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Unity Connection SIP Integration with Communications Manager

Cisco Unity Connection (CUC) and Cisco Communications Manager (CCM) can be integrated using SIP. Instead of multiple
SCCP ports involved with traditional CUC to CCM integrations, SIP uses a single Trunk per CUC server. The SIP integration
eliminates the requirement to configure Directory Numbers for Voicemail Ports and MWI.


SIP Configuration

1. CCM SIP Trunk Security Profile
System > Security > SIP Trunk Security Profile
Copy Non Secure SIP Trunk Profile

SIP Trunk Security Profile Information
a. Name: CUC SIP Trunk Security Profile
b. Device Security Mode: Non Secure
c. Incoming Transport Type: TCP+UDP
d. Outgoing Transport Type: TCP

Check the following:
a. Accept out-of-dialog refer
b. Accept unsolicited notification
c. Accept replaces header





2. CCM SIP Profile
Device > Device Settings > SIP Profile
a. Copy Standard SIP Profile
b. Name new Profile CUC SIP Profile













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3. CCM SIP Trunk
Configure a SIP trunk for each CUC server.
Device > Trunk

System Components
a. Device Pool and Region to support selected codec
b. MRGL with MOH and CFB media resources
c. Calling Search Space with access to all phone DNs

Device Information
a. Name: CUC_PUB (or other descriptive name)
b. Device Pool: System
c. MRGL: PUB_SUB1

Inbound Calls
a. Significant Digits: All
b. Calling Search Space: System

Outbound Calls
Check Redirecting Diversion Header Delivery





SIP Information
a. Enter IP of CUC Server
b. Select SIP Trunk Security Profile and
SIP Profile created earlier





c. Select a Calling Search Space with
access to all phone DNs


d. Save and reset trunk







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4. CUC SIP Port Group and Ports
Configure SIP Port Group and Ports on CUC

Port Group
Telephony Integrations > Port Group
a. Associate with Primary Phone System
b. Set Port Group Type to SIP
c. Enter IP of Primary CCM Call Processor
d. Select Save
e. Select Edit > Servers
f. Enter IP of additional Call Processors
g. Save and Reset Port Group











Ports
Telephony Integrations > Port
a. Add New Ports to equal licensed quantity
b. In HA deployments, associate Port with desired Server
c. Balance Number of Ports required for MWI


















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CCM Dial Plan
The SIP integration eliminates the requirement to configure Directory Numbers for Voicemail Ports and MWI. For
general voicemail use, the only directory number required in CCM is the Voicemail Pilot. If in use, greetings
administrator access will require a unique directory number. Steering users to specific CUC servers within a cluster will
also require unique directory numbers as separate voicemail pilots.



1. Voice Mail Pilot
Advanced Features > Voice Mail > Voice Mail Pilot
a. Enter a voicemail pilot DN
b. Select a CSS with access to all phone DNs





2. Voice Mail Profile
Advanced Features > Voice Mail > Voice Mail Profile
a. Enter a voicemail pilot DN
b. Select a CSS with access to all phone DNs
c. Assign Voice Mail Profile to Phone Line appearance


3. Add SIP Trunk to new Route Group
Call Routing > Route Hunt > Route Group
a. Add the CUC SIP Trunk a new Route Group
b. For cluster environments, create a Route Group for each server


4. Add SIP Trunk Route Group to new Route List
Call Routing > Route Hunt > Route List
a. Add the CUC SIP Trunk Route Group to a new Route List
b. Under Route List Detail, ensure Use Calling Partys External Phone Number Mask is set to Off.


5. Route Pattern
Call Routing > Route Hunt > Route Pattern
a. Add Route Pattern that matches your Voice Mail Pilot

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Unity Connection with SIP PSTN

SIP PSTN Screened ANI
SIP PSTN providers generally require that your outbound ANI match a DID assigned to your SIP Trunk. If you wish to out
pulse a Toll Free or other ANI not assigned to your Trunk, you can insert a Screened Telephone Number (STN) into the P-
Asserted-ID (PAI) SIP header field. CUCM and CUBE offer normalization scripts which replace outbound ANI with carrier
accepted STN.

CUCM SIP Normalization
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b06scrpt.html

CUBE SIP Normalization
http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a0080982499.shtml


Unity Connection with SIP PSTN
Unity Connection offers a Contact Line Name to change the SIP From Address. The Contact Line Name is located in
Unity Connection > Telephony Integrations > Phone System > Port Group. In this example, we will place calls from
513.257.7500 to a Unity Connection Auto Attendant at 513.555.1000. Within the Auto Attendant, we will choose a
DTMF Input which performs a Transfer to the PSTN. We will use a valid SIP DID as for the Contact Line Name value to
alter the From Address that Unity Connection will use in SIP messages.















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Unity Connection Transfers
The two screenshots below display transfers from a Unity Connection (CUC) server which has been integrated to CUCM
using SIP. With the Supervised Transfer, CUC does not originate an ANI (Unknown Number). With the Release to Switch
Transfer, Unity Connection presents the Originating ANI of the PSTN caller (5132577500). If these transfers were placed
to a SIP PSTN with Screened ANI, the outbound ANI of either transfer method would not match a DID assigned to your
SIP Trunk, resulting in call failure for both.

Supervised Transfer Release to Switch














CUCs ANI presentation also presents an issue with the Screened ANI normalizations of CUCM and CUBE scripts as they
both require an ANI to match and replace. We could use a Calling Party Transformation Pattern on the CUC SIP Trunk,
though this would change the ANI of all calls inbound to CUCM from CUC. An alternative method is to assign a valid SIP
DID to the Contact Line Name for the CUC Port Group. Supervised Transfers will now use the Port Groups DID value.
Release to Switch Transfers would continue using the Originating Caller ANI.


Unity Connection Configuration
Within CUC, we configure the Port Groups Contact Line Name setting with a valid DID of 5135551000. The SIP PSTN
(CUBE) in our Lab will recognize this as valid ANI and allow the call to complete. Select save, then reset the Port Group.



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Validate Release to Switch Transfer delivers Originating ANI
Within CUC, configure a Release to Switch Transfer for an On Net IP Phone. Call into the configured CUC extension and
choose the configured Caller Input Option. Our test call has an originating ANI of 5132577500. This ANI is now presented
to the transfer destination.














Validate Supervise Transfer delivers Originating ANI
Within CUC, configure a Supervise Transfer for an Off Net PSTN number. Configure the Rings to Wait For value at 8. This
ensures the PSTN call will complete. Call into the configured CUC extension and choose the configured Caller Input
Option. The Port Groups Contact Line Name setting of 5135551001 is the presented ANI. This is a valid DID and satisfies
the SIP Providers Screened ANI requirement.