________________________________

SIP Trunking Service

Configuration Guide

Cisco Unified Communications Manager PBX

Ver. 10.5

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Confidentiality and copyright statement
The information contained in this document is the property of Videotron Ltd. and must
be kept confidential. The use or distribution of this material without prior consent is
therefore strictly prohibited.

This document was written using gender-neutral language.

The information contained herein is subject to change without prior notice.

Modification history

Edit Date Author Description

1.0 2019-08-19 Pascal Beauregard Original draft

1.1 2019-09-12 Martin Lefrançois Review of the consistency with other manuals
1.2 2019-09-18 A. Marchard Linguistic revision
1.3 2019-09-30 Martin Lefrançois Validation

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Table of Contents
Confidentiality and copyright statement ................................................................................ 2
Modification history............................................................................................................... 2
1 Audience ........................................................................................................................... 5
2 Introduction ....................................................................................................................... 5
3 Network and equipment diagram ....................................................................................... 5
3.1 Physical connection between the CUBE and the customer’s Internet access.............. 6
4 Features ............................................................................................................................ 6
4.1 Supported features ..................................................................................................... 6
4.2 Unsupported or limited features .................................................................................. 8
5 Service requirements ........................................................................................................ 9
5.1 Registering a SIP trunk ............................................................................................... 9
5.2 Responding to SIP INFO messages ........................................................................... 9
5.3 Sending the domain name in the Req URI header of SIP INVITE messages .............. 9
5.4 Configuration settings overview .................................................................................. 9
6 Configuration ................................................................................................................... 10
6.1 Configuring the CUBE (Cisco router 29xx) ................................................................ 10
Step 1: Configuring the physical interfaces ..................................................................... 10
Step 2: IP host section .................................................................................................... 10
Step 3: Voice service VoIP section ................................................................................. 10
Step 3: Sip-ua section ..................................................................................................... 11
Step 4: Voice class sip-profiles section ........................................................................... 11
Step 5: Voice translation section (optional) ..................................................................... 12
Step 6: Voice class URI section ...................................................................................... 12
Step 7: Dial-peer section................................................................................................. 12
6.2 Configuring the CUCM .............................................................................................. 13
Step 1: Login to the Publisher at Cisco Unified CM administration .................................. 14
Step 2: Configuring a Partition and a Calling Search Space (outbound calls) ................. 14
Step 3: Configuring a Calling Search Space (outbound calls) ......................................... 14
Step 4: Applying the CSS to a test telephone (outbound calls) ....................................... 15
Step 5: Configuring a SIP Profile .................................................................................... 16
Step 6: Creating a SIP TRUNK Security Profile .............................................................. 18
Step 7: Configuring the SIP Trunk .................................................................................. 19

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 Step 8: Configuring the Route Group (outbound calls) .................................................... 21
Step 9: Configuring the Route List (outbound calls) ........................................................ 22
Step 10: Configuring a Route Pattern (outbound calls) ................................................... 23
Step 11: Configuring the External Phone Number Mask (outbound calls) ....................... 24
7 Glossary .......................................................................................................................... 25

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1 Audience
The SIP Trunking Service Configuration Guide is intended for service users, technical managers and
authorized integrators.

2 Introduction
The SIP Trunking Service Configuration Guide details the basic steps for setting up a single SIP
trunk between Videotron’s SBC and a Cisco Unified Border Element (CUBE) placed in front of an IP
Cisco Unified Communications Manager (CUCM) PBX. Several SIP trunks may be set up, but this
document does not go over the steps for doing so.

That said, this guide is not intended to help you configure PBX user/application features.

3 Network and equipment diagram

The diagram below is an overhead view of SIP trunking with a Cisco Unified Communications
Manager (CUCM) PBX behind a Cisco Unified Border Element (CUBE).

Videotron
network
Client network

PSTN

The solution includes:

Customer site:
• Cisco Unified Communications Manager (CUCM) servers, version 10.5.
• CUBE: Cisco 29xx router (2901, 2921, 2951), IOS version 15.5(3)M
• IP Cisco telephones (7965, 7821, 7841)

Videotron site:
• Videotron SBC: Oracle (Acme Packet)
• Videotron Softswitch: Genband C20
• PSTN connection

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3.1 Physical connection between the CUBE and the customer’s Internet
access
The CUBE must be linked by a 10/100Mbps network connection toward the customer’s Internet.
Usually, the customer has a router behind the Videotron cable modem that provides a connection to
the Internet.

G0/1

G0/

4 Features
4.1 Supported features
The SIP trunking service supports the following features:

Feature Description Limit(s)

Simultaneous calls The simultaneous calls limit is
established when the SIP trunk
order is placed.
Voice G.711 µ-law standard used
exclusively
Fax G.711 µ-law standard used T.38 standard not supported

Other kinds of data (modem, alarm, etc.) G.711 µ-law standard used

Inbound Caller ID name and number Inbound Caller ID name and

number transmitted from the
Videotron site to the PBX.
Outbound Caller ID name Outbound Caller ID name, as
transmitted via PBX to the
public network.
Outbound Caller ID number Outbound Caller ID number, as
transmitted via PBX to the
public network.
DID display for 911 emergency call centre DID display for 911 emergency
call centre transmitted via PBX
if on the predefined list of
numbers.
Direct trunk overflow Calls are routed to another SIP The other SIP trunk must be on
trunk when the number of the same Videotron telephone
simultaneous calls SIP switch as the primary SIP trunk.
trunking can handle is
exceeded.
Failover to another phone number Calls are routed to another The “Redirect information” or
phone number when the “Original called number” field is
number of simultaneous calls not transmitted. The “Called
that the SIP trunk can handle number” is the actual forwarding
is exceeded. number and not the DID.

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 An overflow to another phone
number requires an additional
service called a “Permanent
Redirect Line (PRL).” This
service is billed according to the
predefined number of
simultaneous PRL calls. If the
phone number is long distance,
charges will apply.
.
Failover to another SIP trunk Calls are routed to another SIP If the PBX responds with a SIP
trunk in the following three message other than “503
cases of failure: Service Unavailable,” there will
1. The customer’s PBX no be no call failover.
longer responds to calls
sent to it on the SIP trunk.
2. The customer's PBX
responds with the message
“SIP 503 Service
Unavailable.”
3. The SIP trunk is faulty.
Failover to another phone number Calls are routed to another If the PBX responds with a SIP
phone number in the same message other than “503
three cases as above. Service Unavailable,” there will
be no call failover.

Same limitation as “Direct trunk

overflow” with respect to the
fields and the need for a
Permanent Redirect Line.
“Redirect number” field (Remote Party ID) The Videotron telephone switch
transmits the original called
number to the Remote-Party-ID
header.
Class of restriction call blocking No blocking for local calls, in 1-976 calls are blocked.
Quebec, Canada, the United
States and abroad, and for
411, 0-, 0+, 00 and 900
numbers.
Number portability Videotron handles the The customer must provide all
transfer of a customer’s required documentation.
telephone number from
their current service to
the SIP trunking service.
SIP-Refer If the external number is long
distance in relation to the
original dialled number, the call
Allows you to free up may be dropped rather than
lines after a call is forwarded. Especially when the
forwarded from an call is transferred through
another Videotron switch.
external number to Routing between Videotron
another external number, switches is subject to change
such as a cellphone. without prior notice.

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4.2 Unsupported or limited features

Our SIP trunking does not support the following features:

Feature Description
Numbers outside our coverage area Only telephone numbers in Videotron service areas will be
accepted.
Fixed 911 This feature allows calls to be forwarded directly to the 911
emergency call centre in the municipality where the caller is
located. Instead, the SIP trunking service uses an intermediary
(“nomad”) 911 emergency call centre to forward calls. See
videotron.com/ip-911 for details.
Emergency call forwarding Allows you to forward calls to different destinations based on a
predefined phone tree for emergency scenarios. This is an
advanced feature reserved for the dedicated fibre optic SIP
trunking service.
Authorization and billing codes The authorization code is used to limit access to long-distance
calls. The billing code is used to count calls per user for internal
billing and customer billing purposes.
These are advanced features reserved for the dedicated fibre
optic SIP trunking service.
Equity of access Allows you to use another long distance provider. This feature is
largely irrelevant considering that Videotron offers unlimited
calling plans for Canada and the United States. This feature is
reserved for the dedicated local fibre optic SIP trunking service.
Occasional calls Used to dial the 101-XXXX code in order to temporarily change
long distance provider. This feature is largely irrelevant
considering that Videotron offers unlimited calling plans for
Canada and the United States. This feature is reserved for the
dedicated local fibre optic SIP trunking service.
Signalling and voice channel encryption Videotron does not currently support signalling encryption (SIP
TLS) and voice channel encryption (SRTP). Encrypted MD5 hash
password.

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5 Service requirements
5.1 Registering a SIP trunk
Once the SIP trunk has been configured at the Videotron site, our technical team will send the
following information to the customer:

• domain name
• username
• password

The customer PBX (in this case the customer’s CUBE) must be registered with Videotron in order to
connect calls via SIP trunking. The customer, or more commonly the integrator-interconnector, must
configure the CUBE such as to register the SIP trunk with Videotron’s switch. The Videotron team will
set up a phone conference with the interconnector to complete the registration and ensure the SIP
trunk is functioning properly.

The CUBE is registered by sending SIP REGISTER messages to Videotron’s SBC IP address that
contains a username, password and domain name.

5.2 Responding to SIP INFO messages

Videotron’s telephone switch periodically sends SIP INFO messages to the customer’s CUBE. If
these messages do not reach the CUBE (i.e., they are blocked by the customer’s firewall), or they are
not answered by the CUBE, the switch will consider the CUBE out of order.

5.3 Sending the domain name in the Req URI header of SIP INVITE messages
The CUBE must be capable of sending a domain name in the Req URI of SIP INVITE messages. If
the domain name is missing, any calls will be rejected.

5.4 Configuration settings overview

Table 4 provides an overview of the parameters required to set up the SIP trunking service.

Domain name Provided by Videotron: <customer

acronym>.sipott.v50.videotron.com
Videotron SBC address 24.200.242.87
SIP communication port UDP 5060
Username Provided by Videotron: s<last 9 numbers of primary telephone
number>
Password Provided by Videotron: 12 alphanumeric characters with at
least 1 lowercase letter, 1 uppercase letter, and 1 number
Number of simultaneous calls on the SIP Provided by Videotron
trunk
Codec G.711 µ-law only
Fax protocol In-Band (T.38 not supported)
DTMF RFC2833
SIP REFER The SIP REFER function must only be activated after
discussion with the Videotron team. If the external number
is long distance in relation to the original dialled number,
the call may be dropped rather than forwarded.
Table 1: Configuration settings overview

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6 Configuration
Putting a SIP trunk in place on a CUCM phone service with CUBE requires the configuration
of the CUCM and the CUBE. These two systems are highly versatile and consequently have
several parameters that could affect communication on the SIP trunk. This guide provides a
configuration example that we have tested and that is fully functional.

6.1 Configuring the CUBE (Cisco router 29xx)

With this proposed configuration, it is possible to test calls that will use the CUBE. The
integrator will modify this configuration to meet the specific and comprehensive needs of the
customer.

Step 1: Configuring the physical interfaces

Configuration that reflects the example in Section 3.1. (The configuration must reflect
the customer’s network.)
interface Port-channel1
ip address 10.4.8.2 255.255.255.248

interface GigabitEthernet0/0
description xxxxxxxx port G1/0/2
no ip address
duplex auto
speed auto
channel-group 1

interface GigabitEthernet0/1
description xxxxxxxxxxx port G1/0/1
no ip address
duplex auto
speed auto
channel-group 1

interface GigabitEthernet0/2
description Toward Videotron’s SBC
ip address 10.4.8.21 255.255.255.252
duplex auto
speed auto

Step 2: IP host section

This section demonstrates how to associate Videotron’s SBC IP address to the domain
dame that will be used for Videotron’s SIP Trunk service. If the CUBE has access to a DNS
server, this line is not required.

ip host hofa01.sipott.v50.videotron.com 24.200.242.87

Note: replace the domain name in the example with the domain name that Videotron has assigned to
you.

Step 3: Voice service VoIP section

All the commands in this section must be configured.
The commands in this section define the SIP communication that enters and exits the
CUBE.
voice service voip
ip address trusted list

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 ##List of the IP addresses that can speak SIP with the CUBE – at least have the CUCMs’ and SBC’s addresses.
ipv4 24.200.242.87

## Le chiffre 100 doit être remplacé par le nombre de licences CUBE achetées
mode border-element license capacity 100
allow-connections sip to sip
fax protocol pass-trough g711ulaw ## Parameters for communications via fax

sip
registrar server expires max 3600 min 3600 ## SIP registration parameters
no update-callerid ##To be copied as it is
early-offer forced ##Force the SDP in the Invite SIP
no call service stop ## Activate the SIP service on the router

Step 3: Sip-ua section

This section demonstrates how to configure the parameters for the registration to
Videotron’s SIP trunk service.

For this section you will need the following information:

• Username
• Password
• Domain name

Videotron’s technical team will give you this information when it has programed the service on its
end. A phone appointment is scheduled with Videotron’s technical team and the
customer/integrator.

Below is an example of programming with the following dummy parameters:

• Username: s383870001
• Password: u12Se3Rf2n53
• Domain name: hofa01.sipott.v50.videotron.com

sip-ua
credentials username s383870001 password u12Se3Rf2n53 realm realm
authentication username s383870001 password u12Se3Rf2n53
retry invite 2
timers keepalive active 10
registrar 1 dns: hofa01.sipott.v50.videotron.com expires 3600
connection-reuse

Step 4: Voice class sip-profiles section

Videotron would like the host part of the SIP URI in the INVITE request sent by the IP PBX
to be a label that resembles a domain name rather than an IP address. Example:

Original Req URI in the SIP INVITE request sent by the CUBE toward Videotron’s
SBC prior to the transformation:

Req URI : : <sip:[email protected]:5060>

Req URI after the transformation in the SIP INVITE request sent by the CUBE to
Videotron’s SBC:

Req URI : : <sip:[email protected]:5060>

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 We require a voice class to replace 24.200.247.87 with
“hofa01.sipott.v50.videotron.com” in the Req URI sent by the CUBE to Videotron.

voice class sip-profiles 1

request INVITE sip-header SIP-Req-URI modify "24.200.247.87:5060"
"hofa01.sipott.v50.videotron.com:5060"

To apply the voice-class, you must insert it in the outbound dial-peer to Videotron with the
voice-class sip profiles 1 command (see dial-peer voice 105 VoIP further in this
document).

Step 5: Voice translation section (optional)

This section only applies if a 9 (for example) has been prefixed to the number dialed for
outbound calls. The 9 must be removed before the called number is sent to the PSTN. The
translation profile “ToPSTN” is called by the dial-peer voice 105.

## Is called by the voice translation-profile ToPSTN

voice translation-rule 2
rule 1 /^9\(911\)/ /\1/
rule 2 /^9\([2-8]11\)/ /\1/
rule 3 /^9\([2-9]..[2-9]......\)/ /\1/
rule 4 /^9\(1[2-9]..[2-9]......\)/ /\1/
rule 5 /^9\(0[2-9]..[2-9]......\)/ /\1/
rule 6 /^9\(011.*\)/ /\1/

## Called by the dial-peer voice 105 VoIP

voice translation-profile ToPSTN
translate called 2 ## Calls voice translation-rule 2 and acts on the called number

Step 6: Voice class URI section

Allows you to form the list of IP addresses for which you wish to establish a match in the
incoming dial-peer from Videotron’s SBC (see dial-peer voice 10 VoIP).
voice class uri 1000 sip
host 24.200.242.87

Step 7: Dial-peer section

Configuring the dial-peers allows you to route the calls when they transit via the CUBE.

The dial-peers presented in this section are only examples. The parameters in bold in the
dial-peers are basic parameters to enter in all the dial-peers you configure.

## Inbound dial-peer for calls from the CUCM

dial-peer voice 1 voip
description Incoming call-leg - Calls from the CUCM
session protocol sipv2 ## Force version 2 of SIP
session transport udp ##Force the SIP signaling to be used with the UDP
incoming called-number 9T ##To match the dial peer on 9 as first of the called
## The “voice-class sip bind” command associates the dial-peer to the control SIP messages and the media that
transit on the po1 (customer network therefore SIP messages of the CUCM) these 2 commands are very important
because SIP messages transit via the port G0/2 (to SBC) and po1 (to CUCM).
voice-class sip bind control source-interface Port-channel1
voice-class sip bind media source-interface Port-channel1
dtmf-relay rtp-nte ## Force RFC2833 for the transmission of DTMF

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 codec g711ulaw ## Force G711 voice without compression
ip qos dscp cs3 signaling
no vad ##Prevents the use of Voice Activity Detection.

## Inbound dial-peer for calls from Videotron’s SBC

dial-peer voice 10 voip
description Incoming call-leg - Inbound PSTN calls
session protocol sipv2
## enables match on SIP requests from addresses that are in the voice class uri 1000 sip
incoming uri via 1000
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte
codec g711ulaw
no vad

## Outbound dial-peer for local calls toward Videotron’s SBC

dial-peer voice 105 voip
description Local calls 10 digits toward the PSTN
## Command that strips the 9 before transmission to the SBC – must also configure an associated translation rule not
showed in this document.
translation-profile outgoing ToPSTN ## Call the translation profile ToPSTN that removes the 9 as prefix (optional)
destination-pattern 9[2-9]..[2-9]......
session protocol sipv2
session target ipv4: 24.200.242.87 ##Videotron’s SBC at the address 24.200.242.87 is the target
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
voice-class sip profiles 1 ## Call voice class sip-profiles 1 that inserts the domain in Req URI
dtmf-relay rtp-nte
codec g711ulaw

## Outbound dial-peer for local calls toward the CUCM

dial-peer voice 1046511 voip
description Calls toward CUCM
destination-pattern [2-9]..[2-9]......
session protocol sipv2
session target ipv4:10.4.65.11 ## The CUCM is the target
voice-class sip bind control source-interface Port-channel1
voice-class sip bind media source-interface Port-channel1
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad

6.2 Configuring the CUCM

The CUCM and the CUBE are linked by a SIP trunk (a different SIP trunk than the one to
Videotron). The configurations presented in this section are configuration suggestions that
have been tested successfully. The integrator will modify this configuration to meet all the
customer’s needs.

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Step 1: Login to the Publisher at Cisco Unified CM administration

Step 2: Configuring a Partition and a Calling Search Space (outbound

calls)

1. Add a partition for the routes intended for outbound calls to the SIP Trunk. Call Routing -
> Class of Control -> Partition -> Add New.
2. Enter a meaningful name (e.g., PSTN_SIP_Local_PT) and a meaningful description.

Step 3: Configuring a Calling Search Space (outbound calls)

1. Add a new CSS: Call Routing -> Class of Control -> Calling Search Space -> Add New.
2. Configure the CSS to at least add the Partition created earlier. Use a meaningful name
for the CSS. E.g., XXX_SIP_Local_Line_CSS. Replace XXX with the site’s acronym,
and the remainder of the name provides the PSTN access level (local, long distance,
etc.).

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Step 4: Applying the CSS to a test telephone (outbound calls)

1. Go to the line of a test telephone and select the CSS created in step 3.

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Step 5: Configuring a SIP Profile

1. Add a new SIP Profile: Device -> Device Settings -> SIP Profile -> Add New.
2. Configure the SIP Profile as indicated in the image below. Use a meaningful SIP Profile name.

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Page | 17
Step 6: Creating a SIP TRUNK Security Profile

1. Add the SIP Trunk Security Profile. Go to the System menu > Security Profile > SIP
Trunk Security Profile.
2. Select the Non-Secure SIP Trunk Profile and click on Copy.
3. Change the SIP Trunk Security Profile name to “PSTN SIP TRUNK Profile,” for
example.
4. Save.

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Step 7: Configuring the SIP Trunk

1. Add the Trunk SIP Device-> Trunk -> Add New.

2. Configure the Trunk SIP parameters as indicated in the image below.

Note: The configuration of the Calling Search Spaces for the “Inbound calls” section and the “Calling party
transformation CSS” of the “Outbound Calls” part must have been done beforehand. The customer must define
his or her call permissions for inbound calls on this SIP Trunk and the way the calling number and called number
of an outbound call can be modified. The same applies for the Device Pool and Media Resource Group List.

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Page | 20
Step 8: Configuring the Route Group (outbound calls)

1. Add a Route Group: Call Routing -> Route/Hunt -> Route Group -> Add New.
2. Configure the Route Group parameters as indicated in the image below.

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Step 9: Configuring the Route List (outbound calls)

1. Add a Route List: Call Routing -> Route/Hunt -> Route List -> Add New.
2. Configure the Route List parameters as indicated in the image below.

3. Click on the Route Group in the Route List to configure the Route Group parameters
when it is used with this Route List.
4. Configure the Route Group parameters as indicated in the image below.

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Step 10: Configuring a Route Pattern (outbound calls)

1. Add a Route Pattern: Call Routing -> Route/Hunt -> Route Pattern -> Add New.
2. Configure the Route Pattern parameters as indicated in the image below (use a different
number).

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Step 11: Configuring the External Phone Number Mask (outbound calls)

Outbound display can be configured in several locations in the CUCM (e.g., Route pattern,
Route-List, on a device’s line).

Here is one of the methods for testing whether the name and number ID are working
properly for outbound calls on the Trunk toward Videotron.

Modify the “ASCII Display (Caller ID) field and the “External Phone Number Mask”
field in the configuration of the telephone line configured in step 4.

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7 Glossary
503 Service unavailable
Server error code.

bursting Feature that allows you to temporarily exceed your calling limit. Simultaneous calls are
billed on a pay-per-use basis. Feature currently in development
called number Number called or requested
called party
Person to whom a call is sent.

calling party
Person sending a call to establish communication.

C20 Videotron telephone switch

CO line central office line

Communication line that connects a PBX to a telephone service provider’s switchboard.

G.711 Digital voice encoding standard

H.323 Standard for transmitting audio, data and images in real time across packet networks. Used for local
networks, like an intranet, or public networks, like the Internet.
Less commonly used than SIP.

IP Internet protocol
IP-GW IP gateway
key system Intercom system, key telephone system
Most commonly used telephone system when few additional extensions are required. Allows users to
call each other directly and to communicate with public network subscribers via outbound and inbound
calls.

original Called Number

PBX Private branch exchange
A company’s private telephone switch

PSTN public switched telephone network

redirect information
REFER SIP method for transferring calls whereby the call is sent to a number indicated in the transfer request.
Allows you to free up lines after a call is forwarded from an external number to another external
number, such as a cellphone.

PSTN public switched telephone network

SBC session border controller
A network element to monitor and protect SIP-based communications from fraud and allowing you to
configure SIP trunk settings.

DID direct inward dialling

Telephone feature allowing an outbound caller to reach a subscriber directly without going through an
operator or dialling an extension. DID number.

SIP session initiation protocol

Logon protocol used in IP telephony. Refers to an IP telephony service allowing a telephone switch to
access the PSTN, thereby supporting the management of call signalling, over IP links using SIP
trunking.

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Softswitch software switch, media gateway controller, call controller, call server
Interconnection equipment that manages the operation of a media gateway that allows signals carrying
voice, data or images to move from a circuit-switched public telephone network to a private packet-
switched network, such as a private IP network—or to go in the reverse.

T.38 Encoding standard for sending faxes across IP networks in a real-time mode.

trunk Circuit
A line that connects switches with each other and is used to route information sequentially.

trunk group; TG
Circuitry starting from a single switch and terminating at one or more switches giving access to the
same subscribers. In the specific case of the Videotron SIP trunking service, TG refers to a SIP trunk.
In certain exceptional situations, there may be more than one TG or multiple SIP trunks between a
PBX and Videotron.

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