CIC ICSystemConfigAndSupport ClassManual

IC System Configuration
and Support
Class Manual
www.inin.com/education
4.0 070715
2015, Interactive Intelligence, Inc.
Copyright and Trademark Information

Interactive Intelligence, Interactive Intelligence Customer Interaction Center, Interaction Administrator,
Interaction Attendant, Interaction Client, Interaction Designer, Interaction Tracker, Interaction Recorder,
ION, Interaction Mobile Office, Interaction Center Platform, Interaction Monitor, Interaction Optimizer,
and the Spirograph logo design are registered trademarks of Interactive Intelligence, Inc. Customer
Interaction Center, EIC, Interaction Fax Viewer, Interaction Server, Interaction Voicemail Player, Interactive
Update, Interaction Supervisor, Interaction Migrator, and Interaction Screen Recorder are trademarks of
Interactive Intelligence, Inc. The foregoing products are 19972015 Interactive Intelligence, Inc.
All rights reserved.
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The foregoing products are 20002015 Interactive Intelligence, Inc. All rights reserved.
Messaging Interaction Center and MIC are trademarks of Interactive Intelligence, Inc. The foregoing
products are 20012015 Interactive Intelligence, Inc. All rights reserved.
Interaction Director is a registered trademark of Interactive Intelligence, Inc. e-FAQ Knowledge
Manager and Interaction Marquee are trademarks of Interactive Intelligence, Inc. The foregoing
products are 20022015 Interactive Intelligence, Inc. All rights reserved.
Interaction Conference is a trademark of Interactive Intelligence, Inc. The foregoing products are
20042015 Interactive Intelligence, Inc. All rights reserved.
Interaction SIP Proxy and Interaction EasyScripter are trademarks of Interactive Intelligence, Inc. The
foregoing products are 20052015 Interactive Intelligence, Inc. All rights reserved.
Interaction Gateway is a registered trademark of Interactive Intelligence, Inc. Interaction Media Server
is a trademark of Interactive Intelligence, Inc. The foregoing products are 20062015 Interactive
Intelligence, Inc. All rights reserved.
Interaction Desktop is a trademark of Interactive Intelligence, Inc. The foregoing products are
20072015 Interactive Intelligence, Inc. All rights reserved.
Interaction Process Automation, Deliberately Innovative, Interaction Feedback, and Interaction SIP Station
are registered trademarks of Interactive Intelligence, Inc. The foregoing products are 20092015
Interactive Intelligence, Inc. All rights reserved.
Interaction Analyzer is a registered trademark of Interactive Intelligence, Inc. Interaction Web Portal
and IPA are trademarks of Interactive Intelligence, Inc. The foregoing products are 20102015
Spotability is a trademark of Interactive Intelligence, Inc. 20112015. All rights reserved.
Interaction Edge, CaaS Quick Spin, Intelligence Marketplace, Interaction SIP Bridge, and Interaction
Mobilizer are registered trademarks of Interactive Intelligence, Inc. Interactive Intelligence
Communications as a ServiceSM, and Interactive Intelligence CaaSSM are trademarks or service marks of
Interactive Intelligence, Inc. The foregoing products are 20122015 Interactive Intelligence, Inc.
All rights reserved.
Bay Bridge Decisions, Interaction Script Builder, Interaction Speech Recognition, and Interaction Quality
Manager are trademarks of Interactive Intelligence, Inc. The foregoing products are 20132015
Interaction Collector and Interaction Decisions are trademarks of Interactive Intelligence, Inc. The
foregoing products are 20132015 Interactive Intelligence, Inc. All rights reserved.
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Inc. The foregoing products are 20142015 Interactive Intelligence, Inc. All rights reserved.
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ii
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(6/24/2009). We hereby agree to indemnify the Initial Developer and every Contributor of the
software licensed under the Common Development and Distribution License (6/24/2009) for any
liability incurred by the Initial Developer or such Contributor as a result of any such terms we offer.
The source code for the included software may be found at http://wpflocalization.codeplex.com.
A database is incorporated in this software which is derived from a database licensed from Hexasoft
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shall not be liable for any damages suffered by the Licensee or any third party resulting from use of
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Other brand and/or product names referenced in this document are the trademarks or registered
trademarks of their respective companies.
DISCLAIMER
Interactive Intelligence, Inc. (Interactive) has no responsibility under warranty, indemnification or
otherwise, for modification or customization of any Interactive software by Interactive, Customer,
or any third party even if such customization and/or modification is done using Interactive tools,
training, or methods documented by Interactive.
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7601 Interactive Way
Indianapolis, Indiana 46278
Telephone/Fax (317) 872-3000
www.ININ.com
DISCLAIMER of REGULATORY COMPLIANCE OBLIGATIONS
Interactive is not responsible for providing, implementing, configuring, and/or coding any software
in a manner that complies with any laws or regulatory requirements that apply to its customers
respective businesses or industries, including, but not limited to, FTC and FCC regulations, the TCPA,
or HIPAA. Customer agrees that Customer is solely responsible for ensuring compliance with all
applicable laws or regulations. Customer agrees to indemnify and hold harmless Interactive and its
resellers from any and all claims, costs, and expenses arising out of or related to such claims. Use of
the Interactive software constitutes acknowledgement and acceptance of these terms.
DISCLAIMER TRAINING ENVIRONMENT
Interactive may teach techniques (for example, handler customization) that could result in
infringement liability. It is solely up to the end user to verify that the applications created by or on
behalf of the end user do not violate the intellectual property rights of third parties.
The company referred to throughout this document, Interactive Innovative Investments, Inc., is
fictitious. Names of people, characters, and/or data mentioned herein are also fictitious and are in no
way intended to represent any real individual, company, or event, unless otherwise noted.
No part of this book may be reproduced or transmitted in any form or by any means without the
advance written permission of Interactive Intelligence, Inc. The material is delivered in a multimedia
format expressly for digital use. Printing the material will lead to content loss.
Materials and processes provided in this class are based on the premise-based versions of
Interaction Center. All features described in this class may not be available or configurable to CaaS
or PureCloud customers.
IC and Interaction Center are abbreviated terms indicating the core Customer Interaction
Center (CIC) product from Interactive Intelligence, and are used only for internal Education
Services purposes.
iii
TABLE OF CONTENTS
Chapter 1: Interaction Center Architecture
Module 1.1 Interaction Center Architecture 1-02
Chapter 2: Session Initiation Protocol (SIP)

Module 2.1 VoIP Protocols 2-02
Module 2.2 Codec Selection 2-03
Module 2.3 SIP Messaging 2-05
Module 2.4 Endpoints 2-08
Chapter 3: Stations
Module 3.1 Managed IP Phone Configuration 3-02
Module 3.2 Auto-Provisioning 3-07
Module 3.3 Managed IP Phone Provisioning 3-10
Module 3.4 Station Configuration 3-12
Chapter 4: License Management

Module 4.1 Licensing Overview 4-02
Module 4.2 Licensing Users and Stations 4-05
Chapter 5: Lines and Regional Dial Plan

Module 5.1 Lines Container 5-02
Module 5.2 Interaction Center Phone Number Process
5-12
Module 5.3 Regional Dial Plan 5-14
Module 5.4 DID/DINS Routing 5-28
Chapter 6: Regionalization
Module 6.1 IC Regionalization 6-02
Chapter 7: Interaction SIP Proxy 4.0

Module 7.1 Interaction SIP Proxy 4.0 7-02
Chapter 8: Interaction Switchover

Module 8.1 Interaction Switchover 8-02
Chapter 9: CIC System Support

Module 9.1 Interaction Support Website 9-02
Module 9.2 Common Issues and Solutions 9-08
Module 9.3 Windows Troubleshooting Tools 9-13
Module 9.4 Interaction Center Troubleshooting Tools
9-18
Module 9.5 Packet Captures 9-22
Chapter 10: CIC System Maintenance

Module 10.1 Preventive Maintenance
Module 10.2 Updating the CIC System
iv
10-02
10-07
Chapter
Interaction Center Architecture

1.1

BENEFITS OF THE INTERACTION CENTER ARCHITECTURE
The Interaction Center Platform is composed of a number of software components that run in the
Microsoft Windows environment. Understanding what the subsystems do and how they work
together will be useful in helping you understand the IC platform. This information may also be
useful insight for troubleshooting.
MODULAR APPROACH
Interaction Centers modular design logically separates each communication element from the other
communication elements. This modular approach allows the system to continue operating even if one
of the subsystems malfunctions or completely ceases to function. Also, if one component requires an
update, only the parts associated with that component are updatedthe entire system does not need
to be upgraded.
The primary components of this modular design are as follows:
Notifier Notifier is considered the heart of Interaction Center, and acts as a communications
hub that all other components of the Interaction Center platform use to communicate.
Notifier makes use of the TCP/IP protocol to communicate with the rest of the Interaction
Center platform and provides critical services. The Notifier component provides tremendous
advantages to the Interaction Center Platform as compared to other communications systems.
These advantages include:
Security Notifier makes it difficult for other applications to infiltrate sensitive corporate
communications.
Scalability Poorly designed systems frequently broadcast events instead of identifying
appropriate applications components. Notifier reduces overall network traffic by sending
event notifications only to components that are actually involved in the transaction. This
allows applications using the Interaction Cener Platform to handle much larger numbrs of
users and interactions.
1-02
Real-time updates Using the Notifier, information can be communicated to the various
Interaction Center platform components in real-time. For example, a supervisory application
can display the rea-time status of every agent in a call center. When an agent finishes a call
and hangs up the phone, an icon on the supervisors screen instantly changes (theres no need
for a Refresh button).
Interaction Processor Interaction Processor, (IP) tells the system how to behave when an
event occurs. IP controls the execution of the handlers (or the system programs) that provide
much of the systems functionality.
Telephony Services Telephony Services (TS) is another main component and is the only
component that communicates directly with the hardware including the gateway. TS allows the
Interaction Center platform to detect telephony events (e.g., incoming calls, DTMF digits, call
disconnects, etc.) and to perform operations on telephone calls (e.g., transfer them, conference
them together, record them, play audio to them, etc.). Recognize that subsystems such as the
Exchange Connector, Web Services, Telephony Services, Client Services and Fax Services
interface with the environment outside of the Interaction Center Server, which allows for
connectivity to the associated devices and media.
QUEUES
In the Interaction Center environment, a queue is a container for interactions that can be processed.
There are two different types of queues in Interaction Center:

System Queue All incoming interactions arrive in the System queue before they are sent on
to a non-system queue. A series of IVR handlers and subroutines collect information from the
caller in order to route the call to the appropriate non-system queue.
Non-System Queue User queues, Station queues and ACD Workgroup queues are nonsystem queues that are created and managed in Interaction Administrator. A new queue is
automatically created every time a user, station, station group, or workgroup is added in
Interaction Administrator.
ADDITIONAL SERVERS IN AN INTERACTION CENTER ENVIRONMENT

The Interaction Center suite of software runs in a windows-based Internet Protocol (IP) network
environment. The software is deployed across two or more servers, and can use existing network
servers and services to provide additional functionality.
1-03
Additional servers and the functionality they add are as follows:

Directory Server Interaction Center uses Windows Active Directory services to authenticate
users. There is minimal configuration required of the Directory Server to enable Interaction
Center to operate.
DHCP Server While DHCP is not directly used by Interaction Center, the service is used by many
of the devices which connect to Interaction Center. Additional configuration information can be
provided via DHCP options including the autoprovisioning functionality of telephone devices.
DNS Server DNS, although not directly used by Interaction Center, can provide functionality to
IC connected devices to support such features as switchover and data security via Transport Layer
Security (TLS) and Secure Real-time Transport Protocol (SRTP). Also like DHCP, DNS is used by the
auto-provisioning functionality.
Database Server Interaction Center can make use of a database to provide many functions that
support call-center operations. Detailed reporting is possible because Interaction Center creates a
logging entry for every interaction that passes through the system. Additionally, the database server
can be accessed to take advantage of advanced features of the IC platform.
Messaging Server Interaction Center provides advanced unified messaging functionality. This is
supported by interacting with a messaging server, such as Microsoft Exchange or a cloud messaging
system such as Google Cloud Messaging (GCM).
Web Server The addition of a web server allows Interaction Center to interact with both users
and external customers via a web browser. This can include the provision of web chat sessions, and
client functionality for computer hardware which either does not have the appropriate software
loaded, or which runs an operating system for which Interactive Intelligence, Inc. has not provided
native software.
Speech Server Interaction Center can make use of advanced speech applications, including both
Automatic Speech Recognition (ASR) and Text To Speech (TTS).
Media Servers The media server is used to handle the audio portion of any call that does not pass
directly between phones and/or gateways. The media server is required in a CIC 4.0 system, and
performs a variety of functions including:
Records calls
Plays prompts when a caller is navigating the IVR (e.g., Please enter your account code.)
Decodes DTMF input when the caller is in the IVR
1-04
Analyzes outbound calls to detect busy signals

Sends and receives fax transmissions
Combines RTP streams for conferencing
When using 3rd Party Speech Servers, diverts RTP stream to an Automatic
Speech Recognition (ASR) server when used with a speech-enabled IVR
Provides real-time word-spotting for speech analytics
Manages the conversion between different codecs
Manages the conversion between different encryption algorithms
Interaction SIP Proxy The Interaction SIP Proxy is an optional additional server that can be used
to load balance endpoints, such as gateways, or to provide business continuity in the form of remote
survivability. When installed at a remote site for survivability, the Interaction SIP Proxy allows users
(at a remote site) to make and receive calls in the event of a WAN outage (i.e., no connection to the
IC server through the network). The proxy can be configured to support inbound, outbound and
intercom calls without IC.
FOR YOUR
INFORMATION
When using the SIP proxy in a remote office, only basic features are available during an outage.
IVR, Interaction Client, Unified Messaging, ACD and other IC features will not be available. You
will be limited to the features available in the Interaction SIP Proxy alone.
Review Questions
Complete the review questions for this chapter in your study guide.
Labs
There are no labs associated with this chapter.
1-05
Chapter
Session Initiation Protocol (SIP)

VoIP Protocols
Codec Selection
SIP Messaging
Endpoints
2.1
VoIP Protocols
VoIP DEFINED
Voice over IP (VoIP) uses Internet Protocol (IP) to transmit voice communications over intranets,
extranets, and the Internet. VoIP reduces costs by reducing or eliminating long distance charges,
and by sharing equipment and bandwidth to transmit voice and data.
Customer Interaction Center (CIC) uses VoIP technology to service calls on an Intranet. Using
gateways, those calls can also travel over the PSTN. Calls may be placed with an IP telephone,
computer and microphone (or USB headset), or an analog telephone adapter.
FOR YOUR
INFORMATION
VoIP is often referred to as IP telephony.
SIP DEFINED
Session Initiation Protocol (SIP) is a standard for call routing and control over IP Networks as
defined by the Internet Engineering Task Force (IETF) in RFC 3261.
The protocol allows devices to identify themselves on a network, find other devices on the network,
and make and receive calls.
SIP is a text-based protocol that utilizes parts of the Hyper-Text Transport Protocol (HTTP).
SIP can use either Transmission Control Protocol (TCP) or User Datagram Protocol (UDP) as its
transport protocol. TCP and UDP operate on a default port setting of 5060. UDP does not suffer
from the overhead imposed by TCP in maintaining a connection. The cost of this is that UDP will not
guarantee delivery and therefore the upper layers where SIP resides have to detect communications
failures and retransmit. SIP is a relatively lightweight protocol and is not time-critical.
TLS is an encrypted form of TCP that uses Secure Sockets Layer (SSL) to provide a version of SIP
known as SIP-Secure (SIPS). This protocol is closely related to HTTPS and operates, by default, on
port 5061.
Interactive Intelligence has chosen SIP as its VoIP solution for communication which offers
interoperability with other SIP-based solutions, and allows us to leverage our proven Interaction
Center Platform to contact centers, enterprises, e-businesses and service providers that wish to
utilize a SIP-based infrastructure.
RTP DEFINED
Real-time Transport Protocol (RTP) is an IETF standard that takes continuous data in real time and
passes it over a packet-switched network.
RTP is often referred to as a stream since it consists of a series of packets that flow from source
to destination device, and is a standard for streaming voice, video, and other multimedia.
Interaction Center uses SIP to setup and tear down VoIP calls, but RTP is used to transport the
audio portion.
RTP takes the stream of data and splits it into fixed-size chunks. It adds header information to these
chunks before passing them to UDP for transmission over the network. UDP is chosen in preference
to TCP as the latter would introduce more delay and bandwidth overhead for no benefit. Voice data
is time-sensitive, by the time TCP detects a packet of data has gone missing, its already too late to
arrange retransmission. Therefore, the additional overhead of maintaining a guaranteed delivery
connection is not justified.
2-02
2.2
Codec Selection
CODEC DEFINED
A codec (which stands for compressor/decompresser or coder/decoder) is the hardware or
software that samples analog sound and converts it to digital bits, and outputs it at a predetermined
bit rate. The codec often performs compression as well, to save bandwidth. There are dozens of
available codecs, each with its own characteristics.
Codecs are part of the standard that corresponds to the naming convention that the ITU
established. For example, the codecs named G.711u and G.711a convert from analog to digital
and back to analog waveform with relatively high quality. As with most things digital, the higher
the quality implies more bits, so these two codecs use more bandwidth than lower bit rate codecs.
Lower bit rate codecs, such as G.726, G.729, and those in the G.723.1 family, consume less network
bandwidth. However, low bit rate codecs can degrade the quality of the audio much more than
high bit rate codecs, because they compress the digital transmission with lossy compression; which
means some of the original data is lost in compression so the audio quality is lowered. Fewer bits are
sent, so the receiving side does its best to approximate what the original audio sounded like.
The table below lists some of the common codecs used in VoIP.
Codec Name
Bit Rate
Voice Played
Duration (ms)
Voice Played Size

(bytes)
G.711u
64.0 kbps
20 ms
160 bytes
G.711a
64.0 kbps
20 ms
160 bytes
G.726
32.0 kbps
20 ms
80 bytes
G.729
8.0 kbps
20 ms
20 bytes
G.723.1 MPMLQ
6.3 kbps
24 ms
24 bytes
G.723.1 ACELP
5.3 kbps
30 ms
20 bytes
The Bit Rate column shows the bandwidth required for a single stream using a particular codec.
Note: The header information (overhead) is not included.
VoIP HEADER OVERHEAD

In a VoIP environment, every single packet containing voice samples that is assembled and sent across
the network will have header information added to it from each of the protocols that are being used.
The standard method of transporting voice through an IP based network requires three headers.
These headers are:
IP An IPv4 header is 20 bytes
UDP A UDP header is 8 bytes
RTP An RTP header is 12 bytes
The total size of this header information is 40 bytes, or 320 bits. The additional bandwidth occupied
by this header information is determined by the number of packets that are sent each second.
2-03
The addition of these headers mean you will need to add 16 Kbps for codecs with 20 ms packetization
rates (such as G.711, G.729, etc) to the bandwidth requirements shown on the codecs table. For
example, if an 8 Kbps algorithm such as G.729 is used, the total bandwidth required to transmit each
voice channel would be 24 Kbps.
AUDIO QUALITY MEASUREMENTSMEAN OPINION SCORING (MOS)

Mean Opinion Scoring (MOS) is a common technique for observing the health or quality of a call, or
more specifically, the stream of packets containing audio payload on your network. The technique is
defined by the International Telegraph Union (ITU-T), and involves a very subjective methodology of
determining audio quality, based on input from a sample of listeners and how they score the quality
of audio samples on a scale of 1-5.
5 Excellent
4 Good (Toll quality)
3 Fair (Cell phone quality)
2 Poor
1 Bad/Unacceptable
VoIP analyzers use this scale to represent call quality by measuring factors that affect the call
such as: the audio codecs used, jitter levels, packet loss levels, etc. Codecs are important in this
technique because each codec has an expected MOS value:
FOR YOUR
INFORMATION
2-04
Codec Name
Mean Opinion Score (MOS)
G.711
4.3
G.729
3.7
G.723.1 MPMLQ
3.9
G.723.1 ACELP
3.8
G.726
3.85
A MOS score of 4.0 is considered toll quality. Why is this important? This will at least be the
expectation of customers installing a new system.
2.3
SIP Messaging
HOW SIP WORKS
SIP is a text-based protocol that uses requests and responses to establish communication among
various components in the network and to establish a conference between two or more endpoints.
Devices in a SIP network are identified by unique SIP addresses. A device ID is similar to an email
address, and is in the format of sip:UserID@Device IP:Port.
An example of a SIP device address is sip:7101@192.168.2.2:5060, or the same device could be
seen as sip:BobStation@edu.inin.com:5060.
When a user starts a call, a SIP request is sent to a SIP server. The request includes the address of
the caller (in the From header field) and the address of the intended callee (in the To header field).
The target client receives the request and the phone rings. If the user accepts the call, the client
responds to the invitation and a connection is established. If the user rejects the call, the session
can be redirected to a voicemail server or another user.
SIP MESSAGES
There are two types of SIP messages:
Requests Sent from the client to the server.
Responses Sent from the server to the client.
REQUEST METHODS
Method
Description
INVITE
Initiates a call, changes call parameters (re-INVITE).
ACK
Confirms a final response for INVITE.
BYE
Terminates a call.
CANCEL
Cancels searches and ringing.
OPTIONS
Queries the capabilities of the other side.
REGISTER
Registers with the Location Service.
INFO
Sends mid-session information that does not modify the session state.
RESPONSES
Response messages contain numeric response codes. The SIP response code set is partly based on
HTTP response codes. There are two types of responses and six classes:
Response Types
Provisional (1xx class) Provisional responses are used by the server to indicate progress, but
they do not terminate SIP transactions.
Final (2xx, 3xx, 4xx, 5xx, 6xx classes) Final responses terminate SIP requests.
2-05
Classes
1xx = Provisional, searching, ringing, queuing, and so on
2xx = Success
3xx = Redirection, forwarding
4xx = Request failure (client mistakes)
5xx = Server failures
6xx = Global failure (busy, refusal, not available anywhere)
SIP CLIENT REGISTRATION PROCESS

Two characteristics of a SIP client are:
They are usually classed as client devices, and they tend to have dynamically assigned IP addresses.
Their availability can be sporadic and their network details can change.
In order for the SIP servers to find the clients, it is necessary for those clients to identify their
location on the network. This process is done using the SIP REGISTER request type.
When a phone is configured, in addition to any other configuration settings that are required for
network access and so on, two parameters are set. These parameters are the address of the SIP
Registrar server and the unique SIP identifier for the phone. As part of its initialization process, the
phone sends a SIP REGISTER to the SIP Registrar. Assuming the SIP Registrar has been configured
to accept the particular registration, it acknowledges the request and records the phones details
(SIP address) in the database maintained by the Location server.
2-06
BASIC SIP TOPOLOGY

The following is a basic SIP diagram including an Interaction Center server, SIP phone, and SIP
gateway connected to the PSTN:
The graphic illustrates the signaling role of SIP in a phone call. SIP messaging is only used to connect
SIP endpoints together. The Interaction Center server acts as a centralized server to service the SIP
endpoints within a network. The RTP stream, which is more resource intensive than SIP, can then be
routed directly between the endpoints.
2-07
2.4
Endpoints
SIP PHONES
Phones are devices that convert voice into VoIP packets. They can be either hardware devices or
deployed as software applications with a headset such as the ININ SIP Softphone.
SIP Phones can by physically similar to an analog phone with a keypad, handset, etc., while others are
used exclusively in a corporate or call center environment, and are accompanied by the Interaction
Client such as the Interaction SIP Station.
Distinguishing features of phones are:
They can usually handle a single call at any one time.
They are usually classed as client devices, and usually support DHCP.
Below you can see examples of SIP Phones that are supported with Interaction Center. For a
complete and up-to-date list of currently supported phones, consult http://testlab.inin.com.
Audiocodes 420HD:
Polycom VVX 500 SIP Telephone:
Interactive SIP Station:
2-08
SIP GATEWAYS
A gateway is a device that converts between different systems. In VoIP terms, a gateway is usually
deployed at the edges of the network to interface with the Public Switched Telephone Network.
Distinguishing features of gateways are:
They can usually handle multiple simultaneous calls.
They are often the target of various calls to multiple destinations.
They are usually classed as network infrastructure devices, so tend to have fixed IP addresses.
A large range of gateways, from various manufacturers are available. Some are used to connect
analog telephone lines to the network, others connect digital lines (ISDN, Euro-ISDN, T-1, or E-1).
Some gateways are modular and can therefore connect to more than one type of line.
For a complete and up-to-date list of currently supported gateways, consult http://testlab.inin.com.
OTHER ENDPOINTS
There are various other devices that can act as the source or destination of VoIP traffic. These
devices can be software-based and hosted on a server or hardware devices.
For a complete and up-to-date list of currently supported endpoints, consult http://testlab.inin.com.
Review Questions
Labs
2-09
Chapter
Stations
Managed IP Phone Configuration

Auto-Provisioning
Managed IP Phone Provisioning
Station Configuration
3.1
Managed IP Phone Configuration

MANAGED IP PHONE CONFIGURATION
The Managed IP Phones container is used to create and/or manage IP phones. Managed IP Phones
can be manually created in this container or the phones can be imported using a .CSV file.
Managed IP phones contain phone-specific configuration per manufacturer and model, a link to the
associated SIP station in the stations container, a link to other SIP stations used in any shared line
appearances, and the registration group that will be used by the phone to contact the IC server.
There are two ways to configure an IP phone in the Managed IP Phone container:
Active Assignment
Select a phone from inventory.
Assign the MAC address of the phone to a Managed IP Phone configuration item. The MAC
address is entered by either scanning the box or manually typing it in.
Transport the specific phone to the specific user.
Passive Assignment
Transport one or multiple phones to a site and distribute.
A delegated user (co-worker, partner, or sub-contractor) with the IP Phone Provisioning
Administrator security right provisions the phone(s) through the TUI.
FOR YOUR
INFORMATION
Users have the rights to provision the default workstation assigned to them.
CREATE A MANAGED IP PHONE

In a SIP implementation, you can reduce implementation time and ongoing maintenance work for
Managed IP phones by managing them completely within Interaction Administrator.
The Provisioning Server subsystem takes care of serving the Managed IP Phone its configuration
(defined in Interaction Administrator) and manufacturer-specific firmware. Using Interaction
Administrator to manage IP phones also eliminates the need to maintain SIP phone configuration
(.cfg) files.
There are two methods for creating Managed IP Phones and SIP Stations:
1. Create individual Managed IP Phones and SIP Stations in the Managed IP Phones container
within Interaction Administrator.
2. Create multiple Managed IP Phones and SIP Stations by importing a .CSV list containing the IP
phones information in the Managed IP Phone Assistant within Interaction Administrator.
The associated SIP stations are automatically created in the Interaction Administrator Stations
container. The new station type will be Managed Workstation. For more information, see
IC Provisioning of Managed IP Phones in the Technical Reference Documents section of the IC
Documentation Library.
3-02
Stations
CREATE INDIVIDUAL MANAGED IP PHONES

You can create individual IP Managed Phones in the IP Managed Phones container.
FOR YOUR
INFORMATION
In the classroom environment, you are going to create individual Managed IP Phones since you
will only create two of them for the development environment. In a production environment, we
recommend that you use the Managed IP Phones Assistant and import from a .CSV list.
MANAGED IP PHONE TEMPLATES

In a production environment where large numbers of phones are involved, it is recommended that
you first create a Managed IP Phone Template based on the IP phone model in the Managed IP
Phones > Templates sub-container.
Give the template a descriptive name, select a type (workstation or stand-alone phone), and choose
the manufacturer, model, and Access Control Group.
3-03
Stations
Specify which Location the phone belongs to, an appropriate Registration Group, Preferred
Language, Audio Protocol, and one or more line appearances.
CREATE MANAGED IP PHONE WITH MANAGED IP PHONE ASSISTANT

The Managed IP Phone Assistant guides you through the process of creating Managed IP Phones
and associated SIP stations. For a new installation, it uses the .CSV import option to create multiple
new Managed IP Phones and associated SIP stations.
EXAMPLE
You have purchased 50 Polycom phones, and you want to create new managed IP phones and
the associated SIP stations for all 50 of the phones at the same time, each having different names
and extensions.
Steps needed to create a Managed IP Phone list for .CSV list import:
1. Create a Managed IP Phone Template based on the IP phone model in the Managed IP
Phones > Templates sub-container. For help filling out the screens, refer to the Interaction
Administrator online help.
2. Create a .CSV file using the correct format and enter the relevant information about the
Managed IP Phones. The .CSV file contains SIP phone name, template, extension, and address
information. Also, it must reference the template you created in the previous step by name.
FOR YOUR
INFORMATION
A sample Managed IP Phone .CSV list is included on the product disc in Additional Files > CSV
Lists. See the Interaction Administrator online help and Importing CSV Lists in IC in the Technical
Reference Documents section of the IC Documentation Library for instructions on how to create a
Managed IP Phones CSV list.
To access the Manage IP Phone Assistant, right-click in the Managed IP Phones container and
select Manage IP Phone Assistant.
3-04
Stations
Fill in the screens to complete these tasks:

1. Select the .CSV file
2. Review .CSV file import results
3. Correct errors if detected, and reload the .CSV file
4. Commit changes
See Interaction Administrator online help for information on each field in the Managed IP
Phone Assistant.
FOR YOUR
INFORMATION
We recommend that you run the Managed IP Phone Assistant after business hours because
the procedure requires significant server resources.
REGISTRATION GROUPS
A registration group is required for every managed IP phone. The registration group controls how
and where the phone registers.
Each registration group consists of a list of registration types. The registration list is ordered. Phones
will attempt to use the first registration, and if it fails the next registration in the list is used, and so on.
REGISTRATION TYPES
Line Select an existing line from the pull-down list for the registration.
Manual Manually enter the address, port, and protocol for the registration.
SIP Proxy A provisioned Interaction SIP Proxy server.
DNS SRV Enter the domain name and protocol for the registration.
CAUTION
3-05
Stations
If you change a line configuration that is used in a registration group, all managed IP phones using it
will need to be reloaded before the changes take effect.
STATION APPEARANCES
Managed IP Phones typically have at least one Station Appearance which corresponds directly to
the physical phone. Station Appearances can be a station extension, or a label.
The station extension is used when the phone is being provisioned and can be used to call the
station directly.
The station access controls determine what classifications of calls can be made when no user
is logged on to the telephone.
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Stations
3.2
Auto-Provisioning
RATIONALE
In most Interaction Center environments, the stations being deployed will require two SIPspecific settings:
The location of the Registrar Server
Unique SIP identity
This identity must match the settings configured in IC on the station object which represents that phone.
More advanced features such as TLS or the use of vLANs with the phones built-in switch will
require further configuration.
The system is currently supported by the ININ SIP Station, the ININ SIP Soft Phone, Polycom SIP
Phones and Audio Codes.
The process is split into two stages, the provisioning of new phones and the management of existing,
provisioned phones.
When a telephone first starts up, it issues a DHCP DISCOVER in order to obtain network configuration
information. In addition to all of the other configuration settings, the DHCP server in an IC environment
also provides the address of the IC server(s) which will be used for provisioning. This is done using
OPTION 160. The phone then contacts this server to request its configuration information. What
happens next depends on whether the phone is recognized by the provisioning server.
PROVISIONING A NEW PHONE

When a phone is unrecognized by the provisioning server, it provides generic information to the
phone. This information is set up using the Default IP Phone object.
This tells the phone to place the label Setup next to the first line button and to send its SIP
REGISTER request to the Proxy defined in the <Stations-TCP> line. (This line is configured to
operate on the non-standard port 8060, or 8061 in the case of TLS.) The phone continues its start
up process and registers with the IC server. At this point a generic SIP identifier is used.
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Stations
When the IC server receives a SIP INVITE request from the phone (due to the user attempting to
place a call), it transfers the call to a dedicated auto-provisioning IVR. This prompts the caller to
enter information which will identify the phone.
There are two methods to provision a phone:
1. Provisioning as an administrator. Delegated users with the IP Phone Provisioning Administrator
right can use this option. They will be prompted to enter the telephones extension, their user
extension and their IC password (PIN).
2. Provisioning as a user. Users with a default workstation defined can provision that workstation.
If this option is chosen, then they will be prompted only for their user extension and their IC
password (PIN).
After successful completion, IC records the MAC address of the phone against the Managed IP
Phone object and the phone restarts.
MANAGEMENT OF AN EXISTING PHONE

If the provisioning server has a record of the phones MAC address recorded against a Managed IP
Phone object, then it uses that object to provide the necessary configuration information to the
phone. This includes at the most basic level, the SIP Registrar to send the SIP REGISTER message
to (defined using the Registration Group).
The name of the label to place next to the button and the SIP Identifier to use. These are both
configured on an appearance-by-appearance basis using the Station Configuration object.
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Stations
The phone is then able to register and be available for use.

Interaction Administrator can be used to make changes to the configurations and remotely restart
the phones.
Individual or all phones requiring a restart can be targeted.
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Stations
3.3
Managed IP Phone Provisioning

MANAGED IP PHONE PROVISIONING
Managed IP Phones rely on the Default Lines (Stations-TCP|UDP|TLS) to send registration requests
and other SIP messages to the IC server. Before this can happen, the Phones must be able to find
the IC server in the network.
Managed IP Phones are setup to use DHCP by default, so when they are connected to a network,
they will contact a DHCP server to receive an IP address. DHCP Options can be configured to
provide additional information to the phones. For example, you can point the phones to the IC
Server using Option 160.
This process enables a centralized point of contact on the network to be used to provide
configuration files, firmware upgrades and even provisioning information to the phones. Once the
phones point to IC for their boot information the IC subsystems provide the appropriate firmware,
and any configuration information.
Below is an example of a DHCP server configured with Option 160:
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Stations
DHCP options for phone provisioning will include the following:

Protocol used for communication with the IC Server for provisioning: HTTP
FQDN of the Interaction Center Server where the phone will obtain configuration information
and files (e.g., IC_Server_Name.edu.inin.com)
Port used for this type of communication: 8088
DHCP OPTIONS
Option 67 carries the filename that is to be downloaded from the server specified in option 160 for
Interaction Center SIP stations. Often data put into option 67 does not actually appear in the DHCP
packet as option 67, but may be moved into the file field of the DHCP packet.
FIRMWARE
Once the phones have been directed to IC, subsystems handle the updating of the phone.
Polycom, Audiocodes, and ININ firmware are placed locally on the IC server during installation
and updated with Service Updates. They are located, by default, in the IC installation directory
(\i3\IC\Server\Firmware).
FOR YOUR
INFORMATION
For more detailed information on SIP lines and stations, review the SIP Application Note in the IC
4.0 Documentation Library.
FINALIZATION
Once the Managed IP Phone has received its firmware and files from the IC server, it will reboot
multiple times, and if it is using a passive or delegated assignment, enter a final provisioning state,
which may require administrator input via the TUI.
The phone will finalize its configuration based on station information in Interaction Administrator,
including, but not limited to:
Station Extension: reachable extension through IC
SIP Address (URL): registered SIP contact address
Access Control: types of calls the station can make, without a user logged on
Licensing: Basic Station, Client Access, ACD media
Multiple line appearances
3-11
Stations
3.4
Station Configuration
NAVIGATING THE STATIONS CONTAINER
The Stations Container in Interaction Administrator is used to display all stations configured in the
system, including Managed IP Phones.
The station types that can be created in the Stations container are:
Workstation A telephone with an associated PC on the network in the same domain as the
Interaction Center Server.
Stand-alone Phone A telephone that utilizes the Interaction Center server for phone calls but
is not associated to a PC.
Stand-alone Fax Machine An independent fax machine with its own line.
Remote Station A station assigned to a user or agent who runs Interaction Client outside of
the network domain of the IC server.
Unified Messaging A third-party unified messaging provider. This station type is only available
when a SIP or third-party UM license is present.
FOR YOUR
INFORMATION
Once a station entry is complete, the only way to change that stations name is to create a new entry
with the new name and configuration attributes and then delete the original station.
CREATING NON-MANAGED STATIONS

Non-managed stations that do not enjoy the full feature set of managed IP phones (e.g. autoprovisioning, reloading from the Interaction Administrator console), can be created individually or
by right-clicking and using the Station Assistant.
The Station Assistant offers the ability to create multiple stations by importing a .CSV file.
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Stations
The Station Assistant offers an example of how this file should be formatted.
3-13
Stations
STATION TYPES
When creating stations individually, the type of the station must be decided at the time you create it
(if using the .CSV method, it is possible to assign a station type for multiple stations at once). Station
types can be chosen from global settings pulled from Templates contained in the Stations container.
SIP CONFIGURATION
For non-managed phones, it is necessary to manually enter the SIP Identification address of the
station in Interaction Administrator. This is the address that will be used for SIP registration. When
a SIP device sends a registration request to IC (for non-managed stations), the system attempts
to match the received SIP registration request with configured stations in IA. For this reason, it is
typically necessary to also configure the same address on the device.
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Stations
FOR YOUR
INFORMATION
The User Portion and the Host combined create the full Address. The Host can be statically
assigned, or can be left blank by selecting Use User Portion Only. This is good for devices that pull IP
addresses from DHCP and may change IP address. Matching against the User Portion Only allows
this to happen without causing the Registration to be lost. When configuring SIP settings for a nonmanaged station, it is necessary to configure the Identification Address. Interaction Administrator
will not create the station without this setting.
CONTACT LINE
A line must be created for non-managed phone registration. These phones will typically be set to
port 5060 for SIP communication by default. The lines created during Setup Assistant use port
8060, and are reserved for Managed IP Phones.
DHCP OPTIONS
In previous versions of IC, option 66 was used to direct SIP phones to the TFTP server containing
configuration files, firmware, etc. for the phones. This option is still supported for IC 4.0, and can
be used to contain boot server host name information (IC server), and a secondary default record
for non-managed phones to find a provisioning server.
DEFAULT STATION CONTAINER

The Default Station container is used to set global options that affect all stations such as choosing
which audio protocol (RTP or SRTP), audio path, or what Contact Line should be selected unless it
is manually set.
TEMPLATES
Templates allow for settings to be applied to a group of similar stations, typically they will be
assigned by station type or model.
The following templates are created during the installation of Interaction Center:
Remote Station
Standalone-Fax Machine
Stand-alone Phone, SIP
Unified Messaging
Workstation, SIP
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Stations
STATION GROUPS
Station groups are a group of stations that share one extension. Station groups can be created using
one of the following types. These types determine how stations within the group are reached:
Group Ring
Sequential
Round-robin
CAUTION
FOR YOUR
INFORMATION
When using group ring, configurations of more than 20 stations will adversely effect the TS Subsystem.
Setting the station timeout allows for control over how long a call will alert at the stations within
the group. Stations already in use will not alert for a station group call, but if a user is logged into a
Interaction Client using a station that is part of the group, the interaction will appear in the users
Interaction Client.
Review Questions
Labs
Complete the following labs in your lab manual:
Lab 1: Create Managed Stations
Lab 2: Provision Phones and Test
Lab 3: Create Managed IP Phone Template
Lab 4: Importing Managed IP Phones
3-16
Stations
Chapter
License Management
Licensing Overview
Licensing Users and Stations
4.1
Licensing Overview
IC LICENSING OVERVIEW
IC uses a system management module for licensing. Customers can manage licenses on the IC
server and allocate those licenses more flexibly than in the past.
Customers who change and grow can easily view the status of their license on a specified
server and understand budgeting concerns to increase counts of features and functions on the
system. Customers can manage their licenses with Interactive Intelligence through the License
Management website.
This section is a summary of some important IC license information for new installations.
Server features and functionality are licensed per server, referred to as Feature Licenses. Each
server is licensed in a multi-server environment. Turn on features and functionality on each
server with the server feature licenses. The CIC server has a base server license with add-on
server features that can be purchased as a package or individually.
Client features are licensed by user or station, or mixed, based on the feature. These licenses
are referred to as Client Access licenses. These licenses are either Assignable or Concurrent.
An ACD Access license is required for ACD functionality. It can be assigned to users, stations,
or both.
The licensing structure includes a simplified port model. There are three types of ports:
External Call Ports, TIE line ports, and Outbound Dialing Ports.
All stations require a Basic Station license. Each client type does include a station.
You can also configure a remote station. All clients that logon to stations that are outside of
the domain will use a Remote License. This allows calls to be directed to their remote station,
such as a cell phone, home phone, or PBX phone.
ACQUIRING AND GENERATING THE LICENSE

We recommend you visit the Interactive Intelligence License Management website at www.inin.com/
license at least two weeks prior to installing IC to confirm that the license information is correct. You
can gain access directly or you can access it from the Partner area of the website. Be prepared by
having the license on your machine, which can save you time during the IC installation process.
Order and generate an appropriate CIC 4.0 license for the version of Interaction Center. The license
defines what version, Server Feature, Basic Station and Access license components, plus any additional
license components, and quantities are included in your agreement. IC Setup Assistant requires an IC
4.0 license downloaded on the IC Server.
FOR YOUR
INFORMATION
Resellers typically order and generate the license. Customers should contact their reseller.
TO ORDER AND GENERATE THE 4.0 LICENSE

1. Place the order for CIC 4.0 product with Interactive Intelligence.
2. Once the companys account is created and the order has been processed, you can generate the
license from the License Management website at www.inin.com/license, using your Interactive
Intelligence web logon. Follow the instructions for generating a new license.
4-02
License Management
3. You will be asked to provide the following machine information:

Host ID (Host ID for the MAC address of the network card on the IC server)*
Machine Name
System Type (telephony platform)
Mail Connector (mail system)
*Interactive Intelligence licensing is based upon a Host ID for the IC server, instead of the MAC
address. (The Host ID is similar, but not identical to the MAC address.) A program called GetHostID
utility may be downloaded from the Support website at https://my.inin.com/support/products/ic40/
Pages/default.aspx (under IC 4.0 Utilities and Downloads) so that you can generate the Host ID
prior to installation.
4. After making the license selections, generate and view the license. The license file will be
hostid.i3lic.
5. Download the license file to the IC server or location accessible from the IC server.
LICENSE MANAGEMENT
The License Management page displays a listing of all licenses on the server. Display this page by
choosing the License icon in the button bar or click License Management on the File menu.
The Load License loads licenses of *.I3Lic or *.I3Lic.xml file types.

The Features tab allows you to view the features included with the loaded licenses.
The View Host ID displays the Host ID of the IC Server.
FOR YOUR
INFORMATION
This is a display list only. You cannot add to, delete from, or copy or paste user or station license
assignments into this list.
This page displays how many days you have to renew the license before expiration in the lower
left corner. If license renewal is due within a specific time period, a message is displayed similar
to Your license is due for its annual registration in ## days. Please visit http://license.inin.com/
LicenseManagement/ReRegister.aspx to re-register the license.
4-03
License Management
LICENSE ALLOCATION
The License Allocation container displays a list of all licenses that have been allocated for use by
Interaction Centers License Manager. For each license, the following information is shown:
Name This is the license name, such as ACD Media 2.
Assignable Allowed This is the number of Assignable licenses purchased.
Assignable Configured This is the number of Assignable licenses assigned to users
or stations.
Concurrent Allowed This is the number of Concurrent licenses purchased.
Concurrent Configured This is the number of Concurrent licenses assigned to users
or stations.
Concurrent in Use This is the number of Concurrent licenses that are currently in use
by users or stations.
Notes This column shows text if the count exceeds the number of licenses.
Right-click the license you want to configure and select Properties. This opens the License Configuration
page. The License Configuration page allows administrators to add and delete users and stations, with a
tab for both Assignable and Concurrent. The bottom of the page displays the number of licenses and the
total number of licenses configured, based on the license allocation type.
4-04
License Management
4.2
Licensing Users and Stations

LICENSING USERS
The Users configuration page under the People container in Interaction Administrator allows you to
configure licenses per user.
License allocation method Assign an Assignable or Concurrent license to the user. Only one
license allocation method can be used per user.
Assignable The license is owned by the user and cannot be used by another user in IC.
Concurrent The license is a shared pool. Assign concurrent license if you want to share
licenses amongst users.
Client Access license Assigning this license to the user allows the client functionality of the
Interaction Client. Without this license assignment, the user cannot run the Interaction Client.
ACD Access license Select this check box if this user is an ACD user, then select the type of
ACD license. There are three types of ACD licenses:
Media 1 This license allows 1 interaction type at a specified time.
Media 2 This license allows 2 interaction types at a specified time.
Media 3 Plus This license allows 3 interaction types at a specified time.
If Media 1 or Media 2 type of ACD licenses is selected, you can click Interaction Types and
select the type of interaction from the list to apply to the license. Interaction Types are
unavailable if Media 3 Plus is selected.
Enable Licenses Select this check box to set the license settings to Active. If cleared, the
license settings on this page are ignored by the system.
FOR YOUR
INFORMATION
Enable License can be used to turn off the license consumption for a user, but keep the
settings intact.
Additional Licenses This is list displays additional licenses that are available. Select the
licenses you want to assign to the user.
4-05
License Management
Click OK to save the changes. These license assignments are immediately reflected in the license
counts in the License Allocation container list.
LICENSING STATIONS
The License tab in Station Configuration for a station allows you to apply licenses per station.
Licensing Stations in IC is similar to licensing Users, with a few exceptions:
Licensed Machine Name This field allows this station to be associated to a specific machine.
Typically the Licensed Machine Name is the station name, but selecting the check box allows the
user to override this setting.
Basic Station License This license represents an audio path between IC and a station. This
license is not required, but without it the audio for the station will not play, and there will be no
dial tone. A non-audio station may be used for non-audio interactions, such as email interactions.
FOR YOUR
INFORMATION
Remote stations must be assigned a Basic Station license.
Review Questions
Labs
4-06
License Management
Chapter
Lines and Regional Dial Plan

Lines Container
Interaction Center Phone Number Process
Regional Dial Plan
DID/DINS Routing
5.1
Lines Container
LINES CONTAINER
In the Interaction Center platform, and more specifically in Interaction Administrator, the term
lines is used as a generalized reference to individual SIP lines. SIP lines are VoIP protocol channels
of communication between the IC server and other SIP devices, such as SIP proxies, SIP gateways,
SIP Providers, and SIP phones. The Lines Container in Interaction Administrator contains all SIP
lines created for the IC Server.
DEFAULT LINES FOR PROVISIONED PHONES

There are three SIP lines created when an IC administrator completes the Setup Assistant
process. These lines are used for Auto-provisioning and Intercom calls. The TCP and UDP lines
communicate on port 8060 by default, and the TLS line uses port 8061.
The default prefix for these lines is Stations, unless the setting is changed during the initial run of
Setup Assistant. The lines appear in Interaction Administrator as Stations-TCP, Stations-UDP and
Stations-TLS in the Lines container.
FOR YOUR
INFORMATION
Any devices that use the standard ports, 5060 and 5061, will need an additional SIP line created for
Registration and communication.
PURPOSES OF ADDITIONAL SIP LINES

Additional SIP lines will need to be created for the following purposes:
SIP registration for non-managed devices
Connections to SIP Gateways or SIP carriers (PSTN)
Connections to other SIP servers
Before adding additional lines to the IC Server, you should first consider the reason or the need
for new lines. The purpose of the line you are creating will help you determine the settings
needed, and where in the call flow the line will be used. Lines can play a part in the call routing,
Regionalization, toll bypass, CODEC selection, reporting, and other features of a VoIP system.
INFORMATION FOR LINE CREATION

Before beginning the process of creating a new line you will need to have the following information
ready to input:
Line Usage What is the purpose of this line: General, Lync, or Stations.
Calling Address This is the combination of a Name and a Phone Number (Address).
Transport Protocol Which protocol will be used: TCP, UDP, or TLS.
5-02
Port What port is paired with the protocol: 5060 or 5061?

Proxy This is the address of the device that IC will route messages to when using this line.
Access These are the addresses of devices allowed to communicate to IC using this line.
FOR YOUR
INFORMATION
Main details of SIP lines are covered in this module. Additional options are covered in the
Documentation Library.
SIP LINE CONFIGURATION LINE PAGE
Field
Active check-box
What It Does
Activates or deactivates the line in the IC system. You can not
deactivate a line if any calls are on the line. If you change line
configuration parameters or perform other line maintenance, you
may have to deactivate a line and then reactivate it in order for
the changes to take effect. For example, if you change the SIP line
transport protocol.
Note: Interactive Intelligence counts only active lines to determine whether
you are in compliance with your license agreement.
Designates the usage of the line.

Line Usage
5-03
The options are: General Purpose, Microsoft Lync, or Station

Connections.
Field
What It Does
Domain
Specifies the domain name that is used to formulate SIPURLs for

IC users and phone numbers. This domain name is automatically
appended to all REGISTER requests that are sent by the
Interaction Center.
This value is used in the From header in outbound SIP calls.
Note: You can set additional identity settings by clicking Identity (In) and
Identity (Out) in the options list.
Maximum Number
of Calls
Designates the maximum number of calls that the SIP line will
process. When the number of calls is reached, this line processes
no more calls.
Fax Protocol
Indicates the fax protocol to use.
Enable Fax Detection
Indicates whether fax tones are detected when the Allow

Deferred Answer check box is selected.
SIP LINE CONFIGURATION IDENTITY (OUT) PAGE
5-04
Field
What It Does
Note: This option is applicable only to TLS lines.
Use sips. scheme
Converts the SIP address in the From header to use SIPS instead
of SIP. SIPS is a secure transmission that uses the URI format. For
example: <sips:+13178723000@ICServer:5061>.
If you do not select this option, IC converts all SIPS to SIP.
Calling Address
FOR YOUR
INFORMATION
Set the local address identification (display name and URI) which
will be attached to all outbound SIP messages on the line. You
can set the values for the two lines by clicking the ellipsis (...) to
open the Configure Line Value dialog box.
Identity (Out) options are covered in more detail in the Documentation Library on the Interaction
Support website.
SIP LINE CONFIGURATION AUDIO PAGE
5-05
Field
What It Does
Audio streams, have two possible routing configurations.
Audio Path
Dynamic The audio stream is routed directly between

endpoints. If the audio is routed directly between endpoints only
an extremely small amount of CPU resources and network traffic
is used by the IC Server for SIP messaging, and no Interaction
Media Server resources are used.
Always-In The audio stream will always be routed through the
Interaction Media Server. Always-In is preferred if the line routes
interactions that rely heavily on Media Server features.
SIP LINE CONFIGURATION TRANSPORT PAGE
Field
What It Does
Transport Protocol
TCP, TLS, or UDP (Default) can be selected depending on which

protocol is supported by the SIP devices using the line.
Audio Protocol
The audio stream can be sent unencrypted using RTP, or it can

be encrypted using Secure RTP.
Note: TLS is required for SRTP.
5-06
Field
What It Does
Security
The two settings, Minimal and End-to-Edge, partly determine

the visibility of the security notifications when calls are made
using the Interaction Client .NET or Web Client on this line.
Note: This drop-down box is only available if the Audio Protocol is set to
SRTP.
Receive Port
The port number on which the SIP line will service requests. For
TCP/UDP the default port is 5060. For TLS the default port is
5061. The valid port range is: 1024 to 65535.
SIP LINE CONFIGURATION PROXY PAGE
5-07
Field
What It Does
Prioritized list of Proxy

addresses
A prioritized list of outbound proxies that are available for the

line to send requests to. All outbound SIP messages sent on this
line will be routed to the first address in the list. The remaining
proxy entries will only be used if the first entry in the list stops
responding to Interaction Centers requests.
SIP LINE CONFIGURATION ACCESS PAGE
Field
What It Does
Access
The Access page is used to protect Interaction Center from abuse

by attackers registering phones from outside the organization.
If the All computers will be Denied Access option is set, IC will
deny all incoming requests on the line until an exception is entered.
Note: All settings are dynamically updated meaning any change to the
values will take effect immediately.
5-08
SINGLE OFFICE LINE CONFIGURATION

In a simple scenario, SIP lines service internal communication (station to IC server) and external
communications (IC Server to Gateway):
Stations-TCP|UDP|TLS Line Used for communication between Managed IP Phones and the IC
server. These three lines are created through Setup Assistant.
Gateway Line This line routes outbound to a PSTN-connected gateway. This line must be
created manually by an IC Administrator.
FOR YOUR
INFORMATION
5-09
More complex scenarios will require additional lines, and will need to be created by an Interaction
Center administrator.
MULTI-OFFICE LINE CONFIGURATION

For this configuration, one additional line is needed for the Remote Office to connect to the PSTN
(IC Server to Las Vegas Gateway across the WAN). The Stations-TCP line can be used to allow
Managed IP Phones in the Remote Office to communicate with the IC Server.
FOR YOUR
INFORMATION
If non-managed IP phones / stations are deployed in the environment, configuration of an additional

Registration line is required.
UNDERSTANDING LINE GROUPS

A Line Group can be created with one or more lines to give the IC Administrator the ability to
dedicate lines for a specific purpose. This can be accomplished by changing it to a Dial Group, and
assigning the group to a Dial Plan pattern. You also have the ability to turn on reporting for each
Line Group so that reports can be run from Interaction Center Business Manager (ICBM) to gather
various call statistics.
When an outbound call is made, Interaction Center refers to the Dial Plan to determine which line
the call should be placed on. Outbound calls that match the pattern will be directed to use a line that
is a member of the assigned Dial Group.
5-10
If an Outbound dialing pattern has no been assigned a Dial Group, IC will try to route the call across
the first active and available line it finds that has a working proxy address. Allowing IC to use any
available line may have an undesired effect, so use the Dial Groups and dialing patterns to properly
manage your outbound call flow.
LINE GROUP CONFIGURATION

There are three primary reasons to use a Line Group:
1. To enable Line Reporting.
2. To create a Dial Group.
3. To dedicate a line for Private Line Assignment.
Use for Reporting Selecting this check box causes Interaction Center to generate trunk group
usage statistics for reporting on the lines in this group.
CAUTION
Any line used in multiple line groups that are also used for reporting, may create duplicate statistics.
Use as Dial Group When this is checked, this line group can be used as a dial group in the Dial Plan
phone number configuration page.
FOR YOUR
INFORMATION
Make sure this line can make outbound calls.
Hunt Selection Method Hunt Selection Method allows for some control over how IC will choose
from the lines within a Line Group (assuming there is more than one line in the line group).
Use for Private Line Assignment Private line assignments (PLA) allow you to assign trunk lines or
trunk line groups to a specific users extension or a phone classification. This provides a trunk line to
always be available to place a call. You might also use this feature to handle 911 emergency calls or
to always have a line available for a specific user.
5-11
5.2
Interaction Center Phone Number Process

INTERACTION CENTER PHONE NUMBER PROCESS OVERVIEW
Outgoing calls placed through IC can originate from a number of places. Because phone numbers
can look very differentsome with area codes, some with letters, some with country codes, etc.,
Interaction Center uses a pattern matching process to dial any phone number in a consistent
manner, regardless of how the user dials it.
WHAT THE PHONE NUMBER CONFIGURATION CONTAINER DOES

The Phone Number configuration container appears to be the most complex container within the
Interaction Administrator. This container is responsible for taking a raw telephone number dialed by
a user and validating the user, formatting the number, and performing other tasks listed below:
Format the phone number into a standard package for simplicity.
Determine the Call Classification (e.g., long distance or local).
Determine if the user is allowed to make this type of call.
Send the appropriate numbers to the telephone company (7, 10, or 11 digits).
Determine which line or lines to use for the call.
Determine what to display on the Interaction Client.
Parse and store segments of the telephone number for statistical reports.
When you open the phone numbers configuration, you will see a number of Input Conversion
objects and Dial Plan objects already configured. These objects are based on the NANP (North
American Numbering Plan). The objects specific to your area were created automatically based
on the information you provided in Setup Assistant.
FOR YOUR
INFORMATION
For international number plans, it is recommended that you import your own dial plan or create
your dial plan in Interaction Administrator.
WHERE THE INITIAL SETTINGS COME FROM

The initial elements of ICs Dial Plan process were created when Setup Assistant was initially run.
Setup Assistant created the correct 7, 10, and 11-digit Dial Plan entries based on the area codes
and telephone exchanges that were entered.
Setup Assistant is focused solely on creating the Local area codes and the corresponding exchanges
that are considered local calls by the IC. By focusing on the Local Dial Plan structure, IC ensures
that any call classified as Local will be treated as such and be billed accordingly. All other calls will
be classified without any modification. That means that IC can use the generic entries in the Phone
Number Configuration container to format and classify the types of calls listed in the table on the
following page without any additional Administrator modification or input.
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Single digit extensions
Long Distance
NxxNxxXXXX
Double digit extensions
XX
Long Distance with a prefix

of 1
1NxxNxxXXXX
Three-digit extensions
XXX
411 or Information
411
Four-digit extensions
XXXX
Toll Free 800 Numbers
Five-digit extensions
XXXXX
900 Numbers
Six-digit extensions
XXXXXX
International Numbers
911
911
Company Operator
Local Telephone
Company Operator
00
Other Numbers
WILDCARD CHARACTERS
Wildcard characters are variables used to represent a range of numbers. Patterns can be represented
with wildcard characters consisting of dialable digits (09) plus any combination of digits and letters.
These digits and characters:
Are represented by this wildcard?
0 through 9
X (or x) represents a single digit character in that range.
1 through 9
Y (or y) represents a single digit character in that range.
2 through 9
N (or n) represents a single digit character in that range.

Z represents zero or more trailing characters of any value.
Any characters after the 7, 10, or

11 required telephone company
digits
0 through 9, #, *,
Using Z allows the dialer to enter extra characters after the

required digits without interfering with the dial string. This
wildcard character must be the last character of the pattern; it
is an error for it to appear anywhere else (which means there
can be only one occurrence of this character in each number).
? represents zero or more of these characters.
(Note that /, @, and + are not included)
$ represents a substitution character used in defining

multiple exchanges from a list.
Phone number patterns can consist of all digits, all wildcard characters, some non-dialing characters
such as + and /, or any combination. These three characters are not actually dialed, but they are
preserved as part of the stored phone number. The spaces these characters hold are also counted
when determining ordinal positions in the formatted number.
The + is used as a prefix to introduce a country code.
The / character is generally used as a prefix to introduce an extension for direct extension dialing.
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5.3
Regional Dial Plan

REGIONAL DIAL PLAN PAGE
The Regional Dial Plan tab is where you will access the pages to add Dial Plan objects, add locations
to objects, add new classifications, add or change exchanges in lists, and use the Simulate Call
feature to test your Dial Plan.
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DIAL PLAN PAGE

When you select the Dial Plan button on the Regional Dial Plan tab, you will see all the objects in
your Dial Plan on a single page.
The initial elements of the Interaction Centers Dial Plan process were created when Setup
Assistant was initially run. The Setup Assistant also created the correct 7, 10, and 11-digit Dial Plan
entries based on the areas codes and telephone exchanges that were entered.
FOR YOUR
INFORMATION
If the Dial Plan portion in Setup Assistant was not completed, you will see only the default Dial Plan
objects, or if you opted to create your Dial Plan in Interaction Administrator, the Dial Plan page will
appear blank.
This view is referred to as the Single Table Format, and it provides a summary list of all Dial Plan
entries. It also allows you to add/remove or edit existing entries.
The Set Filter button (located at the top of the screen) allows you to select the Dial Plan objects to
display based on a location specification. (This will be covered in the Regionalization chapter.)
The Up and Down buttons allow you to order your Dial Plan objects. The order in which your objects
appear is important as it determines what object will be used for the call.
The Import and Export buttons allow you to import or export Dial Plan files (*.i3dplan).
The Simulate Call button allows you to test your Dial Plan changes before saving them.
Select an entry and click the Edit button, for details.
5-15
The fields at the top of the Edit Pattern page allow you to customize this entry by editing the
standardized number, selecting different classifications, and assigning the entry to specific locations.
The Location Filter field contains a list of several default options as well as any locations that you
have defined.
The <All> filter means the entry is applicable to all locations.

The <Skip this item> filter means that this entry is not considered for live Dial Plan processing.
It provides a convenient way to deactivate an entry for testing, or other purposes.
<Many Locations...> allows you to select multiple locations where this entry will apply.
<Default Location> means that this entry applies to all entries that are not already assigned
to other locations. It also defines the SIP Station and SIP Line codec used when no locations
are defined.
<Location Name> allows you to select one location where this entry is applicable.
5-16
ADDING NEW DIAL PLAN OBJECTS

To add more area codes and exchanges after the initial Dial Plan has been created, you may add
them directly in the Dial Plan page or by re-running the Setup Assistant.
If you choose to add a new area code by rerunning Setup Assistant, supply the area codes and the
local exchanges.
If you choose to add a new area code directly through the Dial Plan page, you will need to know
all the specifics of your entry. For example, you will need to supply the correct input pattern, the
standardized number, the default dial string, display string, classification and what lists it uses.
FOR YOUR
INFORMATION
If youre looking to only add new exchanges to existing area codes, this is accomplished via the
Manage Lists button on the Regional Dial Plan tab.
MANAGE LISTS PAGE

The Manage Lists button on the Regional Dial Plan tab is used to add entries to existing number lists.
When you click the Manage Lists button, you see a dialog box displaying the number lists and an
area where you can add numbers.
The primary purpose of this feature is when you need to match many local exchanges to properly
identify local calls. Instead of creating a Dial Plan object for each local exchange, you can create a list
that multiple Dial Plan objects can use.
To add exchanges to existing lists, select the area code in the list, and click Add.
5-17
You can add each exchange individually, or use Dial Plan wildcard characters to indicate the exchange.
EXAMPLE
If all numbers within the 513 area code will be local, you can add Nxx to the list of entry numbers.
This tells the Dial Plan objects to process all calls dialed within the area code 513 as local calls.
EXAMPLE
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If all exchanges between 840 and 849 (in the 513 area code) are local, rather than list each number
consecutively, you can represent all the exchanges between those numbers with the wildcard X
resulting in 84x in your numbers list.
Now, whenever a user dials any of the exchanges listed, the call will go out as a local call. Because we
have identified 10 digit dialing for this site, the caller will always have to dial the area code as well.
MANAGE CLASSIFICATIONS PAGE

The Manage Classifications page allows you to create and manage phone number classifications
such as Internal, Long Distance, International, etc.
Phone numbers that can be identified by a pattern of numbers or an explicit set of numbers with a
purpose are called phone number classifications. For example, internal calls can be identified by the
Internal classification, Toll Free calls (Ex: 1-800) can be identified by the Toll Free classification, and
calls to emergency services (such as 911) can be identified by the Emergency classification.
5-19
These named groups of phone numbers are used to control individual users, workgroups, and
station dialing privileges in IC. For example, stand-alone phones may have only Emergency and
Internal dialing privileges, while members of the Sales workgroup may have full dialing privileges
provided in all classifications.
Interaction Center provides a number of default classifications.
CAUTION
Do NOT change or remove the default classifications or categories!
You may change the descriptive titles, however, categories are internal values that are used by the
system when processing the call. It is recommended that you not change or remove these default
classifications or categories because the default functionality relies on these elements.
SIMULATE CALL PAGE

The Simulate Call page allows you to test your Dial Plan before introducing it into a production system.
5-20
Clicking the Simulate Call button opens the Simulate Call window. Enter a number, and click
Simulate Call. The Call Results area displays how your call will be processed. This screen is very
helpful in simulating the various ways users may dial calls.
EXAMPLE
If a user dials 872-3000, you can simulate how the call will be processed:
The simulation shows the call going out as a local call even though the user dialed 11 digits. The
system will send 7, 10, or 11 digits to the telco depending on the Dial Plan configuration.
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EXPORT THE DIAL PLAN

After you create your Dial Plan, it is good practice to Export the Dial Plan for safekeeping,
particularly if you will be editing the Dial Plan using Setup Assistant.
When you edit the Dial Plan in Setup Assistant, you will be prompted to allow SA to overwrite your
existing plan. If you have a number of customized Dial Plan entries, they will no longer be available if
you allow SA to overwrite the existing Dial Plan.
You can however, export your existing Dial Plan, make your edits in Setup Assistant, then merge your
existing plan back in.
To export your Dial Plan, go to the Dial Plan page.
Click the Export button, and a Save As dialog box appears:
5-22
Select the location where you wish to store the file. It is recommended that you perform this export
anytime you make changes.
EDITING THE DIAL PLAN VIA SETUP ASSISTANT

As mentioned in the previous sections, you can add new area codes to the Dial Plan by re-running
the Setup Assistant.
When you re-run Setup Assistant you will see a welcome screen.
5-23
You can opt to import an existing Dial Plan or create a new one.
If you have a working Dial Plan already (the one you exported), and you are simply going to add
some new area codes and exchanges, select the option to Create a new Dial Plan. You will then
import your previous Dial Plan and merge with the new one.
The area code, number of digits, and local exchanges are defined for the new Dial Plan.
FOR YOUR
INFORMATION
5-24
Only exchanges local to the area code should be configured.
The new Dial Plan completely overwrites the existing Dial Plan.
Now, when you go back into Interaction Administrator and view your Dial Plan, you will see only the
Dial Plan objects for the area codes you just created.
5-25
IMPORT DIAL PLAN

The exported Dial Plan can be imported completely or merged with the current Dial Plan.
The Import Dial Plan dialog box displays a list of Dial Plan objects available in the file.
The option to Replace Entire Dial Plan will completely replace the current Dial Plan with the one
being imported. If this option is not selected, the entries being imported will be merged with the
existing Dial Plan. If you dont want the Classifications merged into your existing Dial Plan, select the
option to Skip Classification Merge.
5-26
The Dial Plan changes are not committed until the dialog is closed.
FOR YOUR
INFORMATION
5-27
Your imported dial plan area code entries will be positioned at the bottom of your dial plan and will
never be used unless you move them higher in your dial plan. You will have to manually do this for
each entry.
5.4
DID/DINS Routing
CALL ROUTING OVERVIEW
IC supports multiple ways to route calls directly to users, workgroups, or stations based on Dialed
Number Identification Service (DNIS) numbers from the Central Office (CO).
Typically, companies buy or lease a block of phone numbers, which are directed to inbound lines or
channels. You can map each phone number to an individual user, a specific station (e.g., phone or
fax), or a workgroup.
Interaction Administrator allows the creation of DNIS map tables to route calls to IC queues
without modifying handlers. The way you choose to build the table typically depends on:
The number of lines you have with DNIS service
Whether you have a block of contiguous or non-contiguous phone numbers
Whether the queues all have the same number of digits in the extensions
Whether the destination extensions are fixed or if they are likely to change
CONFIGURE DINS ROUTING

The options for configuring DNIS routing are:
Simple Mapping Used when only one DNIS route needs to be added.
Substitute Prefix This method is useful in situations where the company has a block of DNIS
numbers that begin with the same prefix, and the dialed numbers suffix matches the extension
for each IC queue.
EXAMPLE
The block of reserved numbers is 565-2100 through 565-2400. The user extension range is planned
using 21012199. DNIS mapping for user extension could result in 565-2101 through 565-2199.
Replace N Digits Enables a standardized, common phone number (e.g., main company
number) to be converted to DNIS routing and mapped to specific internal extensions. This is
useful for extensions with varying lengths.
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EXAMPLE
A single number such as 565-2000 can be used in the Replace N Digits page and mapped to 3-digit
and 4-digit user extensions. 565-2000 could convert to 565-2310 for a user with a 3-digit extension
(310), and 565-2107 for a user with a 4-digit extension (2107).
Defer Substitution This option can only be used when either Prefix Substitution or Replace N
Digits has been configured. This option tells the system to defer the processing of DNIS routing
configured in the Phone Numbers container, and process (via handlers) other possible DID/DNIS
routing elsewhere in the system (e.g., MobileOfficeDID or IVR settings). This option can add
time to DNIS processing.
Review Questions
Labs
Lab 5: Create SIP Line and Line Group
Lab 6: Perform a Basic Functionality Test
Lab 7: Configure Inbound Fax Operations
Lab 8: Export Dial Plan
Lab 9: Add a New Area Code with Specific Exchanges
Lab 10: Add New Exchange to a List
Lab 11: Import Original Dial Plan and Modify
Lab 12: Add Speed Dial Plan
5-29
Chapter
Regionalization
IC Regionalization
6.1
IC Regionalization
REGIONALIZATION
The Regionalization container in Interaction Administrator allows you to configure individual
locations for your environment.
In the Regionalization container you can perform the following functions:

Define a Location
Add stations, lines, or servers as Endpoints to a Location
Define the Codec communication between Locations
Set up Selection Rules
LOCATION
This location defines a set of endpoints (lines, stations, and servers) that share a common dial
plan, and it defines Codec communications for the endpoints. A Codec mapping defines the list
of Codecs for two locations to communicate with each other sharing a common set of bandwidth
requirements. The stations and lines that are members of a Location define the dial plan entries that
are applicable to a locale they are operating in.
EXAMPLE
One example of a location could be a small satellite office either in the same city or another city or
state with all SIP phones serviced by the Interaction Center server in the headquarters office.
Another example of a location is a group of remote employees that use SIP phones over a VPN.
Yet another example is a remote peer-site office with its own IC server in a multi-site configuration.
SOME RULES REGARDING LOCATION MEMBERSHIP

By default, the IC server will always be considered part of the Default Location.
All other location endpoints will be considered a member of the Default Location until assigned
to another location.
A device can only belong to one location.
6-02
Regionalization
EMERGENCY DIALING AMONG REGIONS

One of the first Dial Plan changes you will need to make is to ensure that emergency calls work in
each location. There are two methods to ensure calls route appropriately:
The first method is to create a new Dial Plan entry for 911 (the North American emergency
number), and assign it to the specific location.
The second method allows you to use a single emergency number entry (911 in North America),
and assign the dial group filters for each location.
SELECTION RULES
Use the selection rules to create prioritized lists of locations to select media servers and session
manager servers.
IC uses Interaction Media Server to process audio communications for an interaction between two
or more endpoints, such as a telephone call. To select a media server, IC uses Selection Rules. This
feature enables you to create prioritized lists of locations in Selection Rules configuration. You can
select a media server to service an interaction. (You can also select a session manager server to
service an interaction.)
Within a location, IC selects a media server, if more than one exists, based on the following criteria:
Available CPU resources
Number of resources in use
When an interaction that requires audio processing starts, IC searches, in order, each location in a
selection rules configuration. If a location does not have a media server or all media servers in that
location are busy, IC searches the next location in the Selection Rules configuration. This process
continues until IC finds an available media server.
FOR YOUR
INFORMATION
By default, IC has only one location: <Default Location>. When you create a device in IC, it is
assigned to the <Default Location> location. To use the selection rules effectively, define additional
locations and assign devices, such as stations, SIP lines, and media servers, to those locations.
For more information, see Interaction Media Server Technical Reference in the IC Documentation
Library on the IC Server.
CONFIGURATION
The Configuration page allows you to create a description of the server and identify whether or not
the IC server is housed at this location. You can also set the time zone and a designated time server for
the location.
6-03
Regionalization
CONFIGURATION PAGE SETTINGS

Description This is a descriptor of the location. This is not used for call routing.
Time zone Allows for the time zone for the location to be defined.
SNTP Server Allows the IC server or a network server to be use SNTP
This IC server is in this location Used to specify membership of the IC server to a location. It
is assumed by the system that the IC server is a member of the Default Location.
Enable Regional Dialing Select this check-box to allow dialing of short extensions within the
same location.
EXAMPLE
Station A in Location Indianapolis has an extension of 1500.

Station B in Location Indianapolis has an extension of 1600.
If the Enable Regional Dialing check-box is selected in the Location Indianapolis, and the Significant
Digits are set to 3, Station A can reach Station B by dialing 600, and Station B can reach Station A
by dialing 500. This change notifies managed proxies of changes to regional dialing significant digits
value so that the managed proxy can take that into account during remote survivability.
FOR YOUR
INFORMATION
A warning will be presented if Regional Dialing Significant Digits causes an extension conflict.
COMMUNICATION
The Communication page defines the codec mappings between locations. If you wish to add or
delete a codec to your location, click the Modify button and select the codecs appropriate for
your environment.
6-04
Regionalization
If you wish to change any of the codec parameters, click the Set Parameters button. The SIP Codec
dialog box allows you to further tailor the codecs for your environment.
LINE AND STATION ENDPOINTS

An endpoint is a device that can receive audio. This can be a line, station, IC server, or Interaction
Media Server.
EXAMPLE
An example of an endpoint could be a SIP station located either in the current IC environment or in
a different area code than the IC server is located.
The Line/Station Endpoints configuration page allows you to select individual lines or stations as
endpoints for this particular location.
6-05
Regionalization
LOCATION FILTERING
Once Locations have been created in Interaction Administrator, endpoints (e.g., stations, media
servers, proxies, and line connections) can be assigned to locations. Routing decisions can be made
based on the endpoints involvement in a call. Location Filtering allows for location membership to
be considered for dial plan processing.
Location Filtering Locations are used to restrict or filter access to portions of dial plan entries.
The dial plan can be filtered using locations at two levels of entries:
Input Pattern
Dial Groups (line groups and dialed numbers)
Regionalization/Codec Mapping IC can select a lower-bandwidth codec for inter-location
calls to preserve WAN bandwidth.
A common use for Location Filtering is Emergency dialing. If offices are located in different
geographic locations, yet share one standard emergency dial string (e.g., 911), location filtering
allows dial plan to route that shared dial string (911) to the appropriate office / gateway / SIP line
emergency services, and the correct location is reached.
LOCATION FILTERING EXAMPLE

One entry for 911 in dial plan, but two possible Dial Groups assigned: The Dial Group column
shows which dial group will be used when a 911 call originates from a device within a specific
location (identified in the Filter column).
6-06
Regionalization
When configuring your locations, you must have a clear understanding of the use and application
of IC locations, the gateways, networks, and system software that connect locations, and you must
know how to customize the Regional Dial Plan in order to see the cost savings and benefits of these
Regionalization features.
A checklist is available to help you plan your regional Dial Plan.
Review Questions
Labs
Lab 13: Configure Media Server
Lab 14: Add A New Location
6-07
Regionalization
Chapter
Interaction SIP Proxy 4.0

7.1

SIP PROXY OVERVIEW
Interaction SIP Proxy performs business continuity during disaster recovery and is capable of
performing day-to-day load balancing. The Interaction SIP Proxy software complies with RFC3261.
PROXY LICENSING
All Interaction SIP Proxy licenses have Business Continuity Manager (BCM) capability for remote
survivability. This means all Interaction SIP Proxy licenses include the Managed Proxy feature.
Load Balancing features are not included in the BCM license, but can be purchased as an add-on.
Additional licenses can also be obtained to enable media capabilities for call recording.
INTERACTION SIP PROXY FOR LOAD BALANCING

Load Balancing is often deployed in Interaction Center sites with multiple IVRs, multiple UM servers
(Messaging Interaction Center), or Dialer sites with multiple PSTN gateways configured to receive
and pass calls from IC.
In each of these deployments, the Interaction SIP Proxy is receiving messages from the IC server,
and load balancing the distribution of those sessions across multiple potential endpoints, using
Round Robin or Random load balancing capabilities not available in the other versions of the proxy
(Business Continuity Manager).
SIP PROXY FOR REMOTE SURVIVABILITY

The Interaction SIP Proxy can be installed at a remote site for survivability in the event of a WAN
outage. The proxy can be configured to support inbound, outbound, and intercom calls without IC.
Outbound Calls When users place calls, the SIP Proxy will route them to the Remote Office
PSTN Gateway using the Routing Table.
Intercom Calls User-to-user calls are routed by SIP registration addresses. All stations in the
remote office can register with the SIP Proxy, and those addresses are used for extension dialing.
Registered addresses are based on:
Station Name
Station Extension
User Extension (if the station is assigned as the users default)
Users DID Number (if station is assigned as the users default)
Inbound Calls The Proxy offers source-based routing. Calls sent by the identified gateway to
the proxy (inbound calls) will be sent to the appropriate registered station or group of stations.
The recommended configuration for managed phones at a remote site is to have them dual-register
with the IC server (as the primary) and the SIP Proxy (as the fallback). With this configuration the
phones will register with the IC server and send all calls to it as its first option. If the IC servers are
not available, the phones will fall back to the proxy for basic phone features.
A registration group is created containing both the IC and the SIP Proxy.
7-02
FOR YOUR
INFORMATION
When using the SIP Proxy in a remote office, only basic features are available during an outage. IVR,
Interaction Client, UM, ACD, and other IC features will not be available. You will be limited to the
features available in the Interaction SIP Proxy.
ACCESSING THE INTERACTION SIP PROXY

The Interaction SIP Proxy is installed as a windows service.
ACCESSING THE WEB INTERFACE

Configuration of the Interaction SIP Proxy is performed via the web interface. The default
credentials to log on are:
Username = admin
Password = 1234
Default Ports and URL
http://localhost:8080
https://localhost:442
FOR YOUR
INFORMATION
7-03
In production environments, it is recommended that you immediately change the user name
and password from the Administration tab on the proxy configuration page. This will prevent
unauthorized access and modification.
LAUNCH THE WEB INTERFACE FROM INTERACTION ADMINISTRATOR

Once a Proxy has been provisioned with Interaction Center (discussed later in the chapter), the web
interface for the proxy can be started from the SIP proxies container in IA.
Routing Routing is configured in a routing table using patterns to identify calls presented to
the proxy.
This is similar to how Dial Plan is used in IC, however, the proxys routing table uses regular
expressions instead of the NANPA-based dial plan. The proxys routing table can process patterns
differently based on the originating device. The endpoint can be configured to check (using SIP
Options messages) to see if a Server Plan is connected before deciding to route the call. If the proxy
determines the connection to a destination device is down, the Proxy can continue to search
through the routing table for an Up connection.
The proxy can record calls based on specific routes. In such cases, the recording function is assigned
during the creation of the route entry.
7-04
SOURCE DEVICE ROUTING

Source device, or From-based, routing allows for devices or ranges within the network to be
identified for call routing.
Once identified, these can be introduced into the proxys routing table allowing the proxy to route
the same pattern differently, based on the originating device.
Access Lists Access lists allow or deny access to the SIP proxy service for a specific device, or an
entire network segment. Access lists rely on dotted decimal notation for IP addresses and subnet
masks. Allow lists should be reserved for well-trusted devices.
Protocols Supported protocols are UDP, TCP, and TLS with IC-supported cipher suites. Preferred
cipher suites are enabled by default.
7-05
STATUS FEATURES
The Interaction SIP Proxy allows for real-time updates of key components of SIP session
establishment. The status section allows administrators to monitor different aspects of the proxy
server setup.
Session Status Displays all the currently active sessions or calls. Active calls will have
completed the setup process and be engaged in a conversation.
Registration Status The Registration status page shows all SIP devices that have successfully
registered with the Interaction SIP Proxy.
7-06
Managed Interaction SIP Proxy registrations contain SIP aliases for registered devices such as:
A stations extension number or station name
A users extension
A users DID number
A contact address hyperlinked to a SIP alias
Server Status The server status field displays the SIP operational status of all devices added
into a server plan. Each server is periodically sent a SIP OPTIONS message; if a response is
received, the server is considered up and operational. If no response is received, the server is
considered down.
MEDIA FEATURE
The Interaction SIP Proxy can be configured to record the audio of calls routed through it. These
settings are configured on the Media tab. This includes media formats (codecs), cipher-suites for
SRTP recordings, and assigning a directory for storage.
7-07
The recording feature in the Interaction Proxy is designed for remote survivability. For example,
calls can be recorded when there is a WAN outage, and the IC server cannot be reached for
recording features. Note, however, that the recordings will be kept locally (local to the proxy) until
they are manually moved.
FOR YOUR
INFORMATION
The Media Features of Interaction SIP Proxy require additional licensing beyond the standard BCM
Proxy licensing. Please visit the Interaction SIP Proxy page on the Interactive Intelligence website
for more information: http://www.inin.com/solutions/Pages/SIP-Proxy.aspx.
UNDERSTANDING REGULAR EXPRESSIONS

The Interaction SIP Proxy uses regular expressions to build routing patterns. This makes it more
efficient to configure patterns that satisfy many routing needs.
Regular expressions used within the SIP proxy utilize wildcard characters, substitution characters,
repeat characters, and grouping characters. Combinations of these characters (and others) within a
pattern allow a handful of entries to satisfy many typical calling scenarios.
EXAMPLE
An example of a typical pattern is 10-digit dialing to a PSTN connection:

Pattern = sip:([29][09]{2}[29][09]{6})@.*
User = $1
In the above example, [29] will match a single digit between 2 and 9, and [09] will match a single
digit between 0 and 9, but because of the proceeding {2} this part of the pattern will be repeated
7-08
twice for a total of 3 digits. [29][09]{6} will work as before, but will repeat the second set of
brackets 6 times for a total of 7 digits. The .*after the @ will match any character or digit an infinite
amount of times, allowing any host portion of the address to be matched.
Parentheses are used to identify sub expressions within the pattern. The value contained in
the outermost set of parenteses is represented by $1, while moving inward and left to right are
represented by $2, $3, etc. In this case, $1 would be a 10-digit number matching the NxxNxxXXXX
NANPA pattern.
ROUTING PATTERNS
The combination of routing patterns and server plans function in a manner similar to how
Interaction Center matches dial plan objects to SIP line groups.
Devices in the local network send SIP invitations to the proxy. The invitation will contain call
information which the proxy can use against patterns configured in the proxys routing table. The
proxy will use patterns from top to bottom in an attempt to find a matching pattern.
When a match is found, the associated server plan is referenced to determine the destination for
the SIP invitation. Server plans contain IP addresses of SIP-compliant devices on the local network,
typically an IC server, SIP gateway, or IP phone.
EXAMPLE
911 call routed by the Interaction SIP proxy

Routing pattern = sip:(911)@.* / Server Plan = Local FXO (192.168.2.10).
User dials 911 from a SIP phone.
A SIP invitation is sent to the Interaction Proxy (INVITE sip:911@192.168.2.3:5060).
The proxy will attempt to forward the call to the appropriate device by matching the user
portion of the SIP invitation [911] to a routing pattern entry (e.g., sip:(911)@.*).
The invitation will be forwarded to the endpoint identified in the associated server plan
(e.g., Local FXO server plan / IP address 192.168.2.10).
Note: Only one server plan can be associated with a routing pattern, however, it is possible for
the server plan to contain more than one IP endpoint.
A SIP session will then be established between the callers phone and the FXO gateway.
7-09
PROVISIONING A MANAGED PROXY

The proxy may be provisioned with an IC server and receive updates through the provisioning
subsystem on IC.
PROVISIONING THE PROBY REQUIRES THE FOLLOWING INFORMATION

IC Admin credentials (with Notifier password).
Provisioning address of IC server (i.e., provisioning protocol, name or IP address of IC server, and
port number), or properly configured DHCP server or DNS server to provide the information to
the proxy.
The provisioning process requires administrative input on both the SIP Proxy and the IC server.
Once the above configuration is completed on the SIP Proxy, and a connection is attempted, the
7-10
Managed Proxy status will change to a Pending Approval state. The IC admin must then configure
the proxy as a Trusted device in Interaction Administrator.
Once the Proxy has been trusted by Interaction Center, the Managed Proxy status screen will
change to indicate the connection is Activated.
7-11
Once the provisioning process is complete, the proxy will begin receiving updates from IC, including
Managed IP phone contract address information (i.e., station names and extensions, and user extensions
if users have the station assigned as a default workstation). These SIP aliases can be used to expedite call
routing. For example, if the proxy is supporting a small remote office, and the WAN connection is dropped,
the proxy can support call routing for stations and users based on their registered contact addresses.
CERTIFICATES (TLS/SRTP)
The Interaction SIP Proxy supports TLS / SRTP communication. For secure communication, the
Proxy can generate a certificate signing request (CSR) to be presented to the Certificate Authority.
The CA will generate a new certificate for import to the proxy. This will enable IC to trust the
proxy to establish TLS / SRTP communications.
As discussed earlier, the Media tab allows for SRTP cipher-suites to be enabled for secure audio
processing. SRTP communication in IC relies on secure SIP TLS sessions to be established.
ALERTS
The Interaction Proxy allows for email alerts to be sent in response to conditions that affect the
proxy functionality. Alerts are sent using SMTP or through SNMP traps.
7-12
DIAGNOSTICS
The Diagnostics page allows direct access to the current SIP Proxy configuration file, tracelog data
files, and topic configuration settings.
7-13
Location Indicates which local directory is being used for logs on the proxy server.
CAUTION
Browsing to and opening a log file begins a download of the log file. Downloading large logs can
degrade responsiveness of the SIP Proxy.
Retention The default retention period for logs is seven days. This parameter can be changed,
however, doing so will require a reboot of the server, not just a restart of the proxy service.
Group Presets These items control how much information is generated in the tracelog
files and should be used with caution in a debugging scenario, and with the assistance of an
Interactive Intelligence Support Engineer. Increasing these to higher levels can render the SIP
proxy unusable in production deployments. The default setting for All Topics is Default.
Current Topic Levels This setting allows for finer control over log output than Group Presets.
These should be changed with assistance from Interactive Intelligence support engineers. The
default production setting for all topic levels is Status.
Review Questions
Labs
7-14
Chapter
Interaction Switchover
8.1
INTRODUCTION TO INTERACTION SWITCHOVER
IC supports an automated switchover system. If an IC Server fails, the server can switch control to
another mirror image IC Server in less than 30 seconds with minimal phone disruption. In addition,
the switchover scheme allows administrators to manually switch the active IC server with minimal
phone disruption.
SWITCHOVER TERMINOLOGY
IC Switchover requires a pair of IC servers, as well as the capability of other devices to support
multiple SIP servers. Each IC server has a corresponding server configured identically. One of the IC
servers is installed as the primary server, and the other is the backup server. The backup server
monitors the primary server for TS and UDP pings.
Primary Server The primary server processes all IC interactions, such as phone calls, faxes, web
chats, voicemail, etc.
Backup Server This is the failover system. The backup server is a mirror image of the primary
server, duplicating its hardware and software, including IC registry entries.
All components must be identical including:
An identical amount of RAM, disk space, number, and names of drives.
Operating system software and service packs, including identical services running (e.g., SNMP).
Same Administrator account name and password (e.g., ICAdmin) for logging onto Windows on
each server.
IC components and site names (the site name is case sensitive).
Application software, including email clients that use identical mailboxes, etc.
FOR YOUR
INFORMATION
Both Polycom SoundPoint IP phones and Audiocodes analog and digital gateways support
redundant proxy configuration. Each of these supported devices is able to send SIP messages to
multiple IC servers and failover to an active IC server when necessary, without the presence of a
dedicated SIP proxy. For more information on automated IC switchover, refer to the Automated
Switchover System documentation.
THE SWITCHOVER PROCESS

The backup server regularly monitors the primary server, validating specific IC subsystems and looking
for the appropriate return signal from an attempted call operation. It also monitors and dynamically
copies any changes in the primary servers configuration, such as new user entries, line configuration
changes, handler updates, or any other configuration changes made in Interaction Administrator.
INITIATING SWITCHOVER
By default, the backup server starts the switchover procedure immediately when it detects:
The Primary Server is no longer sending UDP pings.
The Primary Server is no longer responding to TS pings that are sent from the Backup Server.
CAUTION
8-02
It is possible that another subsystem could have severe problems that could render the
server unusable, but switchover would not initiate a switch as the subsystems it monitors are
responding normally.
UDP Ping
In the UDP ping process, the Switchover subsystem on the Primary Server sends datagrams to the
Switchover system on the Backup Server. If a UDP ping is not received for 5 consecutive seconds, a
switchover is triggered.
FOR YOUR
INFORMATION
The UDP ping is not always reliable on high traffic networks. For this reason, many customers prefer
to disable it. In such cases, you can create and set the Switchover UDP Monitor server parameter to
No or 0.
If you want to change the default timeout values, you can create and set the following server
parameters in Interaction Administrator on the active server:
Switchover UDP Maximum Ping Delay A switchover is triggered when the failure count
exceeds 5.
FOR YOUR
INFORMATION
Set this value to between 1 and 3600 seconds. The default value is 5 seconds.
Switchover UDP Initial Ping Delay Specifies the time in seconds that the Switchover system
on the backup server waits before starting to listen for datagrams from the active server. The
default value is 5 seconds.
Often a UDP ping failure shows up in the system event log and indicates that a network problem
occurred or that the active server hung for some reason. The event log can also indicate the
Switchover component failed or restarted.
TS Ping
The Telephony Services (TS) ping specifies the number of seconds that Switchover will wait for a TS
ping response from the active server before signaling a TS failure. The value should be between 5
and 60 seconds. The default is 10 seconds.
In the event Switchover doesnt receive the response in the specified time, it will pause and try
again. Should it fail a second time, it will switch to the backup server.
FOR YOUR
INFORMATION
The pause time is configured using the Switchover TS Failure Retry Delay. This parameter specifies
the number of seconds Switchover will wait after the active server fails to respond to a TS ping
before retrying. The value must be greater than 0 seconds. The default is 1 second.
The Switchover logs are required to diagnose the root cause of the switchover, but it may be
beneficial to scan the system event logs and switchover log to ensure it wasnt a lost network
connection that caused the ping to fail.
THE MIRRORING PROCESS

In a switchover environment, the IC administrator creates identical IC systems to ensure that the
backup server is a mirror of the primary server. However, once the systems are installed, only
certain information regarding the primary IC system is monitored and replicated.
REGISTRY KEYS
Most of ICs key configuration data and all IC process tree information is stored in the IC servers
registry. During normal operation when the primary server is handling interactions, the switchover
system on the backup server monitors the IC part of the registry on the primary server. Anytime
a configuration change occurs or something triggers an IC registry key to change on the primary
server, that change is immediately copied and applied to the backup server. In this way, the backup
server stays synchronized with the primary (active) server and is ready to take over handling calls.
8-03
IC DIRECTORIES
When key files are added, changed, or removed from the primary IC server, that change must be
reflected on the backup server. When the IC switchover service is started on the backup server, it
starts monitoring default directories on the primary server and mirrors all add, remove, and update
file operations to the backup server.
CUSTOM MIRRORED DIRECTORIES

The backup server monitors and mirrors the directories listed in the Custom Mirror directory on the
primary server.
Any time a handler is published on the primary server, the directory containing the published
handler and a copy of the handler source file is automatically updated on the backup server.
That way, if a switchover occurs, the published, primary handlers are identical on both servers.
However, handlers in subdirectories under the Handlers directory are not mirrored on the backup
server by default. To duplicate changes in these other handler directories, specify them in the
CustomMirrorDir server parameter.
Mirrored directories must exist on local drives on the IC server; it is not possible to mirror
directories that exist on a mapped drive.
Each path should be a complete local drive and directory. Separate multiple paths with a semicolon
(e.g., D:\I3\IC\Server\Data1;D:\I3\IC\Server\Data2, etc.).
Once added to the appropriate server parameter on the primary server, the backup server will
mirror this change and immediately start monitoring the specified directories. The maximum
practical length of the sum of all of the paths entered in one server parameter should be no more
than approximately 2000 characters.
CAUTION
Do not mirror the \Work or \Logs directory on the IC server. Several IC processes write temporary
files to these directories that are not needed on the backup server.
INITIALMIRRORDIR PARAMETER
Directories listed for this parameter will only be copied the first time that the backup server
synchronizes with the primary server. Once the initial synchronization has occurred this parameter
will be ignored.
MIRROR EXCEPTIONS
Specifies extensions of any files that you do not want to mirror.
SUBDIRECTORY MIRRORING
To mirror subdirectories, place a + (plus) sign before the drive letter in the path.
EXAMPLE
FOR YOUR
INFORMATION
8-04
D:\I3\Server\Data1;+D:\I3\IC\Server\Data2 would mean that all the subdirectories under Data2

would also be monitored and mirrored on the backup server.
Mirroring may cause an issue when trying to delete directories. If an attempt is made to delete a folder
containing subdirectories that Switchover is monitoring, an Access is denied message will be received.
EFFECT OF A SOFTWARE SWITCHOVER ON SERVERS AND CLIENTS

The entire procedure of switching from the primary to the backup server takes approximately 30
seconds. During switchover, the following will happen:
All calls that were connected when the switchover occurs are switched automatically when the
IC client connects to the backup server. Interaction Client workstations are notified when this
switch occurs and are automatically restarted and pointed to the new primary server. IC users
will see an information box telling them the name of the new server to which they are connected,
but this will not require a configuration change on their part.
Email interactions in queue are re-queued.
Beginning with IC 4.0 SU4, continuity of chats, emails, and Callback objects is maintained.
Emails connected to an agent remain connected.
FOR YOUR
INFORMATION
Emails that contain HTML are converted to plain text, and markup is lost in case of a switchover.
Any attachment that might have been added to a draft email is not synchronized and must be added
again to the draft email after the switchover.
Chat interactions remain connected; the web user wont notice the switch happening. Intercom
chats remain connected as well, maintaining all the history information about the chat.
In case of a file transfer during a switch, the transfer might fail. In that case, the user must
re-initiate the transfer.
Callback requests remain in the system and continue to be routed after the switchover has
been completed.
REVERSING ROLES
After a switchover event, planned or unplanned, the roles of the two servers reverse, with the
former backup server now working as the primary.
When a switch occurs, Switchover notifies Telephony Services on the primary server that it must
stop and notifies Telephony Services on the backup to start.
The failed server is in a non-functioning state until the failure is resolved and the service is rebooted.
When IC starts up, Switchover queries the opposing machine to find out what state it is in,
primary or backup. If the other server is not available, the starting server will start up in primary
mode. If the opposing server is available and already in primary mode, the starting server will
start in backup mode.
SWITCHOVER IN WAN ENVIRONMENTS

It is often desirable to have a duplicate IC server in a geographically distant location to ensure services
will be readily available in the event of a disaster. By design, Switchovers should occur when the keep
alive connection has been lost between IC Switchover Server pairs. However, when relying only on TS
and UDP pinging, it is possible that Switchover services will respond to outages in the network instead
of only responding to a failed server. This is especially true in WAN Switchover setups.
WAN Switchover deployments can use a set of server parameters (Switchover Net Test A/B) to add
another ping process to ensure the chance of false positives as a result of WAN failures are minimized.
8-05
SWITCHOVER CONTROL PANEL

When the Switchover component is selected and installed on an IC server, the Setup program
creates a menu entry in the Programs \ Interactive Intelligence folder called Switchover Control
Panel. Use this to start the graphical interface that monitors and controls the Switchover system.
MANUAL SWITCHOVER
A manual switchover can be initiated to verify that the switchover pair will respond appropriately
when a switch occurs. To force a switch, click the Switch button. The manual switch will cause the
active server to cease taking calls and pass control to the Backup server. The switchover console is
useful for testing, and we recommend running it monthly during the maintenance cycle.
FOR YOUR
INFORMATION
After switchover, the former primary server is in a non-functional state until it has been rebooted.
UPGRADING SWITCHOVER SYSTEMS

You should not attempt to upgrade the IC system without following the steps outlined in the
IC_Automated_Switchover_System document in the Documentation Library.
8-06
APPLYING AN SU IN A SWITCHOVER ENVIRONMENT

Interaction Center allows for a no-downtime Service Update application. This is accomplished by
using Upgrade States to determine if both servers in the switchover pair are on the same SU level,
and how to proceed with replication.
When the primary and backup servers are on the same Service Update version and running in
auto-switch mode, this condition is called the Active state. When the backup server is running in
manual switch only mode because the backup server is on a different SU version than the primary
server, this is called the Upgrade state.
The Upgrade state has two options: Upgrade Higher and Upgrade Lower.
Upgrade Higher When the backup server is on a newer SU than the primary server.
Upgrade Lower When the backup server is on an older SU than the primary server.
Both of these Upgrade states provide limited replication and only allow you to manually transition
directly to the primary state and move the old primary to the failed state (demote it).
CAUTION
The backup server will not become the active server in the event that the primary server
experiences a failure when running in this state.
Manual Switch Only mode means the administrator MUST initiate the switch via the Switchover
Control Panel to make the Backup server become primary. Then, the SU should be applied to the
old primary. When SU installation is complete on the old primary, it will be rebooted and come
up as the backup server.
REPLICATION DURING UPGRADE STATES

If the backup server is not on the same SU version as the primary server:
There is no active monitoring of the primary server, so no automatic switches occur. The
Switchover Control Panel indicates that the backup server is in manual switch only mode.
There is live monitoring of only the DS entries under $CONFIG\Users and $CONFIG\
Passwords. Workgroups arent replicated by default so the Workgroup attribute for users is
not monitored.
If you make changes in Interaction Administrator while in the Upgrade state, only the changes
to users and passwords are replicated to the backup server.
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CONFIGURING POLYCOM PHONES FOR SWITCHOVER

The recommended method for configuring Polycom Managed IP Phones within a Switchover
environment is with the use of DNS Service (SRV) records and DHCP options.
DNS SRV records are used to define the hostname and port number of the servers supporting the
SIP line transport protocols (UDP, TCP, and TLS). A SRV record is created for each server/protocol
combination in the switchover pair. The record with the highest priority/weight combination will be
used to contact the server. If the server doesnt respond, the next highest record will be used.
Two DNS Host (A) records are created with the same name, but pointing to each of the servers
in the switchover pair. This allows the phones to find the active server when booting up and
performing SIP operations.
The 160 DHCP scope option is changed to reflect the name defined in the two DNS Host (A)
records created.
ADDING REGISTRATION INFORMATION TO MANAGED IP PHONES

Once DNS records have been created for the SIP services on the network, this information should
be added to a Registration Group in Interaction Administrator to be applied to Managed IP Phones.
A Registration Group can be used to specify DNS SRV records.
Managed Polycom phones are directed to DNS to find IC servers on the network. In IC 4.0 with
current Polycom firmware, DNS SRV treats both proxies as a primary server and approaches them
as a failover pair (i.e., the phone will park on the responding server). Whichever server responds to
SIP messaging will be first to receive the next call.
8-08
MAINTENANCE AND SWITCHOVER TIPS

Switchover should be included as part of a routine maintenance schedule. It is recommended to test
Switchover on a regular basis, such as every 30 days.
Performing a scheduled manual Switchover accomplishes several things:
Runs the Switchover process, ensuring that clients communicate with the CO to each
server correctly.
Ensures replication of files and configuration is occurring properly.
If you choose to initiate the process by means other than the Switchover Control Panel, you
can also verify that Switchover failure detection is working correctly.
TROUBLESHOOTING SWITCHOVER
During an unplanned switchover, an emergency support ticket (Code Red) should be opened. Be
prepared to supply the following information to support.
INFORMATION GATHERING FOR SUPPORT CASE

What is the NIC configuration for the servers?
Is there an active network connection between servers?
Can both servers resolve hostnames for each server?
Which server was primary and which was backup at time of switchover?
Are the servers in the same LAN or separated by WAN connection?
What version of IC including SU?
Any problems noticed with Switchover prior to unplanned switchover?
Be prepared to get registry information: HKEY_LOCAL_MACHINE\SOFTWARE\Interactive
Intelligence.
Logs required from BOTH servers:
Switchover
SwitchoverFileMonitor
Notifier
TsServer
SIP Engine
RemocoServer
DSServer
IP
Application and System Event logs (Windows Event Viewer)
Review Questions
Labs
8-09
Chapter
CIC System Support

Interaction Support Website
Common Issues and Solutions
Windows Troubleshooting Tools
Interaction Center Troubleshooting Tools

Packet Captures
9.1
Interaction Support Website

TIER 1 SUPPORT EXPECTATIONS
All Direct customer organizations are expected to provide Tier 1 support for their Interactive
Intelligence products and users. These responsibilities include:
Be the interface for all communication with your users.
Provide basic troubleshooting and support for hardware and software issues.
Attempt to reproduce all issues in a comparable test environment.
Keep the system up-to-date with new fixes.
Migrate to new versions as they become available.
Work with network and carrier technicians to ensure a high level of service.
Ensure that third-party software applications that interface with Interactive Intelligence
products are working and available.
Perform all troubleshooting tests requested by Interaction Support.
Gather and deliver all logs and files requested by Interaction Support.
INTERACTION SUPPORT
When an issue needs to be escalated to a higher tier, the Interaction Support team is available to
provide assistance through our Telephone IVR and the iSupport website.
The preferred method for contact is by phone through our global IVR. This contact method ensures
you will be able to quickly reach an engineer who is best equipped to assist with your specific issue.
The Interaction Support website offers a regional list of phone numbers and business hours. While
Standard Support is only available during regional business hours, Priority and Emergency Support
are always available.
BEFORE YOU CALL SUPPORT

When you call Interactive Intelligence Support regarding problems with your IC system, there are
certain questions that will need to be answered. Gathering the answers to these questions ahead of
time will greatly assist the support team in their ability to help you solve your issue.
What version of IC are you using?
What is the customer site?
Is this an existing incident?
Is this a development or production server?
Is this an upgrade or a new install?
Was this working before? In other words, has this ever worked?
Is this problem affecting all users?
What has changed on the system (e.g., hardware)?
Has any new software been loaded to the IC server?
Can the problem be reproduced? If yes, turn on logging for the appropriate subsystem(s), and
note the Call IDs for Support.
Have you looked in the Help?
Have you looked in the Knowledge Base?
9-02
CIC System Support
Be prepared to provide the following to Interaction Support:

IC server Station Configuration Information
Network Topology
Interaction Center Logs
Windows Event Viewer Application Log
Phone Configuration Files
Gateway Configuration Files
Network Captures
Call Recordings / Audio captures
EMERGENCY PHONE SUPPORT

Emergency Phone Support (Code Red) is available after normal business hours and is defined as:
The primary IC server is not running.
A site is running on a backup IC server for any unscheduled reason.
A site is unable to perform its core mission. Here are a few examples:
The customer is a Call Center whose business model is to make outbound calls, and they are
unable to do so.
The customer is an IVR-only installation, and their users are unable to navigate the IVR making
it impossible to route and answer calls.
LOGGING INTO THE SUPPORT WEBSITE

Once you have received your Interactive Intelligence Support user credentials you can logon at the
following address http://www.inin.com/support.
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CIC System Support
iSUPPORT WEBSITE
The iSupport website is available for customers to submit a new Support Incident, access existing
Support Incidents, and to chat with a Support Engineer. The iSupport website can be accessed at
https://isupportweb.inin.com, or through the Interaction Support website at www.inin.com/support
once you have received your logon credentials.
There are also links to download the iSupport App (iOS and Android), which offers a mobile link to
Interaction Support and the ticketing system.
SUBMITTING A NEW INCIDENT ON THE WEB

When opening a support incident, there will be information needed to access your information and
resolve the issue.
End-User Site The site for which this incident is being created.
Contact Person The individual who will be the primary contact for this incident. Generally,
this will be the individual that is submitting the incident. If someone else should be contacted
in regard to the incident, please select them from the list. If the contact individual is not in
the list, mention this in the worknotes of the incident. In addition, include all relevant contact
information for this person such as their email address and phone number.
Description A one-line summary of the incident. Please include keywords to better identify
the topic of the incident. The description is displayed along with the generated incident number
in the search results. The description can be a maximum of 100 characters.
Priority The relative importance or severity of the incident or question. These can be
categorized as Code Red, High, Medium, or Low.
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CIC System Support
CAUTION
If you are dealing with an issue that would constitute a Code Red situation, call Interaction Support
directly through the Global IVR; do not submit an iSupport ticket.
Product The product related to this incident. If the issue or question is not product specific,
select Not Product Specific. If you dont see your product listed, please select Not Product
Specific, and make a note in the worknotes regarding the missing product name. If the
incident is not related to an Interactive Intelligence product, please contact the products
vendor for support.
Product Version The version and software revision of the product selected. If the issue or
question is not version-specific, or if you are not certain what version of the product is involved,
select Not Version Specific. If the appropriate version is not listed, please make note of the
missing version in the incidents worknotes and select the nearest version to proceed.
Worknotes A detailed description of the incidents issue or question. Some general
questions to consider:
Is this a production or development server?
Is this an upgrade or new installation?
Was it working before? What has changed on the system?
Have there been environmental changes (e.g., carrier, network, or power)?
Is this happening to all users?
Is the problem reproducible? What are the steps to reproduce it?
Are there specific error numbers and/or messages?
Have you looked in the Online Help or Support Knowledgebase?
Are there alternate contact persons?
THE INTERACTIVE INTELLIGENCE SUPPORT LANDING PAGE

Once you logon, you will be presented with the Support sites landing page:
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CIC System Support
The Interaction Support website is split into four sections:

Product Information The Product Information section organizes official release information,
documentation, system prerequisites, updates, and recommendations for each Interactive
Intelligence product and their respective telephony platforms. Expect to find installation guides, SU
downloads, supported configurations, and licensing.
Interaction Support (iSupport) This is the web client for our Incident (or Ticket) Tracking System.
Youll be able to view your open incidents, search previous incidents, or report new low-priority
incidents via the web.
Self Help The Self Help section is maintained by Support and provides more in-depth architecturalfocused troubleshooting information categorized by topic or component. Each section will provide
architectural insight, common problems and solutions, what to collect for others problems, and how
to troubleshoot those problems.
Contact Us Not sure how to get in touch with us? This section provides all the information you
need about our efficient telephone IVR and each regions Management Contacts.
INTERACTION CENTER 4.0 SUPPORT PAGE

The IC 4.0 landing page contains the following information:
Release Information This sub-section contains items such as:
Release Notes This document summarizes new features in IC 4.0 and other products.
Critical Considerations This document highlights critical considerations for IC 4.0 related to
known issues and other important information required prior to deployment.
Known Issues This page describes known issues (including Engineering Specials) for IC 4.0
GA and Service Updates.
Localization This page documents which Language Packs are available for IC 4.0 (including
which languages are added with each Service Update), how to obtain them, and how to apply
them to existing installations. The page also offers simple localized dial plans, and localized
quick reference documentation for some languages.
Licensing This sub-section contains relevant links for managing your licensing.
Documentation This sub-section contains links to the Documentation Library, Documentation
Updates, Printable Documentation, and also Localized Quick Reference Documentation.
The importance of the IC Documentation Library is worth noting, as it merges all available help
systems, installation and configuration, and other reference documents into a single searchable unit.
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CIC System Support
Telephony Architecture Links to the Interaction Media Server 4.0 Support site.
System Requirements This section contains links to system hardware and software requirements,
a virtualization technical reference, and a link to the Interactive Intelligence Testlab.
Service Updates The latest Interaction Center Service Update (SU) and any relevant information
can be found in this section. There is also a preview summary link that can be used to see what might
be coming in the next Interaction Center SU.
Utilities and Downloads A majority of the utilities in this section are used during the initial
installation of or migration to a new version of Interaction Center. However there are some useful
post-installation tools such as:
CSV Managed IP Phone Lists Download the zip file containing two types of .CSV lists based
on 1) Template and 2) Type, Manufacturer, and Model. Use one of these lists to create new
managed IP phones and associated SIP Stations in the Interaction Administrator Managed IP
Phone Assistant.
Password Checking Tool Readme To improve system security, a password checking utility
is available to identify users on an IC server who have left their password at the default of
1234. The PWCheckU.exe utility can be found at the \I3\IC\Server root directory on the IC
4.0 Server. See the Readme for details.
THE SELF-HELP SECTION

The Self-Help section contains a wealth of resources that are helpful for troubleshooting and
better understanding your Interaction Center system. This section complements the thorough
Documentation Library by listing information that is specific to the selected topic.
A majority of the topics will also contain a menu item called Triage. This Interactive Intelligence
documentation can be very helpful if you find yourself needing to contact support. They will need
certain logs, configuration files, or even an export of certain Windows Registry keys to be able to
effectively diagnose the issue and assist you. As an example, the screen-shot below contains the
Triage documentation for the Interaction Client.NET edition:
9-07
CIC System Support
9.2
Common Issues and Solutions

INTERACTION CENTER ISSUES
Below is a list of fairly common symptoms, with some common causes and troubleshooting tips
(Note: The frequency of these occurrences should also be monitored to narrow down the causes):
Calls Wont Connect
This could be the result of signaling issues between SIP devices, including IC and SIP Proxy;
dial plan; general network issues; vendor compatibility.
SIP signaling problems Check configuration of SIP lines / line groups, station registration.
Dial Plan SIP routing may require the creation of dial plan objects for specific connections
(e.g., PSTN gateways and peer IC servers). Misconfigured dial plan entries can cause specific
calls to fail. Line groups must be attached to dial plan objects to ensure the correct SIP line is
being used.
Network issues Verify network addresses of affected devices. Also check duplex settings
of affected devices. Should be set to full duplex.
Vendor compatibility Always refer to testlab.inin.com for updated information on validated
network equipment and releases. Also verify recommended settings and configurations for
supported equipment (e.g., DTMF 2833 payload type 101; Disable delayed media for all SIP
carriers, etc.).
No Audio / One-way Audio
It is possible for a call to be established, but the voice stream to be lost. This may be due to a
problem with signaling or how RTP packets are routed.
Check the SIP settings on devices and IC.
Check NAT and Firewall configuration.
Check SIP connection address on affected stations (SIP Registration) in IC.
Use a network analyzer capable of RTP capture and playback. Doing this in various links
within the call path can show where the audio may be lost.
Audio Quality Ticking / Popping
Network congestion (typically at edge routers) causes jitter, sequencing problems, and packet
loss. Packet loss can also come from route flapping. Packet loss should be less than 1%, jitter
should be less than twice the packetization rate.
If load balancing is configured on network routers, it should be set to balance on a per session
basis, not per packet.
Jitter buffers should be configured on gateways to Interactive Intelligences recommendations
(1/30- Optimization factor = 1, jitter buffer = 30ms)
Audio Quality Echo
Echo is never caused within the IP network, but delay may make echo noticeable and annoying.
May be caused by 4-wire digital to 2-wire conversion (local loop).
Typically caused by high-volume on speakerphone, headset, or misconfigured gain settings in
SIP devices.
Adjust or verify gain settings. Only validated firmware and configuration files should be
loaded on to supported 3rd party devices.
Replace problem hardware.
Address with telco if the problem is narrowed down to their circuits.
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CIC System Support
Audio Quality Tinny Voice

Transcoding (changing codecs) multiple times during a conversation can adversely affect the
quality of the call.
Transmitted sound is complete but often altered or distorted.
Codecs should be universal, or number of transcoding operations (devices altering codecs)
is minimized.
Audio Quality Talk Over
Callers talking over each other is typically the result of significant latency. Latency can cause
other issues, such as packet loss and jitter, but if the audio seems to be intact, and users are
talking over each other (conversational timing seems affected), then the delay is substantial.
Optimally, delay should not exceed 150ms end-to-end in any conversation.
Typical causes of high delay levels are network congestion, excessive buffering, and routing
(e.g., excessive hops in path). Each hop in a transmission introduces more opportunity for
packets to wait in buffers.
TROUBLESHOOTING TIPS
It is important to gather information such as:
Types of calls affected and devices involved:
Internal (station-station) Involves phones and IC, maybe Media Server, maybe gateway,
and network.
External calls Can involve same elements as internal calls, but will also involve PSTN
(FXO or digital) gateway.
Conference calls Introduces dependency on conference resources.
Users affected Introduces configuration issues; permissions.
Frequency of problems Intermittent issues differ from constant conditions.
CRITICAL
Interactive Intelligence Support may ask you for a network capture. It is imperative that you or
someone else understands how to conduct a capture, and where in the network the capture should
be taken.
PHONE PROVISIONING ISSUES

Phones are typically set up for DHCP when they are unboxed and put on the network. DHCP
options are used to enable the phones to pull boot server information from the DHCP server. In the
event phones have difficulty provisioning, there are several things that must be checked.
Version of firmware on the phone Is there an upgrade path from the current firmware on the
phone and the firmware level on the IC server?
Firmware is updated on the IC server with Service Updates. Phones receive new firmware
from the IC server during the provisioning process.
Connection to the DHCP server DHCP relies on broadcast traffic to be sent from client to server.
DHCP options configuration
Boot server configuration on the phone Phones should have the correct configuration to
receive boot server options from DHCP servers.
The boot server information is configured in the phones menu. These settings determine which
DHCP options are used for provisioning.
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INTERACTION MEDIA SERVER ISSUES

The Media Server has been configured, but calls do not go through Make sure the Enable Media
Server box is checked under the Media Server container within Interaction Administrator.
Calls are not being recorded and/or compressed on the media server Make sure that you have
configured the correct paths for the recording directories.
RecordingBaseUriLocal Local path on the Media Server where the calls are being stored as
they are being recorded (as .wav files).
RecordingBaseUriRemote Path to the share hosting the recordings on the Media Server.
This path is passed to TS and is used by Interaction Recorder as well, so it must be accessible
to the CIC Server. This path is unique to each Media Server you have. This will have the
recordings stored in the compression format specified on the Media Server.
Recordings are not being compressed in the specified format Specify the recording compression
format by using the RecordingMimeTypeDefault property on the Media Server. Remember, the
value for this must be specifiedthere is no drop-down list from which to choose. So ensure the
correct string has been entered. Currently, the following values are supported:
audio/PCMU
audio/PCMA
audio/L16
audio/GSM
If you do not type this correctly, you will see an error logged on the Media Server log. If you have
specified this correctly and the recording is still in an incorrect format, check what format is
specified in Recorder config.
Recorder will access the compressed recording from Media Server and then re-compress it if
you have specified a compression format on Recorder as well.
If you only want Media Server to compress recordings, and not recorder, then do NOT check
the compress recordings option in Recorder config.
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QoS enablement issues with Media Server To enable QoS on Media Server, you need to set the
RTPQoSTaggingEnabled property on the Media Server. If you see problems after that, please make
sure that the NIC Driver version is correct. More information on this can be found in the Media
Server document under the Configuring Quality of Service section.
Media Server License is expiring soon, how do I create a new one? Please contact support, or
email the licensing team. Media Server licenses cannot be renewed or generated via the License
Management site currently.
How is a Media Server selected for a particular call? If you have more than one Media Server for
an xIC Server:
If you have a Media Server added to a region, then that Media Server will be used for all calls
for that region, as long as resources are available. If the Media Server in that region runs out
of resources, then any other Media Server from outside the region is chosen.
If there are no region-specific Media Servers, then one is chosen based on the load/utilization.
INTERACTION SIP PROXY ISSUES

I have BCM; how do I add load balancing? Obtain the new license, stop the service, apply the new
license file, and restart it.
How do I check the version or license information? The About tab in the status section
displays the version of Interaction SIP Proxy that is running. The version string consists of a major
number, minor number, and patch level separated by periods. Monitor http://www.inin.com/
ProductSolutions/Pages/Interaction-SIP-Proxy.aspx for updates and patches.
The About tab displays information about the status of the license. If the license is expired, then all
SIP functionality is disabled. Follow the on-screen instructions to purchase your license.
How do I see what calls are currently active on the proxy? The session status page displays all
the currently active sessions or calls. Active calls are ones that have completed the setup process
and are engaged in a conversation. Calls that are dialing or waiting for a party to answer are not
shown here.
The Connected column shows the local time when the call setup was completed.
The To column shows the user portion of the dialed SIP address. It comes directly from the To
header of the originating SIP INVITE.
The From column shows caller-ID of the calling party. It comes directly from the From header
of the originating SIP INVITE.
The Call-ID column shows the unique call identifier assigned to this call by the dialing party. It
comes directly from the Call-ID header of the originating SIP INVITE.
The X can be used to clear a call from the Session status page.
How do I get details of the devices registered with the proxy? The registration page is useful
for determining which devices have successfully registered with the Interaction SIP Proxy. Also, if
Registration routing is enabled, then the registration information is used as a target for inbound
calls. This means that if an inbound call failed to match a route item but did match an address of
record, then the call would be routed to the contact address associated with the address of record.
Registrations will be removed from this page when the phone deregisters itself with the appropriate
SIP message or the registration time has expired.
How do I check the status of servers added to the proxy server plan? The server status field
displays the SIP operational status of all servers in the server plans. Each server is periodically sent
a SIP OPTIONS message; if a response is received, the server is considered up and operational. If
no response is received, the server is considered down and not operational. All servers that are not
operational are not used in the dial plan during normal operation. This feature is only available with
the advanced version.
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INTERACTIVE COMMUNITY
The Interactive Community is a free and public service provided by Interactive Intelligence to
promote an online community devoted to the sharing of ideas and information regarding Interactive
Intelligence software. You can access the community here: http://community.inin.com.
Opinions expressed on the Interactive Community are not those of Interactive Intelligence, and
Interactive Intelligence accepts no legal responsibility for their veracity or nature.
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9.3
Windows Troubleshooting Tools

WINDOWS EVENT VIEWER
The Event Viewer is an Administrative tool provided by Microsoft that maintains logs about
programs, security settings, and system devices. It is a valuable troubleshooting tool because the
Operating System, IC, and other programs report errors and warnings to this program.
Often an error or warning in one of the event logs can point an administrator in the right direction for
troubleshooting a problem. The Event Viewer logs should be reviewed by the IC administrator when
the system undergoes maintenance, upgrades, reports errors, or is not functioning as expected.
It can be accessed from the Start > All Programs > Administrative Tools menu of Windows Server
2008 R2. Interaction Center writes event information to the Application Log. The Event Viewer
reports three kinds of event messages:
Error A significant problem, such as loss of functionality. For example, if a service fails to load
during startup, an error will be logged.
Warning An event that may indicate a problem. For example, when disk space is low, a warning
will be logged.
Information An event that describes the successful operation of an application, driver, or service.
For example, when a network driver loads successfully, an Information event will be logged.
WINDOWS EVENT VIEWER LOGS

There are three types of Windows event viewer logs.
Application log
Security log
System log
APPLICATION LOG
The Application Log contains events logged by applications or programs.
For example, a database program might record a file error in the application log. The developer of
the software program decides which events to record. When starting IC, numerous messages will
be written as the Remoco subsystem starts all the other subsystems. It can take several minutes for
all the functions of Interaction Center to start. It is the job of the Remoco subsystem to keep all the
Interaction Center processes running and Remoco will attempt to restart any process that fails. Any
error messages related to Interaction Center will appear in the Event Viewer under the Application Log.
This is the first place to check when trying to isolate a problem with the Interaction Center platform.
FOR YOUR
INFORMATION
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Remoco Message 15008 indicates that all the Interaction Center subsystems and functions were
brought online without error. Look for this message to verify that the Interaction Center is fully started.
SECURITY LOG
The security log can record security events such as valid and invalid logon attempts, and events
related to resource use, such as creating, opening, or deleting files. This log should only be viewed
by Network Administrators, and is not generally useful for troubleshooting Interaction Center.
SYSTEM LOG
The System Log contains events logged by the Windows Server 2008 R2 system components. For
example, the failure of a driver or other system component to load during startup is recorded in
the System log.
WINDOWS PERFORMANCE MONITOR

WHAT IS PERFORMANCE MONITOR (PERFMON)?
Performance monitor (Perfmon) is a tool included in the Administrative Tools group of Windows Servers
that enables monitoring of system resources either through real-time viewing or historic logging.
Perfmon obtains an overview of system health without placing extra load on the monitored server.
Windows is already tracking this information (see Task Manager | Processes tab).
Once Perfmon is configured to capture on a remote computer, the IC server begins sending this
system information over the network to the remote, monitoring computer. The only effect on the
IC server is the increase in packets sent.
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WHAT IS MONITORED?
Perfmon can be configured to obtain specific data regarding the functioning and use of system
resources. There are many items, or counters, that can be monitored including Processors,
Processes, Threads, Memory, and Physical Disks.
WHY GATHER A PERFMON CAPTURE?

Perfmon is the best tool for Interaction Support to diagnose IC server performance issues. IC logs
can only provide clues regarding system health by logging when the individual subsystem is backing
up, creating more threads, and so on. A Perfmon capture provides information on:
Which particular process is overtaxing the CPU, causing server latency.
Whether disks are overtaxed and unable to keep up with the amount of read/write instructions.
Whether a process is eating up all available memory on the machine.
WINDOWS SERVICES PANEL

Windows Server 2008 R2 provides a Services Panel that allows you to monitor services and their
status. There are two columns that provide quick access to this information: the Status Column and
the Startup Type column. We can use this utility to determine whether all necessary services for the
Interaction Center Platform are running.
From this panel, it is also possible to start and stop services, and set services to start manually or
automatically when the server is restarted.
There are several IC-related services that run in the Windows services panel. Some are crucial to
Interaction Center functionality, such as the Interaction Center, Interaction Gateway, Interaction
Proxy, and Interaction Media Server services. Others, such as ININ TFTP, affect only non-crucial
features within IC.
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Starting a Service By double-clicking on a particular service, you can modify the start-up
properties of that service.
CAUTION
Modifications to the logon information should only be done in setup assistant.
WINDOWS TASK MANAGER

The Task Manager allows you to view all currently running processes and programs in Windows
Server 2008 R2.
The Task Manager also provides a quick visual view of the health of the server via the Performance
tab. If your server is nearing 100% usage, investigate whether a process has run away and is
using a large amount of resources. Task Manager allows you to terminate runaway processes by
selecting the specific process and clicking the End Process button. The task manager can also show
if a program is not responding. Interaction Center subsystems will appear as processes in the task
manager, not as applications.
CAUTION
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A program that is not responding can sometimes freeze the server and cause other unrelated
programs and processes to fail. For this reason, we recommend that you do not use the Interaction
Center Server for other nonessential programs.
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9.4
Interaction Center Troubleshooting Tools

IC SYSTEM MANAGER
When you start Interaction Center, each subsystem starts based on its location within the IC
process tree.
IC System Manager is a graphical user interface that can query, stop, restart, and trace the
Interaction Center server subsystems. The window lists information such as process tree
subsystems, startup order, status, PID (Process ID), elapsed time, CPU and memory usage, virtual
memory, thread count, and handler count.
Monitor Returns the most current data from a particular subsystem or all subsystems. You
can also set these subsystems to poll data at set intervals. You can find out the current state of
a subsystem, or view it over time using the Monitor menu.
Control This menu option allows you to stop and restart subsystems.
Trace Tracing is the logging of messages generated by subsystems that details actions
performed by that subsystem. These messages are recorded in a file and can be used to track
performance. Most subsystems start with all tracing set to a default level (which includes status
messages, warnings, and errors). The trace menu allows you to raise or lower the tracing levels
for each subsystem. Use Trace Configuration to set levels on the subsystems in the process tree.
Use All Trace Configuration to set levels on all the subsystems started by the Interaction Center.
If you want (or need) to change the location where tracing logs are stored, this utility can set the
new location.
View Use this to view or hide the toolbar and Status Bar
Help Provides help for the IC System Manager.
FOR YOUR
INFORMATION
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Rights to configure tracing must be set by your Interaction Center Administrator.
CAUTION
Tracing levels should only be changed at the request of Interactive Intelligence Support, or someone
intimately familiar with its operation. All tracing levels can generate many megabytes worth of data,
potentially causing memory issues and filling disk space on your server.
ININ TRACING CONFIGURATION UTILITY

The Trace Configuration Utility can be launched from the Start > Run window (Start >
Interactive Intelligence > Trace Configuration Utility). By default it is present on all Interaction
Center server installations.
One feature of this utility is that it allows you to reset all topics to their default tracing levels (the
levels set at after first installing Interaction Center). By default, some topics are set at 60 while
others are set at 41. That is why this function can be useful.
Also, if you contact Interaction Support with an Interaction Client, Interaction Center Business
Manager, or Interaction Attendant issue on a workstation, you may be required to gather logs for
those client-side applications. You can use Trace Configuration Utility to do this.
FOR YOUR
INFORMATION
Interaction Support recommends raising the relevant subsystem topics to level (80) Notes when
troubleshooting. However, while investigating issues with client applications, it is recommended the
topics be raised to level (100) Verbose.
IC LOG LOCATION
The Interaction Center system has built- in logging that can help you or an Interactive Intelligence
Support team member diagnose system behavior. The IC subsystems write system messages in the
form of logs. These logs are binary files that can be examined using the ININ Log Viewer.
The logs are stored in log files that are associated with the specified subsystem in the \I3\IC\Logs\
dates of the current week folder.
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CAUTION
Opening the logs in the ININ Log Viewer on production servers is not recommended.
ININ LOG VIEWER

The ININ Log Viewer allows administrators to view the logs created by IC subsystems. It is located
in the \I3\IC\Logs folder. The ININ Log Viewer allows you to open, view, and filter the diagnostic
messages written to the IC logs.
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ININ Log Viewer Menu When you first open up a log, it will appear in a new Log Viewer window.
The Log Viewer also has a toolbar that allows you to quickly perform a multitude of functions by
clicking the appropriate icons.
File Allows you to open, reopen, or close log files. You can also snip data from the log or merge
multiple logs together into a new file.
Edit Provides options to copy entries and perform a basic search.
View Allows you to change how the log and messages are viewed.
Filter Allows you to filter the open log to view specific message types or threads. Typically,
filter is only necessary when troubleshooting a problem, as instructed by a support engineer.
You can filter by topic, context, or thread ID, or set a filter based on a search string. You can also
remove all current filters to return to the default view.
Search The Search menu allows you to search for a specific message or to jump to a specific
time stamp in the log.
FOR YOUR
INFORMATION
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The IC SIPEngine log, if turned up to 80, will show all the SIP signaling used in a SIP session. Log
Viewer can be used to view SIP traces in this format.
9.5
Packet Captures
INTRODUCTION TO WIRESHARK
WHAT IS WIRESHARK?
Wireshark is a free, open source packet analyzer. It can be downloaded at www.wireshark.org/
download.html.
WIRESHARK FEATURES
Capture live packet data from a network interface
Display packets with very detailed protocol information
Open and save captured packet data
Import and export packet data to and from many other capture programs
Filter packets on many criteria
Search for packets on many criteria
Colorize packet display based on filters
Generate statistics
FOR YOUR
INFORMATION
You can convert capture files from one format to another by reading in a capture file and writing it
out using a different format. This makes Wireshark captures compatible with analyzers from many
other vendors.
WIRESHARK: GETTING STARTED

NAVIGATING THE WIRESHARK MENU
File Contains items to open and merge capture files, save / print / export capture files in whole
or in part, and to quit from Wireshark.
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Edit Contains items to find a packet, time reference, set your preferences, or mark one or
more packets (note: cut, copy, and paste are not presently implemented).
View Controls the display of the captured data, including the colorization of packets, zooming
the font, showing a packet in a separate window, and expanding and collapsing trees in the
packet details pane.
Go Contains controls to jump to a specific packet.
Capture Allows for starting and stopping captures and creating capture filters.
Analyze Contains items to manipulate display filters, enable or disable the dissection of
protocols, configure user specified decodes, and follow a stream.
Statistics Contains menu-items to display various statistic windows, including a summary of
the packets that have been captured, display protocol hierarchy statistics, and much more.
DISPLAY FILTERS
Wireshark is a useful utility for capturing and analyzing packets. Display filters can add to this
by allowing control over how captured packets are viewed (post-capture filtering). These filters
range from simple to complex and are created using a variety of operators and strings to create
filter expressions.
SIMPLE PROTOCOL FILTERING

This is a simple way of viewing packets from as few as one protocol from a capture session. By
entering one or more protocol names into the Filter field and clicking Apply, your search can be
narrowed down to the desired protocol.
FOR YOUR
INFORMATION
If the Filter field turns green when entering data, the string is valid. If it turns pink, the string is
invalid / incomplete.
Searches can be run for various criteria, including protocols and device addresses.
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The Expression button (next to the Filter field) can be used to display a list of filter expressions and
operators (Relations) to create strings for searching through compiled data.
FILTERING WITH EXPRESSIONS

EXAMPLE
Using Filter Expressions to create a filter for SIP to messages containing the SIP User address
71902 creates a string that Wireshark understands and will use to filter out all other non-relevant
data. The filter was created using:
Field Name = sip.to.addr SIP to address
Relation = Contains
Value (character string) = 71902
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FOR YOUR
INFORMATION
There are many other display filter examples at http://wiki.wireshark.org/DisplayFilters.

Strings can also be created to search specific devices via IP or Ethernet address.
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EXAMPLE
Example: ip.addr == 172.16.1.107 searches packets containing a specific address. Furthermore, this
address can be searched as just a sender, or just a receiver.
A simple way to create expressions to search for specific IP addresses, either as source, destination,
or both, is to right-click in the Packet List Pane and select Prepare Filter or Apply Filter for a specific
packet containing the information you wish to search.
This specific filter displays all captured packets in which IP address 172.16.131.163 was the Source
(ip.src == 172.16.131.163).
VIEW FILTERS
The Menu / View options allow for a variety of custom controls for viewing and analyzing packets.
For many, the default colorization, for example, is difficult to read. Color rules can be applied for
conversations and other criteria to make it more usable.
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A useful utility under the View options is the Time Display Format option. This enables the display
packets in the following ways:
Date and Time of Day
Seconds Since Beginning of Capture
Seconds Since Previous Captured Packet
Seconds Since Previous Displayed Packet
Seconds Since Epoch (1970-01-01)
These settings can be useful in determining delivery time of packets and time stamping packets
during post-capture analysis. If set to Seconds Since Previous Captured Packet or Seconds Since
Previous Displayed Packet, packetization rate and jitter levels can be identified. This example
shows packet number, time (Seconds Since Previous Displayed Packet [filtered for RTP]), Source /
Destination IP addresses, and the Protocol (RTP).
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The Time column displays a reference time at the beginning, then each following packet is based
on packetization rate (approximately 20ms). Slight jitter can be seen in the RTP stream by quickly
analyzing the time column (displayed in milliseconds).
CAPTURE FILTERS
Unlike Display filters, Capture Filters affect the data that is captured, which can minimize the
amount of data you need to store for analysis.
It is possible to use existing capture filters or create new ones. The following steps will walk you
through the creation of a new Capture Filter for SIP:
From the Menu, select Capture / Capture Filters.
Click New, and enter SIP as the name for the filter.
In the Filter String field, enter the string tcp port 5060 and host 192.100.100.101. (This will
capture SIP messages to and from this IP address.)
Click OK to save the filter.
To use the filter, in the Menu select Capture / Options.
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This will display the Capture Options dialog.

Click the Capture Filter button and select the SIP filter from the list.
Click Start, and the filter is applied to a live capture.
VoIP-SPECIFIC TOOLS
Wireshark has integrated some useful VoIP-specific tools into the interface, which makes it easier to
isolate and analyze SIP and RTP traffic above and beyond simple filtering.
The Statistics Menu provides, among other things, the ability to:
Create graphs
Isolate conversations
Gather statistical information per protocol
Analyze RTP traffic and extract audio
View VoIP calls, SIP tracing / session setup, and playback audio
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GRAPHS
Many types of graphs can be created to display a variety of conditions. A simple graph can represent
jitter within an RTP stream.
CONVERSATIONS
It can be useful to drill down on conversations, or sessions between specific devices, or all sessions
for a specific protocol. The Conversations option under Statistics provides this functionality.
Protocol Statistics Provides statistical data per protocol or other filtering process. Can be used to
extract specific information from a data stream.
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ANALYZE RTP AND PLAYBACK AUDIO

Very useful for analyzing RTP packets for arrival time, sequencing, jitter, and packet loss.
From this screen, it is also possible to save the audio payload as an .au file and playback in any
application that can play .wav files, such as Media Player.
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VOIP CALLS, SIP TRACING, AND AUDIO PLAYBACK

The VoIP Calls option under Statistics is an extremely useful utility with Wireshark. This feature
enables users to identify individual VoIP calls by SIP session, along with the associated RTP streams.
It also provides a graphic illustration of SIP sessions and messaging, and the ability to play back the
audio as an entire conversation, or as isolated RTP streams (bi-directional).
From this screen, it is possible to view individual calls, the state of the call as captured, the duration
of the call, parties involved (source and destination), and packets transmitted.
In addition, there is an option to graph any of the sessions using the Graph button.
If the Player button is selected, the associated RTP stream(s) can be isolated and played back with
the embedded audio player. This allows for either or both streams to be played. In addition, statistical
information about the streams is displayed (duration of the call, jitter, and sequencing data).
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WIRESHARK CAPTURES
Wireshark offers a default view for viewing captured packets. This view allows you to see time
stamp information, source and destination addresses, and protocols.
The main Wireshark window for viewing captures is split into three window panes:
Packet List Pane Shows a list of all captured packets. Selecting packets in this pane affects
what displays in the other two panes.
Packet Details Pane Provides a detail of the selected packet. This can include source and
destination port information, domain information within SIP packets, byte length, and other
information.
Packet Bytes Pane Displays data from the selected packet (bit-level information).
Review Questions
Labs
Lab 15: Stopping and Starting Services
Lab 16: System Manager
Lab 17: Installing and Using Wireshark
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10
Chapter
CIC System Maintenance

Preventive Maintenance
Updating the CIC System
10.1
Preventive Maintenance
AVOIDING PROBLEMS
After installation:
Create a change log for the system, and document any changes immediately.
When troubleshooting, be sure to implement only one change at a time. Document each change,
and then document the result.
New audio prompts require CCITT mu-law, 8-bit mono, 8-KHz sampling rate.
Consult the support website for anti-virus software requirements.
Document the telephony configuration, including a wiring diagram.
Label all cabling to the IC server.
Periodically review the consumption of resources to determine server load. Some examples of
resources to monitor include:
Line utilization
Station utilization
License consumption
CPU utilization
Memory Utilization
DO NOT make changes to an IC system during production hours. If you are a 24 x 7 site,
schedule changes for off peak hours.
Test any implementation changes before altering a production system.
Maintain an adequate spare hardware kit.
Do not put IC log files on a separate server. Store Log files on the IC server, and if possible, on
a separate hard disk from the system drive and the RAID array. We recommend RAID 1 for log
drives, however, for more robust sites with large database activity, RAID 10 is recommended.
Perform a server analysis before and after adding more load to the system.
Backup servers are production systems; do NOT treat them as test systems.
DO NOT install third-party software on an IC server unless approved by support. Installation
of third-party software has caused errors by altering and deleting critical files necessary to
the IC system.
Inform IC administrators of any potential changes to domain accounts, such as password changes.
FOR YOUR
INFORMATION
There is a KB article that outlines the necessary steps to ensure a successful change of the IC
Admin password: Reference KB Q100327191209459.
The Interaction Center is a telephony system. This system relies on the proper functioning of the
Central Office (CO) that is servicing any installation.
Peripheral server performance affects the IC system. Interaction Center often works in tandem
with various peripheral systems. The database servers, mail servers, and other systems used
by IC affect the performance of the IC system. Changes to the peripheral systems can cause
performance problems on the IC server.
Contact the IC admin when changes occur on peripheral systems.
Password changes in the messaging server, database server, or network can lead to a failure
on the IC system if the same change is not implemented on the IC server.
Thoroughly research and apply Windows Server 2008 R2 Group policies with caution. Inform
the IC administrator of any changes in Group Policy. Since these policies are reapplied at preset
intervals, a system may continue to function for hours or days and then suddenly fail (when the
new policy hits).
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EXAMPLE
A policy affects the IC server. The problem caused by the policy is corrected only to occur again
when the policy is reapplied. A poorly designed or severely restrictive policy can adversely affect
an IC server.
CHANGE MANAGEMENT
Basic administrative changes like adding users, stations, or roles are expected. High-level changes,
like changes to the messaging server, database server, and the IC server itself, should not be made
without documenting these changes and including a back-out plan.
Examples of changes that could severely impact the IC system:
Changing server names or addresses
Changes to the domain name
Any changes to the database or db server
Loading software on the IC server or any servers used by the IC system
Making hardware changes to the IC system or any servers used by IC
Making service level changes through the telco provider
Restarting the IC server
Restarting a server used by the IC system
Changes to any peripheral device that can affect the system (e.g., changes to switches, gateways,
channel banks, and CSUs)
Changing Windows Server Group Policy Objects
Any permissions changes (includes Domain or IC rights and permissions)
CAUTION
Changes such as those listed can severely impact the performance of the IC system. Be proactive
when considering changes to the IC system or any servers used by the IC system. Plan to
implement changes during low or nonproduction hours. Have a backout plan in case there are
unexpected results.
FAULT TOLERANCE
Fault resilient servers have the following common characteristics.
Redundant, hot-swappable power supplies Internal hot-swappable, redundant power supplies
come in at least two varieties: Load-sharing and non load-sharing power supplies.
Multiple redundant fans We recommend at least three fans. The ventilation system of the
fault-resilient server should force cooling air through a plenum chamber and across all the cards
to cool the entire system.
Redundant or hot-swappable disk drives RAID-10 is the recommended RAID configuration.
RAID 5 is no longer supported.
Self-monitoring system with notification The best fault resilient PCs have self-monitoring
systems to notify the system administrator (and ideally the operating system) of a component failure.
Mean Time Between Failures (MTBF) The predicted elapsed time between inherent failures
of a system during operation. MTBF is the calculation of the arithmetic mean (average) time
between failures of a system component. The MTBF is an important consideration in determining a
preventive maintenance schedule and spare component inventory.
10-03
Other Requirements for a Fault Resilient System The following is a list of additional
considerations to ensure a fault tolerant network:
Spares kit for important hardware components
Additional disk drive
A power supply
A single board computer
Cooling fan
Network card
Memory chips
Video card
PREVENTIVE MAINTENANCE FOR IC SERVERS

MAINTENANCE PROCEDURES FOR IC SERVERS
Proactive steps can reduce system problems and ensure that changes your organization makes
are fully tested and more likely to be successful.
Maintenance recommendations such as the application of Microsoft service updates or
hotfixes, Interactive Intelligence performs an automated update and testing of new hotfixes and
publishes approved hotfixes on Testlab.
The Maintenance Checklist indicates some of the standard, recommended maintenance practices
for the IC server.
Checking these items can avert system problems and prevent problems in advance:
Monthly
Physical check of all cables connected to the IC server. Make sure that there are no loose or
frayed cables. Make sure that they are securely connected.
Conduct a Switchover test.
Weekly
Test the UPS that connects to the IC server.
Check the amount of free space on the hard drive.
FOR YOUR
INFORMATION
If Support asks you to turn on detailed tracing for troubleshooting purposes, you need to check the
available space much more frequently. Check either daily or possibly even hourly. Contact Support
for any unexplained usage of the available space, especially if it is in large increments.
Daily
Back up the hard disks on the IC server.
Run differential nightly backups and a full backup during the weekend.
Monitor the Event logs for any STOP errors.
FOR YOUR
INFORMATION
In a stable system, monitoring the Event logs should only take a few moments because you should
not find many errors.
Use SNMP to monitor for any STOP errors and have the error emailed to the appropriate
personnel for troubleshooting.
10-04
RECOMMENDED BACKUP PROCEDURES

DATA BACKUP RECOMMENDATIONS
There are three basic backup recommendations:
A manual process to create a copy of all vital implementation-specific data.
Dedicated backup system including a local tape drive and appropriate software.
Adding the Customer Interaction Center server to an existing network backup scheme.
FOR YOUR
INFORMATION
Consult the Technical Reference document Data Backup for detailed information on relevant
subdirectories included in the backup of the \I3\IC directory.
ADDITIONAL STORAGE CONSIDERATIONS

It is advisable to create and maintain a directory off the server that contains other important
configuration files and licenses. This directory should include the following:
IC license
Gateway configuration files
IP phone configuration files (non-managed IP phones)
Dial plan backup
Attendant configuration and supporting audio (resource folder)
Custom handlers
Dev / Test server

Use a development / test server to validate system changes before putting them into a production
environment. The following is a list of suggested items for testing on a development server:
Are proposed changes compatible with the current environment?
Will the changes be complete within the expected time frame?
Windows updates
ININ service updates
Attendant changes
Custom Handlers
Domain changes
Dial plan changes
Changes to PSTN services / circuits
30-MINUTE HEALTH CHECK

Performing a health check is a part of CIC system maintenance. The 30-minute health check allows
you to collect the right triage data for support.
In order to perform a quick IC health check, do the following:
1. Review App/Sys Windows Event Viewer logs on the IC server.
2. Check Supervisor
Check for PMQ activity. (You can also check the PMQ folder on IC.)
Are recordings being processed?
Do you see inbound/outbound calls on the lines?
10-05
In the Session Manager section in Supervisor, do you see clients logged on on each listed
Session Manager?
Do they have free drive space?
Lastly, check IC System Manager and see if all the subsystems are up and if any have
significantly different up times?
3. If you are a Microsoft customer, perform the MS Health Check.
4. If you need to check specific IC functionality follow the Health Check page in the appropriate
Self-Help section of the support website.
ANNUAL LICENSE RE-REGISTRATION

Each production license file requires an annual re-registration due on or before the anniversary
date of your account. A customers anniversary date is based on one of the following conditions,
whichever occurs first:
Generation and download date of the first production license for the account
Date of the first Support call or contact for the account
Six months from the date of the first purchase for the account
Production licenses request that the system administrator re-register the license file with
Interactive Intelligence ninety (90) days before the anniversary date. An account can only be
re-registered if it falls within the re-registration range.
A renewal notice is sent to your reseller ninety (90) days before your anniversary date and
includes a note regarding license registration.
Sixty (60) days before the anniversary date, your System Administrator begins receiving
reminders with Interaction Administrator of the need to register.
Thirty (30) days before the anniversary date, the System Administrator receives a message
regarding re-registration.
Thirty (30) days past the anniversary date, Interaction Client pops an alert at start-up with a
message to register the License of Product.
Sixty (60) days past the anniversary date, Interaction Client pops every hour with a message
regarding the license re-registration.
10-06
10.2
Updating the CIC System

SERVICE UPDATES
Interactive Intelligence releases two types of software updates when needed.
Service Updates (SU) These can be found in the Product Information/Service Updates section of
the Interactive Intelligence Support website.
Service updates include the following:
Readme Contains the list of server installs to which the SU can be applied, history, handler
updates, special notices, and instructions.
Summary History of Software Change Requests (SCRs) for this and previous SUs.
History of enhancements to the IC version since GA release.
There are two ways to apply an SU to an Interaction Center system:
Using Interactive Update (Recommended)
Automatic update mode Interactive Update automatically checks for, downloads, and
applies applicable updates in the latest SU to the IC server, workstations, and other devices
on the network at a specified time.
Manual update mode Interactive Update automatically checks for and downloads the
appropriate Service Update components; and the system administrator determines when to
apply the components to the IC server, workstations, and all other devices on the IC network.
Alternatively, Interactive Update just checks for Service Updates, and the IC administrator
(or users, if the IC administrator allows it) determines when to download and when to apply
the components.
Manual Download In the manual download procedure, you download the Service Update
components from the Interactive Intelligence Support site and apply the appropriate components
to the IC server using the SUInstall.exe utility. You can then choose one of two options:
Allow Interactive Update to download and apply the appropriate components to workstations
and other devices on the IC network.
Manually download and apply the appropriate components to workstations and other devices
on the IC network.
Applying Service Updates Each SU contains instructions on how and when to apply the update.
Read the instructions carefully, and follow the directions exactly.
For switchover environments, consult the IC Automated Switchover System Technical Reference
before applying any updates.
Using the Remote Console When installing updates or providing service that requires you to use
Remote Desktop Protocol (RDP), use the administrator tag when logging on.
CAUTION
10-07
Be sure to check the Support website often for Service Updates. Keeping the system up-to-date
reduces the potential for unexpected server downtime. For sites using custom handlers, note that
the SU will ONLY update base handlers.
OPERATING SYSTEM UPDATES

The Interactive Intelligence support portal provides a search engine allowing you to check the
status of Microsoft Updates. Updates tested by the Testlab that appear to be compatible with our
software are marked as tested. The Testlab makes a best effort pass of validating frequent Microsoft
updates, but cannot guarantee every possible issue or full regression testing of older products. We
rate the risks of loading generally available updates from Microsoft as acceptable. You should fully
test all Microsoft updates.
You can access Testlab by following this link: http://testlab.inin.com. Once you have logged on, you can
access the update search engine by clicking the Microsoft Patches link on the left side of the page.
Review Questions
Labs
Lab 18: Perform Hands-on Troubleshooting
Lab 19: Deployment Scenario
10-08

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IC System Configuration

2015, Interactive Intelligence, Inc.

Copyright and Trademark Information

2015, Interactive Intelligence, Inc.

2015, Interactive Intelligence, Inc.

Chapter 2: Session Initiation Protocol (SIP)

Chapter 4: License Management

Chapter 5: Lines and Regional Dial Plan

Chapter 7: Interaction SIP Proxy 4.0

Chapter 8: Interaction Switchover

Chapter 9: CIC System Support

Chapter 10: CIC System Maintenance

2015, Interactive Intelligence, Inc.

Interaction Center Architecture

2015, Interactive Intelligence, Inc.

Interaction Center Architecture

Interaction Center Architecture

2015, Interactive Intelligence, Inc.

There are two different types of queues in Interaction Center:

ADDITIONAL SERVERS IN AN INTERACTION CENTER ENVIRONMENT

Interaction Center Architecture

2015, Interactive Intelligence, Inc.

Additional servers and the functionality they add are as follows:

Interaction Center Architecture

2015, Interactive Intelligence, Inc.

Analyzes outbound calls to detect busy signals

Interaction Center Architecture

2015, Interactive Intelligence, Inc.

Session Initiation Protocol (SIP)

2015, Interactive Intelligence, Inc.

VoIP is often referred to as IP telephony.

Session Initiation Protocol (SIP)

2015, Interactive Intelligence, Inc.

Voice Played Size

VoIP HEADER OVERHEAD

Session Initiation Protocol (SIP)

2015, Interactive Intelligence, Inc.

AUDIO QUALITY MEASUREMENTSMEAN OPINION SCORING (MOS)

Session Initiation Protocol (SIP)

Mean Opinion Score (MOS)

2015, Interactive Intelligence, Inc.

Initiates a call, changes call parameters (re-INVITE).

Confirms a final response for INVITE.

Cancels searches and ringing.

Queries the capabilities of the other side.

Registers with the Location Service.

Session Initiation Protocol (SIP)

2015, Interactive Intelligence, Inc.

SIP CLIENT REGISTRATION PROCESS

Session Initiation Protocol (SIP)

2015, Interactive Intelligence, Inc.

BASIC SIP TOPOLOGY

Session Initiation Protocol (SIP)

2015, Interactive Intelligence, Inc.

Polycom VVX 500 SIP Telephone:

Interactive SIP Station:

Session Initiation Protocol (SIP)

2015, Interactive Intelligence, Inc.

Session Initiation Protocol (SIP)

2015, Interactive Intelligence, Inc.

Managed IP Phone Configuration

2015, Interactive Intelligence, Inc.

Managed IP Phone Configuration

The Load License loads licenses of .I3Lic or .I3Lic.xml file types.