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Cisco Unity Connection SIP Integration with CUCM Step by Step Guide
By UCCollaborationGeek | November 8, 2016 4 Comments
Cisco Unity Connection messaging system is an awesome product that delivers voice messaging feature for small business VoIP and unified
communications infrastructure. Cisco Unity Messaging system voicemail can be supports single server or redundant (Cluster) deployment.
Production network uses two Cisco Unity Connection servers in a cluster along with SIP trunk redundancy from Call manager.
There are multiple ways to integrate Cisco Unity Connection to Cisco Unified Communications Manager either via SCCP or SIP Trunk. The
traditional way to integrate Unity Connection with CUCM is using SCCP, but here I would like to discuss Cisco Unity Connection SIP
Integration with CUCM.
I have seen most of the business phone systems uses Cisco Unity Connection SIP integration with Cisco Unified Communications Manager.
Section 1: CUCM Side Con gurations for Cisco Unity Connection SIP Integration
First, let’s understand the call flow between CUCM and CUC. How do we direct our call to an external server? There are many possible ways
to get this done. In the case of CUC, we will be going for a SIP Trunk that is pointing to the CUC IP address from CUCM. Please refer the
below flow chart to comprehend the call flow and configuration checklist.
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SIP Trunk (SIP Trunk Security Pro le, SIP Pro le, Trunk)
Route Group
Route List
Route Pattern
Voice mail pilot
Voice mail Pro le
The configurations for Cisco SIP Trunk for CUC can be divided into 3 steps. Two prerequisites and last is the actual trunk configuration.
Step 1: SIP Trunk Security Profile Configuration
Go over to System >> Security >> SIP Trunk Security Profile >> Find
Copy the ‘Non Secure SIP Trunk Profile’
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>> Save
Step 2: SIP Profile Configuration
Go to Device >> Device Settings >> SIP Profile
Copy the ‘Standard SIP Profile’
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Confirm the following parameters and values (all these are default)
Under SIP OPTIONS Ping section please check the Enable [ ] OPTIONS Ping to Monitor destination status for Trunks with service
type “None (Default)”
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[Note: This will enable the trunk status monitoring feature from CUCM. You can see whether the SIP Trunk is active or down.]
Device >> Trunk >> Add New >> SIP Trunk >> Next
Provide Name, Description and Device Pool.
Note: The device pool (TEXAS_APP_SERVER_DEVICE_POOL) used to ensure G.711 or G.722 codec between CUCM and CUC.
Look at the Region configured under the device pool (The audio codec set to 64 Kbps between TEXAS_APP_SERVER_REGION and all other
regions.)
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Under Inbound Calls select the Calling Search Space. Check the Redirecting Diversion Header Deliver – Inbound (Important)
Note: This CSS should have visibility to all phones DN. It plays an important role for MWI (Message Waiting Indicator).
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In Outbound Calls section Check the Redirecting Diversion Header Delivery – Outbound
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>> Save
This step is not mandatory, but this will give more flexible and scalable deployment of Cisco Unity Connection. After creating SIP Trunk, you
can directly move to Step 4. By considering the production network design I’m going as per the industry standard.
By the way, What is Route Group? A route group is a logical group in which redundant Gateways and Trunks are mapped.
Go to Call Routing >> Route/Hunt >> Route Group >> Add New
Route Group Name: CUC_ROUTE_GROUP
From Available Devices move the CUC_SIP_TRUNK to Selected Devices and Save
[Note: While adding another Cisco Unity Connection Subscriber to the cluster, the Route List design will make the configurations much
easier.]
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>> Save
Here we have to decide what will be the directory number to be used to reach Cisco Unity Connection from Call Manger. We will intern MAP
this number to MESSAGE button of the IP phone.
In this specific post, I’m taking 1050 as the pattern that would point to CUC.
Go to Call Routing >> Route/Hunt >> Route pattern >> Add New
Configure the Route Pattern as follows and select CUC_ROUTE_LIST in Gatway/Route List section.
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[Note: I kept this number (1050) under TEXAS_INTERNAL_PT partition where all my internal numbers belong to.]
The voicemail pilot number designates the directory number that you dial to access your voice messages. Cisco Call Manager automatically
dials the voice-messaging number when you press the Messages button on your phone. Literally the Voice-mail pilot is exactly the same as
the route pattern that we created in Step 4.
Go to Advanced Features >> Voice Mail >> Voice Mail Pilot >> Add New
Voice Mail Pilot Number : 1051
Calling Search Space : TEXAS_VM_PILOT_CSS
>> Save
[Note: This CSS should have the visibility of Route Pattern’s (1051’s) partition TEXAS_INTERNAL_PT]
This allows you to define any line-related voicemail information that is associated with a directory number.
Go to Advanced Features >> Voice Mail >> Voice Mail Profile >> Add New
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Description : TEXAS_VM_PROFILE
Voice Mail Pilot : 1051/TEXAS_VM_PILOT_CSS
We will apply this profile to all DN’s in the cluster by going to Device >> Phones >> [Search and select phones] >> Go to line page >>
Under Directory Number Settings, Voice Mail Profile select our TEXAS_VM_PROFILE.
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And now you can set Call Forwarding to Cisco Unity Connection check boxes in line page.
If the user did not attend the call, it should forward to CUC after 15 Sec.
Section 2: CUC Side Con gurations for Cisco Unity Connection SIP Integration
There some few configurations left out in Cisco Unity Connection in order to talk CUC to CUCM over the SIP trunk.
Login to Cisco Unity Connection Administration window and go over to Telephony Integration >> Phone Systems.
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No configuration required in this step, keep all the settings as default. The only change which I have done is that the Phone System Name.
Now just below Phone System go to Port Group >> Add New
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>> Save
[Contact Line Name-: This is the FROM address that Unity Connection will use for SIP messages. We basically set this to be the DN or DID of
your voicemail pilot. If this is not a DID you can also use a Calling Party Transformation CSS on the CUCM SIP Trunk to transform the internal
DN to an externally valid DID.
Additionally, there is a field in the Port Group under Edit > Advanced Settings labeled Remote-Party-ID.
This is the Caller ID Name that Unity Connection will set in the FROM address of SIP messages. If you do not set this calls to Unity
Connection will only see the Contact Line Name.]
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These are the logical port which unity will receive calls from CUCM.
The call set up originated via the SIP trunk will terminate at the CUC through logical voice mail ports. Voice mail ports are licensed feature;
the more port you purchase the more concurrent session it will provide.
For example, if you have 2 voicemail port, only two users can access voice mail simultaneously.
Just below Port Group, you can see Port >> Add New
Here only you need to specify how many number of ports required. In this example, I have used 10 voice mail ports.
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>> Save
Good Job! You have successfully integrated CUCM and CUC via SIP Trunk.
Now let’s validate Cisco Unity Connection SIP integration. From CUC, go to Telephony Integration >> Phone System, in the related links
section you can see ‘Check Telephony Integration‘ >> Go
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Now go back to CUCM and check the trunk status, you can see the uptime and service.
Hit voice message button from the IP Phone (keep in mind, the voice mail profile should be selected in line page (Part 6), you will be treated
with “Cisco Unity Connection Greetings”. Explanation of accessing and Sending voice messages will be discussed in later articles.
Please Read:
How con gure and Setup User mail boxes in Cisco Unity Connection
How to Check Voicemail in Cisco IP Phone
How Check Voicemail from Another Phone
Open Visual Voicemail
Troubleshoot MWI Issues
Hope you enjoyed this article, I will be coming up with another Cisco Unity Connection tutorial soon. Please share this post and like our
Facebook page to get the latest updates.
4 thoughts on “Cisco Unity Connection SIP Integration with CUCM Step by Step Guide”
Ritesh
December 30, 2016
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