Setup Guide for 3CX phone system with TieUs SIP Trunk
Before you setup 3CX for TieUs SIP Services, please make sure you already familiar with the 3CX platform and have already done some internal testing on the extension setup, such as how to create a local extension, or how to record a digital voice prompt for incoming calls.
1. Start 3CX Window Management Console.
2. Under VOIP Providers, Add Provider
3. Under Name of Provider, enter TieUs SIP Trunk (or whatever name you want). Select “Generic SIP Trunk”
4. Click Next, Under VOIP Providers Details, enter the SIP server IP address.
Note: Please check the SIP account information we send you, the SIP Server or IP address will be different from the IP address below. It is just an example
5. Click Next. Enter your SIP account information here. Enter 10 digit DID (the number we assigned to you) as External Number, Enter 14 digit authentication ID, Enter 5 digit Authentication Password. Enter the maximum simultaneous call. The number should be matching our system setting.
Note: Your SIP account information will send to you after you sign up our services.
6. Click Next. Here you are required to setup the behavior of 3CX when receiving SIP trunk incoming call. You can connect the call to certain extension, or you can connect to Digital Receptionist (Auto Attendant), provided that you already have recorded the voice message. For initial testing purpose, we recommend you to connect the call to extension, so you can test the incoming call after setup.
7. Here you need to setup the outgoing call behavior. In general, to distinguish the internal call between extension and outgoing call to outside number, you can setup a prefix so 3CX know how to route the call through SIP trunk. For example, you can add Calls to numbers starting with Prefix with “9”. When you want to dial out from extension, simple dial 9+10 digital number you want to dial.
8. Click on “Finish” to compete the initial setup. You can observe if the trunk or extension is registering correctly by looking at Port/Trunks Status or Extension Status.
9. Making incoming and outgoing testing call. Only If encouter problem or one-way voice, check Firewall and Router Setting, port 5060 UDP should open for SIP trunk signaling. Port 5480~ 5486 need to open according to 3CX specs. In General, setup a static map or forward of ports: 5060-5100 (TCP and UDP) for SIP related signal, 9000-9015 (TCP and UDP) for RTP related signal, and 3400-3499 (TCP and UDP) for tunnel related.
NOTE: forwarding ports 5060-5100 covers Port 5090 (TCP) for the 3CX Tunnel. The 3CX Tunnel is so that users can connect to the PBX server remotely to get an extension
Most of our customer using Asterisk opensource platform has different user interface for configuring the Asterisk PBX server. FreePBX and Trixbox are among the most popular one. Those interfaces can vary slightly depending on the version. However, most of the basic settings are the same
Using freePBX/Trixbox you are able to do most of Asterisk's configuration without editing the individual configuration files such as sip.conf or extensions.conf. You can setup most of the features in web interface such as sip trunk, call routing, voicemail and other calling features. Below we will focus on the SIP trunk setup and parameters that will work with TieUs SIP Trunk Services.
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