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.
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?
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 ?
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.
How to trace call ?
We told you here to not set nat=yes. Why did you change it?
I have configured FreePBX on my pc and it’s working on GSM Gateway with Sim Cards but I want now to connect fixed (home) number on it but don’t work.
I have 2 Lan cards (eth0 integrated) and (eth1 non-integrated)
eth0 it’s connected internet connection with IP address 192.168.88.250
eth1 it’s modem of ISP with IP address 192.168.47.178
The IP Address of HOST (phone number) it’s 192.168.15.70
When I try to ping 192.168.15.70 I don’t get any request ?
This is my routing tab…
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.
Is this correct only Outgoing:
Everything it’s okay now the Provider used those RTP Ports: 16000:32000 :).
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