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Microsoft Unified Communications Blog

Archive for the ‘Cisco 6.x Integration’ Category

No Name Appears When Callings from OC to Cisco or vice versa

Posted by Mino on August 10, 2009

One of the frustrating drawbacks when Implementing OCS enterprise voice integrating with Cisco Call Manager or other PBX is that No name would appear in Communicator 2007 when a telephone user calls a Communications Server 2007 user or vice versa .

The only solution by then was to place a media gateway in the middle between mediation and IP-PBX and use a translation feature built in the media gateway to edit the packet header and add Caller Name before sending it to the PBX.

But Finally Microsoft has listened to my prayers and they have released July fix for OCS 2007 R2, once they were released our team started to do the testing in our LABs. At the beginning things didn’t work fine but in the end it worked and it appeared that we only applied the server side updates while the KB 971844 includes Office Communicator update too.

However, even with these fixes, this don’t send display name by default. So, this configuration in KB 972721 will be also necessary. Below are the snapshots taken by our Engineer Amr Nassar who has worked on this and successfully made it work after applying those Fixes.

Calling from Office Communicator R2 with ext 6000 to Cisco IP Phone with ext 10000

From OCS

Calling from  Cisco IP Phone with ext 10000 to Office Communicator R2 with ext 6000 

From Cisco

But let me also share Microsoft explanation on why this problem happens in the first place and what these updates fix?

Problem Explanation:

When a Private Branch Exchange (PBX) telephone user calls a Microsoft Office Communications Server (OCS) 2007 R2 user, the calling party name that exists in PBX is stripped at the OCS 2007 R2 Mediation Server. Because the PBX telephone user does not exist in Active Directory, no name appears in Microsoft Office Communicator 2007 R2

This problem occurs because the OCS 2007 R2 Mediation Server does not forward the display name information from the Unified Communications (UC) side to the gateway side. Therefore, the Communicator 2007 R2 client does not receive the display name information.

Update 970679 introduces functionality for the Mediation Server role of Communications Server 2007 R2 to forward Display Name information that is part of the From header between its gateway side and its proxy side.

A file called MediationServerSvc.exe.config should be created in the Mediation Installation Directory which be default is at  %programfiles%\Microsoft Office Communications Server 2007\Mediation Server

This file should contain.

<?xml version=”1.0″ encoding=”utf-8″ ?>
<configuration>
                 <appSettings>
                                <add key=”forwardDisplayName” value=”True” />
                 </appSettings>
</configuration>

Posted in AVAYA, Cisco 4.x Integration, Cisco 5.x Integration, Cisco 6.x Integration, Cisco 7.x Integration, communicator client, Mediation Server, Nortel CS1000, OCS 2007 R2, PBX Integration, Phone Edition, Quintum's gateways | Tagged: , , , , , , , , , , , , , , | 3 Comments »

How to Integrate Exchange UM Voicemail into Cisco IP Phones

Posted by Mino on March 27, 2009

I am working with a client who is using Cisco CUCM with Cisco Phones, along with Microsoft Exchange 2007 voice mail on the UM , but when you divert the phone to voicemail you are not prompted with the users voicemail prompt – you are prompted with the Subscriber access greeting of “ Welcome , you are connected to Microsoft exchange ,…etc )

Usually when you call someone and there is no answer then you are transferred to the Pilot number, the extension of the person you are calling is sent also in the request so that you would be directly transferred to the users voice mail not to the Welcome greeting.

This Problem Happens when Diverted Calls are not accepted because both sides cannot agree on DTMF handling , the MTP is important, because it deals with differences in how DTMF is signaled between the phones and gateways and the sip trunk

Just make sure the following on the Cisco SIP trunk:

  1. Accept Out-of-Dialog REFER
  2. Accept unsolicited Notification
  3. Accept Replaces Header
  4. Have the SIP trunk configured to use MTP, once I’d configured MTP and MRG/MGRL

The changes detailed below are based on a new installation of Call Manager 5. As this environment been created for the purpose of testing the integration between platforms, it contains only minimum configuration. The required Changes are with:

·         Media Termination Point (MTP)

·         Changes to security profile

Media Termination Point: The Cisco Call Manager installation builds the default media termination point.

Media Resource Group: Create a media resource group “MRG_CCM5” and add the media resource (MTP) to the group. Multicast is not required.

Media Resource Group List: Create a media resource group list “MRGL_CCM5” and add the media resource group “MRG_CCM5” to the list.

Device Pools: By default Cisco Call Manager creates the “default” device pool. Open the device pool “default” and select the new media resource group list “MRGL_CCM5”.

SIP Trunk Security Profiles: Copy the “Non Secure SIP Trunk Profile” to “E2K7 Non Secure SIP Trunk Profile” and enable “Accept Unsolicited Notifications”.

Partition Configuration: Create a Class of Control Partition “Local”.

Calling Search Space: Create a Class of Control Calling Search Space “CCS_Local” and add the Partition “Local” to the calling search space.

Trunk Configuration:

Trunk Configuration

General

Setting

Device Name

E2K7

Description

Exchange UM

Device Pool

Default

Call Classification

Use System Default

Media Resource Group List

<None>

Location

Hub_None

AAR Group

<None>

Packet Capture Mode

None

Packet Capture Duration

0

Media Termination Point Required

Enabled

Retry Video Calls as Audio

Disabled

Transmit UTF-8 for Calling Party Name

Disabled

Unattended Port

Disabled

MLPP Domain Information

<None>

   

Trunk Configuration
Call Routing Information

Setting

Inbound Calls

Significant Digits

All

Connected Line ID Presentation

Default

Connected Name Presentation

Default

Calling Search Space

CCS_Local

ARR Calling Search Space

<None>

Prefix DN

<Blank>

Redirecting Diversion Header Delivery

Disabled

Outbound Calls

Calling Party Selection

First Redirect Number

Connected Line ID Presentation

Default

Connected Name Presentation

Default

Caller ID DN

<Blank>

Caller Name

<Blank>

Redirecting Diversion Header Delivery

Enabled

Trunk Configuration

SIP Information

Setting

Destination Address

<IP Address of E2K7 Server>

Destination Address is an SRV

Disabled

Destination Port

5060

MTP Preferred Originating Codec

711alaw

Presence Group

Standard Presence Group

SIP Trunk Security Profile

E2K7 Non Secure SIP Trunk Profile

Rerouting Calling Search Space

<None>

Out-of-Dialog Refer Calling Search Space

<None>

SUBSCRIBE Calling Search Space

Default

SIP Profile

Standard SIP Profile

DTMF Signalling Method

No Preference

Posted in Cisco 4.x Integration, Cisco 5.x Integration, Cisco 6.x Integration, Cisco 7.x Integration, Good Articles take from Other Blogs, OCS & Exchange07, Unified Messaging | Tagged: , , , , , , , , | 8 Comments »

How to enable inbound fax for OCS 2007 Enterprise Voice and Exchange 2007 UM enabled users?

Posted by Mino on March 9, 2009

Any Post starting with this disclaimer means that this post was not written by me however I have liked it and added to my blog. I will also include the link to the original or Similar post to provide credit to the original author.

http://blogs.technet.com/jenstr/archive/2007/11/13/how-to-enable-inbound-fax-for-enterprise-voice-and-exchange-2007-um-enabled-ocs-2007-users.aspx

Exchange 2007 SP1 UM supports both voice mail and incoming fax to a given extension. However, if the user is both UM-enabled and enabled for Enterprise Voice using OCS 2007, incoming fax is not supported using the same extension. The reason being that OCS 2007 Mediation Server does not currently support T.38.

How is it possible to provide incoming fax support for Enterprise Voice enabled users? The answer is to use a separate extension for fax and route fax calls to this extension directly to Exchange 2007 SP1 UM outside of OCS 2007.

Let’s assume we have a company called Contoso with the environment shown below and we will use that company to explain the issue and the solution

12

The OCS 2007 environment is connected to the PBX via a SIP/PSTN gateway called PSTNOCSGWY. The PBX routes all calls to the DID range +131255xxxxx to OCS 2007. OCS 2007 is integrated with the Exchange 2007 SP1 UM server called UMSRV1. It hosts a UM Dial Plan called OcsUmDialPlan of UriType = SipName (required for OCS 2007 integration). There is a UM Mailbox Policy associated with this UM dial plan called OcsUm. Exchange 2007 SP1 UM is connected to the PBX via OCS 2007.2

The Contoso user Test User is enabled for Enterprise Voice with the DID +13125510001 and SIP URI TestUser@contoso.com. His extension is 10001. His Enterprise Voice configuration is shown below.

 

 

To be enabled for Exchange 2007 SP1 UM the administrator would issue the following Exchange Management Shell command:

Enable-UmMailbox -id TestUser -UmMailboxPolicy OcsUmPolicy -Extensions 10001 -SIPResourceIdentifier TestUser@contoso.com -Pin 1234

Test User is now enabled for Exchange 2007 SP UM, but will not be able to receive incoming fax on extension 10001 or DID +13125510001.

As indicated above the solution is to give Test User a separate extension for fax and the extension needs to be routed to Exchange 2007 SP1 UM directly without going through OCS 2007. Contoso will therefore have to create a configuration as shown below. There is a dedicated SIP/PSTN gateway for connectivity to Exchange 2007 SP1 UM. The PBX routes the DID range +131266xxxxx to this SIP/PSTN gateway. There is a new UM Dial Plan called UmDialPlan with UriType=TelExtn. There is a UM Mailbox Policy associated with this UM dial plan called Um. The UM server UMSRV1 hosts both UM Dial Plans.

The administrator now decides that Test User should have the extra extension 11001 and DID +13126611001 as the fax number.

To enable Test User to receive fax the administrator need to issue the following Exchange Management Shell command:

Set-Mailbox -id TestUser -SecondaryAddress 11001 -SecondaryDialPlan UmDialPlan

With the above configuration Test User is now able to receive fax on DID +13126611001.

3

Posted in AVAYA, Cisco 4.x Integration, Cisco 5.x Integration, Cisco 6.x Integration, Cisco 7.x Integration, Mediation Server, Nortel CS1000, OCS & Exchange07, OCS 2007 R2, PBX Integration, Quintum's gateways, Unified Messaging | Tagged: , , , , , , , , , , , , , , , , | 13 Comments »

Office Communications Server and the Contact Center

Posted by Mino on March 4, 2009

Any Post starting with this disclaimer means that this post was not written by me however I have liked it and added to my blog. I will also include the link to the original or Similar post to provide credit to the original author.

https://blogs.pointbridge.com/Blogs/mcgillen_matt/Pages/Post.aspx?_ID=58

One of the best places to add value to an organization is in the way contacts with customers or partners is handled. For many – this is the call center / contact center. This is how sales are made, support tickets are resolved, new customers are discovered, current customers are retained

Also we as LINK Development were able to integrate these new Contact center features with Microsoft CRM server to provide end to end Helpdesk Contact center solution on the Call handeling and on the Backend activity system.

crm11

 

crm21

There are several new features that can be used for call routing. I thought I’d build a matrix to keep it straight:

R2 Feature Description Ease to Deploy Admin Control Deployment Options
Call Delegates Have an admin answer calls for you, or multiple people Very Easy End User Controlled Built in to OCS R2 client – point and click interface
Team Call Have calls ring a group of users simultaneously Very Easy End User Controlled Built in to OCS R2 client – point and click interface
Basic Hunt Group Ring a group of users in various methods: longest available, circular, serial, round-robin etc. Easy OCS Admin can delegate mgmt. to any user Configured in OCS console, managed via web interface
Enhanced Hunt Group Same as above, but with Welcome Message & Open/Closed Hours Easy OCS Admin can delegate mgmt. to any user Configured in OCS console, managed via web interface
Response Groups Call queueing, music on hold, text-to-speech, speech recognition, open/closed hours Medium OCS Admin can delegate mgmt. to any user Configured in OCS console, managed via web interface
IVR / Speech applications   Somewhat Complex OCS Admin Using the UCMA 2.0 SDK, write “drag-and-drop” Windows Workflow applications / .Net applications
Enterprise Contact Center Functionality Write your own contact center workflows Complex OCS Admin Using the UCMA 2.0 SDK, write “drag-and-drop” Windows Workflow applications / .Net applications

 

IVR / Contact Center Applications with the UCMA 2.0 SDK

This is by far the most exciting feature of OCS R2. MS is giving away the toolkit (the Unified Communications Mangaed API aka UCMA SDK) to write your own IVR apps and contact center apps: anyone with an IM client can be an “agent”! No additional licensing necessary. This is going to change the way people do contact centers.

Posted in AVAYA, Cisco 4.x Integration, Cisco 5.x Integration, Cisco 6.x Integration, Cisco 7.x Integration, Nortel CS1000, OCS 2007 R2, PBX Integration, Quintum's gateways | Tagged: , , , , , , , , , | 1 Comment »

How to configure a SIP trunk between Cisco Call Manager 5.x or 6.x or 7.x and OCS 2007 R1 or R2

Posted by Mino on February 20, 2009

How to configure a SIP trunk between Cisco Call Manager 5.x or 6.x and OCS 2007 R1 or R2

 

Ok you want to ring from MOC to Cisco IP phone and back  , hmmm ok then simple we will deal with it as if OCS is an IP PBX with its extensions 3xxx and you need to connect it with Cisco PBX with extensions 7xxx.

To do that we need a SIP trunk and for the SIP trunk to work fine we need to have some specific configuration on that trunk , remember any of these settings if they not configured right then you will not be able to make a stable calling between Cisco and OCS.

 

First we do the SIP trunk :

 trunk1

 

 

trunk2

 

trunk3

 

Now the SIP trunk which is acting like the bridge between the cisco and the OCS is created , ok then we need now to create a criteria where this trunk is going to be used in.  This is where is Pattern comes in where we will say if a Cisco phone set tries to dial extensions starting with 3xxx then you use the trunk which we have just created .

and from the way back from the OCS to Cisco , when the number is sent in the E164 formate with the + , the Cisco will simply ignore all that and will take only the last 4 Digits which are the 7xxx

pattern1

 

pattern2

 

 

Now you are ready to make the call and Enjoy the Integration

J

Posted in Cisco 5.x Integration, Cisco 6.x Integration, Cisco 7.x Integration, Mediation Server | Tagged: , , , , , , | 62 Comments »

How to strip “Remove” the + from all outgoing SIP communication from an OCS Mediation server ?

Posted by Mino on October 13, 2008

It might be useful in a Cisco direct SIP environment to automatically strip the + from all outgoing SIP communication from an OCS Mediation server.

To do this, create a text (XML) file called MediationServerSvc.exe.config and place it in the location of the MediationServerSvc.exe file. It should be in the ‘C:\Program Files\Microsoft Office Communications Server 2007\Mediation Server’ directory. The contents of this file should be:

<?xml version=”1.0″ encoding=”utf-8″ ?>
<configuration>
<appSettings>
<add key=”RemovePlusFromRequestURI” value=”Yes”/>
</appSettings>
</configuration>

Now, you are free to normalize to E.164 without having to worry about your Cisco devices getting confused!

Srouce : http://jimraymond.wordpress.com/ 

Posted in Cisco 4.x Integration, Cisco 5.x Integration, Cisco 6.x Integration, Mediation Server, Miscellaneous, Nortel CS1000, PBX Integration, Quintum's gateways | Tagged: , , , , , , , , , | 2 Comments »

Cisco Unified Communications Manager 6.0 PBX Configuration Note

Posted by Mino on October 9, 2008

cisco-unified-communications-manager-60

This topic provides link to configuration notes for Cisco Unified Communications Manager 6.0 that have been created and tested by Microsoft . When Microsoft or a partner deploys Exchange 2007 Unified Messaging with a new IP gateway and PBX or IP PBX configuration, the prerequisites and configuration settings are documented. This information is used to create a configuration note.

PBX configuration note contains information about how to deploy Exchange 2007 Unified Messaging with a specific telephony configuration including the manufacturer, model, and firmware version for the IP gateways, IP PBXs or PBXs. In addition, each PBX configuration note also includes other information such as:

  • Contributors in authoring the configuration note.
  • Detailed prerequisites, including the following:
    • Features that have to be enabled or disabled on the PBX.
    • Specialized hardware that has to be installed.
    • Is an IP gateway required?
    • Features that must be present on the IP gateway if one is needed.
    • Specific cabling requirements between an IP gateway and a PBX.
    • A list of Unified Messaging features that may not be available with a given telephony configuration

Posted in Cisco 6.x Integration | Tagged: , , , , , | 2 Comments »

 
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