Free VoIP (SIP) bridging now supported on Extensions

As part of our integration with VoIP we’ve added bridging of extensions using SIP uri’s.

This means that you can now use regular phone numbers OR a SIP uri on the Extension block. To use this feature simply enter the SIP uri in the ‘phone number’ section (see pic. below – the sip uri must be prefixed with “sip:”).

Using a sip uri as extension
Enter the sip uri to bridge to VoIP

Just in case you are wondering a SIP uri is a ‘VoIP address’ used to send a call to a ‘user’, much like an email address is used to identify an email user. The format of a SIP uri is sip:username@hostname (very similar to an email address).

Some important notes regarding this feature.

  • Calls bridging to SIP/VoIP are free.
  • You will need a PUBLIC sip uri to use this feature. This is normally obtained from your VoIP provider (however, hostname could be a static IP address).
  • There’s no authentication mechanism provided, this means your ‘SIP endpoint’ must accept ‘anonymous’ calls.
  • Bridging uses the public internet and regular non-encrypted SIP and RTP. (ie. there’s no special security on the call ).
  • SIP is not compatible with Skype (but it is with thousands of other phones, PBX’s and VoIP providers).

Example of how you might use this.

  1. Get a VoIP phone (can be software or hardware)
  2. Connect your phone to a VoIP provider (iptel.org is a free example).
  3. Enter your the SIP uri provided by the VoIP provider on the Extension block.
  4. Calls routed to the Extension will be received on your VoIP phone (for free).

*We should note, for the sake of clarity, this feature does not allow you to SIP REGISTER with LazyPBX or make outbound calls via. LazyPBX. All outbound calls from the VoIP phone will be through the external  VoIP provider.

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