Nortel Networks Water System NN10043 113 User Manual

[Standard Level - Nortel Networks Confidential]  
Version MCP 1.1 FP1 (02.02)  
Part No. NN10043-113  
April 2003  
SIP Provisioning Client User  
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iii  
Nortel Networks Inc. software license agreement  
This Software License Agreement (“License Agreement”) is between you, the end-user (“Customer”) and Nortel  
Networks Corporation and its subsidiaries and affiliates (“Nortel Networks”). PLEASE READ THE FOLLOWING  
CAREFULLY. YOU MUST ACCEPT THESE LICENSE TERMS IN ORDER TO DOWNLOAD AND/OR USE THE  
SOFTWARE. USE OF THE SOFTWARE CONSTITUTES YOUR ACCEPTANCE OF THIS LICENSE  
AGREEMENT. If you do not accept these terms and conditions, return the Software, unused and in the original shipping  
container, within 30 days of purchase to obtain a credit for the full purchase price.  
“Software” is owned or licensed by Nortel Networks, its parent or one of its subsidiaries or affiliates, and is copyrighted  
and licensed, not sold. Software consists of machine-readable instructions, its components, data, audio-visual content  
(such as images, text, recordings or pictures) and related licensed materials including all whole or partial copies. Nortel  
Networks grants you a license to use the Software only in the country where you acquired the Software. You obtain no  
rights other than those granted to you under this License Agreement. You are responsible for the selection of the  
Software and for the installation of, use of, and results obtained from the Software.  
1. Licensed Use of Software. Nortel Networks grants Customer a nonexclusive license to use a copy of the Software  
on only one machine at any one time or to the extent of the activation or authorized usage level, whichever is applicable.  
To the extent Software is furnished for use with designated hardware or Customer furnished equipment (“CFE”),  
Customer is granted a nonexclusive license to use Software only on such hardware or CFE, as applicable. Software  
contains trade secrets and Customer agrees to treat Software as confidential information using the same care and  
discretion Customer uses with its own similar information that it does not wish to disclose, publish or disseminate.  
Customer will ensure that anyone who uses the Software does so only in compliance with the terms of this Agreement.  
Customer shall not a) use, copy, modify, transfer or distribute the Software except as expressly authorized; b) reverse  
assemble, reverse compile, reverse engineer or otherwise translate the Software; c) create derivative works or  
modifications unless expressly authorized; or d) sublicense, rent or lease the Software. Licensors of intellectual property  
to Nortel Networks are beneficiaries of this provision. Upon termination or breach of the license by Customer or in the  
event designated hardware or CFE is no longer in use, Customer will promptly return the Software to Nortel Networks  
or certify its destruction. Nortel Networks may audit by remote polling or other reasonable means to determine  
Customer’s Software activation or usage levels. If suppliers of third party software included in Software require Nortel  
Networks to include additional or different terms, Customer agrees to abide by such terms provided by Nortel Networks  
with respect to such third party software.  
2. Warranty. Except as may be otherwise expressly agreed to in writing between Nortel Networks and Customer,  
Software is provided “AS IS” without any warranties (conditions) of any kind. NORTEL NETWORKS DISCLAIMS  
ALL WARRANTIES (CONDITIONS) FOR THE SOFTWARE, EITHER EXPRESS OR IMPLIED, INCLUDING,  
BUT NOT LIMITED TO THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A  
PARTICULAR PURPOSE AND ANY WARRANTY OF NON-INFRINGEMENT. Nortel Networks is not obligated to  
provide support of any kind for the Software. Some jurisdictions do not allow exclusion of implied warranties, and, in  
such event, the above exclusions may not apply.  
3. Limitation of Remedies. IN NO EVENT SHALL NORTEL NETWORKS OR ITS AGENTS OR SUPPLIERS BE  
LIABLE FOR ANY OF THE FOLLOWING: a) DAMAGES BASED ON ANY THIRD PARTY CLAIM; b) LOSS OF,  
OR DAMAGE TO, CUSTOMER’S RECORDS, FILES OR DATA; OR c) DIRECT, INDIRECT, SPECIAL,  
INCIDENTAL, PUNITIVE, OR CONSEQUENTIAL DAMAGES (INCLUDING LOST PROFITS OR SAVINGS),  
WHETHER IN CONTRACT, TORT OR OTHERWISE (INCLUDING NEGLIGENCE) ARISING OUT OF YOUR  
USE OF THE SOFTWARE, EVEN IF NORTEL NETWORKS, ITS AGENTS OR SUPPLIERS HAVE BEEN  
ADVISED OF THEIR POSSIBILITY. The forgoing limitations of remedies also apply to any developer and/or supplier  
of the Software. Such developer and/or supplier is an intended beneficiary of this Section. Some jurisdictions do not  
allow these limitations or exclusions and, in such event, they may not apply.  
4. General  
a. If Customer is the United States Government, the following paragraph shall apply: All Nortel Networks  
Software available under this License Agreement is commercial computer software and commercial computer  
software documentation and, in the event Software is licensed for or on behalf of the United States  
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Government, the respective rights to the software and software documentation are governed by Nortel  
Networks standard commercial license in accordance with U.S. Federal Regulations at 48 C.F.R. Sections  
12.212 (for non-DoD entities) and 48 C.F.R. 227.7202 (for DoD entities).  
b. Customer may terminate the license at any time. Nortel Networks may terminate the license if Customer fails to  
comply with the terms and conditions of this license. In either event, upon termination, Customer must either  
return the Software to Nortel Networks or certify its destruction.  
c. Customer is responsible for payment of any taxes, including personal property taxes, resulting from Customer’s  
use of the Software. Customer agrees to comply with all applicable laws including all applicable export and  
import laws and regulations.  
d. Neither party may bring an action, regardless of form, more than two years after the cause of the action arose.  
e. The terms and conditions of this License Agreement form the complete and exclusive agreement between  
Customer and Nortel Networks.  
f. This License Agreement is governed by the laws of the country in which Customer acquires the Software. If the  
Software is acquired in the United States, then this License Agreement is governed by the laws of the state of  
New York.  
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What is not covered . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .xix  
Acronyms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xx  
Selecting a time zone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6  
Creating a new time zone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8  
Modifying or Deleting a time zone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8  
Changing the admin password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9  
Exiting the SIP Provisioning Client . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9  
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vi Contents  
System administrator provisioning role . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16  
Listing provisioning roles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17  
Setting up ban list . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36  
Defining user device status reasons . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38  
Viewing status reasons . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39  
Chapter 4  
IPCMs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41  
What is an IPCM? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41  
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Assigning and modifying IPCMs to a domain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44  
Provisioning a Line-based voice mail server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60  
Assigning a user to a provisioned voice mail server . . . . . . . . . . . . . . . . . . . . . . . . . . 67  
Assigning services to a service package . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83  
Listing/modifying/deleting service packages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87  
Assigning service packages to domains and subdomains . . . . . . . . . . . . . . . . . . . . . . 89  
Upgrading and downgrading service packages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90  
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viii Contents  
Setting up rules for the conference server route . . . . . . . . . . . . . . . . . . . . . . . . . 104  
Listing/modifying/deleting users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105  
Adding an i2004 Internet Telephone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111  
Listing a telephony route . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128  
Setting up an interworking profile with the Communications Server 2000 (CS2000) . 133  
Chapter 11  
Gateways . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135  
Introduction to gateways . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135  
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Listing a gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137  
Listing a gateway route . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 140  
Listing a gateway trunk group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142  
Provisioning Sequence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145  
Step 2: Create the COS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161  
Step 3: Create the subscribers and assign the COS . . . . . . . . . . . . . . . . . . . . . . 162  
Step 4: Create gateways, gateway routes and trunk groups . . . . . . . . . . . . . . . . 165  
Step 5: Create the telephony routes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166  
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Adding a private route to subdomain rich.abc.com . . . . . . . . . . . . . . . . . . . . 167  
Adding a gateway route to abc.com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172  
911 Emergency Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181  
Adding an OSN subscriber . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 192  
Using the SIP Personal Agent to route 911 emergency notifications . . . . . . . . . . . . . 192  
Setting up a route list for the emergency subscriber . . . . . . . . . . . . . . . . . . . . . . 192  
Setting up a rule for the emergency notification route . . . . . . . . . . . . . . . . . . . . . 193  
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195  
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Figure 25 Modify IPCM window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45  
Figure 26 IPCM Domain Relation window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46  
Figure 27 IPCM Delete confirmation dialog box . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47  
Figure 28 Listing physical IPCMs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48  
Figure 29 Delete physical IPCM confirmation dialog box . . . . . . . . . . . . . . . . . . . . . 48  
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xii Figures  
Figure 60 Conference user - route added . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103  
Figure 61 Customize service package link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105  
Figure 62 List Users window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106  
Figure 63 List users search results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106  
Figure 64 User details window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107  
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Figure 95 SIP Application Module routing stages . . . . . . . . . . . . . . . . . . . . . . . . . . 152  
Figure 96 Route list redirect (domain to domain) . . . . . . . . . . . . . . . . . . . . . . . . . . 153  
Figure 97 Route list redirect (between subdomains) . . . . . . . . . . . . . . . . . . . . . . . 154  
Figure 98 Domains/subscribers used for the sample dial plan . . . . . . . . . . . . . . . . 156  
Figure 99 Parent domain provisioning menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160  
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xiv Figures  
Figure 130 Restricted route trace . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180  
Figure 131 Creating a COS for the 911 notification . . . . . . . . . . . . . . . . . . . . . . . . . 185  
Figure 132 Adding a new gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 186  
Figure 133 List of gateways . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 186  
Figure 134 Creating a new gateway route . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 186  
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Figure 137 Change parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 188  
Figure 138 Creating a new route list . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 189  
Figure 139 Create gateway telephony route to PSAP . . . . . . . . . . . . . . . . . . . . . . . 190  
Figure 140 Gateway route options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190  
Figure 141 Add user - emergency subscriber . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191  
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xvi Figures  
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Tables  
Example dial plan data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157  
Dial plan for abc.com and xyz.com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157  
Association between COS and Routes . . . . . . . . . . . . . . . . . . . . . . . . . 158  
Subscribers in rich.abc.com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166  
Data for SIP routes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169  
Table 13  
Table 22  
Table 23  
Table 24  
Table 25  
Table 26  
Table 27  
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xviii Tables  
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xix  
Welcome  
The Session Initiation Protocol (SIP) Provisioning Client is a web-based  
Graphical User Interface (GUI) for service providers to complete administrative  
tasks, for example, provisioning administrators, domains, IP Client Managers  
(IPCMs) and IPCM clusters.  
Overview  
Service providers access the SIP Provisioning Client with any web browser. Being  
a web-based system, the SIP Provisioning Client provides the service providers  
with an highly efficient access from anywhere capability.  
Note: The provisioning abilities visible to an administrator in the SIP  
Provisioning Client GUI depends on the level of authority assigned to  
their account.  
Audience  
This guide is intended for service providers and administrators (with varying  
levels of privileges) assigned to manage customer domains.  
What is not covered  
This document does not cover  
Configuration  
System Management Console  
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xx Welcome  
Text and graphic conventions  
This guide uses the following text and graphic conventions:  
bold text  
Indicates a menu option, link, or command key you  
need to click. Example: Click Ok.  
italic text  
Indicates  
a domain name or subdomain name  
a document title  
a variable name  
separator ( > )  
Shows menu paths.  
Example: Gateway > Add Gateway  
Indicates a top level menu item in the SIP Provisioning  
Client menu hierarchy.  
Indicates a sub menu item in the SIP Provisioning  
Client menu hierarchy.  
Indicates a form in the SIP Provisioning Client menu  
hierarchy.  
Acronyms  
This guide uses the following acronyms  
CAS  
Channel Associated Signalling  
CPL  
Call Processing Language  
Class of Service  
COS  
CS2000  
DSL  
Communication Server 2000  
Digital Subscriber Line  
Greenwich Mean Time  
GMT  
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Welcome xxi  
DHCP  
DIGMAN  
GUI  
Dynamic Host Configuration Protocol  
Digit Manipulation  
Graphical User Interface  
Internet Protocol  
IP  
IPCM  
MAddr  
MCP  
MSD  
MWI  
Internet Protocol Client Manager  
Media Access Control Address  
Multimedia Communications Portfolio  
Most Significant Digit  
Message Waiting Indicator  
Partial Dial  
PDIL  
PRI  
Primary Rate Interface  
PSEIZ  
PSTN  
SA  
Permanent Seizure  
Public Switched Telephone Network  
System Administrator  
SIP  
Session Initiation Protocol  
Simplified Message Desk Interface  
Time Division Multiplex  
SMDI  
TDM  
UNIStim  
URI  
Unified Network IP Stimulus  
Uniform Resource Identifier  
Uniform Resource Locator  
Voice Mail Server  
URL  
VMS  
VPN  
Virtual Private Network  
Related publications  
SIP Personal Agent Getting User Guide (Part No. NN10039-112)  
SIP Multimedia Web Client User Guide (Part No. NN10040-112)  
SIP Multimedia PC Client User Guide (Part No. NN10041-112)  
i2004 Internet Telephone User Guide (Part No. NN10042-113)  
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xxii Welcome  
You can print selected technical manuals and release notes free, directly from the  
Internet. Go to the www.nortelnetworks.com/documentation URL. Find the  
product for which you need documentation. Then locate the specific category and  
model or version for your hardware or software product. Use Adobe* Acrobat  
Reader* to open the manuals and release notes, search for the sections you need,  
and print them on most standard printers. Go to Adobe Systems at the  
www.adobe.com URL to download a free copy of the Adobe Acrobat Reader.  
How to get help  
If you purchased a service contract for your Nortel Networks product from a  
distributor or authorized reseller, contact the technical support staff for that  
distributor or reseller for assistance.  
If you purchased a Nortel Networks service program, contact Nortel Networks  
Technical Support. To obtain contact information online, go to the  
Technical Support.  
From the Technical Support page, you can open a Customer Service Request  
online or find the telephone number for the nearest Technical Solutions Center.  
If you are not connected to the Internet, you can call 1-800-4NORTEL  
(1-800-466-7835) to learn the telephone number for the nearest Technical  
Solutions Center.  
An Express Routing Code (ERC) is available for many Nortel Networks products  
and services. When you use an ERC, your call is routed to a technical support  
person who specializes in supporting that product or service. To locate an ERC for  
your product or service, go to the http://www.nortelnetworks.com/help/contact/  
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1
Chapter 1  
Getting Started  
Topics in this section:  
“SIP Provisioning Client overview” on page 1  
“SIP Provisioning Client features” on page 2  
“System requirements” on page 2  
“Using the online help” on page 5  
“Setting up time zones” on page 5  
“Changing the admin password” on page 9  
“Exiting the SIP Provisioning Client” on page 9  
SIP Provisioning Client overview  
The SIP Provisioning Client is a web-based GUI that you use to perform the  
following administrative and provisioning tasks:  
define administration roles and privileges  
manage (list add, modify, or delete) users and devices  
define multiple service packages  
assign services to service packages  
assign service packages to domains  
assign translation information specific to a domain or subdomain  
provision voice mail servers  
provision 911 emergency notification  
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Chapter 1 Getting Started  
2
SIP Provisioning Client features  
The SIP Provisioning Client gives service providers the capability to create and  
set security, access, and permissions for the following components and roles of the  
user framework:  
administration  
domains and subdomains (users, devices, telephony routes)  
gateways  
IPCM clusters  
voice mail servers  
services  
system  
Note: The provisioning capabilities visible to an administrator in the SIP  
Provisioning Client menu depend on the level of authority (rights)  
assigned to their account.  
System requirements  
The following section describes the minimum and recommended hardware and  
software requirements for the SIP Provisioning Client.  
Minimum hardware and software requirements  
200MHz Pentium class or equivalent processor  
No additional free RAM is required. (The minimum memory requirements of  
your browser must be met. See your browser documentation for more  
information.)  
No additional harddrive space is required. (Local caching of web pages is  
performed by your web browser. See your browser documentation for  
information on the minimum disk space requirements.)  
Mouse (optional)  
640x480 @8bpp (256 colors) VGA graphics card  
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Chapter 1 Getting Started  
3
Microsoft* Windows* 98 Second Edition (SE), Microsoft Windows  
Millennium Edition (Me), Microsoft Windows 2000, Microsoft XP, or  
Microsoft Windows NT* 4.0 with Service Pack 5 (SP5)  
28.8Kbps modem  
Netscape* Communicator 4.77 or Internet Explorer 5.0  
Cookies and javascript enabled  
Recommended hardware and software requirements  
300MHz (or higher) Pentium class  
No additional free RAM is required. (The minimum memory requirements of  
your browser must be met. See your browser documentation for more  
information.)  
No additional harddrive space is required. (Local caching of web pages is  
performed by your web browser. See your browser documentation for more  
information on the minimum disk space requirements.)  
Mouse (required)  
800x600 or higher @16bpp (65,536 colors) VGA or better video graphics  
card  
Microsoft Windows 98 Second Edition (SE), Microsoft Windows 2000,  
Microsoft XP, or Microsoft Windows NT 4.0 with Service Pack 5 (SP5)  
56Kbps modem or other high speed connection (For example, cable modem,  
Digital Subscriber Line (DSL) modem, 10baseT ethernet, an so forth)  
Netscape Communicator 4.77 or greater, or Internet Explorer 5.0 or greater  
Cookies and javascript enabled  
Starting the SIP Provisioning Client  
To start the SIP Provisioning Client from your PC  
1
2
Start any supported web browser.  
Enter the Uniform Resource Locator (URL) for the SIP Provisioning Client.  
The SIP Provisioning Client Login window appears (Figure 1).  
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Figure 1 SIP Provisioning Client login window.  
3
4
Enter your username and password in the respective fields.  
Click LOG IN NOW to log in. Upon a successful login, the SIP Provisioning  
Client Welcome window appears.  
Figure 2 SIP Provisioning Client welcome window  
If the SIP Provisioning Client GUI is inactive for a period of time, your session  
will expire. The following message appears:  
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Figure 3 Session inactivity message  
To restart the SIP Provisioning Client, click on the link to login again. Contact  
your service provider to request a change to the session expiration time limit.  
Using the online help  
There are several ways that you can access SIP Provisioning Client help:  
online help - from the SIP Provisioning Client screen, click the Help button to  
view a task-based help system. The online help provides:  
— help pages containing forward and backward navigation icons  
— procedures that help you use the SIP Provisioning Client  
— a Table of Contents with hypertext links  
— an Index with hypertext links  
question mark help. Click the question mark icon to display a description and  
possible values for parameter fields on a form.  
Setting up time zones  
The SIP Provisioning Client allows you to define and manage as many time zones  
as you require to meet your needs, and to create your own time zones for easy  
reference. Time zones define the time zone from which a subscriber normally  
accesses the system (that is, the subscribers’s local time zone). Time zones are  
used in the SIP Personal Agent Call Manager function to set a screening rule that  
applies during a certain time. This time is dependent on the time zone selected.  
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The SIP Provisioning Client is initially set up with the standard 24 time zones:  
plus or minus 12 hours from the Greenwich Mean Time (GMT) format. However,  
there are more than 300+ time zones. What defines a time zone is not just the  
offset from the GMT. It is also whether or not the time zone uses Daylight Saving  
Time, when Daylight Saving Time occurs, and what the offset is.  
Setting up times zones in the United States  
As an example, the time zones you would set up across the United States from  
East to West are listed in the following table:  
Zone  
Standard Time  
Daylight Saving Time  
Atlantic  
Eastern  
Central  
Mountain  
Pacific  
AST  
EST  
CST  
MST  
PST  
AKST  
HST  
ADT  
EDT  
CDT  
MDT  
PDT  
Alaska  
Hawaii  
AKDT  
Not Observed  
Selecting a time zone  
To select a time zone  
1
2
Click the System menu option.  
Click the Timezones form. The Timezone details window appears.  
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Figure 4 Time Zone details window  
3
4
Select a zone from the list of time zones.  
From the drop down list, select a time zone.  
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Click Save.  
5
Creating a new time zone  
To create a new time zone  
1
2
3
Click the System menu option.  
Click the Timezones form. The Timezone details window appears.  
In the Add new timezone box, enter a meaningful name for the time zone in  
the Name field. (This field is visible to subscribers.) Both the name and the  
time zone need to be unique.  
Figure 5 Add new time zone  
4
Click Save to save the new time zone. Click View details to see the list of  
time zones.  
Modifying or Deleting a time zone  
The SIP Provisioning Client is preloaded with a number of default times zones.  
You may wish to modify or delete a time zone to prevent the time zone from  
showing up in the drop down list that is available when creating or modifying a  
user.  
To modify or delete a time zone  
1
2
3
Click the Services menu option.  
Click the Timezones form. The Timezone details window appears.  
Select the time zone that you wish to delete. Click the Delete link next to any  
time zone that you do not need. In the confirmation dialog box, Click Ok to  
permanently delete the time zone or Cancel to abort the action.  
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Tip: You cannot modify or delete a time zone that is currently in use.  
Changing the admin password  
To change the admin password  
1
2
Click the System menu option.  
Click the icon for the Change Password form. The Change Admin Password  
window appears.  
3
4
5
6
Enter your new password.  
Confirm your new password.  
Enter your current password.  
Click Save. Your new password is activated upon your next login.  
Exiting the SIP Provisioning Client  
To close the SIP Provisioning Client window and exit the SIP Provisioning Client,  
click the Logout menu option.  
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11  
Chapter 2  
Topics in this section:  
“Introduction” on page 11  
“SIP Provisioning roles, rights, and tasks” on page 12  
“Understanding provisioning roles and rights” on page 15  
Introduction  
In the SIP Provisioning Client, a provisioning role is defined as a collection of  
access rights and privileges that allow an administrator to perform various  
provisioning tasks. As a System Administrator, you can create any number of  
administrator roles, depending on your particular needs. For example, you can add  
a provisioning role to allow an administrator to perform multiple tasks, however,  
giving them the rights to only a subset of everything available.  
After logging in to the SIP Provisioning Client, you will see a menu list on the left  
side of your screen. This list varies based on the privileges available to you. The  
types of tasks you can perform as an administrator is based on the scope of your  
defined role.  
Note: The provisioning abilities visible to an administrator in the SIP  
Provisioning Client GUI depends on the rights assigned to their  
provisioning role.  
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SIP Provisioning roles, rights, and tasks  
The following table provides an overview of the SIP Provisioning tasks.  
Table 1 SIP Provisioning tasks overview  
If you have the right  
for  
Use this Provisioning menu option...  
To perform these tasks...  
Full domain access  
Administer all current domains and  
future domains regardless of initial  
provisioning. This allows the  
administrator with this role to see all  
domains, versus other administrators  
that can see only a subset of  
domains.  
Domain management  
List, add, modify, or delete domains  
and subdomains.  
IPCM cluster  
provisioning  
Add, list, modify, or delete an IPCM or  
IPCM cluster. If you add an IPCM  
through the SIP Provisioning Client,  
the Management Console GUI will not  
know that the IPCM exists. It is  
recommended that you deploy an  
IPCM through the System  
Management Console.  
Service Package  
Creation  
Add, list, modify, or delete services  
and service packages to a particular  
domain or all domains. If you have the  
administrator privileges to add a  
domain or multiple domains, then you  
can add or create service packages  
within that domain.  
Device Management  
Add, list, modify, or delete i2004  
devices.  
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Table 1 SIP Provisioning tasks overview  
If you have the right  
for  
Use this Provisioning menu option...  
To perform these tasks...  
Telephony Route/Digit  
Manipulation  
Add, list, modify, or delete telephony  
routes  
Admin  
Add, list, modify, or delete  
administrators and roles.  
Gateway Management  
Add, list, modify, remove, reorder, new  
or existing gateways, routes, and  
trunk groups. When you deploy and  
provision a gateway through the  
management console, you must  
provision the gateway with the similar  
trunk groups and routes through the  
SIP Provisioning Client to ensure that  
the information is saved into the  
database that the SIP Application  
Module uses.  
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Table 1 SIP Provisioning tasks overview  
If you have the right  
for  
Use this Provisioning menu option...  
To perform these tasks...  
Voice Mail Management  
Add, list, modify, or remove new or  
existing voice mail servers. If voice  
mail is enabled in a service package,  
a customer can select from the  
following voice mail server types:  
SIP based - a forward to your  
voice mail system using SIP.  
Trunk based - a Primary Rate  
Interface (PRI) or Channel  
Associated Signalling (CAS) trunk  
is associated with the line.  
Line based - a particular phone  
number is associated to the line.  
User Management  
Add, list, modify, or delete users,  
aliases, blended aliases, and i2004  
Internet Telephone properties  
(available from the domain or  
subdomain menu option).  
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Understanding provisioning roles and rights  
Administrators can create, or define, all the necessary provisioning roles to  
support their system. They can allow or restrict provisioning roles to carry out  
specific actions. The following table provides examples of several administrator  
provisioning roles and the rights based restrictions associated with various  
provisioning tasks.  
Table 2 Example of roles and rights  
Provisioning  
Role example  
Rights given  
Allowed tasks  
User  
administrator  
User management with read,  
write, and delete access  
Can view domain details, and add,  
delete, or modify users. Does not  
have access to other parts of the  
system, for example voice mail,  
service packages, and so forth.  
Domain management with  
read access only  
Device  
administrator  
Domain management with  
read access  
Cannot add or modify users.  
Allowed to add, modify or delete  
devices.  
Device management with read,  
write, and delete access  
System  
Administrator  
Full domain access  
Can see all domains, regardless of  
who created the domain, or the list  
of domains provisioned against the  
administrator.  
Limiting access to a domain  
Complementing the rights-based restrictions, administrators can be provisioned  
against certain domains in order to limit access to within those domains. This  
prevents an administrator from domain xyz from viewing or modifying details in  
domain abc, even though they are hosted from the same system. Administrators  
can then delegate work for subdomains to other administrators without giving  
them access to their own data.  
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For example, if an administrator assigned to the User Administrator provisioning  
role was given access to suba.xyz (a subdomain of domain xyz), this administrator  
can add, modify, and delete users in the subdomain, but does not have access to  
the parent domain, domain xyz. This allows for the delegation of task on both the  
rights-level and the domain-level.  
Note: Access to any domain implies access to all of its descendant  
subdomains. For example, access to domain xyz implies access to  
suba.xyz. Full domain access will override this domain allocation and  
allow any administrator assigned with a provisioning role that includes  
this right, to see all domains regardless of what the administrator is  
provisioned against.  
System administrator provisioning role  
The System Administrator (SA) is the highest level administrator and is the only  
provisioning role that is preloaded when the SIP Provisioning Client is deployed.  
The SA has full access and rights to all provisioning features and functions.  
The SA can designate someone as an Administrative user with the Add Admin  
menu option, and grant specific administrator rights to that administrative user  
with the Add Roles menu option.  
Note: The System Administrator role is the only role that cannot be  
changed.  
Creating a new provisioning role  
To create a new provisioning role  
1
2
Click the Admins menu option.  
Click the Add Roles sub menu option. The Add a New Role window appears  
(Figure 6).  
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Figure 6 Add a New Role window  
3
4
5
Enter a name for the provisioning role in the Role Name field (required).  
Enter a description for the provisioning role in the Role Description field.  
To define the level of administrator access, select the checkboxes for Read,  
Write, or Delete next to each privilege.  
6
Click Add. A message appears that the role has been successfully created.  
Listing provisioning roles  
To view details about a provisioning role to modify or delete the role  
Click the Admins menu option.  
1
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2
Click the List Roles sub menu option. The List System Roles window appears  
(Figure 7).  
Figure 7 List System Roles window  
3
To view details about a provisioning role or description, click the View  
Details link. The Role Details window appears (Figure 8). To delete a  
provisioning role, click the Delete link next to the role.  
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Figure 8 Role details window  
4
5
To change the provisioning role description, enter a new description in the  
Role Description box. To modify privileges, select or deselect privileges in  
the checkbox next to the privilege.  
Click Save to save your changes. A message appears that the provisioning  
role was modified successfully.  
Creating a new administrator  
To add an administrator  
1
Click the Admins menu option.  
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2
Click the Add Admin sub menu option. The Create new admin window  
appears (Figure 9).  
Figure 9 Create new admin  
3
4
Enter the username of the administrator (required) and other information in  
the fields provided.  
Click Add. A message appears that the administrator was successfully added.  
Listing administrators  
To view details about an administrator or to modify or delete an administrator  
1
Click the Admins menu option.  
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2
Click the List Admins sub menu option. The Admins window appears  
(Figure 10). To delete an administrator, click the Delete link next to the  
administrator username.  
Figure 10 Admins window  
3
To modify administrator information, enter the changes in the fields provided  
(Table 11).  
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Figure 11 Admin Details window  
4
Click Save to save your changes. A message appears that the administrator  
was modified successfully.  
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Chapter 3  
Topics in this section:  
“Introduction to domains” on page 23  
“Domain types” on page 24  
“Adding a local root domain” on page 26  
“Adding a subdomain” on page 31  
“Adding a foreign domain” on page 33  
“Listing/modifying a local or foreign domain” on page 33  
“Deleting a local domain” on page 35  
“Banning a user” on page 36  
“Defining user device status reasons” on page 38  
Introduction to domains  
A domain is used as a control mechanism of users, services, devices and  
translation. The SIP Provisioning Client is used to provision a domain. The  
provisioning of domains allows service providers to define a domain and service  
packages available for the domain.  
Once the domain and service packages are defined, service providers can define  
subdomains and assign the following:  
services to service packages  
service packages to domain and users  
translation information specific to a domain or a subdomain  
devices for users to an IPCM (IP Client Manager)  
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If you have full domain access, you can see all provisioned domains. Otherwise,  
you will only be able to see the domains/subdomains you are provisioned against.  
The management rights assigned to a domain administrator dictates what you can  
do with the domains.  
Domain types  
There are three types of domains:  
root domain  
subdomain  
foreign domain  
Root domain  
A root domain is the highest level domain. A service provider can create a root  
domain for each of their customers. An example of a root domain is  
nortelnetworks.com.  
Subdomain  
A subdomain is a category of domains. Subdomains allow you to divide  
subscribers under a domain into smaller groups. The groups can be used to control  
routing and access to services by members of the group.  
Under each subdomain, you can add subdomains, service packages, users,  
devices, and telephony routes. There are no limits to the levels of nesting of  
subdomains. Each subdomain inherits the parameter values from the root domain.  
A typical application of subdomains is based on geographical divisions. For  
example, using Nortel Networks, you could have nortelnetworks.com as the root  
domain, with subdomains of Richardson and Ottawa. Under each of these  
subdomains there could be further subdomains as shown in the following  
illustration (Figure 12).  
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Figure 12 Domains and subdomains  
Foreign domain  
A foreign domain is a domain that is not served by any SIP Application Modules  
in your system. When adding a domain, you must determine whether or not the  
domain is a foreign domain or not. A foreign domain uses foreign routes, while a  
local domain uses SIP routing or telephony routes.  
A domain that is not local is considered foreign. Usually, a foreign domain is  
resolved through DNS and would not need to be provisioned. However, since  
some deployment scenarios may not involve a DNS server, you (as the system  
administrator) can datafill routes for a foreign domain.  
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For example, if the SIP Application Module serving nortelnetworks.com does not  
have a DNS server, and there is a need for users to be able to make calls to the  
xyz.com domain, then you could add the abc.com domain as a foreign domain and  
provide a route to abc’s proxy.  
Note: A foreign domain is not controlled by the system. Therefore it  
will not be displayed under the domain folder in the menu area, and the  
you will not be able to provision users, service packages, devices, and so  
forth to these domains.  
Adding a local root domain  
To add a local root domain  
1
2
Click the Domains menu option.  
Click Add Domain. The Create new domain window appears (Figure 13).  
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Chapter 3 Domain Management 27  
Figure 13 Create new domain window  
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3
Enter the name of the new local domain. The domain name must not be more  
then 60 chars in length and cannot contain the following symbols or  
characters as described in the following table:  
Table 3 Create new domain name restrictions  
$
( )  
_
|
;
~
!
%
^
&
*
[ ]  
{ }  
/
?
@
#
+
=
,
:
< >  
4
Select No. There may be times when you want two domains on different  
systems to communicate with each other. If Yes, enter the URL destination (IP  
address) of where the domain is located in the Foreign Dest: field.  
Tip: Domain parameters are not necessary for foreign domains. When  
adding a foreign domain, the parameters can be ignored.  
5
Enter parameter information in the fields provided. See the following table for  
parameter values and descriptions (Table 4).  
Table 4 Create new domain parameters  
Parameter  
Type  
Range  
Default Description  
i2004 Locale  
String;  
Restricted  
N/A  
Indicates the language in which the i2004  
select from to English  
drop-down in the  
Internet Telephone will display. Future  
releases will include more supported  
languages.  
list  
current  
release.  
Location  
String  
0-50  
characters  
N/A  
N/A  
Text describing location of device. A specific  
format of this string is scheduled for a future  
release.  
Idle Display  
String  
Up to 30  
Idle telephone display heading.  
characters  
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Table 4 Create new domain parameters  
Parameter  
Type  
Range  
Default Description  
WD Phone Timer  
Integer  
13-10000  
150  
Length of time the i2004 Internet Telephone  
waits before determining that is has lost  
connection to the IPCM. Should be greater  
than WD_Timer by X amount. Where X is  
difference between MAX expected network  
latency and MIN expected network.  
WD Timer  
Time FMT  
Integer  
30-10000  
15  
0
How often (in seconds) the IPCM will send  
a “reset watchdog timer” message to a  
connected I2004 device. When the I2004  
receives this watch dog message, it knows  
that it is still connected to the IPCM.  
Select from 0,1,2  
drop down  
list:  
The time format of the proxy used as default  
for the i2004 Internet telephone 0-> 12 hour  
clock  
12-hour  
French  
24-hour  
Date FMT  
Numeric  
Standard  
MM/DD or  
Numeric  
Inverse:  
DD/MM  
2,3  
2
2-numeric standard (09/16); 3-numeric  
inverse (16/09).  
PSEIZ Timer  
PDIL Timer  
Integer  
N/A  
N/A  
15  
4
Time-out for first dialed digit.  
Interdigit time-out (seconds).  
Integer  
(seconds)  
Alpha  
Boolean  
True/False False  
Enables alphanumeric dialing as the default  
dialing style. If disabled, I2004 defaults  
dialing style to numeric.  
Default Proxy Port  
5060  
Any valid  
port  
5060  
Default proxy SIP port.  
No validation is done on this field to ensure  
validity.  
Behind Firewall  
Allow All Codecs  
Boolean  
Boolean  
True/False True  
True/False False  
Boolean indicating whether there is a  
firewall between the IPCM and the i2004  
Internet Telephone.  
Allow all codecs (true) or only selected  
codec (false). See Vocoder parameter  
(below).  
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Table 4 Create new domain parameters  
Parameter  
Type  
Range  
Default Description  
Active Proxy  
Boolean  
True/False True  
Determines if default proxy is used. Used  
for SIP messaging. If yes, calls will route  
through the default proxy. No other routing  
is currently supported for the IPCM.  
Default Proxy Host  
Time Zone  
String (30) Any valid IP N/A  
address  
IP Address of proxy used as default for the  
i2004 telephone. No validation is done on  
this field to ensure validity.  
Integer;  
Up to 30  
N/A  
Time zone of the user.  
select from characters  
drop down  
list  
Vocoder:PacketTime Integer;  
select from  
0,4,8,10,18  
0 - 15  
0
Default Codec Setting: 0 – G711 Mulaw; 4 –  
G723\6.3kbps; 8 – G711\Alaw; 10 – L16; 18  
– G729A.  
drop down  
list  
Contrast  
Integer;  
select from  
drop down  
list  
8
i2004 display contrast setting.  
Server Home  
String(60)  
Up to 60  
characters  
N/A  
Specifies the SIP Application Module that  
will host the users in this domain. This is  
used in an N+M environment. All SIP  
Application Modules that are not the home  
SIP Application Module will forward  
requests for users in this domain to the  
specified address as appropriate.  
(Example: 10.0.0.8) If no host address is  
supplied, or the host address is 0.0.0.0., no  
homing is performed.  
Realm for a domain String (120) Up to 120  
characters  
Realm  
Identifies the domain for subscribers when  
they are being authenticated. Subscribers  
may register in multiple domains. When  
they are required to enter a password  
during authentication, they need to know  
which domain it is so that they can enter the  
appropriate password.  
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Chapter 3 Domain Management 31  
Table 4 Create new domain parameters  
Parameter  
Type  
Range  
Default Description  
Always Use Media  
Portal  
String  
True/False False  
Directs the Application Server to use the  
RTP Media Portal when set to true. The  
purpose is to use the RTP Media Portal  
function in dealing with special SIP  
scenarios. For example, a domain that  
straddles multiple sites contains users who  
do not have Internet Protocol (IP)  
connectivity between them due to firewalls  
at different sites. This domain requires the  
Always Use Media Portal to be set to True  
to negotiate between the firewalls to set up  
SIP sessions.  
Maximum Number of Integer  
Presence  
Subscriptions  
N/A  
0
Maximum number of inbound subscriptions  
to any given user in this domain.The SIP  
Personal Agent, the SIP Multimedia PC  
Client, and the SIP Multimedia Web Client  
will use this limit when allowing a particular  
username to be added to a subscriber's list  
of buddies.  
Accepted  
6
Click Add.  
Tip: In order to see the new root domain added to the list of domains,  
you must close and reopen the domains folder to refresh the list of  
domains.  
Adding a subdomain  
To add a subdomain  
1
2
Click the Domains menu option  
Under the root domain where you want to add a subdomain, click the  
Sub-Domains > Add Sub Domain menu option. The Create new subdomain  
window appears with a list of parameters inherited from the parent domain  
(Figure 14).  
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Figure 14 Create new subdomain window  
3
4
Enter a name for the subdomain.  
Select a COS (Class of Service) for the subdomain from the drop down list of  
routing services.  
5
Click Save.  
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Adding a foreign domain  
To add a foreign domain  
1
Click the Domain > Add Domain menu option. The Create new domain  
window appears (Figure 15). Since parameter fields are not used for foreign  
domains, only the fields at the top of the window need to be completed.  
Figure 15 Create new domain window  
2
3
4
Enter a name for the foreign domain.  
Select Yes from the Foreign drop down box.  
Add a foreign destination into the Foreign Destination box using the  
following format  
SIP<IP_Addr>:port  
5
Click Add.  
Listing/modifying a local or foreign domain  
To list/modify a local or foreign domain  
1
Click the Domains > List Foreign Domains menu option. The Domains  
window appears (Figure 16).  
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Figure 16 List Domains window - step 1  
2
From the Criteria Selection drop down list, select one of the following as a  
search criteria:  
— Domain Name  
— Is a Foreign domain  
— ID  
or enter the name of the domain, foreign domain, or ID in the criteria selection  
text box.  
3
Click Add Criteria. The search criteria you selected appears in the lower  
window and is used to refine your search.  
4
5
Repeat steps 1 -3 to list additional domains.  
In the lower window area, select whether to search by All or Any of the  
criteria specified. Note that you do not have to add any criteria if you want to  
see all the domains in the network.  
6
Click Search to begin the search, or Remove to remove a search criteria from  
the search list. A list of domains (based on the search criteria specified)  
appears (Figure 17).  
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Chapter 3 Domain Management 35  
Figure 17 List domains - step 2  
7
In the Domains window, select a domain and click the Details link. The  
details for the selected domain, or domains, displays.  
8
9
Modify domain information in the parameter fields.  
Click Save.  
Deleting a local domain  
Caution: Exercise extreme caution when deleting a domain as this will  
permanently remove all users, devices, routes, service packages, and so  
forth, associated with the domain. This is an irreversible operation that  
could be a service-affecting change. For example, any calls in progress to  
users or devices in the domain will terminate.  
To delete a local domain or subdomain  
1
2
Click the Domains menu option.  
Click the domain or subdomain you want to delete from the list of domains.  
Tip: In order to delete a domain or subdomain you must first delete all of  
its children subdomains.  
3
4
Click Delete to delete the domain or subdomain you want to delete.  
Click OK in the confirmation dialog box to delete the domain, or Cancel to  
cancel the delete operation.  
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Deleting a foreign domain  
Caution: Exercise extreme caution when deleting a foreign domain as  
this will permanently remove all users, devices, routes, service packages,  
and so forth, associated with the domain.This is an irreversible operation  
that could be a service affecting change. For example, any calls in  
progress to users or devices in the domain will terminate.  
To delete a foreign domain  
1
2
3
Click the Domains > List Foreign Domains menu option.  
Click the Delete link for the foreign domain you want to delete.  
Click OK in the confirmation dialog box to delete the foreign domain, or  
Cancel to cancel the delete operation.  
Banning a user  
You can ban an individual user or a group of users in a domain or from viewing  
the presence information (online status) of a user or group of users in a domain.  
The banning takes effect for all subdomains under the domain.  
Setting up ban list  
To ban a user or a group of users  
1
2
Click the Domains menu option.  
Under the domain where you want to ban a user or users, click the Banned  
Users > Ban User menu option. The Add a party to Domain Ban List window  
appears (Figure 18).  
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Chapter 3 Domain Management 37  
Figure 18 Add a party to Domain Ban List window  
3
Enter the party to ban in the text boxes provided. To ban an individual user,  
use the following format:  
<username>@<domain_name.com>  
To ban a group of users in a domain, use the following wildcard format:  
<*>@<domain_name.com>  
4
5
Enter a reason for banning the user or group of users to the domain list in the  
Description text box.  
Click Save. A message displays that the party was successfully added to the  
ban list.  
Viewing a list of banned users  
To view a list of users banned from viewing the presence information (online  
status) of a user or a group of users in a domain  
1
2
Click the Domains menu option.  
Under domain where you want to view a list of banned users, click the  
Banned Users > List Banned User menu option. The Domain Ban List  
window appears (Figure 19).  
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Figure 19 Domain Ban List window  
3
4
To delete a user or a group of users from the ban list, click the Delete link next  
to the party to ban.  
Click OK at the delete confirmation dialog box, or Cancel to cancel the delete  
operation.  
Defining user device status reasons  
The Status Reason menu option allows you to define a list of reasons for an  
inactive user device status.  
When a user is inactive, the user cannot register to the SIP Application Module.  
This means that the user cannot use any of the services provided by the SIP  
Application Module, including making calls or subscribing to other users'  
presence. However, the inactive user can log in to the SIP Personal Agent and  
view the reason for their INACTIVE status.  
Adding a status reason  
To define a status reason for a user device  
1
2
Click the Domains menu option.  
Under the domain where you want to define a status reason, click Status  
Reason > Add Reason. The Add Status Reason window appears (Figure 20).  
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Chapter 3 Domain Management 39  
Figure 20 Add Status Reason window  
3
4
5
From the Status drop down list, select INACTIVE.  
In the Name field (required) enter a name to identify the status reason.  
In the Description field, enter a description for the inactive status. For  
example, if a user has not paid their bill, the reason for the inactive status  
could be “Inactive for at least 6 months”.  
6
7
In the URL field, enter a web page address. This is the web page that the  
originator of a call to the user will be presented with, along with the status  
reason description, to explain the called user’s inactive status.  
Click Save. A message displays that the status reason was successfully added.  
Viewing status reasons  
To view a list of status reasons for a user device status  
1
2
Click the Domains menu option.  
Under the domain where you want to view the status reasons, click the Status  
Reason > List Reasons menu option. The List Reasons window appears  
(Figure 21).  
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Figure 21 Status Reasons window  
3
To view or modify status reason information, click the Details link for  
information about the status reason. The Modify Status Reason window  
appears (Figure 22).  
Figure 22 Modify status reason window  
4
5
Enter changes in the fields provided to modify information in the status  
reasons fields.  
Click Save. A message displays that the status reason was successfully  
modified.  
6
7
To delete a status reason, click the Delete link.  
At the delete confirmation dialog box, click OK to delete the status reason, or  
Cancel to return to the Status Reasons details window.  
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41  
Chapter 4  
IPCMs  
Note: You must have IPCM provisioning rights to perform the tasks in  
this section.  
Topics in this section:  
“What is an IPCM?” on page 41  
“Adding an IPCM cluster” on page 42  
“Assigning and modifying IPCMs to a domain” on page 44  
“Deleting an IPCM cluster” on page 46  
“Listing physical IPCMs” on page 47  
What is an IPCM?  
The IP Client Manager (IPCM) is a network application capable of managing  
many internal i2004 Internet Telephones while maintaining the call state and  
phone settings. The IPCM also implements features (for example, stock queries  
and personal directory) on the i2004 Internet Telephones. An IPCM Cluster may  
span multiple domains and/or contain multiple domains.  
Note: Users and devices are linked to IPCM Clusters, and not to  
individual IPCMs. An IPCM Cluster may contain a single IPCM or a pair  
of IPCMs. The IPCM cluster contains a primary and standby IPCM for  
each i2004 Internet Telephone. Clustering of IPCMs ensures that each  
IPCM retrieves the same user and device data from the database and that  
each receives notification of database modifications.  
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In order for the IPCM to manage i2004 Internet Telephones, information  
pertaining to the IPCM in general, the i2004 telephones, and users that will use  
i2004s, must be provisioned. For more information, see the SIP IP Client  
Manager Basics documentation.  
i2004 Internet Telephone information can be provisioned in the following ways:  
Provision each individual i2004 using the SIP Provisioning Client.  
Set up default device properties for a domain, assign an activation key to the  
domain, and then use the IPCM to auto-provision each i2004 device whenever  
a user enters the desired activation key. (See the SIP IP Client Manager  
Basics documentation and the i2004 Internet Telephone User Guide for more  
information.)  
Note: Normally, the IPCM is deployed prior to provisioning the IPCM.  
If this is the case, it is still necessary to assign domains to the IPCM  
cluster and to assign a provisioned capacity for the IPCM cluster and  
domain. If the IPCM has been deployed, see “Assigning and modifying  
IPCMs to a domain.” If the IPCM has not been deployed, and it is  
desired to pre-provision the IPCM, see “Adding an IPCM cluster.”  
Adding an IPCM cluster  
Adding an IPCM cluster creates a virtual IPCM to which users, i2004 devices, and  
domains may be assigned. It does not actually create a physical IPCM, nor does it  
deploy IPCM software onto a server. The association of the physical IPCM to this  
IPCM cluster is performed when the physical IPCM is deployed and configured.  
To add an IPCM cluster  
1
Click the IPCM Cluster > Add IPCM Cluster menu option. The Create new  
IPCM window displays. The following figure (Figure 23) shows an example  
of creating an IPCM named LAB IPCM. The maximum provisioned capacity  
is 1000, i2004 Internet Telephones. Note that the IPCM name is case  
sensitive.  
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Chapter 4 IPCMs 43  
Figure 23 Create new IPCM window  
The following table (Table 5) lists the Create new IPCM parameters and values to  
use when creating a new IPCM.  
Table 5 Create new IPCM parameters  
Parameter  
Type  
Range  
Default Description  
IPCM Name  
String(100) Up to 100  
characters  
None  
The name of this IPCM.  
The name of the IPCM  
must be a unique name.  
Prov. Capacity  
Integer  
Maximum  
of 10,000  
per IPCM  
None  
The number of i2004  
Internet Telephones that  
are allowed to be  
provisioned for this IPCM  
(across all domains).  
2
Click Save.  
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Assigning and modifying IPCMs to a domain  
An IPCM is not usable until it is assigned to a domain. To assign an IPCM to a  
domain  
1
Click the IPCM Clusters > List IPCM Clusters menu option. A list of  
deployed IPCMs appears (Figure 24).  
Note: Upon first assignment of an IPCM, the list of deployed IPCMS  
may be empty. However, in many cases, the IPCM would have already  
been deployed using the System Management Console and will appear in  
the list.  
Figure 24 List IPCM Clusters window  
2
Click the Details link for the IPCM you want to assign to a domain. The  
Modify IPCM window appears (Figure 25).  
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Chapter 4 IPCMs 45  
Figure 25 Modify IPCM window  
3
4
If necessary, modify the IPCM Name or Prov. Capacity parameters and click  
Save.  
Tip: The Avail Capacity field is a read only field calculated by adding  
the Assigned Capacity of domains to which this IPCM is assigned and  
subtracting that value from the Prov. Capacity of the IPCM.  
Select a domain from the list of available domains in the Assign New Domain  
area of the window. Enter information in the following fields for the domain  
you are assigning to the IPCM:  
Field  
Assigned Capacity:  
Description  
The number of i2004 Internet Telephones you want  
to allow in this assigned domain.  
Activation Key:  
A unique string (number) per IPCM/domain  
combination that associates default device  
properties with this particular domain. The  
administrator must pick a unique string (number) for  
each domain. If the IPCM is used to auto-provision  
i2004 devices, the installer will be prompted for this  
activation key during the installation process.  
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5
Click Save. The IPCM Domain Relation window appears (Figure 26).  
Figure 26 IPCM Domain Relation window  
6
7
Enter your changes and click Save. A confirmation messages displays  
confirming your changes.  
To assign another domain to the IPCM, select a domain from the list in the  
Assign New Domain area. Enter information in the parameter fields and click  
Save.  
Deleting an IPCM cluster  
Caution: Exercise extreme caution when deleting an IPCM as this will  
permanently remove all i2004 Internet Telephones and users' i2004  
capabilities associated with the IPCM that you are deleting.  
To delete an IPCM cluster  
1
Click the IPCM Clusters > List IPCM Clusters menu option.  
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Chapter 4 IPCMs 47  
2
Click the Delete link for the IPCM cluster you want to delete.  
Click OK in the confirmation dialog box to delete the IPCM cluster, or  
Cancel to cancel the delete operation (Figure 27).  
Figure 27 IPCM Delete confirmation dialog box  
3
Listing physical IPCMs  
The List Physical IPCMs menu option allows you to view the physical IP Address  
of the server on which the IPCM resides.  
To list the physical IP address of the IPCM  
1
Click the IPCM Clusters > List Physical IPCMs menu option. The Physical  
IPCMs window appears (Figure 28) listing the following fields:  
Field  
IPCM Cluster:  
Description  
The name that identifies the virtual IPCM  
cluster.  
Physical IPCM:  
The physical IP Address of the server on  
which the IPCM resides.  
Delete Physical IPCM:  
Allows you to delete the mapping of the  
physical IPCM to the IPCM Cluster.  
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Figure 28 Listing physical IPCMs  
2
To delete a physical IPCM, click the Delete link in the Delete Physical IPCM  
column. At the delete confirmation dialog box, click Ok to delete the physical  
IPCM or Cancel to cancel the delete operation (Figure 29).  
Figure 29 Delete physical IPCM confirmation dialog box  
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Chapter 5  
Voice Mail Servers  
Note: You must have Voice Mail provisioning rights to perform the tasks  
in this section.  
Topics in this section:  
“Introduction” on page 49  
Voice mail server types” on page 50  
“Provisioning a SIP-based voice mail server” on page 50  
“Provisioning a Trunk-based voice mail server” on page 54  
“Provisioning a Line-based voice mail server” on page 60  
“Assigning a user to a provisioned voice mail server” on page 67  
“Listing/modifying/deleting a voice mail server” on page 68  
“Using CPL scripts with voice mail” on page 69  
Introduction  
The SIP Provisioning Client allows you to provision third-party voice mail server  
information to:  
set up and route calls to voice mail  
associate users to a voice mail server  
set up a voice mail server to receive message waiting indication (MWI)  
notifications  
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Voice mail server types  
The following voice mail server types are supported as described in the following  
table (Table 6).  
Table 6 Voice mail server types  
Voice mail server type  
Description  
SIP-based  
An IP based voice mail server that uses the SIP protocol.  
This server type does not need a PSTN gateway.  
Line-based  
A voice mail server with multiple lines, each identified by  
a 10 digit phone number that goes into the voice mail  
server. When people leave messages, the individual  
lines are used in a round robin manner to connect to the  
voice mail server to allow someone to leave a message.  
The SIP Application Module sends an SMDI message to  
the voice mail server to help establish the connection to  
the SIP Application Module that uses SMDI to obtain  
Message Waiting Indication (MWI) information from the  
voice mail server.  
Trunk-based  
A legacy Public Switched Telephone Network (PSTN)  
based voice mail server that uses a Primary Rate  
Interface (PRI) or Channel Associated Signalling (CAS)  
gateway.The SIP Application Module uses SMDI to send  
call setup information using a Call Detail Message to get  
Message Waiting Indication (MWI) information from the  
voice mail server.  
Provisioning a SIP-based voice mail server  
Provisioning a SIP-based voice mail server allows the administrator to configure  
the SIP Application Module to communicate with a voice mail server using the  
SIP protocol.  
The SIP Application Module is given information on what to query to find the  
location of the voice mail server. The SIP Application Module will use this  
information to send SIP calls to the voice mail server when calls are routed to  
voice mail.  
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Adding a SIP-based voice mail server  
To add a SIP-based voice mail server  
1
Click the Voice Mail > Add Voice Mail Server menu option. The Create new  
voice mail server window appears (Figure 30).  
Figure 30 Create new voice mail server window (SIP) - step  
2
Choose SIP-Based Server from the drop down and click Next. The second  
page of the Create new voice mail server appears (Figure 31).  
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Figure 31 Create new voice mail server window (SIP) - step 2  
3
Enter information about the SIP-based voice mail server in the parameter  
fields. The following table (Table 7) lists the SIP-based voice mail  
parameters.  
Table 7 SIP-based voice mail parameters  
Parameter  
Type  
Range  
Default  
Description  
Description  
String(100) Up to 100  
characters  
None  
The description for the server that  
identifies the SIP-based voice  
mail server.  
URI to query to find String(100) Up to 100  
None  
The SIP address of the user that  
the SIP Application Module will  
look up to find the address of the  
voice mail server.  
Voice Mail Server  
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Chapter 5 Voice Mail Servers 53  
Table 7 SIP-based voice mail parameters  
Parameter  
Type  
Range  
Default  
Description  
Client contact  
String(100) Up to 100  
characters  
Voice mail  
The field used to route voice mail.  
In most cases, you will use the  
default (voice mail). If you require  
a different client contact, contact  
your Nortel Networks support  
representative.  
Application Server  
String(60)  
Up to 60  
characters this default, each  
SIP Application  
ServerHome. With  
Specifies the SIP Application  
Module will host the voice mail  
server being provisioned. This is  
used in an N+M environment. All  
Module will ignore  
this field and forward SIP Application Modules that are  
SIP requests to the  
voice mail server  
not the home SIP Application  
Module will forward requests  
instead of forwarding destined to voice mail to the  
them to a "home"  
address specified in this field.  
Application Server.  
Domains  
String(60)  
characters  
Up to 60  
characters  
None  
The domains that will use this  
voice mail server. Use the  
Ctrl-click key to select multiple  
domains to use the same voice  
mail server.  
4
Click Save. The SIP-based voice mail server you added now appears in a list  
of voice mail servers (Figure 32).  
Figure 32 Voice Mail Servers window (SIP)  
Adding a phantom voice mail server user  
After you add a voice mail server, you then need to create a phantom user named  
voicemail for the domain that will use the SIP-based voice mail server.  
To create a phantom user for the SIP-based voice mail server  
1
Click the Domains menu option.  
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54 Chapter 5 Voice Mail Servers  
2
3
4
5
Under the domain where you want to add a phantom user, click the Add User  
menu option.  
In the Add User form, enter the text "voice mail" for the User Name, Last  
Name, First Name, fields. In the password field, enter any password.  
Click Save. The User details screen displays a message that the user was  
added successfully.  
At the top of the window, click the Routes link to specify the route to the  
voice mail server. Enter the SIP address that the voice mail server is  
expecting, along with the domain and IP address (Figure 33).  
Figure 33 User routes window  
6
7
Click Add.  
Click Save.  
Note: For CAS/CallPilot, it is recommended that you do not create a  
phantom user.  
Provisioning a Trunk-based voice mail server  
Provisioning a Trunk-based voice mail server allows the administrator to  
configure the SIP Application Module so that it can communicate with a voice  
mail server using a PRI or CAS gateway.  
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Chapter 5 Voice Mail Servers 55  
The SIP Application Module is given information on what to query to find the  
correct gateway to send SIP calls to the voice mail server when calls are routed to  
voice mail. An SMDI TELNET session is configured to allow the SIP Application  
Module to send call setup information using a SMDI Call Detail Message to  
receive SMDI MWI messages.  
Adding a Trunk-based voice mail server  
To add a Trunk-based voice mail server  
1
Click the Voice Mail > Add Voice Mail Server menu option. The Create new  
voice mail server window appears (Figure 34).  
Figure 34 Create new voice mail server window (Trunk-based) - step 1  
2
Choose Trunk-Based Server from the drop down and click Next. The second  
page of the Create new voice mail server appears (Figure 35).  
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Figure 35 Create new voice mail server window (Trunk-based) - step 2  
3
Enter SMDI parameter information for the Trunk-based voice mail server.  
The following table (Table 8) lists the SMDI parameters.  
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Chapter 5 Voice Mail Servers 57  
Table 8 SMDI parameters  
Parameter  
Type  
Range  
Default Description  
Address  
String(100) Up to 100  
characters  
N/A  
The IP address of the  
terminal (SMDI) server.  
Port  
Integer  
None  
None  
None  
The port for the SMDI  
server to TELNET to.  
SMDI Version  
Float  
None  
The SMDI version. There  
are only 2 versions of the  
SMDI protocol (Version 1  
and Version 2). Version 2  
is the latest draft and is  
supported by most voice  
mail servers.  
Username  
Password  
String(60)  
String(20)  
Up to 60  
characters  
N/A  
N/A  
The TELNET username  
for logging in to the SMDI  
TELNET server. Used by  
the SIP Application  
Module to login in to the  
terminal server.  
Up to 20  
characters  
The TELNET password  
for logging in to the SMDI  
TELNET server.  
4
Click Next to enter information about the gateway connection. The Create  
new voice mail server window appears (Figure 36).  
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Figure 36 Create new voice mail server window (Trunk-based) - step 3  
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5
Enter information about the Trunk-based voice mail server gateway  
connection in the parameter fields. The following table (Table 9) lists the  
gateway connection parameters.  
Table 9 PRI Gateway connection parameters  
Parameter  
Type  
Range  
Default  
Description  
Description  
String(100)  
Up to 100  
characters  
None  
The description for the server that  
identifies the Trunk-based voice  
mail server.  
URI to query to String(100)  
find gateway  
Up to 100  
characters  
None  
For PRI Gateway: The address (in  
either 10 digit phone number and  
domain name format or in SIP:voice  
mail@nortelnetworks.com format)  
of the machine that will host this  
voice mail server. The SIP  
Application Module performs  
translations on the phone number  
and domain to understand which  
gateway to use. You can change the  
translations to use a different  
gateway, and do not have to change  
this field.  
For CAS Gateway (CallPilot): The  
SIP address for the CAS Gateway  
that routes to the CallPilot. The  
address is made up of a DN that the  
gateway routes to the CallPilot, and  
an maddr specification for the CAS  
Gateway. (For example,  
sip:66250@nortelneworks.com;ma  
ddr=46.10.11.219).  
Type of Trunk Drop List  
Client contact String(100)  
NA  
PRI_TRUNK  
Voice mail  
A specification of whether the voice  
mail Gateway is PRI or CAS.  
Up to 100  
characters  
The field used to route voice mail. In  
most cases, you will use the default  
(voice mail). If you require a  
different client contact, contact your  
Nortel Networks support  
representative.  
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Table 9 PRI Gateway connection parameters  
Parameter  
Type  
Range  
Default  
Description  
Application  
Server  
String(60)  
Up to 60  
characters  
ServerHome. With  
this default, each SIP Module will host the voice mail  
Specifies the SIP Application  
Application Module  
will ignore this field  
and forward SIP  
server being provisioned. This is  
used in an N+M environment. All  
SIP Application Modules that are  
requests to the voice not the home SIP Application  
mail server instead of Module will forward requests  
forwarding them to a destined to voice mail to the  
"home" Application  
address specified in this field.  
Server.  
Domain name String(60)  
Up to 60  
characters  
None  
The domains that will use this  
Trunk-based voice mail server.Use  
the Ctrl-click key to select multiple  
domains to use the same voice mail  
server.  
6
Click Save. The Trunk-based voice mail server you added now appears in the  
list of voice mail servers (Figure 37).  
Figure 37 Voice Mail Servers window (Trunk-based)  
Provisioning a Line-based voice mail server  
Provisioning a Line-based voice mail server allows the administrator to configure  
the application so that it can communicate with a voice mail server using a Lines  
Gateway. The SIP Application Module is given information on which Lines  
Gateway to use and which line numbers to use for sending calls to the gateway  
when routing a call to voice mail.  
In addition, an SMDI TELNET session is configured to allow the SIP Application  
Module to directly communicate call setup information to the voice mail server.  
The SMDI link is also used by the SIP Application Module to receive SMDI MWI  
messages.  
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Adding a Line-based voice mail server  
To add a Line-based voice mail server  
1
Click the Voice Mail > Add Voicemail Server menu option. The Create new  
voice mail server window appears (Figure 38).  
Figure 38 Create a new voice mail server window (Line-based) - step 1  
2
Choose Line-Based Server from the drop down and click Next. The second  
window of the Create new voice mail server appears (Figure 39).  
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Figure 39 Create new voice mail server window (Line-based) - step 2  
3
Enter SMDI parameter information about the Line-based voice mail server.  
The following table (Table 10) lists the Line-based voice mail server SMDI  
parameters.  
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Chapter 5 Voice Mail Servers 63  
Table 10 Line-based voice mail server SMDI parameters  
Parameter  
Type  
Range  
Default Description  
Address  
String(100) Up to 100  
characters  
N/A  
The IP address of the  
terminal (SMDI) server.  
Port  
Integer  
None  
None  
None  
The port for the SMDI  
server to TELNET to.  
SMDI Version  
Float  
None  
The SMDI version. There  
are only 2 versions of the  
SMDI protocol: Version 1  
and Version 2. (Version 2  
is the latest draft and is  
supported by most voice  
mail servers.)  
Username  
String(60)  
String(20)  
Up to 60  
characters  
N/A  
N/A  
The TELNET username  
for logging in to the SMDI  
TELNET server.  
Up to 20  
characters  
The TELNET password  
for logging in to the SMDI  
TELNET server.  
4
Click Next to enter general information about the Line-based server  
(Figure 39).  
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Figure 40 Create new voice mail server window (Line-based) - step 3  
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Chapter 5 Voice Mail Servers 65  
5
Enter general information about the Line-based voice mail server in the  
parameter fields. The following table lists the Line-based parameter  
information.  
Table 11 Line-based voice mail server general parameter information  
Parameter  
Type  
Range  
Default  
Description  
Description  
String(100) Up to 100  
characters  
None  
The description for the server that  
identifies the Line-based voice  
mail server.  
Client contact  
String(100) Up to 100  
characters  
Voice mail  
The field used to route voice mail.  
In most cases, you will use the  
default (voice mail). If you require  
a different client contact, contact  
your Nortel Networks support  
representative.  
Application Server  
String(60)  
Up to 60  
ServerHome. With  
Specifies the SIP Application  
characters this default, each SIP Module will host the voice mail  
Application Module  
will ignore this field  
and forward SIP  
server being provisioned. This is  
used in an N+M environment. All  
SIP Application Modules that are  
requests to the voice not the home SIP Application  
mail server instead of Module will forward requests  
forwarding them to a destined to voice mail to the  
"home" Application  
Server.  
address specified in this field.  
Maximum Call  
Duration  
Digit  
1-99  
None  
The period of time that the SIP  
Application Module allows a  
caller to stay connected to the  
Line-based voice mail server  
when someone is trying to leave  
a voice mail message.  
Domain name  
String(60)  
Up to 60  
characters  
None  
The domains that will use this  
voice mail server. Use the  
Ctrl-click key to select multiple  
domains to use the same voice  
mail server.  
6
Click Next to provision the lines for the gateway (Figure 41).  
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Figure 41 Create new voice mail server window (Line-based) - step 4  
7
Enter the Line-based parameter information to provision the lines for the  
gateway. The following table (Table 12) lists the Line-based voice mail server  
parameters.  
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Chapter 5 Voice Mail Servers 67  
Table 12 Line-based voice mail server parameters  
Parameter  
Type  
Range  
Default  
Description  
Gateway  
String 1-100  
domain.com;  
norteltrkgrp=1;ma name and IP address of  
ddr=0.0.0.0  
The trunk group, domain  
characters  
0-38 digits  
the gateway.  
Line  
Digit  
None  
The 10-digit telephone  
number that identifies the  
line from the gateway’s  
point of view.  
Message  
Desk  
3 digits 000-999  
None  
None  
A 3-digit number used by  
the SMDI protocol to  
determine where to store  
an incoming message.  
Position  
Number  
4 digits 0000-9999  
A 4-digit number used to  
determine where to put  
an incoming message.  
8
9
Click Add Line. To add multiple lines, continue to enter the gateway  
parameter information and click Add Line.  
When you are finished adding lines for this gateway, click Set New Line to  
have more than one gateway with lines going into the same voice mail server.  
10 Click Save. The Line-based voice mail server you added now appears in the  
list of voice mail servers (Figure 42).  
Figure 42 Voice Mail Servers window (Line-based)  
Assigning a user to a provisioned voice mail server  
In order for a user to have voice mail, you must associate a voice mail server and  
mail box for that user.  
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To associate a user with a voice mail server that has been previously provisioned  
1
2
3
4
Click the Domains menu option  
Click the domain for the user.  
Click the Users > List Users menu option.  
Click Search at the bottom of the screen to display a list of users in the  
domain.  
5
6
Click the Details link for the user you want to associate with a voice mail  
server.  
Scroll down the screen to the voice mail server field. Select a voice mail  
server to use when sending calls for this user to voice mail from the drop  
down list of provisioned voice mail servers for this domain.  
7
8
Enter the information for the user’s voice mail box in the Voice mail field.  
Click Save.  
Listing/modifying/deleting a voice mail server  
To view details of a provisioned SIP, Trunk, or Line-based voice mail server  
1
Click the Voice Mail > List Voice Mail Servers menu option. The Voice Mail  
Server window appears (Figure 43).  
Figure 43 List of voice mail servers  
2
3
To view the details of a voice mail server, click the link of the voice mail  
server in the Description column. The Create new voice mail server form  
appears where you can modify information.  
Click Save to save your changes or Back to return to the list of voice mail  
servers.  
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4
To delete a voice mail server, click the Delete link for the voice mail server  
you wish to delete. At the confirmation dialog box, click OK to delete the  
voice mail server, or Cancel to cancel the operation.  
Using CPL scripts with voice mail  
A user will only have voice mail service if there is a Call Processing Language  
(CPL) script that specifies a route to sip:voicemail@domain.com (usually the last  
route).  
To specify a route, you need to provision the following voice mail parameters in  
the SIP Provisioning Client:  
Voice mail parameter  
Description  
Client Contact  
Must match the username in the CPL script and  
should always be set to voicemail (unless you are  
working with Nortel Networks technical support).  
If this parameter is set to something else, properties  
in the IPCM and Web Client will require changes.  
URI to Query to Find  
Gateway  
Point to a static route, a user with a static route, or  
have a number that can be translated by the  
Telephony routing to go to the correct gateway.  
To create a CPL script that has sip:voicemail@domain.com as the last route  
1
2
3
4
Log in to the SIP Personal Agent.  
Click the Call Manager icon.  
In the Default Behavior window, select the Try to Find Me option.  
Click Save  
Tip: Unless a user logs into the SIP Personal Agent and clicks Save in the  
Call Manager window, they will not have a CPL Script and thus calls to  
them will never route to voice mail as a final route.  
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Chapter 6  
Defining and Assigning Services  
Note: You must have Services provisioning rights to perform the tasks in  
this section.  
Topics in this section:  
“Defining service parameters” on page 71  
“Assigning services to a domain” on page 75  
Defining service parameters  
The SIP Provisioning Client allows you to define and assign services to domains.  
You can assign all or a subset of the pre-defined services to a domain.  
When you define the service parameters for service package, you are essentially  
defining its contents. Services are set up much like convenience packages for the  
domain. The service provider then has the flexibility to set up services and create  
a standard set of services for their customers according to their own  
specifications.  
To define service parameters  
1
Click the Services > Define Service Parameters menu option.  
The list of available services window appears (Figure 44).  
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Figure 44 List of available services window  
2
From the list of available services window, select a value for a service  
parameter from the drop down list. The following table (Table 13) lists the  
service parameters and options.  
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Table 13 Services parameters and options  
Service Option Parameter  
Description  
Client enforced Server enforced  
Enable/disable None  
Voicemail  
Retrieve voice mail or for other  
subscribers, to leave a voice mail  
message. Voice mail is retrieved  
from a network server.The user is  
aware of deposited voice mail since  
their client's MWI (message waiting  
indicator) is enabled.  
Yes.  
Yes.  
Optional.  
Enabled through  
the SIP  
Application  
Module.  
Enable/disable Maximum  
Initiate an audio conference call  
utilizing a SIP audio server. The  
number of users in the conference  
call is limited by the maximum  
number of ports assigned. Values  
beyond above thirty two are not  
allowed.  
Optional.  
Enabled through  
the SIP  
Application  
Module.  
Audio  
Conference  
number of  
ports  
Enable/disable Maximum  
Specifies the number of people in a Yes  
Presence size of client list of buddies. This is the number of  
list of  
buddies  
people that a user is subscribing to in  
order to view their presence.  
Although a user can be logged into  
numerous devices, the list of buddies  
is a shared resource. For example, if  
the max list of buddies size is set to  
10, there can be 10 buddies on each  
of the client devices (the same 10  
buddies, not 10 per device). The  
client has the option of sending any  
of the buddies.  
Maximum  
number of  
Allows a user to be logged into  
multiple devices (i2004 Internet  
Yes.  
Mandatory.  
subscriptions Telephone, SIP Multimedia PC  
generated  
Client, SIP Multimedia Web Client).  
Each such device performs  
messaging to determine the  
presence (online status) of each of  
the buddies in the list. There is a  
need for limiting the number of  
devices and buddies to limit resource  
contention issues. The maximum  
number of presence subscriptions is  
defined by the following formula:  
maximum number of subscriptions  
generated = (maximum # of clients) *  
(maximum number of allowed  
buddies).  
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Service Option Parameter  
Description  
Client enforced Server enforced  
Enable/disable None  
Video  
Use of point to point video service  
between web clients and PC clients.  
Yes.  
Simultaneous  
Network Call  
Restriction  
Maximum  
number of  
ringlist  
Specifies the maximum number of  
users that are defined in a route in  
the Call Manager portion of the SIP  
Personal Agent. The users in the  
ringlist are simultaneously rung.  
Yes.  
Yes.  
entries  
Enable/disable  
Advanced  
Screening  
-
Enables the ability to perform  
address screening (based on the  
user’s current address book entries)  
and time screening (time of day, day  
of week) in the Call Manager area of  
the SIP Personal Agent.  
Enable/disable QoS DiffServ Specifies quality of service for client Yes.  
QoS  
Code for  
signaling, audio and video (up to 64).  
Signalling  
QoS DiffServ  
Code for  
Audio  
QoS DiffServ  
Code for  
Video  
QoS 802.1p  
Service  
Priority  
Enable/disable None  
Blended User  
Enables the use of a traditional  
phone for voice calls and the SIP  
Multimedia PC for multimedia  
communication and collaboration.  
Yes  
3
To create new values for the service parameter, click [Edit].  
The Add new parameter values window appears (Figure 45).  
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Figure 45 Add new parameter values window  
4
5
6
7
For the selected parameter, select a value from the list, using the UP and  
DOWN buttons to set the priority of the value.  
To add a new value, click Add. The new value appears in the list of parameter  
values.  
Click Set as Default to set the value in the list as the default value for the  
service parameter.  
Click Save to save your service parameter changes.  
Assigning services to a domain  
Once you add a value for a service parameter, you can select a domain to assign  
the services to.  
To assign services to a domain  
1
Click the Services > Assign Services menu option. The Assign Services  
window appears (Figure 46).  
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Figure 46 Assign services window  
2
Select a domain from the drop down list and click Continue. To assign  
services to more than one domain, click the link to assign services to multiple  
domains in the Assign Services window. The Assign services to domains  
window appears (Figure 47).  
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Figure 47 Assign services to a domain window  
3
4
Select the services and service parameter values you want to assign to the  
selected domain or multiple domains (Use the Ctrl-click key to choose  
multiple domains.) To enable all services, check the Enable all Services  
checkbox.  
Click Save. A confirmation message displays that your service was assigned  
successfully.  
Once services have been defined and assigned to domains, you can list all the  
services that are available for a domain (or multiple domains) and create and  
assign packages accordingly. For a subdomain, services are assigned from the root  
domain.  
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79  
Chapter 7  
Assigning Services and Creating Service  
Packages  
Topics in this section:  
“Understanding service packages” on page 79  
“Listing available services” on page 79  
“Assigning services to a service package” on page 83  
“Creating a service package” on page 86  
“Listing/modifying/deleting service packages” on page 87  
“Assigning service packages to domains and subdomains” on page 89  
“Upgrading and downgrading service packages” on page 90  
Understanding service packages  
Once services are provisioned for a domain, you then have the flexibility to create  
service packages for a domain or subdomain based on a set of predefined services.  
Each service package is a collection of predefined services. Including a service in  
a service package enables the service, or feature, for the domains associated with  
the package. You can assign packages that are currently created in the domain  
down to a subdomain.  
Listing available services  
To view a list of all services and parameters available in a service package  
Click the Domains menu option.  
1
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2
Under the domain or subdomain for which you wish to list services, click the  
Service Package > List Services sub menu option. The List of available  
services window appears (Figure 48).  
Figure 48 List of available services  
3
Review the list of services and service parameters. The following table  
(Table 14) lists service packages, service parameters, and the values available  
for each service parameter.  
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Table 14 Services parameters and options  
Service Option Parameter  
Description  
Client enforced Server enforced  
Enable/disable None  
Voicemail  
Retrieve voice mail or for others, to  
leave a voice mail message. Voice  
mail is retrieved from a network  
server.The user is aware of  
deposited voice mail since their  
client's MWI is enabled.  
Yes.  
Optional.  
Enabled through  
the SIP  
Application  
Module.  
Enable/disable Maximum  
Initiate an audio conference call  
utilizing a SIP audio server. The  
number of users in the conference  
call is limited by the maximum  
number of ports assigned. Values  
beyond 32 is not allowed.  
Yes.  
Optional.  
Enabled through  
the SIP  
Application  
Module.  
Audio  
number of  
ports  
Conference  
Enable/disable Maximum  
Presence  
Specifies the number of people in a Yes. Mandatory No.  
size of client list of buddies. This is the number of  
list of  
buddies  
people that a user is subscribing to in  
order to view their presence.  
Although a user can be logged into  
numerous devices, the list of buddies  
is a shared resource. For example, if  
the max list of buddies size is set to  
10, there can be 10 buddies on each  
of the client devices (the same 10  
buddies, not 10 per device). The  
client has the option of sending any  
of the buddies.  
Maximum  
number of  
Allows a user to be logged into  
multiple devices (i2004 Internet  
No.  
Yes.  
Mandatory.  
subscriptions Telephone, SIP Multimedia PC  
generated  
Client, SIP Multimedia Web Client).  
Each such device performs  
messaging to determine the  
presence (online status) of each of  
the buddies in the tree. There is a  
need for limiting the number of  
devices and buddies to limit resource  
contention issues. The maximum  
number of presence subscriptions is  
defined by the following formula:  
maximum number of subscriptions  
generated = (maximum # of clients) *  
(maximum number of allowed  
buddies)  
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Service Option Parameter  
Description  
Client enforced Server enforced  
Report when A checkbox to report a user’s  
Yes.  
Yes.  
No.  
inactive  
inactivity (a user is inactive when  
they do not move their mouse or type  
on their keyboard within a  
configurable amount of time.)  
Inactivity  
Timer (in  
minutes)  
If enabled, an inactivity timer (in  
minutes) that PC-based clients will  
observe before reporting "Inactive" to  
the Presence Module. Values range  
from 10 to 600, with 15 being  
Yes/No (Enabled  
as a privilege,  
lower bound of  
what a user can  
select is  
recommended. If this service  
determined in the  
SIP Provisioning  
Client.)  
package parameter is enabled, then  
the user must explicitly enable the  
feature on the client itself.  
Report when If enabled, then every call made by  
on the phone subscribers can have "On the  
Phone" presence reported to the  
Yes.  
Yes/No. (Enabled  
as a privilege in  
the SIP  
Provisioning  
Client.)  
Presence Module. If this service  
package parameter is enabled, then  
the user must explicitly enable the  
feature on the client itself.  
Enable/disable None  
Video  
Use of point to point video service  
between web clients and PC clients.  
Yes.  
Yes.  
Simultaneous  
Network Call  
Restriction  
Maximum  
number of  
ringlist  
Specifies the maximum number of  
users that are defined in a route in  
the Call Manager portion of the SIP  
Personal Agent. The users in the  
ringlist are simultaneously rung.  
entries  
Enable/disable  
Advanced  
Screening  
-
Enables the ability to perform  
Yes.  
address screening (based on the  
user’s current address book entries)  
and time screening (time of day, day  
of week) in the Call Manager portion  
of the SIP Personal Agent.  
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Chapter 7 Assigning Services and Creating Service Packages 83  
Service Option Parameter  
Description  
Client enforced Server enforced  
Enable/disable QoS DiffServ Specifies quality of service for client Yes.  
QoS  
Code for  
Signalling  
signaling, audio and video (up to 64).  
QoS DiffServ  
Code for  
Audio  
QoS DiffServ  
Code for  
Video  
QoS 802.1p  
Service  
Priority  
Blended user  
Preferred  
Audio  
Device  
If the Blended User service option is  
checked, the Preferred Audio Device  
is required so that the regular clients  
(SIP Multimedia PC Client, SIP  
Multimedia Web Client, i2004  
Yes.  
Internet Telephone controlled via  
IPCM) can call a Blended PC Client  
by their SIP address (not just by the  
PSTN equivalence number).  
4
Select a Service Package sub menu option to continue setting up a service  
package.  
Assigning services to a service package  
The Assign Services menu option allows you to select services and service  
parameter values for a domain or subdomain (or multiple domains or  
subdomains).  
Tip: Changes to service package services and parameter values will take  
effect immediately or upon the next login, depending on the client.  
To assign services to a domain or subdomain  
1
Click the Domains menu option.  
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2
Click the domain or subdomain for which you wish to assign services.  
Tip: When adding a package to a subdomain, you must first have defined  
a service package for the parent domain.  
3
Click the Service Package > Assign Services sub menu option. The Assign  
Services window appears (Figure 49).  
Figure 49 Assign services window  
4
5
From the drop down list, select a domain to assign services and service  
parameters to and click Continue. The Assign services to domains window  
appears (Figure 50).  
To assign services and service parameters to multiple domains, use the  
Ctrl-click key to select multiple domains from the drop down list and click  
the Click here to assign services to multiple domains link.  
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Chapter 7 Assigning Services and Creating Service Packages 85  
Figure 50 Assign services to domains window  
6
To assign services for the selected domain, subdomain, or multiple domains or  
subdomains, select the checkbox next to the service option (see Table 14 on  
page 81 for service and parameter descriptions).  
7
8
9
To modify a service parameter, select a value from the drop down list.  
To select all services, check the Choose All Services checkbox.  
To select particular services for a domain, select the checkbox next to the  
service.  
10 Click Save to assign the services.  
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Creating a service package  
Domains, subdomains, and users are assigned service packages to control their  
access to network services.  
To create a service package  
1
2
Click the Domains menu option.  
Under the domain or subdomain for which you wish to assign services, click  
the Service Package > Create Package menu option. The Create new  
package window appears (Figure 51).  
Figure 51 Create new package window  
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Chapter 7 Assigning Services and Creating Service Packages 87  
3
4
Enter a name for the service package in the text box.  
Select whether or not to make this package the default for the domain or  
subdomain. The default is NO.  
5
Create the service package by selecting:  
a
the checkbox next to the service option to select services for the service  
package (see Table 14 on page 81 for service and parameter descriptions).  
b
c
d
a service parameter value from the drop down list.  
the Choose All Services checkbox to select all services.  
the checkbox next to the service to select particular services.  
6
Click Save to create the service package. A message displays stating that the  
package was successfully created.  
Listing/modifying/deleting service packages  
To view a list of all service packages provisioned in a domain or subdomain  
1
2
Click the Domains menu option.  
Under the domain or subdomain for which you wish to view the service  
package details, click the Service Package > List Packages menu option.  
The Details of the package window appears (Figure 52).  
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Figure 52 List of packages domain window  
3
From the list of packages  
a
Click the Details-Modify link to view service package details and modify  
service parameters and values (see Table 14 on page 81 for service and  
parameter descriptions). If a service package is modified, then all users  
who currently have this service package are affected. If service attributes  
are modified for a domain, then existing packages utilizing this service  
remain as is. Any new packages created for this domain utilize the new  
parameters.  
b
c
Click the Delete link to delete the service package from the domain. A  
service package can only be deleted if no users are associated with the  
service package.  
Click the Make default link to make the selected package the default  
package for the domain.  
4
Click Save from the Details screen to save your changes, or click a menu  
option on the left side of your screen to continue with another operation.  
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Chapter 7 Assigning Services and Creating Service Packages 89  
Assigning service packages to domains and  
subdomains  
To assign a service package to a domain or subdomain  
1
2
Click the Domains menu option.  
Under the domain or subdomain for which you wish to assign a service  
package, click the Service Package > Assign Packages sub menu option.  
The Assign package to domains window appears (Figure 53).  
Figure 53 Assign package to domains window  
3
4
From the drop down list, select a service package.  
To make the selected package the default, check the Make package the  
default for selected domains checkbox.  
5
6
Select the domain from the list of domains in the display area. Use the  
Ctrl-click key to select multiple domains.  
Click Assign.  
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Upgrading and downgrading service packages  
Upgrading a service package is allowed (as in the case when you want to add  
services) as long as the service is available at the domain. Upgrading a service  
package to update parameter values is also allowed, but only to the limit assigned  
to the domain.  
Downgrading a service package (for services with parameters) is allowed only to  
the upper limit of the parameter values for the user and subdomains.  
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91  
Chapter 8  
Topics in this section:  
“Managing user functions” on page 91  
“Adding a user to a domain or subdomain” on page 92  
“Adding a blended user” on page 95  
“Adding i2004 Internet Telephone capability for a user” on page 99  
“Adding a conference server user and route” on page 101  
“Customizing service packages” on page 104  
“Listing/modifying/deleting users” on page 105  
“Listing/modifying/deleting aliases” on page 107  
“Listing/modifying/deleting blended aliases” on page 109  
Managing user functions  
Once a domain or subdomain is provisioned, an administrator is responsible for  
managing the following user functions:  
adding, viewing, modifying, or deleting users (including blended users) from  
the domain or subdomain  
adding, viewing, modifying, or deleting i2004 Internet Telephone device  
properties  
assigning users to i2004 Internet Telephones  
customizing service packages (for example, to add a blended user if available  
as a service package option)  
viewing, modifying, or deleting aliases and/or blended aliases  
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Adding a user to a domain or subdomain  
To add a user to a domain or subdomain  
1
2
Click the Domains option.  
Under the domain or subdomain, click the User > Add User menu option.  
The Add new user window appears (Figure 54).  
Figure 54 Add new user  
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Chapter 8 User Management 93  
3
Enter information about the new user in the parameter fields as described in  
the following table (Table 15).  
Table 15 Add user parameter descriptions  
Parameter  
Type  
Range  
Default Description  
User Name  
String (60)) Up to 60  
characters  
N/A  
Username of this user. Use lowercase. Does  
not have to be the same name as the First  
Name.  
First Name  
String (30) Up to 30  
characters  
N/A  
N/A  
N/A  
N/A  
N/A  
First name of the user. Case sensitive.  
Last name of this user. Case sensitive.  
Password of this user.  
Last Name  
String (30) Up to 30  
characters  
Password  
String (200) Up to 200  
characters  
Confirm Password  
Service package  
String (200) Up to 200  
characters  
Password of this user.  
Select from  
drop down  
list  
-
-
-
The list of service packages assigned to the  
user’s domain or subdomain.  
Aliases  
String (40)  
N/A  
N/A  
N/A  
The list of aliases for this user. Aliases  
associate a PSTN phone number to a  
username.  
Blended Aliases  
Status Reason  
String  
Alias for a blended user, a person who uses  
the traditional phone but is equipped with  
multimedia services at the desktop.  
Select from N/A  
drop down  
list  
The status value of this user.  
email  
String (60) Up to 60  
characters  
N/A  
N/A  
N/A  
N/A  
N/A  
N/A  
Email address of this user.  
Business phone number.  
Home phone number.  
Cell phone number.  
Business Phone  
Home Phone  
Cell Phone  
Pager  
String (60) Up to 60  
characters  
String (30) Up to 30  
characters  
String (30) Up to 30  
characters  
String (30) Up to 30  
characters  
Pager number.  
Fax  
String (30) Up to 30  
characters  
Fax number of this user.  
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Table 15 Add user parameter descriptions  
Parameter  
Type  
Range  
Default Description  
Voicemail Server  
Select from  
drop down  
list  
-
None  
selected  
The voice mail server available to this user.  
Private Charge ID  
Public Charge ID  
String (30) Up to 30  
characters  
N/A  
N/A  
The Private Charge ID of this user.  
String (30) Up to 30  
characters  
The 10-digit national dial plan number  
associated with the subscriber. This is  
required for the interworking with Public  
Switched Telephone Network (PSTN)  
switches in the Time Division Multiplex (TDM)  
network.  
Voicemail  
String (30) Up to 30  
characters  
N/A  
The voice mail address of this user.  
Class of Service  
Select from Up to 30  
None  
The name of the class of service assigned to  
drop down characters selected the domain and user. This is displayed to the  
list  
administrator when modifying or assigning  
the COS to a domain/subdomain/user.  
Time Zone  
Locale  
Select from  
drop down  
list  
-
-
N/A  
N/A  
The time zone that this user belongs to.  
The location of this user.  
Select from  
drop down  
list  
4
Click Save.  
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Chapter 8 User Management 95  
Adding a blended user  
Setting up a user as a blended user allows the end user to use their PCs for the  
multimedia portion of their communication, while using their existing telephony  
system for voice.  
Note: When a user receives their service package (at SIP Multimedia PC  
Client startup and on any changes to the subscriber data made by the  
administrator), and that service package contains Blended User, then that  
PC Client becomes a SIP Multimedia Blended PC Client.  
To add a user as a blended user to a domain or subdomain  
1
2
Click the Domains menu option.  
Under the domain or subdomain where you wish to set up a blended user,  
click the User > Add User menu option. The Add new user window appears.  
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Figure 55 Add new user - blended  
3
4
Complete the fields on the Add new user form.  
Add a blended alias for the user. This alias is used to identify calls that come  
from the SimRing feature that is assigned to the user on an existing switching  
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Chapter 8 User Management 97  
system. It must match the called party number that will be sent to the SIP  
Application Module on these SimRing calls.  
Note: Blended aliases are per subscriber data and are entered as list of  
telephony numbers into the Blended Aliases text box. The blended aliases  
must not match any of the normal aliases entered into the Aliases text box  
on the same screen.  
5
6
Add an alias (not blended alias) that matches the calling party number that  
will be sent to the SIP Application Module when the user calls another  
blended user. This is the calling party number that is sent to the SIP  
Application Module on the SimRing call.  
If calls from users (non-blended) to blended users are routed to the existing  
switching system over a public route, add an alias (not blended alias) that  
matches the public charge ID that you intend to provision for the user. If calls  
from users (non-blended) to blended users are routed to the existing switching  
system over a private route, add an alias (not blended alias) that matches the  
private charge ID you intend to provision for the user.  
Adding these aliases allows the SIP Application Module to identify the  
calling party on SimRing calls that are the result of a call origination, transfer  
or redirect from the SIP Application Module back to a blended user's  
Preferred Audio Device on the existing switching system.  
Note: This step should also be done for all non-blended users that may  
call a blended user in order to identify SimRing calls that were originated  
by non-blended users.  
7
Click Save. A User added successfully message appears.  
Figure 56 Blended User added successfully confirmation  
8
Click the Customize service package link at the top of the form to create a  
service package containing specific services for the blended user. The Service  
package customization window appears (Figure 57).  
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9
Enter the Preferred Audio Device for the user. The number entered here must  
be the number that is required to route a call to the user's phone and trigger the  
user's SimRing feature. This number should be unique for each user. This  
number is required so that regular clients (SIP Multimedia PC Client, SIP  
Multimedia Web Client, and the i2004 Internet Telephone controlled through  
the IPCM) can call a Blended PC Client by their SIP address (and not only by  
the PSTN equivalence number).  
Figure 57 Service package customization for a blended user  
10 Click Save. A message displays that the user was added successfully.  
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Chapter 8 User Management 99  
Adding i2004 Internet Telephone capability for a user  
The IP Client Manager requires the i2004 properties of a user to enable that user  
to log into, receive calls on, or make calls from an i2004 Internet Telephone.  
To add i2004 Internet Telephone capability  
1
After successfully adding a new user, click the i2004 properties link at the  
top of the User details window. The i2004 User details window appears  
(Figure 58).  
Figure 58 User details window (i2004)  
2
Enter i2004 Internet Telephone properties in the parameter fields as described  
in the following table.  
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Table 16 i2004 properties  
Parameter  
Type  
Range  
Default Description  
Display Name  
String (30) Up to 30  
characters  
N/A  
The display string that is  
displayed on the i2004  
Internet Telephone.  
Registration Length Integer  
Max 86400 3600  
seconds or  
1440  
minutes or  
24 hours  
This value, along with the  
Registration Unit  
parameter value, is used  
to determine when the  
i2004 Internet Telephone  
should register with the  
IPCM (see Note). It is  
recommended to use  
Registration Length in the  
range of 24 - 48.  
Registration Unit  
String (10) Up to 10  
Seconds The unit used by the  
i2004 Internet Telephone  
to know when to register  
with the IPCM. It is  
characters  
recommended to use  
Registration Unit as  
Hours.  
IPCM  
String (100) Up to 100  
characters  
N/A  
Drop down list of IPCM  
names to which this user  
belongs.Varies  
depending on which  
IPCMs are associated  
with the root domain.  
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Note: The Registration Length and Registration Unit fields combined  
together make up the registration expiration time. The smaller the value,  
the more SIP traffic generated. If you were to select a registration  
expiration of every 2 minutes, then the IPCM would only register the user  
for 2 minute time periods and would have to re-register the user every 2  
minutes.  
For security reasons, it is recommended to not keep the registration  
expiration time too high. If you selected a registration expiration of 10  
days, then the registration authentication process is only going to occur  
every 10 days. If the user changes a password, the change would not be  
noted for up to 10 days (if only registration on authentication is being  
used). Also, in the unlikely event that the registration is somehow lost, a  
reasonable length registration expiration would allow the IPCM to  
re-register the user.  
3
Click Save.  
Adding a conference server user and route  
To enable conferencing capability, a conference user and route must be  
provisioned using the SIP Provisioning Client or the SIP Personal Agent.  
Use this client...  
To set up...  
SIP Provisioning Client  
a conference user, and, if there is only one conference  
server, a route to the conference server. (Routes  
added through the SIP Provisioning Client are always  
attempted simultaneously.)  
SIP Personal Agent  
sequential routes and rules for multiple conference  
servers that include each conference server listed in a  
separate ringlist.  
Adding a conference server user and route using the SIP  
Provisioning Client  
To add a single conference server user and route using the SIP Provisioning Client  
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1
2
3
4
Click the Domains menu option.  
Click the domain where you want to add a conference server user.  
Click the Add User menu option.  
In the Add User form, enter the text “conference” for the User Name. In the  
password field, enter any password.  
5
6
Click Save. The User details screen displays a message that the user was  
added successfully.  
At the top of the screen, click the Routes link to specify the route to the  
conference server.  
Figure 59 Conference user - add route  
7
8
In the User Routes text box, enter the SIP address that the conference server  
is expecting using the following format:  
sip:conference@<conference_server_IP_address>;nt_info=proxy_foreign;  
nortelservice=conference  
Click Add. The route now appears in the window of selected routes.  
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Chapter 8 User Management 103  
Figure 60 Conference user - route added  
9
Click Save to save the route.  
Adding conference server routes using the SIP Personal  
Agent  
To add multiple conference server routes using the SIP Personal Agent  
1
Log in to the SIP Personal Agent. (The login name and password is the user  
name and password that was provisioned for the conference user in the SIP  
Provisioning Client.)  
2
3
4
5
6
Click the Call Manager icon on the left of the SIP Personal Agent screen.  
The Welcome to Call Manager home screen displays.  
Click the ListRoutes tab from the Call Manager Home screen. The Ring List  
Details screen displays.  
Enter a name for the conference server route in the Ring list name text box  
(for example, conference route).  
From the drop down list, select the maximum number of times to ring the  
phone.  
In the first text box in the column, enter the SIP address that the conference  
server is expecting using the following format:  
sip:conference@<conference_server_IP_address>;nt_info=proxy_foreign;  
nortelservice=conference  
7
Select SIP address from the drop down list.  
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8
9
To set up routes for multiple conference servers, repeat steps 6 and 7 for each  
sequential ring list in columns 2 and 3.  
Click Add Route. The Add Route button updates the online user database  
with the conference server routing information.  
Setting up rules for the conference server route  
Once a route has been set up, a screening rule must be created to define the  
activation of the conference servers. The conference server route is not activated  
until a rule has been applied.  
To set up a screening rule for the first time in the SIP Personal Agent  
1
2
3
4
5
6
Click ListRules from the Call Manager screening and routing screen. The  
Welcome to Call Manager screen displays.  
Click the link, create a screening rule. The Call Manager Rule Details screen  
displays.  
Enter a name for the conference server rule in the Rule name text box (for  
example, conference rule).  
Click the Active check box to select that the rule should appear in the Active  
Rules window.  
Select the conference route from the pull-down menu next to the Active route  
field.  
Click Save Rule.  
Customizing service packages  
Once a user is added, you may want to customize the service package on a per  
user basis.  
To customize a user’s service package  
1
After adding or listing details for a user, click the Customize service package  
link.  
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Figure 61 Customize service package link  
2
3
Customize the options in the user’s service package by completing the  
available selections.  
Click Save to save the changes to the service package. A message appears that  
the user package was customized successfully.  
Listing/modifying/deleting users  
To list, modify, or delete users (including blended users)  
1
Click the Domains menu option.  
Under the domain or subdomain, click the User > List User menu option. The  
2
Users window appears (Figure 62). Select whether to search on Phone  
Number, Last Name, User Name, or First Name from the drop down list  
3
Click Add Criteria. The search criteria you selected appears in the lower  
window and is used to refine your search.  
4
5
Repeat steps 2 and 3 to list additional users.  
In the lower window area, select whether to search by All or Any of the  
criteria specified. Note that you do not have to add any criteria if you want to  
see all the users in the selected domain. This will also find all users in any of  
the selected domain's subdomains.  
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Figure 62 List Users window  
6
7
Define the size of your search list from the drop down list.  
Click Search to begin the search, or Remove to remove search criteria from  
the search list. A list of users appears.  
Figure 63 List users search results  
8
Click the Details link to modify user information. The User details window  
appears (Figure 64). To delete the user from the domain, click the Delete link.  
At the delete confirmation dialog box, click OK to delete the user, or Cancel.  
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Figure 64 User details window  
9
Click Save. A User Updated Successfully message appears.  
Listing/modifying/deleting aliases  
To list all aliases in a domain  
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1
2
Click the Domains menu option.  
Under the domain or subdomain, click the Users > List Aliases menu option.  
The Aliases window appears (Figure 65). Select whether to search on Last  
Name, Alias, or Domain from the drop down list  
3
Click Add Criteria. The search criteria you selected appears in the lower  
window and is used to refine your search.  
4
5
Repeat steps 1-3 to list additional users.  
In the lower window area, select whether to search by All or Any of the  
criteria specified. Note that you do not have to add any criteria if you want to  
see all the aliases in the selected domain. This will also find all aliases in any  
of the selected domain's subdomains.  
Figure 65 List aliases window  
6
Define the size of your search list from the drop down list.  
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7
Click Search to begin the search, or Remove to remove the search criteria  
from the search list.The Aliases search results window appears (Figure 66).  
Figure 66 List aliases search results  
8
9
Click the Details link to modify alias information. The User details window  
appears. Click Save to save your changes.  
To delete an alias, click the Delete link on the List aliases search results  
screen to delete the user from the domain. At the delete confirmation dialog  
box, click OK to delete the user, or Cancel.  
10 Click Save.  
Tip: If you delete an alias, the username associated with the alias is  
deleted.  
Listing/modifying/deleting blended aliases  
To list all blended aliases in a domain  
1
2
3
Click the Domains menu option.  
Select a domain (or a subdomain of a domain).  
Click the Users > List Blended Aliases menu option. The Blended Aliases  
window appears. Select whether to search on Last Name, Alias, or Domain  
from the drop down list  
4
Click Add Criteria. The search criteria you selected appears in the lower  
window and is used to refine your search.  
5
6
Repeat steps 1-4 to list additional last names, aliases, or domains.  
In the lower window area, select whether to search by All or Any of the  
criteria specified. Note that you do not have to add any criteria if you want to  
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see all the blended aliases in the selected domain. This will also find all  
blended aliases in any of the selected domain's subdomains.  
7
Click Save. The Blended Alias search results window displays (Figure 67)  
Figure 67 Blended Aliases search results window  
8
Click the Details link to modify alias information. The User details window  
appears (Figure 64). To delete the user from the domain, click the Delete link.  
At the delete confirmation dialog box, click OK to delete the user, or Cancel.  
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Chapter 9  
Managing i2004 Internet Telephones  
Topics in this section:  
“Introduction” on page 111  
“Adding an i2004 Internet Telephone” on page 111  
“Listing an i2004 Internet Telephone” on page 116  
“Deleting an i2004 Internet Telephone” on page 119  
Introduction  
i2004 Internet Telephones are provisioned by creating an i2004 device entity and  
assigning this device to a domain or subdomain.  
Devices are assigned to a domain based on the domain in which it is created. An  
administrator can use the Devices menu option to add, view, modify or delete  
i2004 Internet Telephones.  
Adding an i2004 Internet Telephone  
1
Click the Domains menu option.  
2
Click the domain or subdomain for which you wish to provision an i2004  
Internet Telephone.  
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3
Click the Devices > Add Device sub menu option. The Adding a Device  
window appears (Figure 68)  
Tip: Several fields in the Adding a Device window are populated with the  
default property values that have been previously set up for the domain or  
subdomain.  
Figure 68 Adding a device  
4
Enter device parameters as described in the following table (Table 17).  
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5
Notice that the most of the values that appear on the form are default values  
that are set from what was entered when the domain was provisioned.  
Table 17 i2004 Internet Telephone parameters  
Parameter  
Type  
Range  
Default Description  
Label  
String (30) N/A  
N/A  
User defined text label that uniquely  
describes this device. This label is used by  
the System Management Console when  
performing maintenance operations on  
queried devices (for example, Tom's Desk).  
Defaults to owner display name.  
MAC Address  
String (100) Up to 100  
characters  
N/A  
N/A  
12 Digit Hex string. Permanently burned into  
the i2004 device at manufacture time. Read  
from tag on back of device. This string is  
converted to lower case. Must be unique per  
IPCM.  
Available IPCM List Select from N/A  
Drop down list of named IPCMs.  
drop down  
list  
Def Proxy Host  
Def Proxy Port  
Active Proxy  
String (30) N/A  
N/A  
IP Address of Proxy used as default for the  
i2004 Internet Telephone.  
Integer  
Any valid  
port  
5060  
Default Proxy SIP Port. No validation is done  
on this field to ensure validity.  
Boolean  
True/False True  
If True, calls will route through the default  
proxy. For future use. No other routing is  
currently supported for the IPCM.  
Allow All Codecs  
Behind Firewall  
Boolean  
Boolean  
True False True  
True/False True  
Allow all codecs (True) or only selected codec  
(False). See Vocoder parameter.  
Indicates whether there is a firewall/NAT  
between the i2004 Internet Telephone and  
the system.  
Alpha  
Boolean  
Integer  
True/False True  
Enables alphanumeric dialing as the default  
dialing style. If disabled, the I2004 Internet  
Telephone dialing style defaults to numeric.  
Contrast  
0-15  
8
Telephone display contrast setting.  
Vocoder:Packet  
Time  
Integer  
(msec)  
10/20/30/  
40/60  
20  
Packet time in milliseconds {10\ms, 20\ms,  
30\ms, 40\ms, 60\ms}.  
PDIL Timer  
Integer  
(seconds)  
N/A  
4
Interdigit time-out.  
PSEIZ Timer  
Integer  
N/A  
15  
Time-out for first dialed digit.  
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Table 17 i2004 Internet Telephone parameters  
Parameter  
Type  
Range  
Default Description  
DATE FMT  
Numeric  
Standard  
MM/DD or  
Numeric  
Inverse  
2,3  
2
0
Only formats 2 and 3 are supported.  
2-numeric standard (09/16); 3-numeric  
inverse (16/09).  
DD/MM  
TIME FMT  
Select from 0,1,2  
drop down  
list:  
The time format of the proxy used as the  
default for the I2004 Internet Telephone 0 >  
12 hour clock.  
12-hour  
French  
24-hour  
Location  
String (50) Up to 50  
characters  
Text describing location of device. A specific  
format of this string is scheduled for a future  
release  
Idle Display  
Time Zone  
String (30) Up to 30  
characters  
N/A  
N/A  
Idle telephone display heading.  
String (20) Any valid  
entry in  
The time zone that this user belongs to.  
table  
TIMEZONE  
INFO  
Locale  
String (20)  
N/A  
The location of this user.  
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Table 18 Click Save. The IPCM Device Created window appears (Figure 69).  
Figure 69 Add device - IPCM device created  
6
The i2004 is now provisioned with the parameters that were entered. The  
i2004 is ready for login. If you want to remotely place a user on an i2004  
Internet Telephone, click Assign Users at the top of the screen. (This is the  
equivalent to a user logging in.) The Device Users window appears  
(Figure 70).  
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Figure 70 Device Users - assigning users  
.
7
Enter the user name in the text field and click Add. (If you do not add a  
domain name for the user, the domain or subdomain of the device is assumed.  
Tip: You can add up to 6 different users per i2004 Internet Telephone.  
8
9
To move users up or down in the selected window, select the user and click  
Up or Down. To remove a name, click Remove or Remove All.  
Click Modify to submit your changes.  
Listing an i2004 Internet Telephone  
To view, or list a particular device (or all devices) in a domain or subdomain  
1
Click the Domains menu option.  
Under the domain or subdomain where you want to view a device, click the  
2
Devices > List Device menu option. The Devices window appears  
(Figure 71).  
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Figure 71 List Devices window  
3
4
Select whether to search on the MACaddress or ID of the i2004 Internet  
Telephone from the drop down list.  
Click Add Criteria. The search criteria you selected appears in the lower  
window.  
5
6
Repeat to list additional devices.  
In the lower window area, check whether to search by All (devices), or Any,  
using the device criteria you specified.  
7
Click Search to begin the search, or Remove to remove the device from the  
search list. The Devices in Domain window appears (Figure 72).  
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Figure 72 List devices search results  
8
Click the Details link to modify device information. The Modifying a Device  
in a domain window appears (Figure 73).  
Figure 73 Modifying a device in a domain window  
9
Click Save.  
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Deleting an i2004 Internet Telephone  
To delete an i2004 Internet Telephone  
1
2
Click the Domains menu option.  
Click the domain or subdomain for which you wish to delete an i2004 Internet  
Telephone.  
3
4
Click the Devices > List Devices sub menu option. The Devices window  
appears. Select whether to search on the MACaddress or ID of the i2004  
Internet Telephone from the drop down list.  
Click Add Criteria. The search criteria you selected appears in the lower  
window.  
5
6
Repeat to list additional devices.  
In the lower window area, select whether to search by All (devices), or Any,  
using the criteria specified.  
7
Click Search. The Devices in Domain window appears with the results of  
your search (Figure 74).  
Figure 74 Devices in Domain window (search results)  
8
Click the Delete link to delete the i2004 Internet Telephone. At the delete  
confirmation dialog box, click OK to delete the device, or Cancel.  
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Chapter 10  
Telephony Routes  
Note: Telephony routing enables customers to configure their dial plans  
and to set Virtual Private Networks (VPNs). You must have telephony  
Topics in this section:  
“Adding Class of Service (COS) information” on page 122  
“Adding a telephony route” on page 124  
“Adding a route list” on page 128  
“Listing a route list” on page 130  
“Setting up an interworking profile with the Communications Server 2000  
(CS2000)” on page 133  
For more information and examples about telephony routing and translation plans,  
see Appendix B, “Understanding Telephony Routing and Translations,” on page  
149.  
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Adding Class of Service (COS) information  
COS is assigned to users, domains, and routing lists. It is an integer value, and the  
lower the number, the greater, or higher, the COS value. For example, if a user has  
a COS value of 1, and a domain has a COS of 2, 3, and 4, the user with a COS of  
1 can use all the domains to make a call. The lower the number, the higher access  
to making calls.  
To create a COS to associate with a domain, user, or route list  
1
2
Click the Domains menu option.  
Under the domain for which you wish to add COS information, click the  
Telephony Routes > Routing COS menu option. The Class of Service  
information window for the domain appears (Figure 75).  
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Chapter 10 Telephony Routes 123  
Figure 75 Class of Service information for domain window  
3
4
5
Enter a name for the COS in the Name text box.  
Enter a description for the COS in the Description text box.  
Click Save.The COS now appears in the window below. Repeat this step for  
each new COS you wish to create.  
6
Click UP or DOWN to reorder the COS in the Current Choices Available  
window.  
Tip: The higher the position of the COS in the reorder list indicates that  
the COS has a lower number, which means a higher COS value, or  
service.  
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7
To delete a COS from the Current Choices window, click Delete.  
Adding a telephony route  
To add a new telephony route  
1
2
Click the Domains menu option.  
Under the domain or subdomain for which you wish to add a new telephony  
route, click the Telephony Routes > Add Telephony Routing menu option.  
The Create New Telephony Route window appears (Figure 76).  
Figure 76 Create New Telephony Route window  
3
Enter telephony route parameter information and values in the fields  
provided. Note that the Route List window will be empty the first time you  
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create a telephony route. The following table (Table 19) lists the telephony  
route list parameter values and descriptions.  
Table 19 Telephony route parameter information  
Parameter  
Type  
Range  
Default Description  
Name  
String (20) N/A  
String (100) N/A  
String (32) N/A  
N/A  
N/A  
N/A  
The name of this telephony route.  
Description  
From Digits  
A description for this telephony route.  
If you take out the most significant digits  
(MSD) from the number to be translated, its  
value should be equal to or greater than this  
field's value. The number of MSDs to be  
compared is the number of digits you have  
entered in this field.  
To Digits  
String (32) N/A  
N/A  
If you take out the MSDs from the number to  
be translated, its value should be equal to or  
less than this field's value. The number of  
MSDs to be compared is the number of digits  
you have filled in this field.  
Min Number of Digits Integer  
N/A  
N/A  
N/A  
N/A  
The number to be translated should have  
digits equal to or more than the Min Number  
of Digits parameter value.  
Max Number of  
Digits  
Integer  
Integer  
The number to be translated should have  
digits equal to or less than the Max Number  
of Digits parameter value.  
Route Type  
Select one of the following route types  
Private (for example, a private dial plan  
not going out to a gateway)  
SIP (does not have to go through a  
gateway, but can go to a different  
application module or domain)  
Gateway  
Remove  
Integer  
N/A  
0
This is the number of MSDs, which will be  
removed while performing digit manipulation/  
translation.  
Prefix  
String (24) N/A  
N/A  
This is the number that would be prefixed to  
the number coming in for translation.  
Recursive  
Boolean  
True  
False  
If set to Y, the digit manipulation process is  
carried out recursively after each iteration.  
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4
Click Save to create the telephony route. The Telephony Route Listing  
window appears with the new route listed added to the domain and parameter  
options based on the telephony route type (Figure 77).  
Figure 77 Telephony Route Listing window  
5
Click the Change Parameters link.  
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6
Provision the parameters for the new telephony route based on the whether  
the route is a Private, SIP, or Gateway route. The following telephony route  
parameters windows appear as described in the following table (Table 20).  
Table 20 Telephony route parameters window (Private, SIP, Gateway)  
Telephony route parameters  
Description  
For a Private Route:  
From the drop down list, specify what to do if the  
subscriber is not found:  
Stop: Stop at the private route. Do not go to SIP or  
Gateway. The call is dropped, since subscriber does not  
exist.  
Continue: Continue to search for SIP or gateway route  
without translating the number.  
Pre Translate: Continue to SIP or gateway route with  
translating the numbers.  
Click Save.  
For a SIP Route:  
From the drop down list, specify a route name and what to  
do if the subscriber is not found:  
Stop: Stop at the SIP route. Do not go to gateway route.  
Since the subscriber does not exist, the call is dropped.  
Continue: Continue to gateway route without translating  
the number.  
Pre Translate: Continue to gateway route and translate  
the numbers.  
SIP Route: Specify the domain you want the call to go to  
or a different appserver address.  
Click Save.  
For a Gateway Route  
OverRide Charge ID: If set to Yes then the charge ID  
described in the OverRide Charge Value field is used  
instead of either public or private.  
Gateway Route: Select the name of the gateway route  
from the list to which you want to route the call to.  
Gateway RouteType Option: Specifies what charge ID to  
be used: Public or Private. If OverRide Charge ID is set to  
Yes, then this is not used.  
OverRide Charge Value: The value of the charge ID to be  
overriden with.  
Click Save.  
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Listing a telephony route  
To view details about a telephony route  
1
2
Click the Domains menu option.  
Under the domain or subdomain for which you wish to view telephony route  
details, click the Telephony Routes > List Telephony Routing menu option.  
The Telephony Route Listing for the domain window appears (Figure 78).  
Figure 78 Telephony route listing window  
3
4
5
To view or modify a telephony route, click the telephony route name. The  
Create New Telephony Route window appears.  
Modify the fields. Click Save to return to the Telephony Route Listing  
window.  
To change the telephony route parameter, click the Change Parameters link  
for the telephony route. The telephony parameters window appears. Based on  
the whether the route is a Private, SIP, or Gateway route, modify the  
telephony parameters (Table 20).  
Adding a route list  
To add a new telephony route list  
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1
2
Click the Domains menu option.  
Under the domain or subdomain for which you wish to add a route list, click  
the Telephony Routes > Add RouteList menu option. The Create a new  
route list window appears (Figure 79).  
Figure 79 Create a New Route List window  
3
4
5
Enter a name for the new route list in the text field provided.  
Enter a text description of the route list in the text filed provided.  
From the drop down list next to the Incoming Foreign Domain Call Routing  
(from another domain or subdomain) and Incoming Local Domain Call  
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Routing (coming from the same domain) fields, choose one of the following  
call routing options as described in the following table (Table 21).  
Table 21 Call routing options  
Call routing option  
Description  
Allow  
Allow all incoming sessions from other domains and  
subdomains.  
Block  
Block all incoming calls from other domains and  
subdomains.  
Redirect  
Redirect sessions to the originator’s domain. Use this  
option to redirect an incoming session from another  
domain that is routing to a restricted route list.  
6
7
From the drop down list, select a Class of Service for the new route list.  
In the Telephony Route(s) window, highlight and select one or more  
telephony routes to add to the route list (each telephony route must go in a  
route list). To select multiple telephony routes, use the Ctrl-click key.  
Listing a route list  
To view details about a route list  
1
2
Click the Domains menu option.  
Under the domain or subdomain you wish for which you wish to view route  
list details, click the Telephony Routes > List RouteLists menu option. The  
List of RouteLists for the domain window appears (Figure 80).  
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Figure 80 List route lists window  
3
4
5
Click the link for the route list name to modify the route list. The Create New  
Route List window appears. Modify route list information and click Save.  
To delete an existing route list, click the Delete link in the Delete column next  
to the route list you wish to delete.  
To delete a telephony route associated with a route list, click the Delete link in  
the Remove column.  
Using the translation verification tool  
The translation tool allows you to verify that a call is provisioned correctly.  
To use the translation verification tool  
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1
2
Click the Domains menu option.  
Under the domain or subdomain, click the Telephony Routes > Translation  
Tool menu option. The Translation Verification Tool window appears  
(Figure 81).  
Figure 81 Translation Verification Tool window  
3
In the Request URI field, enter a number to call using the following format  
<number>@<domain_or_subdomain_name>.com  
4
5
In the To field, enter the same name as in the Request URI field, or leave  
blank.  
In the From field, enter the calling person’s number using one of the following  
formats  
<name>@<domain_or_subdomain_name>.com  
<number>@<domain_or_subdomain_name>.com  
6
Click Execute. The following window appears and lists the translation (read  
only) verification results (Figure 82). If the window displays some, or all, null  
fields, then the route has been misprovisioned and must be provisioned again.  
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Figure 82 Translation tool results window  
Setting up an interworking profile with the  
Communications Server 2000 (CS2000)  
The Set Profile menu option is for interworking with the CS2000.  
To set the profile  
1
2
Click the Domains menu option.  
Under the domain or subdomain that you want to set up to interwork with the  
CS2000, click the Set Profile menu option. The Set Profile for the domain  
window appears (Figure 83).  
Figure 83 Set Profile for Domain window  
3
4
For the domain selected, enter a name for the profile in the text box provided.  
Click Save. A message appears that the domain profile was modified  
successfully.  
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Chapter 11  
Gateways  
Note: You must have gateway management rights to perform the tasks in  
this section.  
Topics in this section:  
“Introduction to gateways” on page 135  
“Adding a gateway” on page 136  
“Listing a gateway” on page 137  
“Adding a gateway route” on page 138  
“Listing a gateway route” on page 140  
“Adding a gateway trunk group” on page 141  
“Listing a gateway trunk group” on page 142  
Introduction to gateways  
A gateway is a server that connects the SIP network with the PSTN. The gateway  
handles the negotiation for calls between PRI signaling and SIP signalling. The  
Gateways menu option is used by an administrator to provision a PRI gateway or  
the gateway’s translations. The data submitted is validated by the provisioning  
system before being stored in the database.  
The data entered in the SIP Provisioning Client should match the data entered of  
the gateway itself done in the System Management Console. For the PRI  
Gateway, the following items need to match:  
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IP address of the gateway  
trunk group on the gateway  
domain assigned to the trunk group  
Adding a gateway  
The Add Gateway menu option allows an administrator to provision a new  
gateway.  
To add a new gateway  
1
Click the Gateways > Add Gateway menu option. The Create new gateway  
window appears (Figure 84).  
Figure 84 Create new gateway window  
2
Enter the IP address or MADDR of the gateway in the text box (required). For  
example:  
nortelnetworks.com;maddr=<IP_address>  
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3
4
Enter a description the gateway in the Gateway Type text box.  
Click Save. (To clear the text fields, click Clear.) The Gateways window  
appears with a list of all provisioned gateways, including the gateway just  
added (Figure 85).  
Figure 85 Gateways window  
Listing a gateway  
The List Gateway option allows an administrator to view information of all the  
available provisioned gateways.  
To list a gateway  
1
Click the Gateways > List Gateways menu option. The Gateways window  
appears (Figure 86).  
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Figure 86 Gateways window (List Gateways)  
2
3
Click the gateway host name to view the details corresponding to that  
particular gateway.  
To modify gateway information, enter the IP address or MADDR of the  
gateway in the textbox (required). For example,  
nortelnetworks.com;maddr=<IP_address>  
4
5
To modify gateway description, enter changes in the Gateway Type text box.  
Click Save. (To clear the text fields, click Clear.) The Gateways window  
appears with a list of all provisioned gateways, including the gateway  
modified.  
6
To delete a gateway, click the Delete link next to the gateway. This removes  
the provisioned gateway from the database and the system.  
Adding a gateway route  
This option allows an administrator to add gateway route information to associate  
the route with a domain.  
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To add a new gateway route  
1
Click the Gateways > Add Route menu option. The Create new gateway  
route appears (Figure 87).  
Figure 87 Create new gateway route window  
2
3
4
Enter a description for the new route in the Description text box.  
From the drop down list, select a domain to associate with the route.  
Click Save. (To clear the text fields, click Clear.) The Gateway Routes  
window appears (Figure 88).  
Figure 88 Gateway routes window  
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Listing a gateway route  
The List Gateways menu option allows the administrator to view routing  
information for a particular gateway.  
To view details for a gateway route  
1
Click the Gateways > List Routes menu option. The Gateway Routes  
window appears (Figure 89).  
Figure 89 Gateway routes window (List Routes)  
2
3
Click on a route to view gateway route information corresponding to that  
particular route.  
To modify the gateway route information, enter a description for the route in  
the Description text box.  
4
5
From the drop down list, select a domain to associate with the route.  
Click Save. (To clear the domain field, click Clear.) The Gateway Routes  
window appears.  
6
A gateway can have multiple trunk groups. To reorder trunk groups associated  
to a route, click the Re-Order link next to the gateway route. The Reorder  
Trunk Groups window appears (Figure 90).  
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Chapter 11 Gateways 141  
Figure 90 Reorder trunk groups window  
7
8
In the Current Choices Available window, click UP or DOWN to reorder the  
trunk group.  
Click Reorder to submit the trunk group reorder to the database.  
Adding a gateway trunk group  
The AddTrunkGroup menu option allows an administrator to add gateway trunk  
group information for a particular gateway. Multiple trunk groups can be assigned  
to a gateway route. If the first trunk group is busy or not available, then the next  
trunk group in the route is used.This option allows an administrator to view  
information for all available gateway trunk groups for a particular gateway.  
To add a new trunk group  
1
Click the Gateways > AddTrunkGroup menu option. The Create new  
trunkgroup window appears (Figure 91).  
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Figure 91 Create new trunkgroup window  
2
3
4
5
Select a gateway from the Gateway drop down list.  
Select a route from the Route drop down list.  
Enter a description for the new trunk group in the text box provided.  
Click Save. (To clear information from any of the fields, click Clear.) The  
TrunkGroups window appears (Figure 92).  
Figure 92 TrunkGroups window  
To view trunk group information  
1
Click Gateways > List TrunkGroups. The TrunkGroups window appears  
(Figure 93).  
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Chapter 11 Gateways 143  
Figure 93 TrunkGroups window (List TrunkGroup)  
2
3
To delete a gateway trunk group, click the Delete link next to the gateway.  
At the confirmation dialog box, click OK to delete the gateway trunk group  
information from the database for that particular gateway, or Cancel to return  
to the Trunk Groups window.  
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145  
Appendix A  
Provisioning Sequence  
This section describes the sequence of provisioning tasks.  
Sequence of tasks  
The following table provides a step by step provisioning sequence.  
Table 22 SIP Provisioning Client provisioning sequence  
Step Description  
Task(s)  
Menu option  
1.  
Define roles and rights and  
use to create a new  
Administrator  
Add role and assign rights  
Admins  
Add Role  
Add Admin  
Admins  
Add Admin  
2.  
Define new domain(s)  
Add domain(s)  
Domains  
Add Domain  
Add sub domain(s) if required Domains  
domainxx.com  
Add Sub-domain  
Services  
Define Service Parameters  
Services  
Assign Services  
3.  
Define service parameters  
and assign to domain(s)  
Define service parameters  
Assign services to primary  
domain  
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146 Provisioning Sequence  
Step Description  
Task(s)  
Menu option  
4.  
Define domain service  
package  
Create service package  
Domains  
domainxx.com  
Service Package  
Create Package  
Assign service package(s) to Domains  
domain(s) and sub-domain(s)  
domainxx.com  
Service Package  
Assign Package  
Assign services to  
subdomain(s)  
Domains  
domainxx.com  
Service Package  
Assign Services  
5.  
6.  
Define voice mail servers and Add voice mail server (SIP-/  
Voice Mail  
Add Voice Mail Server  
assign to domain(s)  
Trunk-/Line-based) and  
assign to domain(s)  
If not deployed using the  
System Management  
Add IPCM cluster  
IPCM  
Add IPCM Cluster  
IPCM  
Console, add IPCM cluster  
and assign to domain  
Assign IPCM to domain  
List IPCM  
Cluster>Details>Assign to  
Domain>Assign Domain  
7.  
8.  
Add domain status reason(s) Add reason  
Domains  
domainxx.com  
Status Reasons  
Add Reason  
Add users to domain(s)  
Add user(s)  
Domains  
domainxx.com  
Users  
Add User  
Add user(s) to subdomains  
Domains  
domainxx.com  
SubDomain  
SubDomainYY  
Users  
Add User  
Add i2004 device properties  
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Provisioning Sequence 147  
Step Description  
Task(s)  
Menu option  
Domains  
domainxx.com  
9.  
If not autoprovisioning, assign Add device  
devices to domain(s)  
Devices  
Add Device  
Assign users to a device  
Domains  
domainxx.com  
Devices  
List Device  
Assign User  
10.  
11.  
Define gateway, gateway  
routes, and trunk groups  
Add gateway  
Gateways  
Add Gateway  
Gateways  
Add Route  
Gateways  
Add TrunkGroup  
Add gateway route  
Add trunkgroup  
Define domain telephony  
routes and parameters  
Add routing Class of Service Domains  
(COS)  
domainxx.com  
Telephony Routes  
Routing COS  
Add telephony routes: Private, Domains  
SIP, or Gateway  
domainxx.com  
Telephony Routes  
Add Telephony Route  
Change routing parameters  
Domains  
domainxx.com  
Telephony Routes  
List Telephony Route  
Add route list  
Ban users  
Domains  
domainxx.com  
Telephony Routes  
RouteList  
12.  
Define banned users for a  
domain  
Domains  
domainxx.com  
Banned Users  
Ban User  
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148 Provisioning Sequence  
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149  
Appendix B  
Understanding Telephony Routing and  
Translations  
This section provides a basic understanding of how to provision telephony style  
translations interworking with traditional circuit-switched network routing.  
Topics in this section:  
“Telephony routing concepts” on page 149  
“Telephony routing terminology” on page 152  
“Sample dial plan for telephony routes” on page 155  
“Telephony route provisioning scenario and examples” on page 158  
Telephony routing concepts  
The basic function of any communication equipment is routing, and SIP-based  
systems add additional requirements to routing when compared to traditional  
circuit-switched network-based routing. Routing in the VoIP networks adds the  
complexity of integrating with the traditional circuit-switched networks and  
domain-based VoIP/SIP networks.  
Subdomains  
The concept of subdomains allows the domain administrator to group users into a  
smaller routing groups. For example, Nortel Networks can have a single system  
supporting a Richardson domain and an Ottawa domain. Users within Nortel  
Networks are assigned to subdomains, but can be reached by just using the top  
domain. The concept of SIP subdomains is similar to the email domains used  
within an enterprise.  
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The following figure shows an example of the subdomain concept (Figure 94).  
Figure 94 Example domain tree  
Tip: Subscribers do not have to be aware of subdomains. Subdomains  
are used only to group subscribers for partitioning telephony  
translations.  
Class of Service  
Class of Service (COS) screens and restricts users from accessing certain  
telephony routes within a domain and subdomain. The following restrictions are  
imposed on the use of COS:  
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Appendix B Understanding Telephony Routing and Translations 151  
A single COS value can be assigned to a list of routes, domain, subdomain or  
a subscriber.  
A domain, subdomain or subscriber has access to the routes if the COS value  
equal or higher is assigned to a list of routes.  
The domain based COS value is used to determine the telephony routes for  
incoming sessions from gateways or other SIP networks. In this scenario the  
domain will not host the originator of the call so the domain’s routing  
behavior is applied.  
If the From: header and Request URI: do not have the same domain, then the  
domain specified in the Request URI: is used to determine the telephony  
routes.  
If the From:, To:, and Request URI: domains are the same, then the  
subscriber’s COS and the domain assigned to the subscriber is used to  
determine the telephony routes. If the user does not have any COS value then  
the domain’s COS value in which the user actually exists is applied.  
The same COS values can be assigned to a domain, subdomain and a  
subscriber. This allows or restricts a subscriber in the subdomain to access  
routes in the parent and other subdomains.  
SIP Application Module routing stages  
The following figure (Figure 95) shows the relationship between the telephony  
routing stages provided by the SIP Application Module. If the COS value of the  
subscriber and subdomain route do not match, then the parent and subsequent  
subdomains are checked for routes with the same, or higher, COS value.  
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Figure 95 SIP Application Module routing stages  
Telephony routing terminology  
This section describes the following telephony routing terms:  
route lists  
routes  
Route lists  
Route lists consist of specific telephony routes for a local domain, gateways, or  
another SIP domain. The following additional options can be applied to routes in  
the route list:  
allow all incoming sessions from other domains  
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Appendix B Understanding Telephony Routing and Translations 153  
block all incoming sessions from other domains  
allow all incoming sessions from other subdomains  
block all incoming sessions from other subdomains  
Example of telephony route redirection between domains  
The following figure (Figure 96) shows an example of a telephony redirect  
imposed by a route list in the xyz.com domain when a session originates from the  
abc.com domain.  
Figure 96 Route list redirect (domain to domain)  
Example of telephony route redirection between subdomains  
The following figure (Figure 97) shows an example of a telephony redirect  
between two subdomains. In this scenario the originating subscriber’s subdomain  
is checked when a route list in the subdomain otta.abc.com sets the subdomain  
redirect option.  
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Figure 97 Route list redirect (between subdomains)  
Routes  
Telephony routes provide the actual dial plan to a particular destination. The  
telephony routing includes routes for a private digit dial plan, routes to gateways,  
and telephony style routing between SIP domains and subdomains. The three  
types of telephony routes available are (in order of telephony route selection):  
Private  
SIP  
Gateway  
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Appendix B Understanding Telephony Routing and Translations 155  
Private telephony routes  
Private telephony routes are used for abbreviated telephony style digit dial plans.  
For example, private routes can be set up for a 5-digit dial plan between selected  
subscribers in a domain.  
SIP and gateway routes  
SIP and gateway routes provide access to a gateway, and SIP routes pointing to  
other SIP domains and subdomains for inter-SIP domain routing using telephony  
style dial plans.  
Sample dial plan for telephony routes  
This section provides a sample dial plan for provisioning the following telephone  
routes:  
Private telephony route for using 5-digit translations between subscribers.  
SIP telephony route for using PSTN style number plan to call between  
different domains.  
Gateway route to make local (PSTN) calls using 9+ prefix.  
Provisioning domains, subscribers, and COS  
Before implementing the dial plan a list of domains, subscribers, and COS must  
be provisioned for the following domains in the example:  
abc.com - (includes the rich.abc.com and otta.abc.com subdomains)  
xyz.com  
The following figure (Figure 98) shows the domains and subscribers in the sample  
dial plan.  
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Figure 98 Domains/subscribers used for the sample dial plan  
The domain abc.com defines the following three types of COS:  
lobby_dial_plan (lowest COS)  
local_dial_plan  
company_dial_plan (highest COS)  
A subscriber that has company_dial_plan COS will also have access to routes  
defined for lobby_dial_plan and local_dial_plan. The domain xyz.com defines a  
COS, all_dial_plans, that is available to all subscribers in this domain.  
Tip: It is assumed that each subscriber is assigned a 10-digit alias in all  
the domains.  
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Example dial plan assignments  
The following table (Table 23) provides an example list of subscribers, subscriber  
aliases, and COS assignments in the domains.  
Table 23 Example dial plan data  
Subscriber  
Alias  
Domain  
Subdomain  
COS  
lobby  
paul  
john  
fred  
9726850000  
9726851000  
9726852000  
2146842000  
abc.com  
abc.com  
abc.com  
xyz.com  
rich.abc.com  
rich.abc.com  
otta.abc.com  
lobby_dial_plan  
local_dial_plan  
company_dial_plan  
all_dial_plans  
Dial plan description  
The dial plan for the two parent domains (abc.com and xyz.com) uses the various  
digit ranges as described in the following table (Table 24).  
Table 24 Dial plan for abc.com and xyz.com  
Number  
digits  
Type of  
Subdomain Route  
Route Name Description  
From To  
Domain  
5_digit_plan 5+4 digit to reach  
subscribers in  
5
5
5
abc.com  
rich.abc.co Private  
m
rich.abc.com  
pstn_route  
9+10 digits to reach a  
PSTN subscriber. Strip  
the 9 before reaching  
the subscriber  
9
9
11  
abc.com  
Gateway  
route_to_xyz. 6+7 digits to reach  
com xyz.com from abc.com  
6
6
6
6
8
8
abc.com  
xyz.com  
SIP  
SIP  
route_to_abc 6+7 digits to reach  
.com abc.com from xyz.com  
The following provides further description of the dial plans:  
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The 5+4 digit dial plan allows the subscribers in subdomain rich.abc.com to  
reach other subscribers in the same subdomain using only 5 digit dial plan (for  
example, 5+1000). Since the 5 digit dial plan is only required in the  
subdomain, it is assigned as a private route.  
The 9+10 digit dial plan is used to access the SIP PRI Gateway to terminate  
calls to a PSTN network, so this is assigned a gateway route.  
The 6+7 digit dial plan is used for communicating between two parent  
domains, therefore this digit range is assigned to SIP routes.  
Associating COS and the route  
The COS and routes described earlier in this section are tied together to provide  
the user restrictions in the following table (Table 25). This table shows that the  
lobby_dial_plan COS will only have access to the 5+4 digit dialing, therefore,  
restricting phones in a lobby to only 5 digit dialing only. The company_dial_plan,  
having the highest COS, will have access to all three routes..  
Table 25 Association between COS and Routes  
COS  
Routes  
lobby_dial_plan  
local_dial_plan  
5+4 digit to reach subscribers in rich.abc.com  
5+4 digit to reach subscribers in rich.abc.com  
9+10 digits to reach a PSTN subscriber. Strip the 9 before  
reaching the subscriber  
company_dial_plan  
5+4 digit to reach subscribers in rich.abc.com  
9+10 digits to reach a PSTN subscriber. Strip the 9 before  
reaching the subscriber  
6+7 digits to reach xyz.com from abc.com  
all_dial_plans  
6+7 digits to reach abc.com from xyz.com  
Telephony route provisioning scenario and examples  
The steps in this section provide a scenario and sequence to follow when using the  
sample dial plan to provision telephony routing.  
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“Step 1: Create the parent and subdomains” on page 159  
“Step 2: Create the COS” on page 161  
“Step 3: Create the subscribers and assign the COS” on page 162  
“Step 4: Create gateways, gateway routes and trunk groups” on page 165  
“Step 5: Create the telephony routes” on page 166  
“Step 6: Create the telephony route lists and assign COS” on page 174  
“Step 7: Test the telephony routing configuration” on page 178  
Step 1: Create the parent and subdomains  
To create the parent and subdomains  
1
2
3
Login in to the SIP Provisioning Client.  
Click the Domains > Add domain menu option.  
Create a domain called abc.com. See the following figure (Figure 99) and  
locate the parent domain in the SIP Provisioning Client.  
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Figure 99 Parent domain provisioning menu  
parent domain  
(abc.com)  
menu option to create  
subdomains  
menu option to create  
and reorder COS  
menu option to add  
telephony routes  
menu option to add  
a route list  
4
Create the following subdomains:  
— rich.abc.com  
— otta.abc.com  
— xyz.abc.com  
The subdomains assume the default parameters from the parent domain. After  
adding the subdomains, the SIP Provisioning Client will look similar to the  
following figure (Figure 100).  
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Appendix B Understanding Telephony Routing and Translations 161  
Figure 100 Subdomain provisioning  
subdomains  
Step 2: Create the COS  
To create the COS  
1
Click the Domains > abc.com > Telephony Routes > Routing COS menu  
option.  
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2
Add three COS to the parent domain abc.com, and re-order the COS as shown  
in the following figure (Figure 101). The highest COS is company_dial_plan.  
Tip: The COS list is only created in the parent domain.  
Figure 101 Creating COS  
3
Click Save.  
Step 3: Create the subscribers and assign the COS  
To create the subscribers and assign the COS to the subscribers  
1
Create a service package and assign the service package to the parent and  
subdomains. An example service package assignment is shown in the  
following figure (Figure 102).  
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Appendix B Understanding Telephony Routing and Translations 163  
Figure 102 Assigning service packages to subdomains  
2
Create the subscribers for rich.abc.com, otta.abc.com and xyz.com. The  
following figure (Figure 103) shows a user called lobby added to the  
subdomain rich.abc.com.  
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Figure 103 Adding a subscriber with COS  
SIP user  
10 digit alias  
10 digit calling  
number display used  
for PSTN calls  
COS assigned to this  
user  
3
4
Click Save.  
Add the remaining subscribers to rich.abc.com, otta.abc.com and xyz.com  
domains.  
5
Click Save.  
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Step 4: Create gateways, gateway routes and trunk groups  
Before a gateway route can be assigned to a dial plan, the gateway and trunk  
groups must be created. A SIP PRI gateway is used with following parameters  
provisioned on the gateway:  
the gateway’s IP address is 192.12.12.12.  
the trunk group assigned to domain abc.com is pstn_trunk_group for facing  
with the PSTN network.  
This example uses a gateway route to terminate 9+10 digits to PSTN subscribers.  
To create a gateway, gateway route and trunk group  
1
Click the Gateways > Add Gateway menu option to create a logical link to  
the PRI Gateway. Use the data above to create the gateway as shown in the  
following figure (Figure 104).  
Figure 104 Adding a gateway to abc.com  
2
Click the Gateways > Add Route menu option to provision a gateway route  
for abc.com (for example, Route to abc.com). This addition is shown in the  
following figure (Figure 105). Note that the route created here is used later in  
the dial plan to route calls to this gateway.  
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Figure 105 Adding a gateway route  
3
Click the Gateways > Add TrunkGroup menu option to assign the gateway  
trunk group to this route as shown in the following figure (Figure 106).  
Figure 106 Adding a trunk group to a gateway route  
4
Click Save.  
Step 5: Create the telephony routes  
Using the dial plan defined earlier (Table 24), you can now create the telephony  
routes. First, create the private routes so that subscribers in subdomain  
rich.abc.com can reach each other using 5+xxxx digits.  
Using the subscriber information in the following table (Table 26), a lobby phone  
(user = lobby) can reach user the user named paul by dialing 51000.  
Table 26 Subscribers in rich.abc.com  
Subscriber  
Alias  
Subdomain  
COS  
lobby  
paul  
9726850000  
9726851000  
rich.abc.com  
rich.abc.com  
lobby_dial_plan  
local_dial_plan  
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Adding a private route to subdomain rich.abc.com  
Note: Privates routes in this example are for the rich.abc.com subdomain  
only.  
To add a private route  
1
Click the Domains > abc.com > rich.abc.com > Add Telephony Route sub  
menu option as shown in the following figure (Figure 107).  
Figure 107 Menu to add routes in subdomain  
2
Provision a 5 digit dial plan as shown in the following figure (Figure 108).  
The prefix 97268 is used to create the 10 digit alias assigned to the subscriber.  
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Figure 108 Adding a private route to rich.abc.com  
3
Click Save.The following window appears (Figure 109).  
Figure 109 Private route created  
4
5
Click the Change Parameters link.  
Set the Subscriber Not Found option to STOP. The STOP option terminates  
the call if a subscriber with the prefix 97268+5 digits cannot be found in the  
abc.com domain.  
Figure 110 Setting the Subscriber Not Found option  
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6
Click Save. The following window appears (Figure 111) indicating a 5-digit  
private route is created for subdomain rich.abc.com.  
Figure 111 Listing a private route  
Adding SIP routes to abc.com and xyz.com  
In this example, two SIP routes are added. The routes use the dial plan defined in  
the previous sections to communicate between abc.com and xyz.com domains  
using PSTN style dialing. The two SIP routes created to reach each domain are  
described in the following table (Table 27).  
Table 27 Data for SIP routes  
Number  
digits  
Type of  
Route  
Route Name  
Description  
From To  
Domain  
route_to_xyz.com  
6+7 digits to reach  
xyz.com from abc.com  
6
6
8
abc.com  
SIP  
route_to_abc.com  
6+7 digits to reach  
abc.com from xyz.com  
6
6
8
xyz.com  
SIP  
Tip: The SIP routes are added to the parent domains and not to the  
subdomains. This option allows the subdomains to access the common  
routes defined in the parent domain. For this example, rich.abc.com and  
otta.abc.com can use the same route defined in abc.com.  
To create a SIP route  
1
Click the Domains > abc.com > Telephony Routing > Add Telephony  
Route menu option. In this route, the 6 prefix is removed and 214 is prefixed  
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to determine the user alias in xyz.com. The following figure shows a SIP route  
added to the abc.com domain (Figure 112).  
Figure 112 Adding a SIP route to abc.com  
2
Click the Change Parameters link to set the route list options (Figure 113).  
Figure 113 SIP route change parameters  
3
Provision the SIP route to point to the xyz.com domain as shown in the  
following figure (Figure 114). The STOP option terminates the call if a  
subscriber with the prefix 214+7 digits cannot be found in xyz.com.  
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Figure 114 SIP route options  
4
5
Click Save.  
Click the Domains > abc.com > Telephony Route > List TelephonyRoute  
menu option to view a list of the telephony routes as shown in the following  
figure (Figure 115).  
Figure 115 Listing SIP routes in abc.com  
6
Add a SIP route to xyz.com to point to abc.com using the data shown in the  
following figure (Figure 116).  
Figure 116 SIP route to abc.com  
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Adding a gateway route to abc.com  
Add the dial plan to gateway route as shown in Table 24 to domain abc.com (9+  
10 digits).  
To add a gateway route to abc.com  
1
Click the abc.com > Telephony Routing > Add Telephony Route menu  
option to assign the gateway route to a dial plan, In this route the 9 prefix is  
removed before sending the request to the PRI Gateway as shown in the  
following figure (Figure 117).  
Figure 117 Adding a gateway route  
2
3
Click Save.  
Click the Change Parameters link on the route listing window (Figure 118).  
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Figure 118 Gateway change parameters  
4
Provision the gateway route and options as shown in the following figure  
(Figure 119). By selecting the PUBLIC option, the public charge ID assigned  
to a subscriber in abc.com is sent as the calling number to the PSTN.  
Figure 119 Gateway route options  
5
Click the List Telephony Routes menu option to view the list of telephony  
routes. Notice that there are now two routes in this listing (Figure 120).  
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Figure 120 Listing abc.com routes  
Step 6: Create the telephony route lists and assign COS  
At this point, all the subscribers with COS and telephony routes have been  
created. However, there is no association between the subscribers and telephony  
routes. The route lists provide the association between subscribers COS and  
telephony routes that can be used by the subscribers.  
To create the telephony route lists and assign COS to each list  
1
2
3
Create a route list named abc_route_list using the COS company_dial_plan  
and assign the two routes in abc.com to this route list.  
Click the Domains > abc.com > Telephony Routes > Add RouteList menu  
option.  
Enter the data as shown in the following figure (Figure 121). The BLOCK  
option blocks calls from other domains accessing the routes in this Route List.  
For example, a subscriber in xyz.com will be blocked if  
9972685121@abc.com is sent. Since the COS of this route list is  
company_dial_plan, only the subscriber named john in the abc.com domain  
will have access to this route list.  
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Figure 121 Adding a route list to abc.com  
block calls from other  
domains accessing  
this route  
telephony route  
selected  
4
Create a route list for the local calling plan and assign it the local_dial_plan  
COS. This route will be available to subscribers named john and paul in the  
abc.com domain. Since local_dial_plan is a lower COS than  
company_dial_plan, both route lists are available to the subscriber named  
john (Figure 122).  
Figure 122 Adding a route list with PSTN route to abc.com  
5
Click the Domains > abc.com > Telephony Routes > List RouteLists menu  
option to view the route list as shown in the following figure (Figure 123).  
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Figure 123 Listing of abc.com route lists  
6
Create a route list in subdomain xyz.com and assign it the COS of  
all_dial_plans. The Route List of xyz.com is shown in the following figure  
(Figure 124).  
Figure 124 Listing of xyz.com route lists  
7
Click the Domains > abc.com > rich.abc.com > Telephony Routes > Add  
RouteList menu option to add a route list that contains the private route  
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Appendix B Understanding Telephony Routing and Translations 177  
created earlier for the 5 digit dialing between the subscribers in the  
rich.abc.com subdomain. The following figure (Figure 125) shows the new  
route list.  
This route will be available to subscribers named paul and lobby in the  
subdomain rich.abc.com only. Since the COS for lobby_dial_plan is lower  
than the COS for local_dial_plan, subscriber paul has access to this route list.  
Figure 125 Adding a route list to rich.abc.com  
8
Click Save. The following figure appears (Figure 126).  
Figure 126 Listing of rich.abc.com route lists  
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Step 7: Test the telephony routing configuration  
At this point all subscribers, COS, route, and route lists have been provisioned.  
Before making any calls, telephony routing can be verified using the Translations  
Tool menu option available in the domains and subdomains.  
To check the telephony routing  
1
Click the abc.com > rich.abc.com > TelephonyRoutes > Translations Tool  
menu option. Verify that the subscriber named lobby can reach the subscriber  
named paul by dialing 51000.  
The data for translations trace and results are shown in the following figure  
(Figure 127).  
Figure 127 Subdomain 5 digit translations trace  
Paul’s 5 digit  
extension  
Calling from  
subscriber lobby  
2
Verify that the subscriber paul can reach the PSTN network using the 9+10  
digit dial plan. The data for translations trace and results are shown in the  
following figure (Figure 128).  
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Appendix B Understanding Telephony Routing and Translations 179  
Figure 128 Gateway route translations trace  
to PSTN  
PRI gateway  
trunk group  
IP address of the PRI  
gateway  
3
Verify that subscriber named john in subdomain otta.abc.com can reach the  
SIP domain xyz.com subscriber named fred with the 214684200 alias using  
the 6+8 digit dial plan. The data and results are shown in the following figure  
(Figure 128).  
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Figure 129 SIP route trace from abc.com to xyz.com  
Fred’s 10 digit alias  
4
Verify that subscriber named paul in subdomain rich.abc.com cannot reach  
the SIP domain xyz.com subscriber named fred with the 214684200 alias  
using the 6+8 digit dial plan.The data and results are shown in the following  
figure (Figure 130). Paul only has a COS of local_dial_plan, therefore the  
route_to_xyz route list is not available.  
Figure 130 Restricted route trace  
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181  
Appendix C  
911 Emergency Notification  
Note: You must have telephony routing and domain management  
provisioning rights to perform the tasks in this section.  
This section describes how to set up 911 emergency notification using the SIP  
“911 emergency notification terms and concepts” on page 181  
“Setting up Instant Message Notifications” on page 182  
“Using the SIP Provisioning Client to add emergency subscribers and  
telephony routes” on page 183  
“Using the SIP Personal Agent to route 911 emergency notifications” on  
page 192  
911 emergency notification terms and concepts  
The SIP Application Module supports sending Instant Message (IM) notifications  
to a specified On Site Notification (OSN) subscriber whenever a user makes a call  
to an emergency number such as 911. This service is provided by using the same  
mechanism that allows users to push web pages and/or email links back to the  
originator of a call.  
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A Public Safety Answering Point (PSAP) is an agency or group of agencies  
designated and authorized to receive and respond to 911 emergency calls  
requiring one or more public services: Police, Fire, Emergency Medical Service,  
or all three.  
The OSN is required so that the OSN staff can prepare for the arrival of the  
emergency response team, and direct them to the location of the 911 caller when  
they arrive on site.  
An emergency subscriber is the logical representation of a PSAP/OSN pair. When  
a user dials an emergency number (for example, 911), the call is mapped (using  
telephony routing rules) to the correct emergency subscriber for that subdomain.  
The routing rules for that emergency subscriber (created in the SIP Personal  
Agent) then instruct the SIP Application Module to both route the call to the  
correct PSAP and to send an Instant Message notification to the correct OSN  
subscriber. There is, therefore, a one-to-one relationship between a subdomain  
and the PSAP/OSN pair.  
Setting up Instant Message Notifications  
In order to properly set up IM notifications to an OSN subscriber, the domain  
administrator should divide the domain into geographical areas. A subdomain  
should then be created for each of these areas. If there is only one geographical  
area, no subdomains need to be created.  
When a subscriber places an emergency call, the OSN subscriber assigned to the  
same subdomain (area) as the originator is notified. It is up to the administrator to  
decide how large or small to make the subdomain (area).  
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Appendix C 911 Emergency Notification 183  
To set up Instant Message (IM) notifications, the following information must be  
provisioned using both the SIP Provisioning Client and the SIP Personal Agent.  
Use this client...  
To set up...  
SIP Provisioning Client  
an emergency subscriber per subdomain  
an OSN subscriber per subdomain  
the 911 route in each subdomain to point to each  
emergency subscriber for that subdomain  
the PSAP route to route to the correct PSAP for  
each subdomain  
SIP Personal Agent  
routing rules to the emergency subscriber for  
an emergency telephone number to route the call  
to the proper gateway serving the PSAP.  
the OSN subscriber to receive the IM notification.  
See the SIP Personal Agent Getting Started Guide  
for more information on using the SIP Personal  
Agent.  
Tip: Since telephony routes are only unique within a subdomain, you cannot  
have more than one OSN location per subdomain.  
subscribers and telephony routes  
“Creating a COS for the 911 notification” on page 184  
“Adding a gateway, gateway route, and trunk group” on page 185  
“Adding a private telephony route” on page 187  
“Adding a gateway telephony route” on page 189  
“Adding an emergency subscriber” on page 190  
“Adding an OSN subscriber” on page 192  
“Setting up a route list for the emergency subscriber” on page 192  
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“Setting up a rule for the emergency notification route” on page 193  
Creating a COS for the 911 notification  
As a first step, create a new COS for the 911 notification and assign it to the parent  
domain.  
To create the COS  
1
Click the Domains menu option. Click the subdomain for which you want to  
set up the COS (for example, rich.abc.com)  
2
3
4
Click the Telephony Routes > Routing COS menu option.  
Enter a name and a description for the COS.  
Click Save.The COS appears in the Current Choices Available window  
(Figure 131).  
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Appendix C 911 Emergency Notification 185  
Figure 131 Creating a COS for the 911 notification  
5
Click UP or DOWN to reorder the COS in the Current Choices Available  
window. The COS for 911 emergency notification should be the highest in the  
reorder list. This indicates that the COS has a lower number, which means a  
higher COS value, or service, and ensures that everyone will receive 911  
emergency notification support.  
Adding a gateway, gateway route, and trunk group  
To create a PSAP gateway, gateway route, and trunk group per domain or  
1
2
Click the Gateways > Add Gateway menu option to create a logical link to  
the PRI gateway.  
Enter the IP address or MADDR (short address) of the gateway  
(Figure 132).  
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Figure 132 Adding a new gateway  
3
Click Save. The new gateway appears in the list of gateways (Figure 133).  
Figure 133 List of gateways  
4
Click the Gateways > Add Route menu option to provision a gateway route  
for the parent domain, for example, Route for abc.com (Figure 134). The  
route created here will route 911 calls to this gateway.  
Figure 134 Creating a new gateway route  
5
6
Click Save.  
Click the Gateways > Add TrunkGroup menu option to assign the gateway  
trunk group to this route (Figure 135).  
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Figure 135 Creating a new trunkgroup  
Adding a private telephony route  
An emergency telephony route must be set up that is based on the unique  
emergency subscriber name provisioned per domain or subdomain.  
To create a private telephony route for when a user dials 911  
1
2
Click the Domains menu option.  
Click the domain subdomain for which you want to set up the private  
telephony route (for example, rich.abc.com)  
3
Click the Telephony Routes > Add Telephony Route menu option.  
Complete the fields for the new telephony route in the The Create New  
Telephony Route window (Figure 136). The prefix field (this is the number  
that goes in front of the 911) must be changed based on the username of the  
pseudo emergency subscriber. The prefix field is only field that is different  
between every telephony route in every subdomain.  
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Figure 136 Create new telephony route window  
4
5
Click Save. The Telephony Route Listing for the domain appears (for  
example, abc.com domain).  
Click the Change Parameters link and select PRE-TRANSLATE from the  
drop down list. Click Save (Figure 137).  
Figure 137 Change parameters  
6
Click the Telephony Routes > Add RouteList menu option. Complete the  
fields for the new telephony route in the The Create New Telephony Route  
window (Figure 138).  
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Figure 138 Creating a new route list  
Adding a gateway telephony route  
To add a gateway telephony route that routes to the proper gateway serving the  
PSAP for the OSN location when a user dials 911  
1
2
Click the abc.com > Telephony Routing > Add Telephony Route menu  
option to assign the gateway telephony route for the PSAP route.  
Complete the fields for the telephony gateway route to the PSAP in the Create  
New Telephony Route window (Figure 139). The number in the From Digits  
and To Digits fields (for example, 911) corresponds to the number that will be  
provisioned in the SIP Personal Agent route list.  
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Figure 139 Create gateway telephony route to PSAP  
3
Click Save.  
Click the Change Parameters link on the route listing window.  
4
5
Provision the gateway route and options as shown in the following figure  
(Figure 140). By selecting the PUBLIC option, the public charge ID assigned  
to a subscriber in abc.com is sent as the calling number to the PSTN.  
Figure 140 Gateway route options  
6
Click the List Telephony Routes menu option to view the list of telephony  
routes. Notice that there are now two routes in this listing.  
Adding an emergency subscriber  
To add an emergency subscriber to a domain or subdomain  
1
Click the Domains menu option.  
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Appendix C 911 Emergency Notification 191  
2
3
Select a domain (or a subdomain of a domain).  
Click the User > Add User sub menu option. The Add new user window  
appears (Figure 141).  
Figure 141 Add user - emergency subscriber  
4
5
6
7
Enter a name for the emergency subscriber (for example, 55911).  
Enter and confirm a password for the subscriber.  
Enter additional information for the emergency subscriber (optional).  
Select the COS for the subscriber that you created earlier.  
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8
Click Save. A message appears that the user was added successfully.  
Adding an OSN subscriber  
To add an OSN subscriber to a domain or subdomain  
1
2
3
Click the Domains option.  
Select a domain (or a subdomain of a domain).  
Click the User > Add User sub menu option. The Add new user window  
appears.  
4
5
6
7
Enter a name for the OSN subscriber (for example, guarddeskA).  
Enter and confirm a password for the OSN subscriber (required).  
Enter additional information for the OSN subscriber (optional).  
Click Save. A message appears that the user was added successfully.  
Using the SIP Personal Agent to route 911 emergency  
notifications  
The instant message sent to the Guard Desk is provisioned through the Personal  
Agent just as you would if you wanted to push a web page back to the originator  
while the terminator is being alerted.  
Use the SIP Personal Agent to add a new rule to the emergency subscriber’s Call  
Manager route list.  
Setting up a route list for the emergency subscriber  
To set up a route and ring list to send an IM to a specific OSN location and to call  
an emergency number at the same time  
1
2
Log in to the SIP Personal Agent.  
Click the Call Manager icon on the left of the SIP Personal Agent screen.  
The Welcome to Call Manager home screen displays.  
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3
4
5
6
Click ListRoutes tab from the Call Manager Home screen. The Ring List  
Details screen displays.  
Enter a name for the emergency route in the Ring list name text box (for  
example, Emergency notification).  
From the drop down list, select the maximum number of times to ring the  
phone.  
In the first text box in the column, enter a unique emergency number. This  
number must correspond to the number provisioned in the gateway telephony  
route that routes to the proper gateway serving the PSAP for the OSN location  
(for example, 6911).  
7
8
Select SIP address from the drop down list.  
In the second text box in the first column, enter the SIP address of the OSN  
subscriber. This is name of the target user who receives the IM notification,  
for example:  
sip:guarddeskA@abc.com.  
9
Select Web address from the drop down list.  
10 Click Add Route.The Add Route button updates the online user database  
with your the emergency subscriber and routing information.  
Setting up a rule for the emergency notification route  
Once a route is set up, a screening rule must be created that uses the defined  
emergency notification ring list to define the activation of 911 calls.  
Tip: The 911 notification is not activated until a rule has been applied.  
To set up a screening rule for the first time  
1
Click ListRules from the Call Manager screening and routing screen. The  
Welcome to Call Manager screen displays.  
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2
3
4
5
6
Click the create a screening rule link. The Call Manager Rule Details screen  
displays.  
Enter a name for emergency rule in the Rule name text box (for example, 911  
rule).  
Click the Active check box to select that the rule should appear in the Active  
Rules window.  
Select the emergency route from the pull-down menu next to the Active route  
field.  
Click Save Rule.  
Tip: If you selected the Active check box in the Call Manager Rule  
Details screen, then the rules you just programmed now appear in the  
Active rules window in the Call Manager Home screen. Every time 911  
is called, the Call Manager checks the rules to match and route the 911  
calls appropriately.  
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195  
Numbers  
B
911 emergency notification 181  
ban list  
viewing a ban list 36, 37  
Behind Firewall parameter 29  
A
Activation 45  
activation key 45  
C
Active Proxy parameter 30  
adding an administrator 19  
Administration  
overview 11  
Class of Service  
administrator  
adding 19  
deleting 21  
viewing details 20  
Client contact parameter 53  
aliases  
deleting 107, 109  
listing 107, 109  
text xx  
Alpha parameter 29  
COS (Class of Service) 122  
CPL scripts 69  
Always Use Media Portal parameter 31  
Application Server parameter 53  
assigned capacity 45  
Creating a new provisioning role 16  
CS2000 133  
Audience xix  
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196 Index  
D
gateway  
adding a server 136  
listing 140  
Default Proxy Host parameter 30  
Description parameter 52  
listing a server 137  
trunk groups  
domain  
foreign 25  
listing 142  
reordering 140  
gateway connection parameters  
listing 33  
Description 59  
URI to query to find gateway 59  
listing users 105  
modifying 33  
parameters 28  
root 24  
H
types 23  
Domains parameter 53  
i2004 Internet Telephone  
adding 99, 111  
E
listing 116  
emergency notification 181  
F
Fax parameter 93  
Idle Display 28  
foreign domain  
adding 33  
IPCM  
adding 42  
definition 25  
deleting 36  
example 26  
listing 33  
assigning to a domain 44  
definition 41  
deleting 46, 47  
modifying 45  
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Index 197  
IPCM Name parameter 43  
L
how to use 5  
Line based voice mail server general parameters  
Pager parameter 93  
Maximum Call Duration 65  
parameters  
user 93  
Line 67  
Address 63  
Password 63  
Port 63  
Prefix parameter (telephony route) 125  
product support xxii  
Listing 128  
listing a user 105  
listing administrators 20  
Locale parameter 94  
provisioning  
creating a new role 16  
rights 11  
roles 11  
sequence of tasks 22  
tasks 11, 12  
M
provisioning rights  
examples 15  
Managing 91  
Max Number of Digits parameter (telephony  
route) 125  
understanding 15  
Maximum Number of Presence Subscriptions  
Accepted parameter 31  
PSEIZ Timer parameter 29  
Public Charge ID parameter 94  
Min Number of Digits parameter (telephony  
route) 125  
publications  
hard copy xxii  
modifying a user 105  
SIP Provisioning Client User Guide  
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[Standard - Nortel Networks Confidential]  
198 Index  
defining 71  
R
service parameters and options 73  
Registration Unit parameter (i2004) 100  
related publications xxi  
SIP based voice mail server 50  
adding 51  
setting up rules for emergency notifications 193  
SIP Provisioning Client xix  
exiting 9  
Remove parameter (telephony route) 125  
Routes 154  
features 2  
launching 3  
overview 1  
S
Status Reason parameter 93  
search criteria  
subdomain 77  
adding users 92  
ServerHome parameter 65  
support, Nortel Networks xxii  
System administrator provisioning role 16  
service pacakge  
service package 79  
technical support xxii  
listing 87  
listing services 79  
modifying 87  
parameters and options 81  
upgrading 90  
Service Package paramter 93  
definition 121  
service packages 104  
assigning to subdomain 89  
deleting 88  
dial plan description 157  
example dial plan assignments 157  
examples 158  
gateway route 127  
listing 126  
downgrading 90  
modifying 88  
service parameters  
NN10043-113  
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modifying 128  
Line based 50  
listing 68  
private route 127  
provisioning 50  
SIP based 50  
redirection between subdomains 153  
sample dial plan 154, 155  
Telephony routing concepts 149  
telephony routing concepts 149  
Class of Service (COS) 150  
subdomains 149  
way 50  
telephony routing terminology 152  
WD Phone Timer parameter 29  
WD Timer parameter 29  
routes 154  
Time FMT parameter 29  
time zones 5, 7, 8  
To Digits parameter (telephony route) 125  
translation verification tool 131  
Trunk based voice mail server  
adding 55  
U
parameter 52  
user management  
adding a user 105  
adding users 92  
deleting a user 105  
listing users 105  
V
Vocoder  
PacketTime parameter 30  
voice mail server types 50  
adding a SIP based voice mail server 51  
SIP Provisioning Client User Guide  
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[Standard - Nortel Networks Confidential]  
200 Index  
NN10043-113  
Download from Www.Somanuals.com. All Manuals Search And Download.  

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