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

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

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

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>

How to Integrate Exchange UM Voicemail into Cisco IP Phones

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

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

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

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.

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.

Communicator 2007 R2 Phone Accessibility

A common request for RFP is support for TTY. What is it? A TTY is a device that enables people who are deaf, hard of hearing, or speech-disabled to use the telephone by typing messages back and forth to one another instead of talking back and forth. In order to communicate, a TTY is required at both ends of the conversation, unless the call is placed through Relay. OCS now supports integration with TTY with OCS R2.

Under options you can turn on tty mode and connect a keyboard to a device such as the Catalina phones for OCS.  I haven’t seen documentation on this yet but I’ll post when I know more.

Office Communications Server and the Contact Center

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.

 

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.

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

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 :

 

 

 

 

 

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

 

 

 

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

J

OCS Integration with AVAYA ( 1 of 2 )

Hi all,

As i have promissed you that I will be back with lots of great information regarding the AVAYA integration.

here we go , i was working with a huge client in Dubai where he has just purchased an AVAYA IP-PBX along with Sip Enablement Server ( SES ) and AVAYA Application Enablement Server ( AES ) .

ok first of all lets us all memorize these two terms , AVAYA SES and AVAYA AES because we will be talking about them both allot.

There are two stories for the AVAYA-Microsoft integration , the AVAYA story and the Microsoft story. we will talk about both stories and the integration idea from each vendor , the limititions from each way of integration along with the pros and cons.

Microsoft Story :

For Microsoft the integration with AVAYA is simply by dealing with them as two separate IP-PBX where Microsoft has an extensions of 3xxx for example and AVAYA with extensions of 7xxx .where you need to create a SIP trunk between both of them to be able to call from here to there and vise verse.

This assures microsoft that the call processing will happen in the Mediation server , which is the target of Microsoft in order to help its users to move to the Microsoft IP PBX concept bit by bit.

for this type of integration you will need AVAYA SIP enablement server only which shall give the AVAYA communication manager the SIP capability to communicate with other system , in this SES we will configure a sip trunk for the 3xxx extensions to route to the mediation Server over TCP-IP port 5060 or through TLS over 5061 after you configure the certificate on the SES Server.

below is a diagram for this type of integration .

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Good Thing :

1- AVAYA SES is not costy at all  , 2-Full featured OCS capabilities including conferencing,  3- you can do dual forking workarround by adding AVAYA extensions as the similtaniously ring another number in OCS

Bad thing:

no dual forking between AVAYA and OCS which means that each person will have two extensions .

in my oppenion , no Big deal …the client was so happy with this scenario because the second AVAYA scenario will cost him 60$ per users license plus it will not provide him with all capabilities of OCS ( the conferencing for example is dimmed )

in the Second article we will discuss the second way of integration through the AVAYA AES Server

Cisco Unified Communications Manager 7.0 PBX Configuration Note

This document describes the simultaneous ring feature interoperability and documents the steps and configurations necessary for Cisco Unified Communications Manager (Cisco UCM) Release 7.0(1) to interoperate with Microsoft Office Communications Server (OCS) 2007 Enterprise Edition. It aims to provide a good understanding of what works and what does not work in terms of the feature interaction between a Cisco UCM device and Microsoft Office Communicator. It also provides guidance to deployment participants regarding the limitations, expected behaviors, and known issues. Please note that this document does not address performance and scalability, which are part of broader criteria for a deployment-ready solution.

 

· The simultaneous ring feature allows a Cisco UCM endpoint (IP Phone, IP Communicator, etc…) to simultaneously ring its remote destination(s) when a call is placed to that endpoint. The remote destination(s) could be Microsoft Office Communicator and/or any other device(s) including mobile phones, PSTN phones or even local PBX extensions. The remote destination feature uses the Cisco Unified Mobility functionality and can be configured to allow or block the receipt of certain numbers. For the simultaneous ring configuration, Cisco UCM has a Direct SIP connection to Microsoft Mediation Server.

cisco-unified-communications-manager-7011

Application Notes for Microsoft Office Communicator Clients with Avaya Communication Manager Phones

These Application Notes describe the configuration procedures for enabling Microsoft Office

Communicator (MOC) clients to operate Avaya Communication Manager phones. The MOC

client and Microsoft Office Communications Server (OCS) provide the user with multiple

modes of communications, such as Instant Messaging (IM), voice, and video. The

configuration steps described herein focus on the TR/87 integration between Avaya

Application Enablement Services and Microsoft OCS. The TR/87 integration allows MOC

clients to make and answer calls on Avaya Communication Manager phones, as well as

perform basic call control operations such as hold/retrieve, transfer, and disconnect.

ocs-acm-rcc

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

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/