UC for All ;) : UM

Exchange 2007 UM - Transfer to Voicemail

gclark — Mon, 30 Mar 2009 15:10:00 GMT

I was recently working with a Customer who wanted to transfer calls to voicemail directly without ringing the phone first. We discussed a couple different options.

1. Adding additional EUM Proxy Address to each users mailbox.

This would give the flexibility of routing a call directly to Voicemail for any user without ringing the phone line.

The down side to this approach is having to assign and remember two extensions for each user. Also having to route additional extensions thru the phone switch / PBX to UM for each user.

2. Route calls through the PBX with Diversion / History-Info to UM.

This can be implemented by creating an extension such as 7XXXX. So any extension with a preceding '7' would get routed with Diversion / History-Info to UM.

Example from a Asterisk PBX might look like this:

exten => _7XXXX,1,Answer
exten => _7XXXX,n,SipAddHeader (Diversion:<tel:${EXTEN:1}>)
exten => _7XXXX,n,Dial(SIP/EXCH_UM/${SA-NUM})
exten => _7XXXX,n,Busy

Again this solution would require some programming from the PBX in order to implement this solution. This would not require any additional configuring on Exchange UM to implement this solution.

3. Create an Auto Attendant that only forwards calls to Exchange UM for voicemail.

This is done by creating an Auto Attendant, using your normal process. But I will take you through how this gets created and what the customer discovered that works for them.

New-UMAutoAttendant -Name VoiceMail -SpeechEnabled $False -Status Enabled -UMDialPlan PBXDial -PilotIdentifierList 4606

This will create your Auto Attendant, but we will still need to make a couple more modifications to Auto Attendant.

Set-UMAutoAttendant -Identity VoiceMail -AllowExtensions $False -CallSomeoneEnabled $False -SendVoiceMsgEnabled $True -NameLookupEnabled $False

When calling into the Auto Attendant now you will hear the following:

"Welcome to the Microsoft Exchange Auto Attendant. Enter the extension for the person you are calling."

When you dial the Extension / Voicemail Number. You will then be taken directly to Voicemail with out ringing any phone lines.

In the customers environment, they now call an extension associated with the newly created Auto Attendant. They are prompted for a users extension. Then transferred to the users Mailbox to leave a message.

I must give credit to the Customer for coming up with Option 3. This seemed to answer their needs and work in their environment.

Asterisk 1.6.0 + Exchange 2007 SP1 Unified Messaging

gclark — Wed, 22 Oct 2008 07:32:00 GMT

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 http://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.