STEP 1 – Trunk Configuration In the context of this guide a trunk is used to route calls between your Asterisk PBX and your desired VSP(Voice Service Provider), in this case IsraelNumber. In this section we will configure a SIP trunk. Login to FreePBX administrative interface Click on Setup in top right of page. MyNetFone take the complexity out of business telephone systems and make it easy to connect to the world. MyNetFone business customers can choose between a Virtual PBX or SIP Trunking voice services, depending on what equipment they have and how many phone lines are required. PBX is a hosted phone. FreePBX is an open source IP Telephony system. If you can use home and office for communication. At first install FreePBX on Ubuntu 14.04. After installation completed then setup CHAN SIP TRUNK on your server. Browse your FreePBX server via any browser. Now you follow this step by step configure CHAN SIP TRUNK. ★ How To Setup CHAN SIP Trunk.

• Asterisk Sip Trunk Setup
• Ip Office Sip Trunk Setup
• 3cx Sip Trunk Setup

Configuring an Affordable SIP Trunk

Setting up a SIP Trunk is a straightforward process. When configuring your PBX or device, there are two types of authentication mechanisms commonly used when setting up the SIP trunk – “Username / Password authentication” or “IP address authentication”.

Username / Password Authentication

This is the most commonly used setup for SIP trunks and is enabled by default on SIP.US trunks that are automatically provisioned. There are three key pieces of information we provide that will need to be entered into your PBX or device:

• Address of the SIP.US Server (we use two for redundancy)
• Trunk Username (this is a numeric string we provide you)
• Trunk Password (this is an alphanumeric string we provide that you can modify)

These trunk credentials can be found in the SIP.US control panel under the Trunks tab. Once the trunk credentials are setup in your PBX or device, the trunk will then ‘register’ to SIP.US and be ready to make and receive calls. SIP.US supports both G.711 ulaw and G.729 voice codecs.

IP Address Authentication

For additional security, if your PBX or device has a static IP address, we recommend enabling IP Address authentication for your SIP trunk. In the SIP.US control panel, you have the ability to turn on IP Address authentication and specify the static IP address of your PBX or device. Once enabled, you can then make and receive calls without username/password authentication. For outbound calls, our systems will trust all call traffic coming from that specific IP address. For inbound calls, we will automatically deliver all calls to that IP address as well.

If your PBX or device is behind a NAT on an internal IP address, you’ll want to make sure that you forward the appropriate ports in your router. SIP Trunks operate with a signaling layer on port 5060 UDP and an RTP media stream commonly starting at port 10000 UDP. (Asterisk-based systems use ports 10000-20000 UDP by default for the RTP media streams).

For additional information on specific configuration setups for certain types of PBXs and devices, be sure to check out our support page.

With SIPStation SIP Trunks, you can be making calls in just a few minutes. SIPstation Sip Trunks provide telephony services using your high-speed Internet connection, eliminating the need for traditional phone service. No contracts, no fuss. Our services work with your phone system and are compatible with the majority of IP PBXs. They are also compatible with virtually any TDM PBX or legacy CPE phone system using a gateway device.

SIPStation Services are backed by our World-Class customer support staff which are available online or by phone.

All SIPStation service is built into FreePBX and PBXact direct for easy setup. Install the SIPStation module and follow our guide here and have your service setup in minutes and placing calls. Who better to bring you phone service then the company that also manages and builds FreePBX and PBXact.

Available Services:

High Volume Voice Trunk with Inbound and Outbound calling ability (1 Trunk = 1 Simultaneous Call) $24.99 /mo each

• Trunks provide high volume inbound calling
• Includes calling to the lower US 48 states & Canada

Asterisk Sip Trunk Setup

Local numbers available in most areas of the USA

• Only $1.00 /mo each (Add as many as you need from across the US).
• Port your existing telephone numbers to our service: $10 port charge then $1/mo per DID charge.

Includes e911 address support
We provide Caller ID with Name on all inbound calls to any DID provisioned on SIPStations network.

Ip Office Sip Trunk Setup

Supported Codecs
FreePBX module that is the fastest and easiest way to set up telephony service with FreePBX. Sipstation also can be used with just about any VoIP PBX, Softphone or Hardphone.

3cx Sip Trunk Setup

• Add SIP Trunking to your FreePBX installation.
• Add SIP Trunking to your existing VoIP PBX.
• Add SIP Trunking to your legacy PBX.