SIP / VoIP Connection to our Teleconferencing Service

We have added the ability for customers to connect to our teleconferencing service through the SIP protocol.

Because its an IP based technology, phone lines are not used in the process of the call and there for reducing the cost for delivering the service.

This is not a separate conference room or service, its the same service. You can have participants connected through the 1800 Toll-Free number, Our standard inbound lines, Called through the Invite System and through SIP at the same time. all while having access to our feature range such as recording and scheduling.

Not all VoIP is the same. You need to be able to dial our service directly through the VoIP protocol and not just use a VoIP phone to call our telephone numbers.

The methods below cover how to set this up:

IP PBX
If you have an IP based PBX (EG: Asterisk), you can create an extension:
exten => 139,1,Answer
exten => 139,2,Dial(SIP/conference@greymouse.com.au)
exten => 139,3,Hangup

With FreePBX, add a other (custom) device from the extensions page:
In the dial field enter: SIP/conference@greymouse.com.au

Softphone dial directly: conference@greymouse.com.au

Other Hardware Devices:
Some devices can be configured to have a speed dial.
In this case we advise to read the manual on how to configure this for each phone.

You would be setting SIP:conference@greymouse.com.au as the speed dial number.

(Codecs available: G729, GSM, iLBC) - Your phone must support these codecs to connect.

At 4¢ per minute per user its a cost effective for company's who already have a VoIP system/Softphones that want the ability to have the feature set a professional service delivers. (Recording, Scheduling, Automatic Invite, 1800 participant Toll-Free numbers, etc)

This system is available immediately for both standard and trial users.

Clint Davis
CEO, Greymouse Teleconference