To ease the process and get the most from your SIP Trunking solution, we recommend the following SIP Trunking best practices:
If you have a modern PBX, it is almost certainly SIP enabled. If it has a data or ethernet port, then it is SIP enabled. Check your user manual to make sure. Or call one of our consultants for help. If your PBX is not SIP enabled, you can still use SIP Trunking. You’ll need to get an Analog Telephone Adapter.
Your voice calls will be routed over your data network. If you don’t have sufficient bandwidth, your call quality will suffer.
Your router controls your access to your data network. It should allow you to enable QoS, which stands for Quality of Service. This will prioritise your voice traffic over data. If your router doesn’t have such a setting, you might want to get a new router.
Each SIP channel can handle one voice call. You could decide you need one channel per employee, but that’s likely to be wasteful for most businesses. It all depends how many of them are on the phone at any one time. If you’re not sure, talk to one of our consultants. We have tools to help you work this out. Don’t forget that you can add and remove channels at any time using a simple web portal, so if you have too many channels or too few, it’s easy to change.
You’ll need to make some changes to the settings in your PBX. Our consultants can talk you through this. Alternatively, all the information you need is available in our manual.
Test the system the make sure your SIP trunk, PBX, router and internet connection are all getting along and co-operating with each other. It’s simple to do this without making any commitment: all you need to do is take advantage of our free 30-day trial.
Typically, with most SIP trunking providers, you would need to monitor your call quality on a daily basis. With GoTrunk it’s all automated. Our engineers have a designed tools to fully automate the control of your call quality – and if you’re ever in any doubt, you can even monitor it in real time.
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