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Asterisk 1.6.0 + Exchange 2007 SP1 Unified Messaging

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Last week I discussed how to connect Asterisk with OCS Mediation Server over TCP. I figured since we can connect Asterisk to OCS, why not to Exchange UM. I will go over the configuration changes I made to Asterisk for this, but will not go into detail – since most of it was covered last week.

** Just wanted to add a side note. This configuration is not supported, but again I like to play – so here you go. **

Asterisk Configuration Changes

 

SIP.CONF

[general]

context = default

bindport = 5060

bindaddr = 0.0.0.0

tcpbindaddr = 0.0.0.0

tcpenable = yes

promiscredir = yes ** This is new line added **

promiscredir -> If set to yes, allows 302 or REDIR to non-local SIP Address

[SIP_VM]

type = peer

host = 10.100.16.20

qualify = yes

transport = tcp

[50012]

type = friend

callerid = Exchange User <50012>

secret = 50012

host = dynamic

canreinvite = no

dtmfmode = rfc2833

mailbox = 50012

disallow = all

allow = ulaw

transport = udp

You have seen most of this configuration before. We created a new Trunk/Peer that will connect to the Exchange UM Server. I also created a new Extension since I setup my Exchange DialPlan to support 5-Digit extensions.

Now we have to make some changes to the Asterisk Dialplan. There are two different configurations.

1. If you are going to call into Exchange UM.

;Exch VM

exten => 79999,1,Answer

exten => 79999,n,Dial(SIP/SIP_VM/${EXTEN})

exten => 79999,n,Busy

This just gets added to the outbound context in your dialplan.

 

2. Here we will create a simple Macro to dial the extension then redirect to Exchange UM if no one answers.

; User Extension Macro

[macro-stdexten]

exten => s,1,Answer

exten => s,n,Set(MBEXT=${ARG1})

exten => s,n,Dial(SIP/${ARG1},20)

exten => s,n,SipAddHeader(Diversion:<tel:${MBEXT}>)

exten => s,n,Dial(SIP/SIP_VM/${EXCHUM})

exten => s,n,Busy

exten => _5XXXX,1,Macro(stdexten,${EXTEN})

So your extension.conf file will now look like this:

[general]

static=yes

writeprotect=no

[globals]

EXCHUM = 79999

[default]

exten => _5XXXX,1,Macro(stdexten,${EXTEN})

include => outbound

[macro-stdexten]

exten => s,1,Answer

exten => s,n,Set(MBEXT=${ARG1})

exten => s,n,Dial(SIP/${ARG1},20)

exten => s,n,SipAddHeader(Diversion:<tel:${MBEXT}>)

exten => s,n,Dial(SIP/SIP_VM/${EXCHUM})

exten => s,n,Busy

[outbound]

exten => _NXXNXXXXXX,1,Set(EXT=+${EXTEN})

exten => _NXXNXXXXXX,n,Dial(SIP/SIP_TRUNK/${EXT})

exten => _NXXNXXXXXX,n,Busy

;Exch VM

exten => 79999,1,Answer

exten => 79999,n,Dial(SIP/SIP_VM/${EXTEN})

exten => 79999,n,Busy

Now this is done we will move onto the Exchange UM configurations.

The first step is to create a UM Dial Plan. Open up Exchange Management Console -> Under Organization Configuration -> Unified Message. Select New UM Dial Plan. There will be a wizard that walks you thru the configuration. The important pieces are: 

 

- URI Type: Telephone Extension

- VoIP Security: Unsecured

 

 

Once you have the Dial Plan configured -> edit the dial plan and create a Subscriber Access Number.

 

This is the number that is dialed from Asterisk to reach your VM Box.

Next we need to configure a UM IP Gateway. This is done by using the New UM IP Gateway wizard.

You can leave the default UM Mailbox Policy that was created when you created the Dial Plan. The only thing left is it to associate the Dial Plan with your UM Server.

Under Server Configuration -> Unified Message -> Properties.

Add your new Dial Plan to the UM Settings.

Last is to apply the Dial Plan to a User Mailbox. You can use the Wizard to enable the User for Unified Messaging.

Now you are ready to test making a call from your SIP Client to Exchange UM.

When you make a call from Asterisk and if it fails with the following error, you will need to patch asterisk and recompile.

The formatting of the SIP call is incorrect. It is showing SIP/::::TCP@79999@10.100.16.20:5065.

To download the patch https://bugs.digium.com/view.php?id=13523

Download the second patch ‘13523v2.patch’ and apply the patch. To apply the patch you can follow these instructions.

- Change to install directory (/usr/src/pbx/asterisk-1.6.0)

- Download patch

- patch –p0 < 13523v2.patch

- You will get a success once this patch is installed

- /etc/init.d/asterisk stop

- make clean

- make

- make install

- /etc/init.d/asterisk start

Now when you make the same call you should see the following output:

You can see the difference in how the SIP call is formatted.

Well that’s it. Have fun – enjoy playing.

Comments

  • Anonymous
    January 01, 2003
    Yes that would be the preferred method if using OCS.  But if you just wanted to use Exchange with existing Asterisk deployment, then this shows you how to configure the environment.

  • Anonymous
    January 01, 2003
    Why wouldn't you just integrate Asterisk 1.6.0 into an OCS 2007 R2 environment then have Exchange UM integrate with OCS?

  • Anonymous
    December 20, 2010
    I can make a call to Subscriber Access Number successfully. but I can't make a call UM auto attendant, number is 79998. of course, I change the dial pattern. can you tell me how to configure dialpattern to make a call to UM auto attendant? thanks.

  • Anonymous
    December 21, 2010
    Tim - do you have any spaces in the Auto Attendant name?  If so - recreate the AA with no spaces. Geoff