...
Goal
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(short
...
description)
This recipe describes tying Microsoft Office Communications Server 2007 (OCS) to Asterisk. This enables users that have a hard phone connected to an Asterisk PABX, to
- make calls from their Microsoft Office Communicator to the PSTN
- be dialled from the PSTN and answer the call on either the hard phone or Office Communicator (for instance when working from home)
- let them control forwarding and simultaneous ringing options from the Communicator
- have an archive of call records and missed calls notifications in Microsoft Outlook
- click to dial on telephone numbers within Outlook
Applicability
OCS is meant as a value adding service platform that offers Unified communications by integrating IPTelephony, presence, instant messaging and e-mail. The power lies in the connections. A powerfull connection, and also a difficult, is to connect it to a PABX. In this case, the PABX is based on Asterisk. Asterisk takes care of the connectivity with the PSTN and the number plan.
OpenSER acts as a proxy between Asterisk and OCS, since OCS only supports SIP over TCP and Asterisk only SIP over UDP.
When a call is made, the message flow is as follows:
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SIP:
Communicator 2007 <-TLS--> OCS <-TLS--> OCS med serv <- TCP -> OpenSER <- UDP -> Asterisk <-> PSTN
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Media: Communicator 2007 <-SRTP-> OCS <-SRTP-> OCS med serv <---- Tying Microsoft Office Communications Server 2007 (OCS) to Asterisk, to enable users to call from OCS through Asterisk and be called on their (extension) number. h1. Applicability OCS is meant as a value adding service platform that connects to PABX-es. In this case, the PABX is based on Asterisk. Asterisk takes care of the connectivity with the PSTN and the number plan. OpenSER acts as a proxy between Asterisk and OCS, since OCS only supports SIP over TCP and Asterisk only SIP over UDP. When a call is made, the message flow is as follows: {noformat} SIP: MS Office Communicator 2007 <- TLS -> MS OCS 2007 < TCP > OpenSER < UDP > Asterisk <> PSTN {noformat} {noformat} RTP: MS Office Communicator 2007 <- SRTP > MS OCS 2007 <-/-------- RTP ---------/-> Asterisk <><-> PSTN {noformat} h1. Prerequisites * Linux machine with Asterisk, a number plan and PSTN connectivity * Linux machine with OpenSER |
This recipe needs one bug to be resolved concerning RTP flowing from Asterisk to the Mediation server. Solution coming soon.
Prerequisites
- Linux machine with Asterisk, a number plan and PSTN connectivity
- Linux machine with OpenSER 1.2.1
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- or
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- higher
...
...
- Microsoft
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- Office
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- Communications
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- Server
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- 2007 core installation
- Windows 2003 machine to install OCS Mediation Server
OCS can be either Enterprise Edition or Standard Edition. For the basic installation guides of OCS, see http://technet.microsoft.com/en-us/library/bb676082.aspx
...
.
...
You
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need a second windows machine to install an OCS Mediation Server, which is the designated node for external SIP connectivity.
For specifics on installing OCS Mediation Server, see http://www.microsoft.com/downloads/details.aspx?FamilyId=24E72DAC-2B26-4F43-BBA2-60488F2ACA8D&displaylang=en
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Configuration of Asterisk
Assuming you have a working Asterisk setup, only few things have to be taken care of.
Create a SIP trunk, see http://blog.lithiumblue.com/2007/12/interim-information-on-integration-ocs.html for an example if you use Trixbox as your Asterisk server. Use the following settings in your SIP.conf:
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host=<your OpenSER IP address>
port=5060
qualify=no
type=peer
context=from-internal
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Forward all incoming calls from the PSTN destined for known extensions to the SIP trunk that connects to OpenSER.
Asterisk should always send a full E.164 number to OCS, starting with a plus sign 
!
Configuration of Microsoft Office Communications Server 2007
Your basic Enterprise Edition or Standard Edition server should be running and you have enabled users to log on to OCS. PC-to-PC calls should run smoothly. If they are outside the firewall, you should use an Edge Server (which supports STUN and rate limiting).
In the OCS MMC snap-in, right click on the OCS forest and select 'properties -> voice properties'.
Add a Localization Profile
The dialog opens in the 'Localization Profiles' tab.
Edit the default location profile or add a new one and edit the profile, so you can add a localization rule.
Add a normalization rule
Within the localization rule, add a normalization rule. The rule translates numbers dialed by users to a standard format. Here we always translate to E.164 format, namely
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+<countrycode><area code><subscriber number>
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Fortunately, the Microsoft Communicator translates numbers automatically to this format, leaving out the +, (), spaces and dashes, even when starting a call from Microsoft Outlook. However, we have to add at least one rule. The normalization rule is written in .net regular expression format. See http://www.regular-expressions.info/tutorial.html
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for
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more
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explanation
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of
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regular
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expressions.
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In
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our
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case,
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we
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translate
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any
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number
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starting
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with
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a
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zero
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to
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a
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Dutch
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number:
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} \^0(\d*)$ \-> \ +31$1 |
It
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basically
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means
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that
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for
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any
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new
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translation
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(^)
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of
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a
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number
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that
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starts
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with
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a
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0
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and
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then
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containing
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any
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number
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of
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digits
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\d
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*
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it
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should
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take
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the
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part
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of
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the
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number
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between
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brackets
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()
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and
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use
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it
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as
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variable
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in
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the
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translation.
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The
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translation
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adds
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+31
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to
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it.
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THE
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'+'
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SIGN
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IS
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ABSOLUTELY
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CRUCIAL
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!
...
!
...
!
...
!
...
!
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Don't
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forget
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to
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put
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it
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there
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!
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Please
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double
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check
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whether
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it's
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in
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the
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config
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!
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Don't
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say
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we
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didn't
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warn
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!
Add phone usages and policies
When you're
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done
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with
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the
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'Localization
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Profiles'
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tab
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you
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can
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add
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'phone
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usages'
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in
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the
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'voice
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properties'
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dialog,
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which
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are
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useful
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for
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assigning
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to
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users
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and
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logging
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these
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types
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of
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use.
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Be
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sure
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to
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also
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create
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at
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least
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one
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Policy
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in
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the
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Policy
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tab. A collection of phone usages form a 'policy', and a user will later on be assigned a policy. This way, you can discern which users are allowed what type of calls (for instance disallow service numbers).
Add a Route
Finally add a 'route' under the 'routes' tab.
Here is defined that for numbers matching a certain regular expression, the call should be routed to our mediation server. The regular expression states in this case that any number should be forwarded:
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Most important, however, is to add a 'route' under the 'routes' tab Here is defined that for numbers matching a certain regular expression, the call should be routed to our mediation server. The regular expression states in this case that any number should be forwarded: {noformat} \^(\d*)$ See the Microsoft |
Part of the route definition are the phone usages that are allowed for this route.
See the Microsoft 'OCS_VoIp_Guide'
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document
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for
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further
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reference.
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Enabling
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users
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Even
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though
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they
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cannot
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really
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make
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external
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calls
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yet,
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you
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can
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enable
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users
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for
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Enterprise
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Voice.
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This
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can
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be
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done
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in
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various
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places.
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Let's
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go
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to
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the
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'Active
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Directory
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Users'
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MMC
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snap-in,
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go
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to
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the
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entry
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of
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a
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test
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user
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and
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select
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its
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properties.
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In
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the
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'Communications
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Tab'
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choose
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'additional
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options'
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->
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Configure
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,
and
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in
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the
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new
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dialog
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select
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'Enable
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Enterprise
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Voice': 
Fill in the line URI. which can be of the form
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tel:+<nr> (yes, the '+' again!)
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Install and configure OCS Mediation Server
Now install an OCS Mediation Server on a machine running Microsoft Windows 2003 server. The Mediation Server software can be found in the OCS installer under 'additional server roles'.
Configure the Medation Server as follows:
Next Hop:
Gateway Listening IP address: the external IP address of the Mediation Server
Location Profile:
choose the location profile as chosen before (i.e. 'Utrecht')
Gateway IP addres:
<your OpenSER IP address>
Configuration of OpenSER
Edit your openser.cfg as follows or download openser.cfg for your convenience:
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# lower debug level for debugging purposes: Fill in the line URI. which can be of the form h2. Install and configure OCS Mediation Server Now install an OCS Mediation Server on a machine running Microsoft Windows 2003 server. The Mediation Server software can be found in the OCS installer under 'additional server roles'. Configure the Medation Server as follows: Next Hop: Gateway Listening IP address: the external IP address of the Mediation Server Location Profile: choose the location profile as chosen before (i.e. 'Utrecht') Gateway IP addres: <your OpenSER IP address> Phone usages: 'PSTN out' h2. Add a Route Create a route that points to the Mediation Server. For instance, to forward calls to any number: \^+(.*)$ h2. Configure users h1. Configuration of OpenSER Edit your openser.cfg as follows: {noformat} {noformat} {noformat} {noformat} debug=5 log_stderror = no fork=yes children=4 alias=<your OpenSER proxy FQDN> listen=udp:<your OpenSER proxy IP address>:5060 listen=tcp:<your OpenSER proxy IP address>:5060 check_via=no dns=no rev_dns=no mpathnompath="/usr/local/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "maxfwd.so" loadmodule "uri.so" loadmodule "uac.so" modparam("uac","from_restore_mode","auto") loadmodule "textops.so" loadmodule "rr.so" modparam("rr", "enable_full_lr", 1) modparam("rr", "enable_double_rr",0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 1) route { if (\!mf_process_maxfwd_header("10")) { { sl_send_reply("483","Too Many Hops"); exit; } ; if (msg:len >= 2048 ) { { sl_send_reply("513", "Message too big"); exit; }; ; if (loose_route()) { { append_hf("P-hint: rr-enforced\r\n"); route(21); }; ; if (\!uri==myself) { { append_hf("P-hint: outbound\r\n"); route(21); }; ; route(1); } route[1] { # If coming from OCS if (src_ip==<your OCS Mediation Server IP address>) { if (method=="INVITE|BYE") { xlog("L_INFO", "*** invite from OCS M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } } t_relay("udp:212.0.234.219<your Asterisk Server IP Address>:5060"); } # else coming from Asterisk else { { t_relay("tcp:213.154.233.13<your Asterisk Server IP Address>:5060"); } ; exit; } h1. |
OS
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specific
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help
Validation,
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confirmation
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tests
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Configure
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MS
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Office
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Communicator
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2007
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to
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log
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on
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to
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OCS
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and
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provide
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the
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login
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account
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with
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a
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'line
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URI'
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as
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mentioned
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above.
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Start
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typing
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a
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telephone
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number
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or
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(Outlook)
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contact
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in
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the
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search
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bar
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or
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right-click
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on
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one
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of
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the
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contacts
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to
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dial
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the
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number.
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The
...
communicator
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should
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ring,
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as
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well
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as
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the
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chosen
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number.
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A
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useful
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tool
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to
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debug
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SIP
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message
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flows
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can
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be
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downloaded
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from
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the
...
MS
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site:
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http://www.microsoft.com/downloads/details.aspx?familyid=7b6ab4f3-2949-4e97-856e-9c4ae323c75a
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You
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can
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use
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it
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by
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right-clicking
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on
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your
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OCS
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pool
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or
...
server
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and
...
choose
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'Logging
...
tool->Start
...
debugging
...
session'
...
.
...
Choose
...
'SIPstack'
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in
...
the
...
options
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and
...
click
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'start'
...
before
...
setting
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up
...
a
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testcall.
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After
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pressing
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'stop'
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and
...
'analyse'
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you
...
can
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see
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a
...
SIP
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messageflow
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between
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your
...
OCS
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nodes
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and
...
find
...
problems.
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For
...
debugging
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on
...
the
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OpenSER
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machine,
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you
...
can
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use
...
ngrep
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or
...
wireshark
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> (tshark \-i any \-n \-V \-R sip). \\ |
Acknowledgements
Thanks to the NLnet Foundation for sponsoring, 2AT and Rob Kruit.