Analogue PBX to VoIP converter - PSTN to VoIP gateway
Your old phone system like the Panasonic KX Series can make and receive calls over VoIP, it just need an Analogue to SIP converter. If you have ISDN lines then you need our ISDN to SIP converter, if you have traditional analogue phone lines then you need this Analogue PBX to VoIP converter. Keep your existing phone system and move your lines to VoIP and save £££ pounds! Our Analogue PBX to VoIP converters can pay for themselves in just a few months. You can connect up to two different providers and you can use your preferred VoIP supplier, or one / both of ours. This is possible as the unit generates a single registration per trunk, making it easy to mix and match. Select the correct unit to match your number of lines. | ||||||
| ||||||
The above prices include configuration to your chosen provicer, wiring kit as well as the Analogue PBX to VoIP converter |
Deal 1£6 per month per simultaneous call and you get 2500 mins of Free calls to UK landline / mobile numbers for each channel. For example, if you have 6 lines you would pay £36 per month for your phone bill and that would be it. Unless each month you make for more than 15,000 mins of calls or call International or premium numbers. |
Deal 2This is best for people who get lots of incoming calls on a single number as your monthly bill would be just £1.20. Outgoing calls are at very low rates, but we recommend that you make all out going calls via Deal 1 unless you only call out for a few hours a month. |
How Analogue PBX to VoIP converter worksYou just take the BT line cable out of your junction box, and connect to our Analogue PBX to VoIP converter instead. Plug the Analogue PBX to VoIP converter into to your office / home network and then via the Internet the Analogue to VoIP converter will recive it's configeration and then register at the VoIP SIP exchange. Your existing phone system doesn't know anything has changed, and you have saved the cost of buying lots of new phones and signing up for an expensive hosted VoIP service. |
Set the Firewall Rules
You will need to enable SIP and RTP port forwarding in your firewall, and lock them to your VoIP exchange. In a Draytek 2860 or newer this takes 20 minutes, if you want us to do this for you we can and will charge our normal support rates.
For updates and provisioning Port 443 needs to access fm.grandstream.com & firmware.grandstream.com. On other routers it may take longer to set things up.
When ready to test
- Unpack the Analogue to SIP converter and cables.
- Plug the SIP to Analogue converter into a port on your network, watch the lights start flashing on the SIP to Analogue converter.
- Scan your network to find the VoIP SIP to Analogue converter - look for the MAC address.
- Bind that MAC address to the IP address you want to use so the VoIP SIP to Analogue converter always receives the same IP address, if necessary, reboot the SIP VoIP to Analogue converter.
- Check the port forwarding rules in your firewall are set to this IP address.
- Look at the phone status LEDs and check they are solid blue showing it is registered at the exchange.
- Unplug the BT line cables to your phone system from your phone system's sockets.
- Plug a new cable(s) from the the adaptor into each of your phone systems line sockets.
- Try a test call, you should see a phone status LED is blinking every second.