Information included in this document is dedicated to customer equipment (IPBX, TOIP ecosystems)

connection to Business Talk IP service : it shall not be used for other goals or in another context.

Document Version
Version of 06/10/2017

TECHNICAL GUIDE to access

Business Talk IP SIP
CUCM

versions addressed in this guide: 11.5

 Technical guide to access Business Talk IP service
CUCM IPBX

Table of contents

Goal of this document...................................................................................................................... 3

2 Architecture overview ............................................................................................................... 4
2.1 CUCM without CUBE (standard) ................................................................................................... 4
2.2 CUCM with CUBE in “flow through” mode (standard but on demand) ........................................... 4
2.3 CUCM with CUBE in “flow around” mode (NOT standard - on demand only)................................. 5
3 Parameters to be provided by customer to access service .......................................................... 6
3.1 CUCM without CUBE ................................................................................................................... 6
3.2 CUCM with CUBE (flow through) .................................................................................................. 6
4 Certified software and hardware versions ................................................................................... 7
4.1 CUCM certified versions ............................................................................................................... 7
4.2 CUCM certified applications and devices versions......................................................................... 7
4.3 CUBE certified versions ................................................................................................................ 8
5 Cisco Call Manager configuration .............................................................................................. 9
6 Cisco Unity Connection configuration....................................................................................... 23
7 Unified Contact Center Express configuration ........................................................................... 24
8 Cisco Unified Attendant Console configuration .......................................................................... 25
9 CUCM with Cisco Unified Border Element configuration ............................................................. 27
9.1 General CUBE configuration (flow-through mode by default) ........................................................27
9.2 Configuration for a CUCM cluster and two CUBEs .......................................................................28
9.3 Configuration for a single CUCM server and one CUBE ...............................................................31
9.4 Configuration for a CUCM cluster and one CUBE ........................................................................33

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Goal of this document

The aim of this document is to list technical requirements to ensure the interoperability between Cisco
CUCM IPBX with Business Talk IP SIP, hereafter so-called “service”.

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2 Architecture overview
2.1 CUCM without CUBE

CUCM
SBC
Nominal

Business Talk

CE
SBC
Backup

signalization
media
All SIP trunks attached directly to CUCM

2.2 CUCM with CUBE in “flow through” mode

CUCM
SBC
Nominal

Business Talk

CE
SBC
Backup

signalization
media

All SIP trunks attached to the CUBE. Signaling and media flows cross the CUBE.

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2.3 CUCM with CUBE in “flow around” mode

SBC
Backup

signalization

All SIP trunks attached to the CUBE. Signaling flows cross the CUBE, but media flows go
directly towards endpoints.

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3 Parameters to be provided by customer to access service

IP addresses marked in red have to be indicated by the customer, depending on customer
architecture scenario.

3.1 CUCM without CUBE

Customer IP addresses
used by service
Head Quarter (HQ) or Branch Office
(BO) architecture Level of Service Nominal Backup
CUCM Business Edition (1 server ) No reduncdancy (1 Publisher) CUCMBE IP@ N/A
Local redundancy Subscriber (Nominal) /
Publisher (Backup)
Publisher and Subscriber are on different
CUCM (1 Publisher + 1 Subscriber) servers) Subscriber IP@ Publisher IP@
- Local redundancy Subscriber1 (Nominal)
/ Subscriber2 (Backup)
- If more than 1 Subscriber, the SIP trunks
CUCM (1 Publisher + 2 Subscribers) are held by the Subscribers. The Publisher Subscriber1 Subscriber2
Subscribers Nominal/Backup holds the database. IP@ IP@
- Local redundancy and Load Sharing
Subscriber1 / Subscriber2 Subscriber1
- The Subscribers share the load in a IP@
CUCM (1 Publisher + 2 Subscribers) round robin fashion Subscriber2
Subscribers Load Sharing (Also applicable with N Subscribers) IP@ N/A
- Site redundancy: Subscriber and
CUCM with clustering over WAN (1 Publisher servers hosted by 2 different
Publisher + 1 Subscriber) physical sites Subscriber IP@ Publisher IP@
- Site redundancy: the 2 Subscribers are
hosted by 2 different physical sites
(Subscriber1(Nominal) /
Subscriber2(Backup))
CUCM with clustering over WAN (1 - If more than 1 Subscriber, the SIP trunks
Publisher + 2 Subscribers) are held by the Subscribers. The Publisher Subscriber1 Subscriber2
Subscribers Nominal/Backup holds the database. IP@ IP@
- Site redundancy: the 2 Subscribers are
hosted by 2 different physical sites Subscriber1
CUCM with clustering over WAN (1 (Subscriber1 + Subscriber2) IP@
Publisher + 2 Subscribers) - The Subscribers share the load in a Subscriber2
Subscribers Load Sharing round robin fashion IP@ N/A
Nominal Backup
Remote site without survivability No survivability, no trunk redundancy N/A N/A
Local site survivability and trunk
SRST redundancy via PSTN only N/A N/A

3.2 CUCM with CUBE (flow through)

Customer IP addresses
Head Quarter (HQ) or Branch Office used by service
(BO) architecture Level of Service Nominal Backup
CUCM + Single CUBE No redundancy CUBE IP@ N/A

CUCM + 2 CUBES - Local redundancy:

warning: if both CUBES are hosted by the same site
- Site access capacity to be sized (CUBE1+CUBE2)
adequately on the site carrying the 2nd - Geographical redundancy:
CUBE in case both CUBEs are based if each CUBE is hosted by different sites
on different sites (CUBE1+CUBE2) CUBE1 IP@ CUBE2 IP@
Nominal Backup
Remote site without survivability No survivability, no trunk redundancy N/A N/A
Local site survivability and trunk
SRST redundancy via PSTN only N/A N/A

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4 Certified software and hardware versions

4.1 CUCM certified versions

Cisco IPBX
Equipment validation IPBX
Equipment status
Version Version
R10.5 / R10.6  Load 10.5.2.12901-1 min
CUCM
R11  Load 11.0.1.10000-10 min
CBE5000/6000 R11.5  Load 11.5.1.11900-26 min

4.2 CUCM certified applications and devices versions

Cisco ecosystems
Equipment validation IPBX
Equipment Comment
Version status Version
10.5.2.10-15  R10.5
Attendant
Console
CUxAC 11.0.1.3058  R11.0 Standard and Advanced editions

11.5.x  R11.5

10.5.2.10000-15  R10.5
Unity Connection 11.0.1.20000-2  R11.0
Voice Mail 11.5.x  R11.5

Unity Express
11.0.x  R11.0
11.5.x  R11.5

CUPS
10.5.2.10000-9  R10.5
As a component of CUPC only
not supported R11.x
Unified
Communica- not supported  R10.5
tions
Meeting place 11.0.x  R11.0
11.5.x  R11.5

10.6.1.10000-39  R10.5
Contact
center
UCCX 11.0.1.10000-75  R11.0
11.5.x  R11.5

Cisco IOS Cascaded

MediaGateway (ISR
15.1(4)M7  R10.5

28xx/38xx) not supported  R11.x

Cisco IOS Cascaded

15.4(3)M  R10.5
MediaGateway (ISR 15.5(3)M2  R11.0
29xx/39xx)
15.5(3)M4  R11.5

Cisco IOS Cascaded

15.5(2)S  R10.5
MediaGateway (ISR 3.16.01a.S  R11.0
43xx/44xx)
MGW
3.16.1a.S  R11.5
Analog GW Cisco R10.x
ATA187
9.2.3.1  R11.x

Audiocodes MP112 6.20A.058.x  R10.5

FXS
on demand R11.x
Analog GW Cisco VG 15.1(4)M7  R10.5
224 15.1(4)M10  R11.x

Analog GW Cisco VG 15.1(4)M7  R10.5

202-204 15.1(4)M10  R11.x

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Analog GW Cisco VG 15.4(3)M  R10.5

202-204 XM 15.5(3)M2  R11.x

Analog GW Cisco VG 15.4(3)M  R10.5

310-320-350 15.5(3)M2  R11.x

Analog GW Cisco 1.1.2(002)  R10.5

ATA190 1.2.1(004)  R11.x

R10.x
Cisco VoIP GW  R11.x
VOIP
OneAccess VoIP GW R10.x
(Business Livebox)  R11.x

Cisco Unified
10.5.2.12901-1  R10.5
Communication
Manager Assistant
11.0.1.10000-10  R11.0
(IPMA)
11.5.1.11900-26  R11.5
All Cisco SCCP R10.x
phones (skinny)  R11.x
R10.x
All Cisco SIP phones  R11.x
Phones
IPCommunicator R10.x
SCCP  R11.x

Jabber
10.5.2  R10.5
11.5.0  R11.x

CUCILync
9.7.4  R10.5

 R11.x

IP DECT ASCOM  R10.x

4.3 CUBE certified versions

Cisco CUBE
Equipment validation IPBX
Equipment Comment
Version status Version
R10.5 CUBE authorized on demand with
Cisco Unified Border Element on demand versions CUCM 10.5 and 11.0
R11.0
(CUBE) - “flow thru” mode
15.5(3)S4a  R11.5 IOS 15.6 authorized on demand
Cisco Unified Border Element R10.5
on demand
(CUBE) - “flow around” mode R11.x

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5 Cisco Call Manager configuration

The checklists below present all the configuration steps required for interoperability between the service
and CUCM.

Cisco Call Manager Service

Codec and payload configuration
Menu Value
System > Service Parameters > Appropriate server > Cisco CallManager (Active) > Advanced >
Clusterwide Parameters (System – Location and Region)
Preferred G.711 Millisecond Packet Size 20
Preferred G.729 Millisecond Packet Size 20
G.722 Codec Enabled Enabled for All Devices
Cisco CallManager Service
Codec and payload configuration
System > Service Parameters > Appropriate server > Cisco CallManager (Active) > Advanced
Clusterwide Parameters (Service)
Duplex Streaming Enabled True
Media Exchange Timer 5
Silence suppression False
Silence suppression for Gateways False
Media Exchange Timer True
Cisco CallManager Service
SIP Parameters
System > Service Parameters > Appropriate server > Cisco CallManager (Active) > Advanced
Clusterwide Parameters (Device - SIP)
Retry Count for SIP Invite 1
SIP Session Expires Timer 86400
Cisco CallManager Service
System – QOS Parameters
System > Service Parameters > Appropriate server > Cisco CallManager (Active) > Advanced
Clusterwide Parameters (System - QOS)
DSCP for Video Calls 34 (100010)
Cisco CallManager Service
Enterprise Parameters
System > Enterprise Parameters
Advertise G.722 Codec Enabled
Cisco CallManager Service
Cisco IP Voice Media Streaming Application service
System > Service Parameters > Appropriate server > Cisco IP Voice Media Streaming App (Active)
MTP Run Flag False
Supported MOH Codec G711alaw, G729 Annex A

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Cisco CallManager Service

Region configuration
Menu Value
System > Region Information > Region
Regions configuration for customer using G.729
From HQ RS WAN
To
HQ G711 G729 G729
RS G729 G711 G729
WAN G729 G729 G729

Regions configuration for customer using G.711

From HQ RS WAN
To
HQ G711 G711 G711
RS G711 G711 G711
WAN G711 G711 G711

Cisco CallManager Service

Device Pool Configuration
System > Device Pool > Add new
New Device Pool Device Pool configuration:
 The number of Device Pools at least should be the
same as the number of site
 Every Device Pool should have appropriate Region and
Location value

Note: MOH server requires a separate Device Pool

configuration.
Cisco CallManager Service
Locations (Call Admission Control)
System > Location Info> Location > Add new
New Location Warning! RSVP locations are not supported!

Create the necessary locations and configure the

bandwidth for each.

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Media Resources
Transcoder configuration : Warning! Hardware MTP resources on IOS Gateway and software MTP
resource on CUCM are NOT SUPPORTED. Software MTPs on
IOS Gateway are SUPPORTED in BT/BTIP SIP Trunking.

Menu Value
Media Resources > Transcoder > Add new
Transcoder Type Cisco IOS Enhanced Media Termination Point
Device Name Use the name configured in sccp ccm group in the IOS
Device Pool Use the appropriate Device Pool
Trusted Rely Point Unchecked
Media Resources
Conference Bridge configuration
Media Resources > Conference Bridge > Add new
Conference Bridge Type Cisco IOS Enhanced Media Termination Point
Device Name Use the name configured in sccp ccm group in the IOS
Device Pool Use the appropriate Device Pool
Device Security Mode Non Secure Conference Bridge
Media Resources
Multicast Music on Hold
CUCM configuration - Region
System > Region Information > Region > Add new
New Region Please refer to chapter on Region configuration for
additional information.
With this configuration, all devices in “MoH Multicast”
region will use G.711 as codec for sending RTP packets
to devices to all other regions and also for the “WAN”
region where codec G.711 will be used.
Media Resources
Multicast Music on Hold
CUCM configuration – Device Pool
System > Device Pool > Add new
New Device Pool Choose a name and associate the Region “MoH
Multicast” to this new Device Pool.
Media Resources
Multicast Music on Hold
CUCM configuration - Audio Source Configuration
Media Resources > Music On Hold Audio Source > Add new
Play continuously (repeat) Checked
Allow Multicasting Checked

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Media Resources
Multicast Music on Hold
CUCM configuration - Multicast MoH server configuration
Menu Value
Media Resources > Music On Hold Server
Device Pool Checked
Enable Multi-cast Audio Sources on this MoH Server Checked
Base Multi-cast IP Address 239.1.1.1 (example)
Base Multi-cast IP Port 16384 (example)
Increment Multi-cast on IP Address
Max Hops (per Audio Source in Selected Audio 1
Sources configuration area)
Media Resources
Multicast Music on Hold
CUCM configuration - Multicast MoH server configuration
Media Resources > Media Resource Group
Appropriate Media Resource Group Check the Use Multicast for MoH Audio checkbox to
allow multicast with this resource group.
Media Resources
Multicast Music on Hold
Router configuration – Audio file
Frequency 9kHz
Coded with 8bit
Audio mode Mono
Codec type CCITT u-law
Media Resources
Multicast Music on Hold
Router configuration – IOS Commands
Commands ccm-manager music-on-hold
call-manager-fallback
max-conferences 4
ip source-address 10.108.105.254 port 2000
max-ephones 24
max-dn 48
moh TheJourneyAndTheWind.alaw.wav
multicast moh 239.1.1.1 port 16384 route 210.72.240.13 10.108.105.254
Media Resources
Multicast Music on Hold
Media Resource Group Lists configuration
Media resources Warning! Media Resources, which are not associated with any MRG are
available to every device in the cluster by default.

Media Resources > Media Resource Group > Add new

Resources > Media Resource Group List > Add new
Off-net calling via BT/BTIP
Diversion Header manipulation
Partition
Menu Value
Call Routing -> Class of Control -> Partition -> Add new

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Name DIV-HEADER-PT
Off-net calling via BT/BTIP
Diversion Header manipulation
Called Party Transformation Pattern
Call Routing -> Transformation -> Transformation Pattern -> Called PartyTransformation Pattern ->
Add New
Pattern XXXX
Prefix digits Site Prefix
Off-net calling via BT/BTIP
Diversion Header manipulation
Calling Search Space
Call Routing -> Class of Control -> Calling Search Space -> Add New
Name DIV-HEADER-CSS
Selected Partitions DIV-HEADER-PT
Off-net calling via BT/BTIP
Basic Configuration
Sip Trunk Security Profile
System > Security > SIP Trunk Security Profile, select “Non Secure SIP Trunk Profile” from SIP Trunk
Security Profile List
Incoming Transport Type TCP + UDP
Outgoing Transport Type UDP
Off-net calling via BT/BTIP
Basic Configuration
SIP Profile
Device > Device Settings > SIP Profile
User-Agent and Server header information Send Unified CM Version Information as User-Agent
Header
Version in User Agent and Server Header Full Build
SIP Rel1XX Options Send PRACK for 1xx Messages
Early Offer support for voice and video Mandatory (insert MTP if needed)
Send send-receive SDP in mid-call INVITE Checked
Ping Interval for In-service and Partially In-service 300
Trunks (seconds)
Ping Interval for Out-of-service Trunks (seconds) 5
Version in User Agent and Sever Header Full build
Session Refresh Method INVITE or UPDATE

Version in User Agent and Sever Header - inject info about full version of CUCM
Session Refresh Method - since CUCM 10.0 there is additional method – “UPDATE”. “INVITE” should be
used by default.

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Off-net calling via BT/BTIP

Basic Configuration
SIP Normalization Script
Device > Device Settings > SIP normalization script > Add new
SIP Normalization Script is applied to SIP trunk and is required to adapt
the SIP signaling to the form expected by BT/BTIP infrastructure.
The content of the script is given below:

-- Orange SIP Normalization Script v10

-- this is normalization script for uc 11.x
M = {}

-- This is called when an INVITE message is sent

function M.outbound_INVITE(msg)
local sdp = msg:getSdp()
if sdp
then
-- remove b=TIAS:
sdp = sdp:gsub("b=TIAS:%d*\r\n", "")
-- store the updated sdp in the message object
msg:setSdp(sdp)
end
end

--modifying of Server header in 183 messages

function M.outbound_183_INVITE(msg)
-- change 183 to 180 if sdp
local sdp = msg:getSdp()
if sdp
then
msg:setResponseCode(180, "Ringing")
end
end

--modifying of Server header in 488 messages

function M.outbound_488_INVITE(msg)
-- change 488 to 503 if sdp
msg:setResponseCode(503, "Service Unavailable")
end

--handling of 400 errors

function M.inbound_400_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=27")
else
msg:addHeader("Reason", "Q.850; cause=27")
end
end

--handling of 403 errors

function M.inbound_403_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=2")
end
end

--handling of 408 errors

function M.inbound_408_INVITE(msg)
local reason = msg:getHeader("Reason")

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if reason
then
msg:removeHeader("Reason")
end
end

-- handling of 480 errors

function M.inbound_480_INVITE(msg)
local reason = msg:getHeader("Reason")
if not reason
then
msg:addHeader("Reason", "Q.850; cause=20")
end
end

--handling of 481 errors

function M.inbound_481_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=27")
else
msg:addHeader("Reason", "Q.850; cause=27")
end
end

--handling of 487 errors

function M.inbound_487_INVITE(msg)
local reason = msg:getHeader("Reason")
if not reason
then
msg:addHeader("Reason", "Q.850; cause=16")
end
end

--handling of 488 errors

function M.inbound_488_INVITE(msg)
local reason = msg:getHeader("Reason")
if not reason
then
msg:addHeader("Reason", "Q.850; cause=127")
end
end

--handling of 500 errors

function M.inbound_500_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=2")
else
msg:addHeader("Reason", "Q.850; cause=2")
end
end

--handling of 501 errors

function M.inbound_501_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=2")
else
msg:addHeader("Reason", "Q.850; cause=2")
end
end

--handling of 502 errors

function M.inbound_502_INVITE(msg)

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local reason = msg:getHeader("Reason")

if reason
then
msg:removeHeader("Reason")
end
end

-- handling of 503 errors

function M.inbound_503_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=38")
else
msg:addHeader("Reason", "Q.850; cause=38")
end
end

-- handling of 505 errors

function M.inbound_505_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=38")
else
msg:addHeader("Reason", "Q.850; cause=38")
end
end

-- handling of 513 errors

function M.inbound_513_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=38")
else
msg:addHeader("Reason", "Q.850; cause=38")
end
end

-- addition of PAI header if incoming INVITE includes Privacy

header
function M.inbound_INVITE(msg)
-- get Privacy header
local privacy = msg:getHeader("Privacy")
if privacy
then
-- get From and Pai
from = msg:getHeader("From")
pai = msg:getHeader("P-Asserted-Identity")
--check if Pai header is not present
if pai==nil
then
-- add Pai header filled with From URI value
local uri = string.match(from, "(<.+>)")
msg:addHeader("P-Asserted-Identity", uri)
end
end
end

return M

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Off-net calling via BT/BTIP

Basic Configuration
SIP Trunk Configuration
Menu Value
Device > Trunk > Add new
Device Pool Choose Device Pool which include Region and Location
value
Media Resource Group List MRGL
Redirecting Diversion Header Delivery - Inbound Checked
Redirecting Diversion Header Delivery - outbound Checked
Destination Address SBC IP Address
SIP Trunk Security Profile SIP Trunk Security Profile name
SIP Profile Standard SIP Profile with PRACKs, EO, Send-recv
DTMF Signaling Method RFC 2833
Normalization Script SIP Normalization Script name (currently v8)
Enable Trace Unchecked
Redirecting Party Transformation CSS DIV-HEADER-CSS
Off-net calling via BT/BTIP
Basic Configuration
Route Group
Call Routing > Route/Hunt > Route group > Add new
Distribution algorithm Top Down
Selected devices both SIP trunks to ACMEs
Off-net calling via BT/BTIP
Basic Configuration
Route List
Call Routing > Route/Hunt > Route list > Add new
Selected Groups Route Group with SIP trunks to BT/BTIP
Off-net calling via BT/BTIP
Basic Configuration
Route Pattern
Call Routing > Route/Hunt > Route Pattern > Add new
Route Pattern Specific Route Pattern
Gateway/Route List Route List name
Call Classification OffNet
Discard Digits PreDot Trailing#
On-net calling
Basic Configuration
The configuration of such intercluster SIP Trunk is the same as the one described for off-net calls
except that on trunk between sites there is no SIP Normalization Script.

SME Architecture (ON CUSTOMER DEMAND)

Off-net calling via BT/BTIP
SIP Trunk Security Profile (at CUCM SME and CUCM)
Menu Value
System > Security > SIP Trunk Security Profile > Add new
Incoming Transport Type TCP + UDP
Outgoing Transport Type UDP

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SME Architecture
Off-net calling via BT/BTIP
SIP Trunk Security Profile (at CUCM SME and CUCM)
Device > Device Settings > SIP Profile
User-Agent and Server header information Send Unified CM Version Information as User-Agent
Header
Version in User Agent and Server Header Full Build
SIP Rel1XX Options Send PRACK for 1xx Messages
Early Offer support for voice and video calls (insert Checked
MTP if needed)
Send send-receive SDP in mid-call INVITE Checked
Ping Interval for In-service and Partially In-service 300
Trunks (seconds)
Ping Interval for Out-of-service Trunks (seconds) 5
SME Architecture
Off-net calling via BT/BTIP
SIP Normalization Script (at CUCM SME)
Device > Device Settings > SIP normalization script > Add new
SIP Normalization Script is applied to SIP trunk at CUCM SME and is required to adapt the SIP
signaling to the form expected by BT/BTIP infrastructure. Create the script.
The content of the script is given below:

-- Orange SIP Normalization Script v10

-- this is normalization script for uc 11.x
M = {}

-- This is called when an INVITE message is sent

function M.outbound_INVITE(msg)
local sdp = msg:getSdp()
if sdp
then
-- remove b=TIAS:
sdp = sdp:gsub("b=TIAS:%d*\r\n", "")
-- store the updated sdp in the message object
msg:setSdp(sdp)
end
end

--modifying of Server header in 183 messages

function M.outbound_183_INVITE(msg)
-- change 183 to 180 if sdp
local sdp = msg:getSdp()
if sdp
then
msg:setResponseCode(180, "Ringing")
end
end

--modifying of Server header in 488 messages

function M.outbound_488_INVITE(msg)
-- change 488 to 503 if sdp
msg:setResponseCode(503, "Service Unavailable")
end

--handling of 400 errors

function M.inbound_400_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason

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then
msg:modifyHeader("Reason", "Q.850; cause=27")
else
msg:addHeader("Reason", "Q.850; cause=27")
end
end

--handling of 403 errors

function M.inbound_403_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=2")
end
end

--handling of 408 errors

function M.inbound_408_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:removeHeader("Reason")
end
end

-- handling of 480 errors

function M.inbound_480_INVITE(msg)
local reason = msg:getHeader("Reason")
if not reason
then
msg:addHeader("Reason", "Q.850; cause=20")
end
end

--handling of 481 errors

function M.inbound_481_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=27")
else
msg:addHeader("Reason", "Q.850; cause=27")
end
end

--handling of 487 errors

function M.inbound_487_INVITE(msg)
local reason = msg:getHeader("Reason")
if not reason
then
msg:addHeader("Reason", "Q.850; cause=16")
end
end

--handling of 488 errors

function M.inbound_488_INVITE(msg)
local reason = msg:getHeader("Reason")
if not reason
then
msg:addHeader("Reason", "Q.850; cause=127")
end
end

--handling of 500 errors

function M.inbound_500_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then

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msg:modifyHeader("Reason", "Q.850; cause=2")

else
msg:addHeader("Reason", "Q.850; cause=2")
end
end

--handling of 501 errors

function M.inbound_501_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=2")
else
msg:addHeader("Reason", "Q.850; cause=2")
end
end

--handling of 502 errors

function M.inbound_502_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:removeHeader("Reason")
end
end

-- handling of 503 errors

function M.inbound_503_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=38")
else
msg:addHeader("Reason", "Q.850; cause=38")
end
end

-- handling of 505 errors

function M.inbound_505_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=38")
else
msg:addHeader("Reason", "Q.850; cause=38")
end
end

-- handling of 513 errors

function M.inbound_513_INVITE(msg)
local reason = msg:getHeader("Reason")
if reason
then
msg:modifyHeader("Reason", "Q.850; cause=38")
else
msg:addHeader("Reason", "Q.850; cause=38")
end
end

-- addition of PAI header if incoming INVITE includes Privacy

header
function M.inbound_INVITE(msg)
-- get Privacy header
local privacy = msg:getHeader("Privacy")
if privacy
then
-- get From and Pai
from = msg:getHeader("From")

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pai = msg:getHeader("P-Asserted-Identity")
--check if Pai header is not present
if pai==nil
then
-- add Pai header filled with From URI value
local uri = string.match(from, "(<.+>)")
msg:addHeader("P-Asserted-Identity", uri)
end
end
end

return M

SME Architecture
Off-net calling via BT/BTIP
SIP Trunk Configuration to offnet (at CUCM SME)
Menu Value
Device > Trunk > Add new
Device Pool Choose Device Pool which include Region and Location
value
Media Resource Group List None
Redirecting Diversion Header Delivery - Inbound Checked
Destination Address SBC IP Address
SIP Trunk Security Profile SIP Trunk Secure Profile name
SIP Profile Standard SIP Profile with PRACKs, EO and Send-recv
Normalization Script SIP Normalization Script name
Enable Trace Unchecked
SME Architecture
Off-net calling via BT/BTIP
Route group (at CUCM SME)
Call Routing > Route/Hunt > Route group > Add new
Distribution algorithm Top Down
Selected devices both SIP trunks to ACMEs
SME Architecture
Off-net calling via BT/BTIP
Route list (at CUCM SME)
Call Routing > Route/Hunt > Route list > Add new
Selected Groups Route Group with SIP trunks to BT/BTIP
SME Architecture
Off-net calling via BT/BTIP
Route pattern (at CUCM SME)
Call Routing > Route/Hunt > Route Pattern > Add new
Route Pattern Specific Route Pattern
Gateway/Route List Route List name
Call Classification OffNet
Discard Digits PreDot Trailing#

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SME Architecture
On-net calling
The configuration of such intercluster SIP Trunk is the same as the one described for off-net calls
except for:
 Media Resource Group List – should be set to the group containing following resources:
conference, transcoder, annuciator (Subscribers), MOH Server (Subscribers), software MTP
 SIP Normalization Script should not be added to this trunk
SIP Trunks should be between CUCM of independent site and CUCM SME (there is no direct
SIP Trunks between independent sites in SME Architecture – all on-net calls are managed by
CUCM SME).

Emergency number support for Extension Mobility

Partitions
Menu Value
Call Routing > Class of Control > Partition > Add Create a partition for emergency numbers for each
new site, for example: EN_HQ_PT, EN_RSA_PT,
EN_RSB_PT.
Route Patterns
Call Routing > Route/Hunt > Route Pattern > Add new
Route Partition Choose Partition for appropriate Route Pattern
Urgent Priority Checked
Calling Party Transform Mask Enter valid office attendant phone number (unique for
each site)
Calling search spaces
Call Routing > Class of Control > Calling Search Space > Add new
Create a CSS for emergency numbers for each site and another one for non-emergency numbers.

 CSS_LINE associated to the line deals with general call right except emergency numbers.
 CSS_PHONE associated to the phone deals with emergency calls. This CSS should be unique for
each site.
Device > Phone > Calling Search Space
Associate the calling search spaces for emergency numbers with particular phones (deivces), and
calling search spaces for non-emergency numbers with lines.
Device > Phone -> find a phone ->Calling select the proper CSS
Search Space field
Device > Phone -> find a phone ->select the line select the proper CSS
on the left menu -> Calling Search Space field
Survivable Remote Site Telephony configuration
SRST mode is not supported with BT/BTIP infrastructure but with local PSTN gateway configured on
CE router

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6 Cisco Unity Connection configuration

Cisco Unified Communication Manager Configuration
Menu Value
System > Device Pool > Add New Add new Device pool
Advanced FeaturesVoice Mail > Cisco Voice Create a new Cisco Voice Mail Server and add
Mail Port Wizard > ports to it
Call Routing > Route/Hunt > Line Group add/configure the Answering Voice Mail Ports to a
Line Group
Call Routing > Route/Hunt > Hunt List > Add include the Line Group created earlier
New
Call Routing > Route/Hunt > Hunt Pilot > Add include the Hunt List created earlier
New
Advanced Features > Voice Mail > Message add one number for turning MWIs on and one for
Waiting turning MWIs off
Advanced Features > Voice Mail > Voice Mail Configure the voice mail pilot
Pilot > Add New
Advanced Features > Voice Mail > Voice Mail Associate Voice Mail Pilot number created earlier
Profile > Add New with this profile
Cisco Unity Connection Configuration
Telephony Integrations > Phone System Configure the phone system
Phone System Basics > Related Links drop- Port group configuration
down box > Add Port Group > Go
Port Group Basics > Related Links drop-down Add and configure required number of ports
box > Add Ports > Go
Cisco Unity Connection Administration > On Search Port Groups page click the display
Telephony Integrations > Port Group name of the port group that you created with the
phone system integration
Port Group Basics page > Edit > Servers > add backup CUCM servers if needed
BT/BTIP specific parameters
Telephony Integrations -> Port Group -> choose change the codec list used for calls to CUC - select
appropriate -> Edit -> Codec Advertising G.711 A-law / G.711ulaw/G.722 or G.729 codecs in
advertised codecs.
System Setting > General Configuration Select G.711 Alaw or G.729 codec as specified for
Recording Format parameter

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7 Unified Contact Center Express configuration

Post Installation
Menu Value
UCCX administrative interface Follow Post-installation tasks
http://<UCCX_IP_address>/appadmin

CUCM Administration > User management > End Create UCCX administrative user in CUCM
users
UCCX administration > Tools > User Management select this user as administrator
> Administrator capability view
Tools > Plugin > Desktop suite > Client Run the Unified CCX Desktop Client Configuration
Configuration Tool Tool
Applications > Application Management Add a new Cisco script application
From the Application Type drop-down menu, choose Cisco Script Application and select your script or
the standard ICD script SSCRIPT[ics.aef].
Set the value for CSQ variable: check the checkbox next to it and enter “mycsq” (will be created later)
into the text area. (quotation marks are required)
Configure a new trigger for this application (Add new trigger), this is the CTI route point, which will
route calls to this application
Testing your System and the Unified CCX Script
Use one of your IP phones to call the Unified CM Telephony trigger.
If you get the welcome prompt, then the icd.aef script is working
Provisioning UCCX - Configuring IP Phone Agent service
User Management > End User In the Controlled Devices list box below the
Device Information section, select the agent’s
phone device.
Phone > Subscribe services dropbox Subscribe all agents’ phones to this newly
created service
Create an application user named “telecaster” with “telecaster” as the password (or whatever BIPPA
user ID and password was specified in the CAD Configuration Setup utility).
Assign the telecaster application user to all the IP agent phones
Provisioning UCCX - Provisioning Call Control Group
Subsystems > Unified CM Telephony > Call Provision Unified CM Telephony call control
Control Group groups
Synchronize Cisco JTAPI Client and Unified CM Telephony Data (this creates all necessary CTI devices
on CUCM using AXL interface)
Provisioning UCCX - Configuring Customer Service Queues
Subsystems > RmCm > RmCm Provider Provision the RmCm Provider to allow the RmCm
Subsystem to be in service.
Subsystems > RmCm > Skills Create skills..
Subsystems > RmCm > Resources Assigning skills to agents
Subsystems > RmCm > Contact Service Queues Creating Contact Service Queues
Subsystems > RmCm > Teams Creating teams and assigning agents to teams

Menu Value
UCCX administration > System > System G711
Parameters > Media Parameters > Codec
DTMF support An MTP or transcoder is needed to translate the
DTMFs sent through a SIP trunk.

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8 Cisco Unified Attendant Console configuration

CISCO UNIFIED COMMUNICATION MANAGER
Device>CTI Route Point>Add New
Menu Value
User ID CUDAC
Password Enter password
Confirm Password Confirm entered password
User Management > Application User > Add new
User ID CUDAC
Password Enter password
Confirm Password Confirm entered password
BLF Presence Group Standard Presence Group
Permissions Information -Standard Access AXL API
-Standard CTI Allow Car Park Monitoring
-Standard CTI Allow Calling Number Modification
-Standard CTI Allow Control of All Devices
-Standard CTI Allow Reception of SRTP Key Material
-Standard CTI Enabled
-Standard CTI Allow Control of Phones supporting
Rollover Mode
-Standard CTI Allow Control of Phones supporting
Connected Xfer and conf
CISCO UNIFIED ATTENDAND ADMIN
Menu Value
Installation  When asked enter the IP address of the
machine server is being installed on
 If SQL Server Express is already installed enter
the SQL Server name, User Name, ale
password. If you don’t have SQL installed it will
be installed automatically
 Enter the IP address of CUCM
 Enter port number (443)
 Enter Application User credentials created
before
 If certificate security alert from CUCM will be
displayed it means connection was successful,
accept the certificate
 Follow on screen instructions
Database Wizard  Once installation is completed the database is
started, let the wizard to perform necessary
configuration, when done, click finish, and
restart the computer.
http://<<ip.address.of.Unified.Attendand.Server>>/w Login to the Attendant Server administration
ebadmin/login.aspx User name: ADMIN
Password: CISCO
Engineering > Administrator Management Let’s you change default password
Engineering > Database Management Parameters for the SQL server, if blank enter IP address
of machine where SQL server is installed, specify user
name, and password,

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Menu Value
Engineering > CUCM connectivity CUCM parameters, if blank, enter CUCM IP address in
name field, port number (443), and user name and
password of application user.
Engineering > Database Management Parameters for the SQL server, if blank enter IP address
of machine where SQL server is installed, specify user
name, and password of application user
System Configuration > System Device Menagment
CT Gateway Devices> From 6301 (example)
CT Gateway Devices> To 6302 (example)
Service Devices> From 6401 (example)
Service Devices>To 6402 (example)
Park Devices>From 6501 (example)
Park Devices>To 6502 (example)
System Configuration > System Device Menagment Synchronize with CUCM (Devices will be added
automatically to CUCM)
User Configuration > General Properties
Minimum internal device digit length 1
Maximum internal device digit length 7
External access number 8
Note! Such configuration is necessary to perform successful delayed transfer. Although etting external access
number makes it impossible to perform onnet connections to numbers beginning with 8 (i.e LO BLB) as even
though they are seven digits numbers, they are traeted as external numbers. Refer to mantis ticket 2462.
User Configuration > Queue Management
Team Dev1
DDI 6100 (example)
Synchronize with CUCM Will be automatically added to CUCM as CTI port
User Configuration > Operator Management
Login Name OPERATOR1 (example)
Password Set password
Confirm Password Confirm password
Associated Queues Associate queue created in previous step
CISCO UNIFIED ATTENDAND CONSOLE
Menu Value
Installation  When asked enter the IP address of Cisco
Unified Attendant Server
 Select the language for application
 Follow on screen instruction until installation I
completed
Login Login with credentials created in previous step
CISCO UNIFIED COMMUNICATION MANAGER
User Management > Application User > CUDAC
Controlled Devices Associate devices added by CUDAC Admin
Device > CTI route point > Route point created by CUDAC Admin
Media Resource Group List MRGL_MTP_XCODE

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9 CUCM with Cisco Unified Border Element configuration

9.1 General CUBE configuration (flow-through mode by default)

network interface
Note : for two SIP trunks two IP addresses must be configured.

interface GigabitEthernet0/0
description CUBE Voice Interface
no ip address
duplex auto
speed auto
!
interface GigabitEthernet0/0.<INTERFACE>
description *** CUBE ***
encapsulation dot1Q <INTERFACE>
ip address <IP_ADDR> <Mask>

SNMP Server

snmp-server community public RO

snmp-server manager

Global settings

voice service voip

mode border-element license capacity [session count]
allow-connections sip to sip
sip
header-passing
error-passthru
pass-thru headers unsupp
no update-callerid
early-offer forced
midcall-signaling passthru
sip-profiles 1
ip address trusted list
ipv4 A.B.C.D ! primary SBC IP address
ipv4 E.F.G.H ! backup SBC IP address

Codecs
For customers using G.711 alaw codec:

voice class codec 1

codec preference 1 g711alaw

For customers using G.729 codec use following configuration:

voice class codec 2

codec preference 1 g729r8

SIP User Agent

sip-ua
retry invite 1
retry response 2
retry bye 2
retry cancel 2
reason-header override
connection-reuse
g729-annexb override
timers options 1000

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Support for Privacy and P-Asserted Identity

To enable the privacy settings for the header on a specific dial peer, use the voice-class sip privacy id
command in dial peer voice configuration mode:

dial-peer voice tag voip

voice-class sip privacy id

To enable the translation to PAID privacy headers in the outgoing header on a specific dial peer, use
the voice-class sip asserted-id pai command in dial peer voice configuration mode:

dial-peer voice tag voip

voice-class sip asserted-id pai

9.2 Configuration for a CUCM cluster and two CUBEs

CUBE needs to be configured with physical interface will be configured with a secondary IP
address.

interface FastEthernet 0/0.<INTERFACE>

ip address <PRIMARY_IP_ADDR> <Mask>

ip address <SECONDARY_IP_ADDR> <Mask> secondary

CUCM cluster will be configured with 4 different SIP trunks :

 1st SIP trunk pointing to the primary address of Primary CUBE

 2nd SIP trunk pointing to the secondary address of Primary CUBE

 3rd SIP trunk pointing to primary address of Secondary CUBE

 4th SIP trunk pointing to secondary address of Secondary CUBE

CUCM will be configured with a Route List composed of (at least) 4 Route Groups. Each route
group will include SIP trunk to one of CUBE IP Address (Primary or Secondary). On each route
group parameters, a specific prefix should be defined (one prefix for each RG). This way the
CUBE will be able to route the outgoing calls to the right SBC, depending on this prefix value:

For incoming and outgoing calls for CUCMs side

dial-peer voice 1 voip

description ** to/from site devices - Primary CUCM **

answer-address <INTERFACE>....

destination-pattern <INTERFACE>....

session protocol sipv2

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session target ipv4:<PRIMARY_CUCM_IP_ADDR>

voice-class codec 1

voice-class sip options-keepalive up-interval 300 down-interval 300 retry 5

dtmf-relay rtp-nte

no vad

dial-peer voice 2 voip

description ** to/from site devices - Backup CUCM **

preference 1

answer-address <INTERFACE>....

destination-pattern <INTERFACE>....

session protocol sipv2

session target ipv4:<SECONDARY_CUCM_IP_ADDR>

voice-class codec 1

voice-class sip options-keepalive up-interval 300 down-interval 300 retry 5

dtmf-relay rtp-nte

no vad

!For outgoing calls (with a prefix to select the target SBC)

dial-peer voice 102 voip

description ** Outgoing calls - Outbound dial peer - Primary SBC side **

translation-profile outgoing 113

huntstop

destination-pattern 113T

session protocol sipv2

session target ipv4:<PRIMARY_SBC_IP_ADDR>

voice-class codec 1

voice-class sip options-keepalive up-interval 300 down-interval 300 retry 5

voice-class sip send 180 sdp

dtmf-relay rtp-nte

no vad

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dial-peer voice 103 voip

description ** Outgoing calls - Outbound dial peer - Backup SBC side **

translation-profile outgoing 114

huntstop

destination-pattern 114T

session protocol sipv2

session target ipv4:<SECONDARY_SBC_IP_ADDR>

voice-class codec 1

voice-class sip options-keepalive up-interval 300 down-interval 300 retry 5

voice-class sip send 180 sdp

dtmf-relay rtp-nte

no vad

!For incoming calls

dial-peer voice 100 voip

description ** Incoming calls - Inbound dial peer - SBC side **

answer-address +.T

session protocol sipv2

voice-class codec 1

voice-class sip send 180 sdp

dtmf-relay rtp-nte

no vad

The prefix should be stripped using voice translation rules before sending the call to the
infrastructure.

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9.3 Configuration for a single CUCM server and one CUBE

CUBE needs to be configured with physical interface will be configured with a secondary IP
address.

interface FastEthernet 0/0.<INTERFACE>

ip address <PRIMARY_IP_ADDR> <Mask>

ip address <SECONDARY_IP_ADDR> <Mask> secondary

CUCM will be configured with 2 different SIP trunks :

 1st SIP trunk pointing to the primary address of the CUBE

 2nd SIP trunk pointing to the secondary address of the CUBE

CUCM will be configured with a Route List composed of (at least) 2 Route Groups. Each route
group will include one of the SIP trunk configured. On each route group parameters, a specific
prefix should be defined. This way the CUBE will be able to route the outgoing calls to the right
SBC, depending on this prefix value:

dial-peer voice 1 voip

description **CUCMBE**

answer-address 227....

destination-pattern 227....

session target ipv4:<CUCMBE_IP>

[…]

!For outgoing calls (with a prefix to select the target SBC)

dial-peer voice 11 voip

description ** Outgoing calls - Outbound dial peer - SBC1 side **

answer-address 227....

destination-pattern 11T

session-target <SBC1_IP>

[…]

dial-peer voice 12 voip

description ** Outgoing calls - Outbound dial peer - SBC2 side **

answer-address 227....

destination-pattern 12T

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session-target <SBC2_IP>

[…]

dial-peer voice 101 voip

description ** Incoming calls - Inbound dial peer - SBC side **

answer-address +.T

voice-class codec 1

voice-class sip send 180 sdp

session protocol sipv2

dtmf-relay rtp-nte

no vad

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9.4 Configuration for a CUCM cluster and one CUBE

CUBE needs to be configured with physical interface will be configured with a secondary IP
address.

interface FastEthernet 0/0.<INTERFACE>

ip address <PRIMARY_IP_ADDR> <Mask>

ip address <SECONDARY_IP_ADDR> <Mask> secondary

CUCM cluster will be configured with 2 different SIP trunks :

 1st SIP trunk pointing to the primary address of the CUBE

 2nd SIP trunk pointing to the secondary address of the CUBE

CUCM will be configured with a Route List composed of (at least) 2 Route Groups. Each route
group will include one of the SIP trunk configured. On each route group parameters, a specific
prefix should be defined. This way the CUBE will be able to route the outgoing calls to the right
SBC, depending on this prefix value:

For incoming and outgoing calls for CUCMs side

dial-peer voice 1 voip

description **CUCM SUB**

preference 1

answer-address 227....

destination-pattern 227....

voice-class codec 1

session target ipv4:<CUCM2_IP>

[…]

dial-peer voice 2 voip

description **CUCM PUB**

preference 2

answer-address 227....

destination-pattern 227....

voice-class codec 1

session target ipv4:<CUCM1_IP>

[…]

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For outgoing calls (with a prefix to select the target SBC)

dial-peer voice 11 voip

preference 1

answer-address 227....

destination-pattern 11T

session-target <SBC1_IP>

[…]

dial-peer voice 12 voip

preference 2

answer-address 227....

destination-pattern 12T

session-target <SBC2_IP>

[…]

For incoming calls

dial-peer voice 101 voip

description ** Incoming calls - Inbound dial peer - SBC side **

answer-address +.T

voice-class codec 1

voice-class sip send 180 sdp

session protocol sipv2

dtmf-relay rtp-nte

no vad

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