Instructions for ControlX/DMCDivert

Set where controls to this number should be diverted. Always click Save to activate any changes you make.

Note: changes to the type of divert destination (PSTN/VoIP) may take a few seconds to take effect

VoIP forwarding

This is where you set ip forwarding of your calls via VoIP. Check the box to enable the feature.

Protocol

• SIP - A widely used protocol for VoIP. Fairly simple to setup but can struggle to get through firewalls / routers.
• IAX2 - A protocol most closely associated with the Asterisk Open Source PABX package (and derivatives thereof). Slightly more complex setup but designed to improve access through firewalls / routers.

SIP Parameters

• Host - the address of your VoIP server/PABX in dot-decimal notation (e.g. 87.215.32.121) or hostname (e.g. pabx.acmeproducts.com).
• DTMF - Whether DTMF tones should simply appear in the audo stream (in-band) or as explicit protocol signalling (RFC 8322). This setting often doesn't matter, but if you are having trouble with a tone-driven application on your system, changing this setting may help.

IAX2 Parameters

• Username - the IAX2 username (sometimes referred to as "user context")
• Password - the IAX2 password (sometimes referred to as "secret")
• Host - the address of your VoIP server/pabx in dot-decimal notation (e.g. 87.215.32.121) or hostname (e.g. pabx.acmeproducts.com).

PSTN Forwarding

This is the telephone number to forward calls to via the PSTN (Public Switched Telephone Network). You can enter this in international format (e.g. +44-20-7060-2000, +1-212-123-3214), but it will always be displayed as if dialled from the UK (e.g. 020-7060-2000, 00-12121233214).

You can have both VoIP and PSTN enabled, but VoIP will always be tried first. Only if the VoIP call failed (e.g. your Internet connection is down or the PABX is not working) will the PSTN destination be tried.

Troubleshooting VoIP

VoIP can be tricky. If you want to check the number itself is working and you have doubts about your server setup or its connectivity, try our test service:

SIP Testing Parameters

IAX2 Testing Parameters

Calls should be answered immediately with a "congratulations" message followed by an echo-test (the system plays back to you what it hears as quickly as it can - usually within about 0.2s).

For more information on VoIP diverts and configuring your SIP server, see dmNote 3511: Getting Started with VoIP.