OCS 2007 R2 => Mediationserver => Asterisk PBX RRS feed

  • Question

  • Hi Volks,

    We have done the setup between OCS 2007 R2 => Mediationserver => Asterisk PBX.

    - We can do outbound calls (via sip provider that is managed by Asterisk)
    - We can receive inbound calls from PSTN (via sip provider that is managed by Asterisk)
    - We can do internal calls (between two MOC's)

    Strange is that when logon to MOC we have the error message "no phone system connection" - but why? Can anybody explain that to me? Unfortunately i have no glue why this happens ...

    thanks in advance,

    Thursday, March 19, 2009 7:48 AM

All replies

  • Hi,

    On your user's OCS telephony configuration, did you check "pbx integration" ? because in your case you should not, Entreprise Voice is the only mode you need for this kind of integration.
    Saturday, March 21, 2009 7:35 AM
  • Matthias,

    can you explain the configuration you did for the integration of OCS 2007 R2 and Asterisk PBX??

    Wednesday, April 1, 2009 2:46 AM

  • Marck - it depends on the version of Asterisk you have.

    !.4 only does SIP over UDP - so it needs 'something' inbetween to convert SIP over UDP to SIP over TCP - OpenSER is a good choice.

    This is a good guide - https://confluence.terena.org/display/IPTelCB/3.2.7.+Tying+MS+OCS+with+Asterisk+through+OpenSER

    NOTE: Config of OpenSER can be tricky.  Also - OpenSER can be set to make 'on-hold' work correctly between the two systems.

    1.6 can use SIP over TCP - so can go directly to a mediation server.

    Geoff's guide is excellent - http://blogs.technet.com/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx

    NOTE: This is much easier to setup then the 1.4 / OpenSER way - AND - on-hold works fine.


    Paul Adams

    Wednesday, April 8, 2009 5:51 PM
  • To fork the call between the Asterisk and MOC there needs to be a mechanism in place to handle the UDP/TCP Conversion.  We used an AudioCodes Mediant 1000 gateway configured without a T1 but with IP:IP routing.  http://www.audiocodes.com/products/mediant-1000-uc

    This configuration is good for a small/medium size environment. 

    Friday, April 10, 2009 9:01 PM

  • At the risk of saying the same thing over & over, you only need a UDP/TCP conversion for Asterisk 1.4

    Asterisk 1.6 works without the need for this - it does SIP/TCP natively.  We run Asterisk 1.6 direct to MOC and it works fine - forking, redirects, on-hold - no issues.


    Paul Adams
    Tuesday, April 14, 2009 5:19 PM
  • Dear Paul,

    Can you please let us know how you did configuration of Asterisk 1.6 to work with MOC.

    Thanks in advance
    Mantu Jha
    Sunday, May 31, 2009 8:49 AM

  • Just so there's no confusion - I'm not using Office Communicator directly against Asterisk.

    I'm using office communicatior with an OCS R2 server - which in turn is using an OCS R2 Mediaition server - which in turn goes to an Astewrisk server - which in turn passes calls out to either the PSTN or to a Nortel Option 11C (using a RedFone 2 x T1 device).

    Everything you need to get this working is on Geoff's website



    Friday, June 12, 2009 11:19 PM
  • mantu,

    I've also a full write up of OCS to Asterisk 1.6 (trixbox) talking across 2 trunks.

    This may help you - http://blogs.breezetraining.com.au/mickb/2009/07/31/FinallyConnectedOCS2007R2ToTrixboxAsteriskToAPSTNPBX.aspx


    Mick Badran - http://blogs.breezetraining.com.au/mickb
    Friday, July 31, 2009 10:24 PM
  • I am sorry to get involved in this conversation, but what puzzles me why you guys are still trying to use very complex third party Linux passed solutions as middleware between OCS and SIP Trunk provider? The only justification I see if one already have Asterisk/Trixbox/etc. IP-PBX as 100% Voice solution and wants to integrate with or migrate to OCS EV.

    It is well known fact that even OCS 2007 RTM will connect to SIP Trunk provider via Mediation Server directly. R2 does the same. The two conditions is you need IP Trunk which does not require registration and must be SIP over TCP. This indeed limits the scope of providers but, in this post: http://social.microsoft.com/Forums/en-US/communicationsservertelephony/thread/0d5e8ca6-1b8a-4ed7-ac4f-816e5c814921 I have explained how we could connect OCS environment to ANY provider via TCP or UDP and with or without trunk registration.

    I continued experimenting with FreeSwitch in different scenarios for the past few months. As of now, using FS I have OCS R2 connected to Callcentric (single trunk, SIP over UDP, require registration), Broadvox (single trunk, SIP over UDP, require registration), and… MagicJack service (don’t ask me how, I will not tell youJ). I also connected Exchange 2007 UM to the above providers with success. This includes Fax-to-Email using the Exchange’s T.38 detection feature.

    On my opinion, Microsoft intentionally limited the Mediation server capabilities to SIP over TCP (to prevent packet loss) and IP Trunks only (to increase the security).


    Saturday, August 1, 2009 4:26 PM
  • Drago - thanks for the response and buy-in to this discussion.

    Sounds like you had a couple of early mornings too.

    From what I read in this thread - no one is debating the age old debate of why doesn't ms support sip/udp etc etc.

    I did have a look at FreeSwitch (spent several weeks evaluating what's out there) and in my solution I leant towards Trixbox/Asterisk (several factors, one of them being you can get dedicated embedded Trixbox devices)

    Glad you got it working with your solution.

    OCS is not my area of speciality - I was amazed that (like yourself) everyone was saying "how easy...OCS<->(the world)...done!"

    I then started digging and there's a gaping hole on HOW and WHAT pieces are needed to integrate with PSTNs(without spending $1000s of dollars).

    In the US, MS can point you to several providers that talk SIP/TCP. Here in OZ we're not so lucky.

    Being in training myself, I spoke to several MS OCS instructors, Product Managers, ex-product managers all pointing to either 3rd party boxes, or changing different providers (one MCS consultant wrote his own translator).

    I just thought this is all too unnecessarily hard......

    Next to implement a Skype trunk and then'll we'll be cooking!

    Thanks for your input.

    Mick Badran - http://blogs.breezetraining.com.au/mickb
    Monday, August 3, 2009 3:42 AM
  • “…Sounds like you had a couple of early mornings too…” In my native country we call this “Late beers” J


    Will you contact me via email: (dnt @ drago.ws) or federation: (dragomir @ gmc.cc.ga.us)


    Monday, August 3, 2009 11:12 PM
  • Drago - you assume we want to talk to a sip end point from OCS (which I don't all the time).

    I have an aging Nortel Option 11C (which sends & receives using a PRI) & my PSTN connection is a PRI T1

    My choices were / are - twin T1 'device' - or Asterisk using a twin T1 card or device.

    I used Asterisk because it was half the cost AND I can connect it to what ever I want AND it can talk to SkyPe AND I can make a non-OCS phone ring from a response group AND I can connect it to whatever sip provider I choose (including MagicJack - again don't ask) AND it does auto failover to another Asterisk box AND it's FREE! AND it's very stable AND it's efficent AND etc...

    Asterisk is a love it or hate it thing.  People who love it could go on for hours about how great they feel it is.  For me - Asterisk provides a lot of flexibility that OCS is missing.  You are wondering why I bother using it.  I am wondering how you function without it.


    Tuesday, August 4, 2009 3:58 PM
  • Paul,

    My post was rather toward the people that are devastated form the fact that in their countries/regions/etc. getting SIP over TCP VOIP trunk is impossible and/or getting new gateway/IP-PBX capable of TCP – too expensive and are or will soon look for a solution.

    Those with past experience with Asterisk and Trixbox will not be “converted” J for obvious reason. To convince you is same as convincing MAC user to go PC…

     I remember the old days when was struggling to find a solution and without any experience on either Linux based IP-PBX or even Linux OS was driving myself to the verge of insanity… I am just trying to save time and frustration to some colleagues.

    I’m working on complete manual and preconfigured environment which is to be installed on either Mediation server (co-existence) or standalone machine. It will provide dynamic registration and static SIP Trunking, routing and provider fallover etc.  I see it done before end of August.





    Wednesday, August 5, 2009 2:36 AM