Assistance Required: SIP Trunk Integration with FreePBX and Networking Configuration

Hi All,

I am new to this platform and I apologize in advance for any gaps in my networking knowledge.

I have successfully installed FreePBX on a local server and set up internal communication using desk phones. The internal call functionality is working as expected.

Recently, I purchased a SIP Trunk from a local service provider. They have installed a Huawei AR617vw Router at my premises and provided the following configurations:

  • Pilot number: 11763XXXX
  • Number series: 11763XXXX – XXXX
  • Router LAN IP: 192.168.1.100 (to be used as the default gateway for SIP trunk calls)
  • ISP end SBC Block: 12.12.12.88/29
  • SIP Proxy IP: 12.12.12.89 (reachable through the LAN IP 192.168.1.100)
  • However, I am facing some challenges with the integration and would appreciate your assistance.

Currently, my FreePBX server and desk phones are connected to a fiber connection from another ISP, which is used for internet access. I need to bridge the two routers and configure the Huawei AR Router to handle SIP Trunk communications, while the other router continues to provide internet access.

Here are the specific issues I am encountering:

  1. Connectivity Question: Should the FreePBX server and desk phones be directly connected to the SIP Trunk Router (Huawei AR617vw) to make calls? If so, I am unable to access the FreePBX server remotely, as my computers are connected to the other router, which provides internet connectivity.

  2. Network Integration: Is there a way to configure both routers so that the FreePBX server, desk phones, and my computers are on the same network, allowing the SIP Trunk to function alongside internet access?

  3. Audio Issue: When connecting the server and phones directly to the SIP Trunk Router, I can make calls, but the recipient cannot hear my voice. Additionally, inbound calls disconnect automatically after a few seconds, even though the internal desk phone still shows the call as active.

I would greatly appreciate any guidance or solutions you can provide to resolve these issues.

Thank you in advance for your assistance.

You need 2 IP addresses on FreePBX. One for each network. Connect FreePBX server to both networks. Either with 2 network ports or 1 network port and a switch.

IP 1 = whatever your regular LAN IP is for internet access.
IP 1 default gateway = whatever your regular LAN gateway is

IP 2 = 192.168.1.101 (or whatever in that SIP router range)
IP 2 = No default gateway

Then, after that, since the IP of the SIP Provider Server is in a different subnet than the network of the Sip Providers LAN (which in your case it is), then you need a route to the SIP via the LAN of the SIP router, like this:

/etc/sysconfig/network-scripts/route-eth1
12.12.12.88/29 via 192.168.1.100 dev eth1

(change eth1 to whatever it is in your case)
and reboot to make the route take affect.

Now, any traffic going to 12.12.12.88/29 will go through the SIP router, and all other traffic will go through your regular lan.

That should also fix your audio issues. Your audio is probably currently trying to use the wrong network so it gets lost.

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