted there, once you followed the instructions, please post your updated config. Thank you.

Okay,

Is this correct only Outgoing:

type=peer
port=5060
host=192.168.15.70
nat=no
dtmfmode=rfc2833
context=from-trunk
allowguest=yes
allow=ulaw&alaw

Everything it’s okay now the Provider used those RTP Ports: 16000:32000 :).

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