How to get each call to route through their own FXO ports in GXW41XX

Grandstream Networks GXW410x series is an FXO gateway that converts SIP/RTP IP calls to traditional PSTN calls and vice versa. FXO gateway is extremely useful if you are trying to venture into SIP-based calling, but is still on an analog PSTN technology.

As much as I want to talk more about GXW41XX, this article will be focusing on guiding any new users on how to configure GXW41XX to have calls routed through their own dedicated FXO ports. Here is a guide from Grandstream Networks on this, but it seems a little confusing for new users. You may follow my steps here instead.

Scenario:

Ext 101 can make outgoing calls via FXO port 1 and FXO port 1 all incoming calls will be routed to Ext 101.

Ext 102 can make incoming & outgoing calls via FXO port 2.

…. and the same goes on till FXO port 8.

The user is using the Grandstream UCM series along with GXW41XX.

Steps:

Stage 1 method –

Create each FXO line in GXW410X as own trunk in the UCM server. Each FXO should have its number.

For Inbound calls –

1. Settings > Channels Settings > Calling to VoIP > Unconditional Call Forward to Following > User ID: enter ch1: EXT1; ch2:EXT2; … ch8:EXT8
(For instance ch1:101;ch2:102;ch3:…)

2. Settings > Channels Settings > Calling to VoIP > Unconditional Call Forward to Following > SIP Server: set to ch1-8:p1 (profile for Acount 1);
3. FXO Lines > Settings > Port Caller ID Setting > Number of Rings Before Pickup: set to ch1-8:1;
4. In UCM, do the following –

Create 8 inbound routes with:
(1) ExtX DID as the pattern and
(2) ExtX as the destination

For Outbound calls –

1. FXO lines > Dialing > Port Scheduling Schema > Prefix to Specific Port: 99
2. FXO Lines > Dialing > Dialing to PSTN > Wait for Dial-Tone(Y/N): set to ch1-8:N;
3. FXO Lines > Dialing > Stage Method(1/2): set to ch1-8:1;

Configure the following in UCM –
1. For each extension an outbound route with the following pattern 991 for port 1 (EXT1), 992 for port 2… 998 for port 8

For example, in order to send the call via port 1 on the gateway, you need to dial 991xxxxxxx, where x is the number to dial; if you want to select port 2 you will need to dial 992xxxxxx, and so on.

Stage 2 method –

The configuration will be the same as Stage 1 but you will have to configure the following:

1. Channel setting > Local SIP Listen Port > change every single channel UDP
2. Setting the round-robin only (rr:1-8) for ch under FXO Lines > Settings > Dialing

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