Вы находитесь на странице: 1из 7

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”

c. Check “Enable OPTIONS Ping”

“CUC SIP Profile” c. Check “Enable OPTIONS Ping” Morgan Stepp CCIE #12603 | morgstepp@gmail.com Page 1
“CUC SIP Profile” c. Check “Enable OPTIONS Ping” Morgan Stepp CCIE #12603 | morgstepp@gmail.com Page 1
“CUC SIP Profile” c. Check “Enable OPTIONS Ping” Morgan Stepp CCIE #12603 | morgstepp@gmail.com Page 1

CUC3. 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 DN’s

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”

Calls Check “Redirecting Diversion Header Delivery” SIP Information a. Enter IP of CUC Server b. Select

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 DN’s

d. Save and reset trunk

with access to all phone DN’s d. Save and reset trunk Morgan Stepp CCIE #12603 |

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

h. Select Edit > Advanced Settings

i. Enter Voicemail as Remote-Party-ID

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

desired Server c. Balance Number of Ports required for MWI Morgan Stepp CCIE #12603 | morgstepp@gmail.com
desired Server c. Balance Number of Ports required for MWI Morgan Stepp CCIE #12603 | morgstepp@gmail.com
desired Server c. Balance Number of Ports required for MWI Morgan Stepp CCIE #12603 | morgstepp@gmail.com
desired Server c. Balance Number of Ports required for MWI Morgan Stepp CCIE #12603 | morgstepp@gmail.com

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 DN’s

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 DN’s

c.

Assign Voice Mail Profile to Phone Line appearance

3.

Add SIP Trunk to new Route Group

Line appearance 3. Add SIP Trunk to new Route Group Call Routing > Route Hunt >
Line appearance 3. Add SIP Trunk to new Route Group Call Routing > Route Hunt >

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

Route List Call Routing > Route Hunt > Route List a. b. Add the CUC SIP

a.

b.

Add the CUC SIP Trunk Route Group to a new Route List

Under Route List Detail, ensure “Use Calling Party’s 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

a. Add Route Pattern that matches your Voice Mail Pilot Morgan Stepp CCIE #12603 | morgstepp@gmail.com

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

CUBE SIP Normalization

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.

From Address that Unity Connection will use in SIP messages. Morgan Stepp CCIE #12603 | morgstepp@gmail.com

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).

Supervised Transfer

ANI of the PSTN caller (5132577500). Supervised Transfer Release to Switch If a CUC Supervised Transfer

Release to Switch

caller (5132577500). Supervised Transfer Release to Switch If a CUC Supervised Transfer call were placed to

If a CUC Supervised Transfer call were placed to a SIP PSTN, the outbound ANI would be Anonymous and would not match a DID assigned to your SIP Trunk. The SIP carrier would not recognize the Calling Party as a registered SIP DID and would reject the call.

INVITE sip:916145552001@10.0.4.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.8:5060;branch=z9hG4bK19dcf4077feda From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=183332 To: <sip:916145552001@10.0.4.10>

We resolve the Supervised Transfer issue by assigning a valid SIP DID to the Contact Line Name for the CUC Port Group. Supervised Transfers will now use the Port Group’s DID value. Release to Switch Transfers would continue using the Originating Caller ANI.

Unity Connection Configuration Within CUC, we configure the Port Group’s 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.

call to complete. Select save, then reset the Port Group. Morgan Stepp CCIE #12603 | morgstepp@gmail.com

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.

This ANI is now presented to the transfer destination. Validate Supervise Transfer delivers Originating ANI Within
This ANI is now presented to the transfer destination. Validate Supervise Transfer delivers Originating ANI Within

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 Group’s Contact Line Name setting of 5135551001 is the presented ANI. This is a valid DID and satisfies the SIP Provider’s Screened ANI requirement.

satisfies the SIP Pro vider’s Screened ANI requirement. Normalize E164 Deployment with SIP If using E164
satisfies the SIP Pro vider’s Screened ANI requirement. Normalize E164 Deployment with SIP If using E164

Normalize E164 Deployment with SIP If using E164 deployment, ensure your outgoing SIP Trunk to Cube has a redirecting party CSS set to include a Call Party Transformation of \+1.XXXXXXXXXX with strip predot. This ensures the redirecting party is not E164 and matches your registered DID block.

CCM Ringback During Supervise Transfer, the remote calling party may not hear ringback. In CCM Service Parameters, under CallManager, set "Send H225 User Info Message" to “Use ANN for Ring Back”.