Talkaphone VOIP-8 User Manual

Rev. 7/2/2014

Copyright 2014 Talk-A-Phone Co. • 7530 North Natchez Avenue • Niles, Illinois 60714 • Phone [phone] • [e-mail] • www.talkaphone.com.

All prices and specifications are subject to change without notice. Talkaphone, Talk-A-Lert, and WEBS are registered trademarks of Talk-A-Phone Co. Windows is a trademark

of Microsoft Corporation. All other trademarks are the property of their respective owners.

Installation and Operation Manual

for

Talkaphone Voice over IP Interface

VOIP-1-2-4-8

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CHAPTER 1

Introduction to Voice over IP Interfaces
(VOIP-1, VOIP-2, VOIP-4, and VOIP-8)

The Voice over IP (VoIP) Interface allows all Talkaphone Emergency Phones to be used over an IP data
network. The VOIPs integrate seamlessly with existing VoIP phone systems, and support standard VoIP
protocols. For sites without existing VoIP systems, two VOIPs can be used in conjunction to send
emergency calls over the IP network and then remotely “jump off” onto an existing PBX or PSTN phone
network.

Figure 1-1: VOIP-1 Chassis

Figure 1-2: VOIP-2 Chassis

Figure 1-3: VOIP-4/VOIP-8 Chassis

Capacity. Talkaphone’s VOIP-8 model is an eight-channel unit, the model VOIP-4 is a four-channel unit,

the model VOIP-2 is a two-channel unit, and the VOIP-1 is a single-channel unit. All of these VoIP units
have a 10/100Mbps Ethernet interface and a command port for configuration.

Mounting. Mechanically, the VOIP-4 and VOIP-8 units are designed for a one-high industry-standard EIA
19-inch rack enclosure. By contrast, the VOIP-1 and the VOIP-2 are not rack mountable.

Phone System Transparency. These VOIP-1-2-4-8’s interoperate with a telephone switch or PBX,
acting as a switching device that directs voice and fax calls over an IP network. The VOIP-1-2-4-8 units
have “phonebooks,” directories that determine to whom calls may be made and the sequences that must
be used to complete calls through the VOIP-1-2-4-8. The phonebooks allow the phone user to interact
with the VOIP system just as they would with an ordinary PBX or telco switch. When the phonebooks are
set, special dialing sequences are minimized or eliminated altogether. Once the call destination is
determined, the phonebook settings determine whether the destination VOIP unit must strip off or add
dialing digits to make the call appear at its destination to be a local call.

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H. 323, SIP, & SPP. Being H.323 compatible, the VOIP-1-2-4-8 units can place calls to telephone
equipment at remote IP network locations that also contain H.323 compatible voice-over-IP gateways. It
will interface with H.323 software and H.323 gatekeeper units. H.323 specifications also bring to VoIP
telephony many special features common to conventional telephony. H.323 features of this kind that have
been implemented into the VOIP-1-2-4-8 units include Call Hold, Call Waiting, Call Identification, Call
Forwarding (from the H.450 standard), and Call Transfer (H.450.2 from H.323 Version 2). The fourth
version of the H.323 standard improves system resource usage (esp. logical port or socket usage) by
handling call signaling more compactly and allowing use of the low-overhead UDP protocol instead of the
error-correcting TCP protocol where possible.

The VOIP-1-2-4-8 is also SIP-compatible. (“SIP” means Session Initiation Protocol.) However, H.450
Supplementary Services features can be used under H.323 only and not under SIP. It can register with
SIP proxy servers and call managers that are 100% SIP-compliant.

SPP (Single-Port Protocol) is a non-standard protocol that offers advantages in certain situations,
especially when firewalls are used and when dynamic IP address assignment is needed. However, when
SPP is used, certain features of SIP and H.323 will not be available and SPP will not interoperate with
VoIP systems using H.323 or SIP.

Data Compression & Quality of Service. The VOIP-1-2-4-8 unit comes equipped with a variety of data
compression capabilities, including G.723, G.729, and G.711 and features DiffServ quality-of-service
(QoS) capabilities.

PSTN Failover Feature. The VOIP-2-4-8 can be programmed to divert calls to the PSTN temporarily in
case the IP network fails. Enabling this feature will require a dedicated channel, therefore a VOIP-1 does
not have the PSTN failover feature.

Management. Configuration and system management can be done locally with the VOIP-1-2-4-8
configuration software via a serial connection. After an IP address has been assigned locally, other
configuration can be done remotely using the Web Interface GUI. All of these control software packages
are included on the VOIP-1-2-4-8 CD.

While the Web GUI’s appearance differs slightly, its content and organization are essentially the same as
that of the Windows GUI (except for logging).

The primary advantage of the Web GUI is remote access for control and configuration. The controller PC
and the VOIP-1-2-4-8 unit itself must both be connected to the same IP network and their IP addresses
must be known.

The Windows GUI gives access to commands via icons and pulldown menus, whereas the Web GUI
does not. The Web GUI, however cannot perform logging in the same direct mode done in the Windows
GUI. However, when the Web GUI is used, logging can be done by e-mail (SMTP).

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Figure 1-4: VOIP Interface Windows GUI (left) and Web Interface GUI (right)

Once you’ve begun using the web browser GUI, you can go back to the Windows GUI at any time.
However, you must log out of the web browser GUI before using the Windows GUI.

Logging of System Events. The software for the VOIP-1-2-4-8 units has SysLog Server functionality.
SysLog is a de facto standard for logging events in network communication systems.

Figure 1-5: Syslog Functionality in VOIP-1-2-4-8 Interface Units

The SysLog Server resides in the VOIP-1-2-4-8 unit itself. To implement this functionality, you will need a
SysLog client program (sometimes referred to as a “daemon”).

Supplementary Telephony Services. The H.450 standard (an addition to H.323) brings to VoIP
telephony more of the premium features found in PSTN and PBX telephony. VOIP-1-2-4-8 units offer five
of these H.450 features: Call Transfer, Call Hold, Call Waiting, Call Name Identification (not the same as
Caller ID), and Call Forwarding. (The first four features are found in the “Supplementary Services”
window; the fifth, Call Forwarding, appears in the
Add/Edit Inbound phonebook screen.) Note that the first three features are closely related. All of these
H.450 features are supported for H.323 operation only; they are not supported for SIP or SPP.

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VOIP-1-2-4-8 Front Panel LEDs

LED Types. The VOIP-1-2-4-8 units have two types of LEDs on their front panels:

(1) general operation LED indicators (for power, booting, and Ethernet functions), and
(2) channel operation LED indicators that describe the data traffic and performance in each VoIP
data channel.

Active LEDs. On both the VOIP-4 and VOIP-8, there are eight sets of channel-operation LEDs.
However, on the VOIP-4, only the lower four sets of channel-operation LEDs are functional. On the VOIP­8, all eight sets are functional.

Figure 1-6. VOIP-4/VOIP-8 LEDs

Similarly, the VOIP-2 has the general-operation indicator LEDs and two sets of channel-operation LEDs,
one for each channel.

Figure 1-7. VOIP-2 LEDs

Finally, the VOIP-1 has the general-operation indicator LEDs and a set of channel-operation LEDs for its
single VoIP channel.

Figure 1-8. VOIP-1 LEDs

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VOIP-1 LED Description

VOIP-1 Front Panel LED Definitions

LED NAME

DESCRIPTION

General Operation LEDs

Power

Indicates presence of power.

Boot

After power up, the Boot LED will be on briefly while the
VOIP-1 is booting. It lights whenever the VOIP-1 is booting or
downloading a setup configuration data set.

Ethernet

FDX. LED indicates whether Ethernet connection is half-

duplex or full-duplex (FDX) and, in half-duplex mode,
indicates occurrence of data collisions. LED is on constantly
for full-duplex mode; LED is off constantly for half-duplex
mode. When operating in half-duplex mode, the LED will flash
during data collisions. LNK. Link/Activity LED. This LED is lit if
Ethernet connection has been made. It is off when the link is
down (i.e., when no Ethernet connection exists). While link is
up, this LED will flash off to indicate data activity.

Channel-Operation LEDs

TX

Transmit. This indicator blinks when voice packets are being

transmitted to the local area network.

RX

Receive. This indicator blinks when voice packets are being

received from the local area network.

XS

Transmit Signal. This indicator lights when the FXS-

configured channel is off-hook or the FXO-configured channel
is receiving a ring from the Telco or PBX.

RS

Receive Signal. This indicator lights when the FXS-

configured channel is ringing or the FXO-configured channel
has taken the line off-hook.

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VOIP-2-4-8 LED Descriptions

VOIP-2/VOIP-4/VOIP-8 Front Panel LED Definitions

LED NAME

DESCRIPTION

General Operation LEDs (one set on each VoIP Interface model)

Power

Indicates presence of power.

Boot

After power up, the Boot LED will be on briefly while the
VOIP-2-4-8 is booting. It lights whenever the VOIP-2-4-8 is
booting or downloading a setup configuration data set.

Ethernet

FDX. LED indicates whether Ethernet connection is half-

duplex or full-duplex (FDX) and, in half-duplex mode,
indicates occurrence of data collisions. LED is on constantly
for full-duplex mode; LED is off constantly for half-duplex
mode. When operating in half-duplex mode, the LED will flash
during data collisions. LNK. Link/Activity LED. This LED is lit if
Ethernet connection has been made. It is off when the link is
down (i.e., when no Ethernet connection exists). While link is
up, this LED will flash off to indicate data activity.

Channel-Operation LEDs (one set for each channel)

XMT

Transmit. This indicator blinks when voice packets are being

transmitted to the local area network.

RCV

Receive. This indicator blinks when voice packets are being
received from the local area network.

XSG

Transmit Signal. This indicator lights when the FXS-

configured channel is off-hook, the FXO-configured channel
is receiving a ring from the Telco, or the M lead is active on
the E&M configured channel. That is, it lights when the VOIP­2-4-8 is receiving a ring from the PBX.

RSG

Receive Signal. This indicator lights when the FXS-

configured channel is ringing, the FXO-configured channel
has taken the line off-hook, or the E lead is active on the
E&M-configured channel.

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Computer Requirements

Minimum Requirements for Windows GUI:

The computer on which the VOIP-1-2-4-8 units’ configuration program is installed must meet these

requirements:

• must be IBM-compatible PC with MS Windows operating system

• must have an available COM port for connection to the VOIP-1-2-4-8 unit

However, this PC does not need to be connected to the VOIP-1-2-4-8 unit permanently. It only needs to
be connected when local configuration and monitoring are done. Nearly all configuration and monitoring
functions can be done remotely via the IP network.

You will need an available COM port on the controller PC. You’ll need to know which COM port is
available for use with the VOIP-1-2-4-8 (COM1, COM2, etc.).

Minimum Requirements for Web GUI

• Local Windows GUI must have been used to assign IP address to VOIP-1-2-4-8.

• Internet Explorer 6.0 or higher; or Netscape 6.0 or higher

• Java Runtime Environment version 1.4.0_01 or higher

Placement

Mount your VOIP-1-2-4-8 in a safe and convenient location where cables for your network and phone
system are accessible.

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Specifications for VOIP-1-2-4-8 Units

Contents: The VOIP-1-2-4-8 includes the following:

• VOIP unit

• 120VAC power supply

• 19” EIA rack-mount brackets (VOIP-4 and VOIP-8 only)

Model

VOIP-1

VOIP-2

VOIP-4

VOIP-8

Operating
Voltage/Current

100-240VAC

1.0 A

External
transformer:
3A @5V

100-240 VAC

1.2 - 0.6 A

100-240 VAC

1.2 - 0.6 A

Main Frequencies

50/60 Hz

50/60 Hz

50/60 Hz

50/60 Hz

Power Consumption

9.7 watts (with
phone off hook)

19 watts

29 watts

46 watts

Mechanical
Dimension

4.3" W x 5.6" D

1.0" H

6.2” W x
9” D x

1.4” H

1.75” H x

17.4” W x

8.5” D

1.75” H x

17.4” W x

8.5” D

10.8 cm W x

14.2 cm D x

2.95 cm H

15.8cm W x

22.9cm D x

3.6cm H

4.5cm H x

44.2 cm W x

21.6 cm D

4.5cm H x

44.2 cm W x

21.6 cm D

Weight

8 oz. (23 g)

1.8lbs (.82kg)

2.6lbs (1.17kg)
with transformer

7.1 lbs. (3.2 kg)

7.7 lbs. (3.5 kg)

Identify Remote VOIP Site to Call

When you’re done installing the VOIP-1-2-4-8, you’ll want to confirm that it is configured and operating
properly. To do so, it’s good to have another VoIP unit that you can call for testing purposes. You’ll want
to confirm end-to-end connectivity. You’ll need IP and telephone information about that remote site. If this
is the very first VoIP unit in the system, you’ll want to coordinate the installation of this VOIP-1-2-4-8 with
an installation of another unit at a remote site.

Identify VOIP Protocol to be Used

Will you use H.323, SIP, or SPP? Each has advantages and disadvantages. Although it is possible to
mix protocols in a single VOIP system, it is highly desirable to use the same VOIP protocol for all VOIP
units in the system.

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Hookup for VOIP-1

Figure 1-9: Sample hookup diagram for VOIP-1 Interface Unit

Hookup for VOIP-2

Figure 1-10: Sample hookup diagram for VOIP-2 Interface Unit

CH1

CH2

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Hookup for VOIP-4 and VOIP-8

Connect the VOIP-4 or VOIP-8 as indicated in the following diagram. Connect the RJ-11 cables from the
emergency phone(s), PSTN line(s), or analog extension(s) of the PBX to the ports labeled “FXS/FXO”.

Make sure to connect the chassis to Earth Ground at the grounding screw as indicated (VOIP-2, VOIP-4,
and VOIP-8 only)

Figure 1-11: Sample hookup diagram for VOIP-4/VOIP-8 Interface Unit

Operation

When the VOIP unit is powered on, it will take approximately one minute to boot up. The red LED
(second from the left) indicates that the unit is still booting. After the red LED clears, allow an extra
twenty seconds to ensure the unit has fully booted before attempting to initiate a call.

The emergency phones will need to be programmed in accordance with the instructions in the Quick

Start section in Chapter 2.

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VoIP System Design

Before you begin programming the VOIP Interface units, it is recommended that you plan out your system
layout. You should begin by choosing a setup type. There are two basic types of VoIP setups we can
design for an emergency phone system:

Figure 1-12: One-to-One Configuration

(1) The first setup type is a one-to-one configuration. In this scenario, each emergency phone has
its own PBX extension or phone line. The number of calls that the head end is capable of receiving
is equal to the number of emergency phones in the field.

Figure 1-13: Many-to-One Configuration

(2) The second setup type is a many-to-one configuration. In this scenario, many Emergency Phones
share PBX extensions or phone lines. The number of calls that the head end is capable of receiving is
less than the number of emergency phones in the field.

Once you have chosen a setup type, it is recommended that you assign phone numbers/PBX extensions
to the emergency phones and IP addresses to the VOIP units before programming any of the VOIP units.
Keep in mind that the PBX extension assignments are separate from the VOIP phone book extensions.
Please reference Phone Book Design (p. 18) for more information.

When designing your system layout, please keep in mind that All VOIP units must have fixed IP
addresses. Also, ensure that the proper routing and switching hardware (routers, hubs, firewalls, VPNs,
etc.) are in place for the VOIP units to communicate. It is critical that ensure network reliability, which
includes sufficient bandwidth and minimizes packet loss and packet delays.

IMPORTANT NOTE: For the Emergency Phone System to work through a power outage, all components of
the data path (i.e. the VOIP units, routers, hubs, switches, etc.) must be on back-up power.

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The following are examples of other types of VoIP setups you can design.

Figure 1-14: Many-to-One IP-Native Head End Configuration

(3) Figure 1-14 is an example of a configuration with a completely IP-native head end. With the IP-native
PBX, there isn’t a need for VOIP units at the head end.

Figure 1-15: One-to-One Closed Configuration

(4) Figure 1-15 is an example of a closed configuration with no PBX or PSTN lines. This configuration
type relies solely on the network infrastructure for call routing.

IMPORTANT NOTE: For the Emergency Phone System to work through a power outage, all components of
the data path (i.e. the VOIP units, routers, hubs, switches, etc.) must be on back-up power.