Running OCS 2007 R2 with a soft PBX aka Aterisk 1.6

September 27th, 2009 | Tags: , , ,

Most IT pros will not be in possession of an OCS/UC capable PBX from the day they decide to look into evangelising their businesses comms infrastructure. But now that OCS 2007 R2 supports UDP based SIP trunks and third parties like JAHJAH offer off-site trunking services there are alternatives worth considering before making the “bigger” investment.

The other option for smaller organisations or test labs is Asterisk (be aware that this is not an officially supported PBX but will get you on your way prior to making a full investment or suffice for braver souls!). Asterisk is an open source software based PBX, created in 1999 by Digium, a now mature application developed for Linux (there is a Win32 port but untested by me). For those interested I would recommend you give “Trixbox” (previously known as asterisk@home) a whirl as it comes as a pre-packaged easy to install ISO – that can be Virtualised within Hyper-V or VMware.

Previously this configuration required a SIP proxy (sipX), but now an MVP named Mick Badran has posted instructions on how a directly attached setup is now possible with Trixbox CE 2.8!

My intention is to take Mick’s configuration and perform further integrations with, Exchange 2010 Unified Messaging and an analogue based terminal adaptor.

No promises, but watch this space!

  1. October 10th, 2009 at 00:25
    Reply | Quote | #1

    Good luck Adam – I’m keen to hear all about your findings.

    Looking back at my ‘journey’ – in the OCS world, 90% of what asterisk does we don’t use. I was just amazed at how hard getting a solution was, without having to create 50 asterisk entities (E.g. extensions etc)

    One thing to note – for our Exchange UM (2007 Sp2) piece in the solution, the incoming call rule through asterisk I had to add the ‘tr’ otherwise the call dropped out when it was redirected to UM.

    I also took out the Answer from Trixbox, so if the OCS user doesn’t pick up or go to UM Voice mail, we have external options available (from our provider)



    exten => _X.,1,Set(numDialed=+61${EXTEN})
    exten => _X.,2,Set(__FROM_DID=${EXTEN})
    ;exten => _X.,3,Answer
    exten => _X.,3,Dial(SIP/${numDialed}@Connect-with-OCS,,tr)