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1: How to Verify That Your PBX is SIP-Enabled

These days, most new PBX systems are SIP-capable. There are several ways to check and see if your PBX is SIP-enabled. First, if your PBX has a data jack or Ethernet jack on the back, there is a good chance that it is SIP-capable. Older PBX or key systems just have analog lines to connect to the PSTN, so if your system does not have a data jack or Ethernet jack, it is probably not SIP-capable. However, in those cases, you can still use what is called an ATA (analog telephone adapter) that will convert SIP over to analog. The ATA will front-end your legacy PBX and allow you to use SIP.US trunks. ATAs come in a variety of sizes, from single port all the way up to 24 analog ports. ATA manufacturers include Cisco/Linksys, ObiHai, Grandstream, and others.

If your PBX has a data jack and you are still unsure if it’s SIP-capable, you can check the user manual. You’ll want to look for a section on ‘configuring a SIP Trunk’ or you might find it in the specifications section, typically located at the end of the manual. Look for words like SIP or SIP-enabled IP calling.

You can also contact us with the particular make and model of the PBX. We’ll check it out for you and let you know for sure.

2: Can you connect to a legacy analog PBX or key system?

Absolutely. This is done via a multi-port analog telephone adapter. We extensive experience and recommend the Grandstream multi-port ATA’s. They come in 4,8 and 24 port increments. These multi-port ATA’s are very easy to configure and have proven to be very reliable with our customer deployments. They can be purchased from Amazon.

3: Which PBX systems and devices do you support?

We support any PBX system that is SIP-enabled. This includes popular VoIP IP-PBX’s like Asterisk, FreePBX®, Trixbox, Switchvox, PBX in a Flash, Elastix, Bluebox, FusionPBX, 3CX, sipXecs, Go Auto Dial, Vicidial, Thirdlane, and more. Many traditional PBX manufacturers also support SIP trunking with their latest software releases. These SIP-capable PBX vendors include Toshiba, Panasonic, NEC, Avaya, Cisco, Nortel, Intertel, and others. For those systems that do not support SIP trunking, we support SIP-T1 gateway devices where it’s SIP-in and T1-out to the PBX. Examples of these types of gateways are the Digium G100 and G200 gateways. In addition, we fully support analog telephone adapters (ATAs) which can interface to legacy analog PBX’s and key systems. We recommend the Cisco SPA-2102 for a single port ATA interface and the Grandstream GXW400X series multi-port ATAs for interfacing with analog systems that require more than one line.

Learn how to Setup SIP Trunk for FreePBX, Cisco SIP Trunk, Asterisk and others in our Knowledge Base.

4: Does your SIP trunking service work with Asterisk, Elastix, FreeSwitch, PiAF, or other popular Graphical User Interfaces to configure and control Asterisk?

Our SIP trunking service works perfectly with Asterisk, FreeSwitch and other open source telephony applications including popular Graphical User Interface applications used to configure and control Asterisk. We provide detailed configuration instructions for these systems and will even help you configure the trunks if provided remote login credentials for your server.

5: What VoIP protocols and voice codecs do you support?

We currently only support the SIP protocol. We support both G.711 and G.729 voice codecs for calling.