cisco unified communications manager sip integration

CISCO UNIFIED COMMUNICATIONS MANAGER
SIP INTEGRATION
Validated Integrations:
8.5 with xIC version 3.0 SU-10 and greater
INTEGRATION DOCUMENT
Version 2.03
7601 Interactive Way
Indianapolis, IN 46278
Telephone/Fax: (317) 872-3000
www.ININ.com
Cisco Unified Communication Manager SIP Integration
App Note
Table of Contents
CISCO UNIFIED COMMUNICATIONS MANAGER SIP INTEGRATION ....................................................1
TABLE OF CONTENTS ..........................................................................................................................................1
REVISION CONTROL ............................................................................................................................................2
INTRODUCTION .....................................................................................................................................................3
CONFIGURATION SUMMARY ............................................................................................................................4
Extension Details ..................................................................................................................................................... 5
CONFIGURING A SIP TRUNK ON THE CUCM ................................................................................................6
SIP Trunk Security Profile (Option 1 and 2)........................................................................................................ 6
SIP Profile (Option 1 and 2)................................................................................................................................... 8
SIP Trunk (Option 1 and 2) ................................................................................................................................... 9
Route Pattern (Option 1) ...................................................................................................................................... 12
Route Pattern (Option 2) ...................................................................................................................................... 12
Voicemail Pilot (Option 1 and 2) ......................................................................................................................... 13
Fault Tolerance Configuration (Option 1 and 2) ............................................................................................... 14
Creating additional SIP Trunks ...................................................................................................................... 14
Creating a Route Group and Route List .............................................................................................................. 14
Cisco CUCM Cluster Support (Option 1 and 2) ................................................................................................ 18
Interaction Message Indicator: ............................................................................................................................ 18
CIC and MIC: ..................................................................................................................................................... 18
Additional CUCM Feature Configuration (Option 1 and 2)............................................................................. 20
Putback Transfer (All call control and RTP out of IC) ....................................................................................... 20
MULTIPLE CUCM REGIONS ............................................................................................................................. 20
IC SERVER CONFIGURATION .......................................................................................................................... 21
Media Server ......................................................................................................................................................... 21
Line Configuration ............................................................................................................................................... 23
MWI ....................................................................................................................................................................... 25
Station Configuration ........................................................................................................................................... 26
User Configuration ............................................................................................................................................... 28
AUDIO PATHS ........................................................................................................................................................ 29
NETWORK ARCHITECTURE FOR CUCM SIP INTEGRATION ................................................................. 29
Architecture 1- H.323/MGCP gateway connected to CUCM and SIP gateway connected to IC server ....... 29
Architecture 2- H.323/MGCP gateway connected to CUCM............................................................................ 31
Architecture 3- SIP gateway connected to IC server ......................................................................................... 32
Architecture 4- CUCM station to station call ..................................................................................................... 33
CUCM STATIONS: TRANSFER SCENARIOS – CALL FLOW ..................................................................... 34
Call from CUCM station to station - Transfer From Phone ............................................................................. 34
Call from PSTN to CUCM Station -Transfer from Phone ................................................................................ 34
Call from CUCM station to station - Transfer from IC Client ......................................................................... 35
Call from PSTN to IC Client - Transfer from IC Client ................................................................................... 35
CUCM STATIONS: CONFERENCE SCENARIOS – CALL FLOW .............................................................. 36
Conference between CUCM Stations – Conference from Phone...................................................................... 36
Conference between PSTN and CUCM Stations - Conference from Phone .................................................... 36
Conference between CUCM Stations - Conference from IC Client ................................................................. 37
Call from PSTN to IC Client - Conference from IC Client ............................................................................... 37
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Revision Control
Version
Date
Change Description
Author
2.00
3/27/11
Initial draft.
Jason Probala
2.01
4/25/11
Modified for 8.5
Jason Probala
2.02
4/26/11
Added Audio Paths
Jason Probala
2.03
4/29/11
Added Codec support with CUCM regions
Jason Probala
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Introduction
Integrating with Cisco Unified Communications Manager via SIP opens the door for leveraging existing
flexibility and product support without relying on the Cisco TSP. It allows for customers to bring
converged architectures onto a single platform without sacrificing important product functionality. The
integration will continue to use CUCM station configuration, MGCP and H.323 gateway connections, as
well as add support for a distributed environment with audio flowing directly between the IC Media
Server and the two endpoints. Remote offices, with their local MGCP, H.323, or SIP gateways can take
full use of Media Server capabilities, by keeping the audio local yet maintaining call recording.
Stations are configured as standard IC workstations, allowing for MWI and consistency of dialing from
either the CUCM phone or the IC client. Phone functionality, such as hold and conference, are also
persevered when done either from the phone or the IC client. Inbound calls, both from another CUCM
station or the PSTN via an MGCP/H.323 gateway, are sent over from CUCM and routed to the intended
user. The user, if logged on to a CUCM workstation configured in IA, will be alerted on their CUCM
controlled phone. Outbound calls (from the phone) can be configured to appear in the IC client as if it
were dialed from the client itself.
Products that have been limited in a TAPI environment, such as Interaction Dialer, because of the lack of
call analysis, can now be integrated into a CUCM environment. Using an Interaction Intelligence
Gateway for outbound dialing is now an option with agents using CUCM stations.
Testing has been completed for the use of SIP stations connected directly to the IC server concurrently
with CUCM stations. The testing required a significant number of additional conference, transfers, hold,
coaching, and monitoring scenarios that were proved to integrate well when using a media server.
Interactive Intelligence has completed interoperability testing with Cisco Unified Communication
Manager version 8.5 with xIC version 3.0 SU-10. This document is provided to show configuration,
limitations, additions, and best practices for a successful integration.
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Configuration Summary
This section highlights the major configuration settings, requirements, and limitations that are important
to a successful integration. These items are described in more detail in the appropriate sections later in
the document.
IC Configuration
•
IC Station and Line audio settings need to be configured to use TCP and Always-In and with
Normal Media. (pages 15-16)
•
The connection address is manually configured because the CUCM SIP trunk does not
dynamically register. (page 18)
•
The connection address is set to the Cisco Phone extension. The full connection address will be
set when the proxy configured on the SIP line is added to complete the address. (page 18)
•
The Station configuration will have Use Proxy for Station Connection selected. (page 20)
•
The Line configuration will use TCP and the proxy on the line will have a list of the CUCM’s for
the Proxy address. (pages 15-16)
•
Interaction Media Server should be used in this configuration and is used on all calls because
the line setting is configured as Always-In. (page 14)
CUCM Configuration
•
In the CUCM SIP Trunk configuration, MTP should to be unselected to allow audio to flow
directly from endpoint to endpoint; bypassing CUCM as an intermediary. (page 10)
•
SIP Trunk Security Profile needs to have TCP+UDP for Incoming Transport Type and TCP for
Outgoing Transport Type. This profile also needs Accept Presence Subscription, Accept Out-ofDialog REFER, Accept Unsolicited Notifications, Accept Replaces Header selected. (page 7)
•
Create a Route Pattern on the CUCM to route any ININ Extensions (ex. 7XXX, 8XXX) over the
SIP Trunk to the IC Server. (page 12)
•
The Cisco phones are created as SIP workstations in IA.
•
In the SIP Profile, select “Conference Join Enabled, RFC 2543 Hold, and Semi Attended
Transfer.” (page 8)
•
Setting the Cisco Phone to Auto Answer to Headset is recommended in an agent environment.
The SIP Trunk does not support the Talk Event.
Additional Information
•
•
•
•
•
•
•
Faxing will need to be directed to the HMP IC server directly from the gateway if possible.
Off hook and dialing call states from the Cisco phone will not be indicated on the IC client. This
is standard for any SIP implementation, but differs from a TAPI integration.
Pressing Hold from the Cisco Phone will provide Cisco Hold Music. Selecting Hold from the IC
client will provide IC hold Music.
If the CUCM station is unplugged or loses network connectivity, IC will continue to send calls to
the user and then roll to VM. If the user is an agent, the first ACD routed call will alert, then
change the user’s status to ACD-Agent Not Available.
Auto Answer must be set on the CUCM to auto off-hook a connection call.
CUCM does not do a Putback transfer, but can accept it
Transfer from the phone keeps the call in the IC client until the call has ended
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Extension Details
It is important to understand that there are two significant configuration options a customer has to
choose from.
Option 1 (Preferred Option):
This option sends all dialing from CUCM phones across the SIP trunk to IC. This allows every call
to appear in the IC client. This provides dialing consistency for the end user, but requires a
customer to be reliant on the IC server being active for outbound calls. Dialing from the phone
and the IC client are all consistent, IC station extensions will match the CUCM directory numbers,
and IC can provide call analysis, reporting, as well as on the phone presence.
Option 2:
This option is for customers that have existing CUCM configurations and do not want to modify
CUCM call paths. This would be for integrations that consist mainly of inbound calling and using
the Interaction Center as the ACD application. This option will keep CUCM station to station calls
from appearing in the IC client and kept local to CUCM.
The following extensions would be used for this option:
ININ User Extension
9000
ININ Station Extension
8000
Cisco Phone Extension
7000
•
ININ users Extensions [9000] will be used for internal dialing and external callers via the IVR to
connect to the user. The user logs onto a configured ININ station to receive the alerting call.
The ININ station is configured with the Cisco phone as the connection address.
•
ININ station Extension [8000] has an Identification address configured as the Cisco phone
extension [7000]. This is to allow outgoing calls from the Cisco phone to be identified as a
configured ININ station. Calls needing to connect to external lines will need to be identified as
known ININ stations before they will be routed.
•
Cisco Phone Extension [7000] is configured on the Cisco Unified Communication
Manager(CUCM). It is the directory number given during the CUCM configuration.
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Configuring a SIP trunk on the CUCM
The configuration of a SIP trunk on CUCM consists of three major components:
SIP Trunk Security Profile
Configures the Protocol of the SIP Trunk
SIP Profile
Configures RFC 2543 Hold
SIP Trunk
Configures MTP and Proxy Destination address
These three components are needed for a successful SIP Trunk configuration. Cisco has split these
configuration settings into three components to allow an administrator to reuse the SIP Trunk Security
Profile and the SIP Profile for every SIP Trunk that is configured. If multiple SIP Trunks were needed,
only the SIP Trunk portion of the guide below would need to be re-executed. A step by step guide is
provided below.
SIP Trunk Security Profile (Option 1 and 2)
1.) Under the Systems tab, highlight Security Profile and then select the SIP Trunk Security Profile
option.
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2. Select the default profile or create a new one and check to see that Accept Presence
Subscription, Accept Out-of-Dialog Refer, Accept Unsolicited Notifications, and Accept Replaces
Header are all selected. Also, setup TCP on the SIP Trunk in CUCM for faster failover to
additional CUCM servers.
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SIP Profile (Option 1 and 2)
3. Under the DEVICE tab, inside DEVICE SETTINGS, select the SIP Profile option. Select
“Conference Join Enabled, RFC 2543 Hold, and Semi Attended Transfer.”
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SIP Trunk (Option 1 and 2)
4. Under the DEVICE tab, select the TRUNK option.
5. Click on "Add a New Trunk."
6. Set the Trunk type to "SIP Trunk." Be sure the Device Protocol automatically sets itself to SIP
and then click Next.
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7. Under the Trunk Configuration dialog, create a Device Name for the trunk and set the Device
Pool to Default.
8. Also in the SIP Trunk configuration, MTP needs to remain unselected. This will allow the media
to travel from endpoint-to-endpoint (station) instead of forcing the CUCM to be in the audio path.
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9. In the Destination Address field, type the IP address of the IC server or SIP Proxy
10. Set the SIP Trunk Security Profile to the one created in steps 1 and 2 above.
Select Redirecting Diversion Header Delivery for both Inbound and Outbound. This is selected
to support voicemail support on calls forwarded from the Cisco endpoint.
Set the appropriate Rerouting Calling Search Space to the partitions containing your CUCM
phones and/or gateways.
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Route Pattern (Option 1)
This option routes all dialing through the IC server with minimal route pattern configuration.
1.
2.
3.
4.
5.
6.
Create a partition for the Trunk (LinePartition)
Create a partition for phones (PhonePartition)
Create a Calling Search Space for the phones (PhoneSS) - only include the LinePartition
Create a Calling Search Space for Inbound calls from IC (InboundSS) – include all partitions
Set the phones’ Calling search space to LinePartition
Create a wildcard route pattern to send all dialed traffic to the SIP trunk and set its route
parting to LinePartition
7. On the SIP Trunk itself, set the Inbound Calls \ Calling Search Space to PhoneSS
Route Pattern (Option 2)
11. Setup a Route Pattern in CUCM to allow dialed IC User extension numbers to pass through the
CUCM to the IC server across the SIP Trunk . (Ex. 9XXX will route over the SIP Trunk and then
back to the CUCM station that the agent is logged into.)
Route Pattern 8888 is setup to allow for the Voicemail Pilot number as well as the Message
button on the CUCM station to route across the SIP Trunk. Checking voicemail via the message
button is supported.
Route Pattern 8XXX is to allow routing of standalone CUCM stations. This will be rare in most
integrations.
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Route Pattern 9XXX is configured to allow IC user extensions dialed from CUCM phones to
route across the SIP Trunk to IC. This is a very important configuration step not to be
overlooked.
Route Pattern 90XXXXXXX is configured if dialing external numbers from CUCM phones to
PSTN gateways controlled by CUCM is to be represented in the IC client. The CUCM user will
dial 91 + ten digit number. IC will need a matching entry in it’s dial plan. In the IC Dial-plan
entry, the number will be modified to 91XXXXXXX and routed back to CUCM. CUCM will match
the 91XXXXXXX dial plan entry and route it to the assigned gateway.
Configure more Route Patterns in CUCM to allow additional IC extension numbers to pass
through the CUCM to the IC Server using the SIP Trunk.
Voicemail Pilot (Option 1 and 2)
12. Configure a Voice Mail Pilot number in the CUCM.
Add the Voice Mail Pilot number to the CUCM Routing Pattern to forward to the IC Server over
the SIP Trunk. In IC, setup the Message button under Server>Telephony Parameters>SIP.
To Allow MWI usage, ensure “ Accept Unsolicited Notifications” is selected in the CUCM
SIP Profile as indicated in step 2.
If using MWI in a CUCM cluster environment, ensure the cluster is set up as indicated in
the “Cisco CUCM Cluster Support” section of this document.
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Fault Tolerance Configuration (Option 1 and 2)
13. Configuring fault tolerance to allow for redundant IC servers has two parts
a. Creating a second SIP Trunk as was done in steps 4-10 above.
b. Creating a Route Group and Route List that contains both these SIP Trunks
Keep in mind that this is not limited to (2) SIP Trunks, this can be scaled increased to any
number of SIP Trunks. The testing was completed with 7 SIP Trunks.
Creating additional SIP Trunks
As indicated above, steps 4-10 show the creation of a single SIP Trunk. Repeating this
process, selecting the same configuration settings (including protocol and port number) is
followed. The difference is on Step 9; entering the secondary IC server’s IP address instead of
the primary server allows for two SIP Trunks, each pointing to a single IC server.
Creating a Route Group and Route List
The last two remaining configurations are to create a Route Group and Route List.
A Route Group is where the x number of SIP trunks that have been created will be grouped
together.
A Route List is where the Route Groups can be grouped together. In most cases it will be the
one Route Group that was created for the x number of IC server SIP Trunks.
The final step is selecting the Route List for the Route Pattern directed to the IC servers.
Configuring a Route Group:
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Name, select Top Down as the algorithm, select the two or (x) number of SIP Trunks to
include in the group and save.
Note that any SIP Trunk that is already associated with a Route Patter will not show up in the list.
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Configuring a Route List:
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Name, select the Route Group, and save.
The last step is to select the configured Route list containing the Route Group of SIP Trunks to
the IC servers as the destination of any Route Patterns needing to be directed to IC servers.
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Cisco CUCM Cluster Support (Option 1 and 2)
Supporting a cluster of CUCM servers consisting of multiple subscribers and a publisher is
supported with the following requirements:
Interaction Message Indicator:
•
The gateway/route configured in IMI must point to a CUCM server that is in each of the
endpoint Device Pool. For example, if the Cisco phone is using Subscriber 1 as its
primary and Subscriber 2 as its backup, a 503 error will be returned if the gateway/route
in IMI is configured to send the MWI event to the publisher. The Publisher will need to
be added as a tertiary server in the group or the IMI gateway/route will need to be set to
either of the subscriber servers. Also, the same CUCM group assigned to the phone (via
Device Pool) will need to be set on the SIP Trunk as well.
CIC and MIC:
•
The SIP line configured in IA must point to CUCM server(s) that are in each of the
endpoint Device Pools. For example, if the Cisco Phone is using Subscriber 1 as its
primary and Subscriber 2 as its backup, a 503 error will be returned if the Proxy on the
SIP line is pointed to the Publisher. The server indicated on the IC SIP line will need to be
added to the CUCM group in the Device Pool, in this example as a tertiary, that the
endpoints are associated with. Also, the SIP Trunk will need to be part of this same
Device Pool.
Setup a CUCM group for the endpoints to receive messaging:
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Setup a Device Pool that includes the server group defined above:
Set the Device Pool to the DP created above for both the SIP Trunk and the endpoints receiving
the MWI and/or SIP messaging: (Screen capture below only shows the SIP trunk)
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Additional CUCM Feature Configuration (Option 1 and 2)
Putback Transfer (All call control and RTP out of IC)
•
•
Verify Step 2 in the SIP Trunk Security Profile configuration is set
o Accept Replaces Header is selected
Verify Step 10 in the SIP Trunk configuration is set
o Set the appropriate Rerouting Calling Search Space to the partitions containing
your CUCM phones and/or gateways receiving the transfer
Multiple CUCM Regions
Some environments have multiple regions configured in the CUCM architecture. Using different codecs
between regions is supported with IC 3.0 SU11. The SIP trunk may be in a region that uses G.711 while
a remote phone may be across a WAN link in a separate region using G729. With SU-11, the invite
message sent to CUCM will include all available codecs configured in the Regionalization container for
the Location.
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IC Server Configuration
Media Server
1. The Interactive Media Server is required in the CUCM integration. This is configured using the
same process as a traditional SIP solution. For a distributed architecture, consisting of multiple
locations and multiple media servers, create locations in the Regionalization container and
place the media servers in the appropriate location. This will allow Audio plays to flow from a
local SIP gateway to the CUCM phone via the local media server. IC 3.0 SU6 includes the
support for play prompts and is also supported in the CUCM integration.
The Media Server is required for:
• RFC 2833 DTMF Negotiation issues
• Conferencing from the CUCM phone
• Transferring from the CUCM phone to an IC only controlled phone.
• Keeping the audio local at remote sites if using a SIP enabled gateway
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Line Configuration
2. The Audio settings on the Stations and SIP Line need to be configured with Always IN and
RFC2833 enabled.
3. The SIP Line and SIP Station Session field should be configured with Normal Media.
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4. Audio should also be set to TCP to match the CUCM. This allows for more efficient failover by
reducing the time it takes to contact the backup CUCM.
5. Setting the Proxy to the Primary and backup CUCM will allow for redundant CUCM support. If
there are more CUCMs in the cluster continue to add them in priority order. This order should
be consistent with the CUCM order configured on the stations.
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MWI
6. Enable the MWI light on each IC User and Station.
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Station Configuration
7. The Station Extension is set to the ININ Station Extension [8000] to allow calls that are made
to an ININ station that does not have a User logged into it. This setting allows a potential
intercom/lobby phone to be used to place and receive calls and it is not required to be
associated with an agent.
The Identification Address [7000] is configured as the Cisco Phone Extension. The “Use User
Portion only” is selected because in a CUCM cluster architecture the Host portion will change
depending on the active CUCM server.
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The station’s Connection SIP Address uses the Cisco Phone Extension [7000] in the “Use an Alternate
format” textbox to statically assign the CUCM’s phone extension. This is done manually because
CUCM does not dynamically register its phones with IC.
8. To add the host portion to the station connection address the “Use Proxy for Station
Connections” must be selected in the Default Station container in IA. This is located under the
Global SIP Station tab in the Transport container.
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User Configuration
The User Extension will be set to a number that will be dialed to contact the user. This extension
number cannot be the same as any other configured extensions. When a user extension is dialed,
the call will route to the user, indicate in the IC client, and then will alert the station that the user is
logged into.
A user dialing from a CUCM station will always dial the user extension to contact an IC user. The
route pattern configured in CUCM will allow for the dialing of 9xxx user extensions to be routed
across the SIP trunk and alert the station the intended user is logged into; in this case it will be a
CUCM station. The intent of this call flow is to get IC in the call signaling path and Media Server in
the Audio Path.
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Audio Paths
Inbound/Outbound calls to the PSTN
If using an MGCP gateway:
RTP:
Gateway <-> CUCM <-> Media Server <-> Phone
Signaling: Gateway <-> CUCM <-> IC Server <-> CUCM <-> Phone
If using an h.323 gateway:
RTP:
Gateway <-> CUCM <-> Media Server <-> Phone
Signaling: Gateway <-> CUCM <-> IC Server <-> CUCM <-> Phone
If using an SIP gateway:
RTP:
Gateway <-> Media Server <-> Phone
Signaling: Gateway <-> CUCM <-> IC Server <-> CUCM <-> Phone
Network Architecture for CUCM SIP Integration
Architecture 1- H.323/MGCP gateway connected to CUCM and SIP gateway connected to IC server
1. Incoming call to (555-9XXX) from PSTN to MGCP gateway or PSTN to SIP gateway.
2. Call from MGCP gateway is caught by CUCM Routing Pattern (5559XXX) and is routed
to IC Server
3. IC Server processes the call from the SIP gateway or from the CUCM and routes it back
to the CUCM
4. CUCM routes call to Cisco Phone 7XXX (User 9XXX)
5. Station connects call
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Architecture 2- H.323/MGCP gateway connected to CUCM
6.
7.
8.
9.
10.
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Incoming call to (555-9XXX) from PSTN to MGCP gateway
Call is caught by CUCM Routing Pattern (5559XXX) and is routed to IC Server
IC Server processes the call and routes it back to the CUCM
CUCM routes call to station 7XXX (User 9XXX)
Station connects call
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Architecture 3- SIP gateway connected to IC server
1.
2.
3.
4.
5.
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Incoming call to (555-9XXX) from PSTN to SIP gateway
Call is routed from the SIP gateway to the IC Server
IC Server processes call and routes it to the CUCM
CUCM routes call to station 7XXX (User 9XXX)
Station connects call
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Architecture 4- CUCM station to station call
1.
2.
3.
4.
5.
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Station 1 places call to Station 2 connected to the CUCM
Call is caught by CUCM Routing Pattern (9XXX) and is routed to IC Server
IC Server processes call and routes it back to the CUCM
CUCM routes call to station 7XXX (User 9XXX)
Station connects call
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CUCM Stations: Transfer Scenarios – Call Flow
Call from CUCM station to station - Transfer From Phone
Scenario – CUCM Station 1 calls CUCM Station2 then transfers to CUCM Station3
1.
2.
3.
4.
5.
6.
7.
8.
9.
CUCM Station1 places call to CUCM Station2.
Call is caught by CUCM Routing Pattern (9XXX) and is routed to IC Server.
IC Server processes call and routes it to the CUCM.
CUCM routes call to CUCM Station2 (7XXX)(User 9XXX.)
CUCM Station1 presses “Transfer” on the phone and then places a call to CUCM
Station3.
Call is caught by CUCM Routing Pattern (9XXX) and is routed to IC Server.
IC Server processes call and routes it to the CUCM.
CUCM routes call to CUCM Station3 (7XXX)(User 9XXX.)
CUCM Station1 presses “Transfer” on the phone again and CUCM Station2 is
connected to CUCM Station3.
Call from PSTN to CUCM Station -Transfer from Phone
Scenario – PSTN calls CUCM Station1 then CUCM Station1 transfers call to CUCM Station2
(Architecture 2 above)
1.
2.
3.
4.
5.
6.
7.
8.
9.
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A call comes from the PSTN to CUCM Station1.
Call is caught by CUCM Routing Pattern (9XXX) and is routed to IC Server.
IC Server processes call and routes it to the CUCM.
CUCM routes call to CUCM Station1 (7XXX)(User 9XXX.)
CUCM Station1 presses “Transfer” on the phone and then places a call to CUCM
Station2.
Call is caught by CUCM Routing Pattern (9XXX) and is routed to IC Server.
IC Server processes call and routes it to the CUCM.
CUCM routes call to CUCM Station3 (7XXX)(User 9XXX.)
CUCM Station1 presses “Transfer” on the phone again and PSTN call is connected to
CUCM Station2.
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© 2011 Interactive Intelligence, Inc.
Cisco Unified Communication Manager SIP Integration
App Note
Call from CUCM station to station - Transfer from IC Client
Scenario – CUCM Station 1 calls CUCM Station2 then transfers to CUCM Station3
1.
2.
3.
4.
CUCM Station1 places call from the IC Client to CUCM Station2.
IC Server processes call and routes it to the CUCM.
CUCM routes call to CUCM Station2 (7XXX)(User 9XXX.)
CUCM Station1 presses “Transfer” on the IC Client and types CUCM Station3’s
number into the Transfer dialog.
5. IC Server processes call and routes it to the CUCM.
6. CUCM routes call to CUCM Station3 (7XXX)(User 9XXX.)
7. CUCM Station1 presses “Transfer Now” on the IC Client’s Transfer dialog and CUCM
Station2 is connected to CUCM Station3.
Call from PSTN to IC Client - Transfer from IC Client
Scenario – PSTN calls CUCM Station1’s IC Client then CUCM Station1 transfers call to CUCM Station2
(Architecture 3 above)
1. A call comes from the PSTN to CUCM Station1’s IC Client.
2. IC Server processes call and routes it to the CUCM.
3. CUCM routes call to CUCM Station1 (7XXX)(User 9XXX.)
4. CUCM Station1 presses “Transfer” on the IC Client and types CUCM Station2’s
number into the Transfer dialog.
5. IC Server processes call and routes it to the CUCM.
6. CUCM routes call to CUCM Station2 (7XXX)(User 9XXX.)
7. CUCM Station1 presses “Transfer Now” on the IC Client’s Transfer dialog and PSTN
Call is connected to CUCM Station2.
April 27, 2011
Page 35 of 37
© 2011 Interactive Intelligence, Inc.
Cisco Unified Communication Manager SIP Integration
App Note
CUCM Stations: Conference Scenarios – Call Flow
Conference between CUCM Stations – Conference from Phone
Scenario – CUCM Station1 calls CUCM Station2 then creates conference from phone with CUCM Station3
1.
2.
3.
4.
5.
6.
7.
8.
9.
CUCM Station1 places call to CUCM Station2.
Call is caught by CUCM Routing Pattern (9XXX) and is routed to IC Server.
IC Server processes call and routes it to the CUCM.
CUCM routes call to CUCM Station2 (7XXX)(User 9XXX.)
CUCM Station1 presses “Conference” on the phone and then places a call to CUCM
Station2.
Call is caught by CUCM Routing Pattern (7XXX) and is routed to IC Server.
IC Server processes call and routes it to the CUCM.
CUCM routes call to CUCM Station3 (7XXX)(User 9XXX.)
CUCM Station1 highlights existing call and presses “Conference” on the phone again
and is connected to CUCM Station2 and CUCM Station3.
Conference between PSTN and CUCM Stations - Conference from Phone
Scenario – PSTN calls CUCM Station1 then CUCM Station1 conferences with CUCM Station2
(Architecture 2 above)
1.
2.
3.
4.
5.
6.
7.
8.
9.
April 27, 2011
A call comes from the PSTN to CUCM Station1.
Call is caught by CUCM Routing Pattern (9XXX) and is routed to IC Server.
IC Server processes call and routes it to the CUCM.
CUCM routes call to CUCM Station1 (7XXX)(User 9XXX.)
CUCM Station1 presses “Conference” on the phone and then places a call to CUCM
Station2.
Call is caught by CUCM Routing Pattern (9XXX) and is routed to IC Server.
IC Server processes call and routed it to the CUCM.
CUCM routes call to CUCM Station3 (7XXX)(User 9XXX.)
CUCM Station1 highlights the existing call and presses “Conference” on the phone
again and the PSTN call is connected to CUCM Station1 and CUCM Station2.
Page 36 of 37
© 2011 Interactive Intelligence, Inc.
Cisco Unified Communication Manager SIP Integration
App Note
Conference between CUCM Stations - Conference from IC Client
Scenario – CUCM Station 1 calls CUCM Station2 then creates Conference with CUCM Station3
1.
2.
3.
4.
CUCM Station1 places call from the IC Client to CUCM Station2.
IC Server processes call and routes it to the CUCM.
CUCM routes call to CUCM Station2 (7XXX)(User 9XXX.)
On CUCM Station1’s IC Client place the current call on Hold by pressing the “Hold”
button on the IC Client and then place another call to CUCM Station3 from the IC
Client.
5. IC Server processes call and routes it to the CUCM.
6. CUCM routes call to CUCM Station3 (7XXX)(User 9XXX.)
7. CUCM Station1 creates the conference by dragging and dropping the current
connected call onto the existing call that is on Hold. CUCM Station1 and CUCM
Station2 are both connected to CUCM Station3.
Call from PSTN to IC Client - Conference from IC Client
Scenario – PSTN calls CUCM Station1’s IC Client then CUCM Station1 creates a conf. with CUCM Station2
1. A call comes from the PSTN to CUCM Station1’s IC Client.
2. IC Server processes call and routes it to the
3. CUCM routes call to CUCM Station1 (7XXX)(User 9XXX.)
4. On CUCM Station1’s IC Client place the current call on Hold by pressing the “Hold”
button on the IC Client and then place another call to CUCM Station2 from the IC
Client.
5. IC Server processes call and routes it to the CUCM.
6. CUCM routes call to CUCM Station2 (7XXX)(User 9XXX.)
7. CUCM Station1 creates the conference by dragging and dropping the current
connected call onto the existing call that is on Hold. CUCM Station1 and CUCM
Station2 are both connected to the PSTN call.
April 27, 2011
Page 37 of 37
© 2011 Interactive Intelligence, Inc.
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