VOIP-500 and VOIP-600 Installation Guide

VOIP-500 and VOIP-600 Installation Guide

Call Station Installation Guide for Model VOIP-500 and Model VOIP-600



To jump to the INSTALLATION Steps, use the Table of Contents links.

Overview and Intended Audience

This guide provides detailed instructions for the configuration and operation of VOIP-500 and VOIP-600 Series Call Stations. It is recommended to read this instructional manual completely before performing any configuration.

This installation guide is targeted towards systems administrators, or any person who would configure and maintain VOIP-500 and VOIP-600 Series Call Stations. Fundamental knowledge in computer networking and Voice over Internet Protocol (VoIP) technologies is strongly recommended for understanding this guide.

The VOIP-500 series phones are outdoor-rated, ADA-compliant hands-free Voice over IP (VoIP) Emergency/Information Phones for use in locations such as: parking facilities, college campuses, medical centers, and industrial parks. VOIP-500 and VOIP-600 series call stations are SIP
* compliant.

The VOIP-500 and VOIP-600 series of call stations are categorized into various device modes based on the number of buttons, keypad, handset and speaker/microphone present on the phone’s face plate.

These device modes are also given different marketing names. A model with “Keypad” is denoted with a letter “K” in its model number. A model with two buttons is denoted with a letter “D” in its model number. For example, VOIP-500 corresponds to “Single button Emergency VoIP phone”. VOIP-500D is “Dual button Emergency VoIP phone”. VOIP-500K is “Two-button Emergency VoIP Phone with Keypad”.

Some of the functionality described below may not be available depending on the phone device mode. The rest of the document may use the term “phone” to refer to a “VOIP-500 series” call station.

NOTE:  Not all features of the VOIP-500 and VOIP-600 series call stations are available when used in conjunction with analog telephony adapter devices or with the Public Switched Telephone Network (PSTN).
*RFC 3261 and RFC 2833 for DTMF delay

Acronyms and Abbreviations Used In This Guide

The following acronyms and abbreviations are commonly used throughout the guide:
Acronyms
Definitions
ADA
Americans with Disabilities Act
AEC
Acoustic Echo Cancellation
AGC
Automatic Gain Control
ANSI
The American National Standards Institute a private non-profit organization that oversees the development of voluntary consensus standards for products, services, processes, systems, and personnel in the United States.
AUX
Auxiliary Input/Output. An Auxiliary Input accepts a contact closure from an external device, such as a Vehicle Detector, Door Switch, Scream Alert, and card swipe. An auxiliary output produces a contact closure to an external device, such as a strobe light and motorized garage gate.
BABT
British Approvals Board of Telecommunications
CNG/VAD
Comfort Noise Generator/Voice Activity Detector. It is used to reduce the transmission rate during inactive speech periods while maintaining an acceptable level of output quality.
CSA
Canadian Standards Association
CE
The CE marking certifies that a product has met EU consumer safety, health or environmental requirements.
DHCP
Dynamic Host Configuration Protocol — protocol for assigning dynamic IP addresses to devices on a network.
DNS
Domain Name Server
DTMF
Dual Tone Multi Frequency signaling is used for telecommunication signaling over telephone lines.
FCC
Federal Communications Commission
FTP
File Transfer Protocol
GUI
Graphical User Interface
G.711
G.711 is codec also known as Pulse Code Modulation (PCM). It is the ITU-T international standard for encoding telephone audio on a 64 kbps channel.
G.723
G.723 is an ITU-T standard speech codec.
G.729
G.729 is an audio data compression algorithm. It is the ITU-T international standard for encoding telephone audio on 8 kbps channel.
IVR
Interactive Voice Response
IP-PBX
It is an IP based switch for call handling through public and private exchanges.
IE
Internet Explorer
IETF
The Internet Engineering Task Force (IETF) develops and promotes Internet standards.
PCBA
Printed Circuit Board Assembly
PoE
Power over Ethernet, IEEE 802.3af standard.
QoS
Quality of Service is of particular concern for the continuous transmission of high-bandwidth video and multimedia information.
SIP
Session Initiation Protocol is a signaling protocol, widely used for setting up and tearing down multimedia communication sessions over network.
TIA
Telecommunications Industry Association
UL
Underwriters Laboratories
VoIP
Voice over Internet Protocol


Objective

This guide provides a detailed examination of the features included in VOIP-500 and VOIP-600 Series Call Stations. It guides an installer and or an administrator through the configuration and optimization of call station features. While configuration of the VOIP-500 and VOIP-600 Series Call Stations is covered in detail, configuration of other peripheral VoIP network elements is beyond the scope of this document.

Typographic Conventions

The following guidelines are used as typographic conventions in this user manual:
Item
Convention
Sample
Acronyms
All uppercase
SIP
Chapter titles
Title caps
See Chapter 3 Getting Started
Command-line commands and options (switches)
All lowercase, bold
ifconfig command
/a option
Device names
All uppercase
VOIP-500
Directories
All lowercase
/flash
Error message names
Initial caps
Update failed
File names
Title caps (internal caps in short file names are acceptable for readability)
MainLogFile.txt, BackupLogFile.txt
Menu names
Bold; title caps
Insert menu
Programs and applications
Usually title caps
HyperTerminal
Toolbar button names
Usually title caps (follow the interface); bold
Apply
Reset

URLs
All lowercase; break long URLs before a forward slash, if necessary to break; do not hyphenate.
User input
Usually title caps; bold
Enter Password


Prerequisites

Prior to configuring a VOIP-500 and VOIP-600 Series Call Station, ensure the unit is powered on and connected to the network.

The VOIP-500 and VOIP-600 can be configured from a computer with either a TCP/IP network connection or a DB-9 Serial port. The VOIP-500 and VOIP-600 supports straight-through serial connections for basic programming. For access to the full configurable feature set, a modern web browser will be required.

The VOIP-500 currently supports access from:
  • Internet Explorer 8 or greater
  • Firefox 3.5 or greater


Configuration Using the Web GUI

Ensure both the VOIP-500/VOIP-600 and your PC are connected to the Local Area Network. A direct connection to the VOIP-500/VOIP-600 will require the use of a crossover network cable.

VOIP-500 and VOIP-600 Series Call Stations are pre-configured with the following default settings:
IP Address:
192.168.1.10
Username:
admin
Password:
admin@123


NOTE: It is HIGHLY recommended you change the default password to a strong password

Configure the IP address of your PC to be on the same subnet as the VOIP-500/VOIP-600. For example, 192.168.1.3

Open a supported web browser and direct it to the IP address of the VOIP-500/VOIP-600. For example, enter the following URL: http://192.168.1.10/
You will be prompted to enter the username and password. (Example in the following image:)

NOTE: Some versions of Internet Explorer, the login prompt window will be behind the window giving the impression this is not working. This is an issue with Internet Explorer and not the VOIP-500/VOIP-600 device. We recommend using Firefox to prevent this anomaly with Internet Explorer.
Enter the default Username and Password. After authentication is successful, you are redirected to the Home page.
Further configuration of VOIP-500/VOIP-600 settings will be accomplished using this interface.


Configuration Using the Serial Console

Basic settings can be configured on VOIP-500 Series Phones through a serial console connection. Knowledge of Linux shell commands is recommended for serial console configuration.

Most commonly, the serial console connection can be used to set the IP address, as described below.
  1. Connect a straight-through serial cable (DB9 Male to DB9 Female) from the serial port on the phone to an available COM port on the PC, noting which port was utilized.
  2. Open your preferred serial console application (e.g., HyperTerminal or TeraTerm) on the PC and specify the following settings in table below:
  3. Once connected, you will be presented with the following prompt: root:/>
  4. The /flash directory contains persistent files. It is highly recommended to switch to the /flash directory by entering the following command, and pressing Enter.
    1. root:/> cd /flash
    2. (Note the space between ‘cd’ and ‘/flash’)
  5. Create a file where configuration parameters can be entered. For example, we create the config.txt file by entering the following command, and pressing Enter.
    1. root:flash/> vi config.txt
    2. (Note the space between ‘vi’ and ‘config.txt’)
  6. While in the vi application, press i to force vi into “Insert Mode.”
  7. The body of the config.txt file can now be created.
    1. To set a new static IP address enter the following data shown below in the config.txt Settings table
  8. Once the configuration options are entered, save the changes and exit vi by pressing Escape, then entering :wq
    1. To exit without saving changes press Escape, then enter :q!
  9. With the configuration file created, the final step is to apply these settings to the VOIP-500. Enter the following command, and press Enter.
    1. root:flash/> configApp admin admin@123 config.txt
  10. A reboot is required after serial console configuration. Enter the following command, and press Enter.
    1. root:flash/> reboot
Serial Settings:
Communications Port: 
<Determined in Step 1 above>
Baud rate (bits per second):
115200
Data bits:
8
Parity:
None
Stop bits:
1
Flow control:
None

config.txt Settings:
NetworkMode = 3
IPAddress = <IP address, e.g. 192.168.1.10>
Netmask = <netmask, e.g. 255.255.255.0>
Gateway = <default gateway, e.g. 192.168.1.1>


WEB GUI Layout (Overview):

NOTE: Detailed WEB GUI Layouts will be in the appendices below.
This section describes the layout of the Web GUI. The Web GUI layout consists of four (4) general areas. The Navigation, Page Title, Action buttons and Content Area. All of these sections dynamically update based on the action you are performing. The action buttons have additional buttons when you are in a configuration section of the call station. (See reference below) - To enlarge the image, tap or click on it.



Action button APPLY:


When a configuration change is in the process of being applied, various messages will show next to the apply button:
When a config is being updated:

A successful config update:

A failed config update:




INSTALLATION

THIS PRODUCT MUST BE INSTALLED IN ACCORDANCE WITH THE APPLICABLE INSTALLATION CODE BY A PERSON FAMILIAR WITH THE CONSTRUCTION AND OPERATION OF THE PRODUCT AND THE HAZARDS INVOLVED

It is recommended that this instruction set be read completely prior to the start of any installation. Additional requirements (Network ports) can be found in the appendices below.

Ensure the following components of the VOIP-500/VOIP-600 are included:
( TAKE NOTE OF THE DIFFERENT SCREW TYPES FOR THE VOIP-600 )
QTY
Part Number
Description
1
VOIP-X00 (500 or 600)
VOIP-500/600 Series Call Station
6 (VOIP-500)
42935
#10-24 T20 Torx® Pin-Head Stainless Steel Screws
3 (VOIP-600)
42935
#10-24 T20 Torx® Pin-Head Stainless Steel Screws
3 (VOIP-600)
42936
#10-24 T20 Torx® Pin-Head Black Anodized Stainless Steel Screws
1
86392
Ferrite Core

Power Requirements:
The VOIP-500 Series Phone can be powered over Ethernet or through a dedicated, line-regulated power supply that meets the following specifications:
Power Input
Description
Acceptable Voltage
Power Consumption
Ethernet WAN
PoE - IEEE 802.3af Class 3
+36-57 VDC
150 mA
Power Supply Input
12 VDC
+10-14 VDC
500-800 mA
Power Supply Input
24 VDC
+21-27 VDC
300-500 mA
Power Supply Input
24 VAC
+21-27 VAC
300-500 mA

Installation Steps:
  1. Unbox the VOIP-500/VOIP-600 and verify all components are accounted for. (See list above for parts included in box)
  2. Remove the back box from the VOIP-500/VOIP-600 Series Phone assembly by unfastening the four (4), #6-32 nuts and washers.
  3. Install the Ferrite Core as shown in Figure 1 below.
  4. Provide network connectivity to the VOIP-500/VOIP-600 - For PCB (Printed Circuit Board) wiring details, refer to Figure 2 below.
    1. The VOIP-500/VOIP-600 Series Phone is equipped with two (2) Ethernet ports. The WAN Ethernet (PoE) port should be used as the primary port for data communications. The WAN Ethernet port can be used to power the phone via an IEEE 802.3af, Class 3 compliant PoE supply.
    2. The VOIP-500/VOIP-600 Series Phone has a layer 2 network switch port labeled LAN Ethernet which can provide network connectivity to an auxiliary device e.g. (IP Camera, Card Reader, etc). The LAN Ethernet port is a pass-through port and cannot be used to access the phone.
    3. When installing the Network cable, it is essential to have a ferrite core (provided with the phone), installed on to the cable as close as possible to the connector. Make sure the network cable is passed twice through the ferrite core, forming a loop of not less than one (1) inch in diameter.
      1. NOTE: A cable with a rating of Cat5e or higher with an RJ45 connector is typical for all network cables.
  5. Connect any auxiliary Inputs or Outputs - see Figure 5 below.
  6. If applicable, connect a Line Level Audio connector on the PCB as shown in the Figure 2 below, to the respective device on the other end, such as the WEBS® paging amplifier, recording device, speaker, etc.
  7. The faceplate of the phone must be connected to earth ground. Install a #10 ring terminal to the ground wire before connecting it to the earth ground terminal on the rear of the faceplate as shown in Figure 2 below.
  8. Connect power to the VOIP-500/VOIP-600
    1. PoE (Power Over Ethernet) - Connect the Ethernet cable (Power + Data) to the WAN Ethernet port on the PCB as shown in Figure 2 below.
    2. Local Power
      1. For 12 VDC power mode place the internal jumpers on the two outermost terminals as shown in Figure 3 below. Once the jumpers are correctly set, connect the power supply to the Power Supply Input using a two conductor, No. 24 to 12 AWG cable as shown in  Figure 4 below. For 12 VDC, ensure the Negative supply is connected to terminal A, and the Positive supply is connected to terminal B.
      2. For 24 VDC/VAC power modes, place the internal jumper on the two innermost terminals as shown in Figure 3 belowOnce the jumpers are correctly set, connect the power supply to the Power Supply Input using a two conductor, No. 24 to 12 AWG cable as shown in Figure 4 below. For 24 VDC/VAC, ensure the Positive supply is connected to terminal A, and the Negative supply is connected to terminal B.
  9. After completing the field wiring installation, fasten the back box on to the faceplate with the six (6) hardware nuts and washers.
    1. NOTE: It is the installer’s obligation to ensure that the wiring should pass through the cable entry hole at the bottom of the back box. Please exercise caution when reinstalling the back box and ensure that no cables are crushed during the process.
  10. Replace the backbox onto the VOIP-500/VOIP-600 and Install the VOIP-500/VOIP-600 Series Call Station into its appropriate mount (e.g. ETP-SM, as shown in Figure 6 below), with the screws provided.

Figure 1: Installation of the ferrite core on the network cable (Tap or Click the image for enlarged view)


Figure 2: Internal PCB (Printed Circuit Board) view (Tap or Click the image for enlarged view)


Figure 3: Internal Jumper position for external power modes. Phone default setting is 24 VDC/VAC. (Tap or Click the image for enlarged view)


Figure 4: Power Supply Input connection polarity. (Tap or Click the image for enlarged view)


Figure 5: Auxiliary Inputs / Outputs - Use a two conductor, No. 28 to 16 AWG cable size for all auxiliary connections.
The VOIP-500 Series Phone has three (3) Auxiliary Inputs (dry contact, 10 mA @ 8 VDC) and three (3) Auxiliary Outputs (dry contact, 120 mA @ 60 VAC/DC).

Auxiliary Outputs allow peripheral equipment such as strobe lights, PTZ cameras, door entry systems, etc. to be activated when the push button is pressed.
Two (2) removable 6-pin connector plugs are provided for the auxiliary input and output as shown.
The Auxiliary Input and Auxiliary Output connections are as follows:

Auxiliary INPUT Connections
Connector Plug Pin Position
Aux. Input 1
Position 7 and 8
Aux. Input 2
Position 9 and 10
Aux. Input 3
Position 11 and 12
Auxiliary OUTPUT Connections
Connector Plug Pin Position
Aux. Output 1
Position 1 and 2
Aux. Output 2
Position 3 and 4
Aux. Output 3
Position 5 and 6

(Tap or Click the image for enlarged view)


Figure 6: Installation of VOIP-500/VOIP-600 Call Station. (Tap or Click the image for enlarged view)



Operating the VOIP-500 / VOIP-600

REFERENCE ITEMS ARE FOUND IN THE APPENDIX

The phone takes approximately one minute to boot. While the first and second LED’s are flash, the unit is still booting (Refer to Figure 1 Front View). After all LED’s clear, wait for an additional twenty seconds before initiating a call.
The phone needs to be programmed in accordance with the instructions provided in Chapter 4 (Refer to 4 Using the Web GUI).

Calling Emergency/Information Numbers
In case of an emergency, the phone can dial a list of numbers (Refer to Number Lists) in round robin fashion. Depending on the device mode and programming, a call is automatically initiated when:
  • Push button is pressed
  • Auxiliary Input is activated
  • Handset goes off-hook (0BH device mode only)
If the first number does not answer within a specified time (Refer to Call Parameters) the phone dials the next number from the number list. It continues to dial in round robin fashion until the call is answered, or the call conversation timer expires (Refer to Call Parameters: Call Conversation Timer).

Answering the Emergency/Information Call
One of the following scenarios occurs upon answering the emergency call on the remote side:
  • The VOIP-500 Series Phone may transmit its location by playing a recorded message (Refer to Played to Remote Side). If no such feature is enabled or when the voice message completes playing, the caller may talk and listen hands-free  without using any controls.
  • The VOIP-500 Series Phone may transmit a voice message and request that the remote side user to press a specific key on the keypad of the remote side phone (Refer to Call Parameters: answer a call) to answer this call. Providing the incorrect key sequence within three attempts will terminate the call. Once the call is terminated, a new call is made to the next number on the list. On providing the correct key sequence, the VOIP-500 Series Phone may transmit its location by playing a recorded message (Refer to Played to Remote Side). If no such feature is enabled, the caller may talk and listen hands-free  without using any controls.
Terminating Calls
Round robin calls can be terminated only from the remote side by going on-hook or by pressing a specific call termination key on the remote keypad (Refer to Call Parameters). Calls placed from the keypad can be terminated locally using button(s) programmed as a hook switch (Refer to Buttons) or from the remote side by going on-hook/pressing a specific call termination key.

For phones with just a handset and no speaker, calls (both round robin and keypad calls) can be terminated locally by going on-hook.

Activating/Deactivating Auxiliary Outputs
To provide additional visible or audible alerts upon an event, the phone is equipped with 3 Auxiliary Outputs. The Auxiliary Outputs can be used to connect devices such as a siren, a strobe light, a PTZ camera, etc. These devices can then be activated and deactivated locally or remotely (Refer to Auxiliary Outputs).

Activating/Deactivating the “Help on the Way” LED
The phone is equipped with 3 LEDs; each of the LEDs has a specific function assigned to it. The third LED is used for indicating “Help on the Way” and is activated/deactivated through DTMF operation codes on the remote side (Refer to LEDs in the appendices below).

Activating/Deactivating Speaker and Microphone
During a call, the speaker (both Hands-free and Handset) and microphone (both Hands-free and Handset) can be enabled/disabled using DTMF operation codes *5*/*6* & *7*/*8* respectively.

Note: The settings made above are temporary, which are valid for current call only. Once the call is terminated, the settings will reset to the values defined in the Web GUI (Refer to Audio Settings: Enable or disable the Speaker and/or Microphone).

Adjusting Volume
During a call, the speaker gain and the microphone gain can be adjusted remotely using DTMF operation codes *32*level* and *31*level* (where level ranges from 1-20), respectively.

During a call, the User can also locally adjust the speaker volume using Button #3 and Button #4 (Refer to Buttons). This option is only available for phones with four buttons.

Note: The volume adjustments made above are temporary and are valid for a single active call instance. Once the call is terminated, the settings will reset to the values defined in the Web GUI (Refer to Audio Settings: Speaker Gain and the Microphone Gain).

Playing Voice Messages
This feature is used to play pre-recorded voice messages on the phone’s built-in speaker/handset or to the remote side upon occurrence of certain events (e.g., Call Initiation events, local/remote key press events). For configuration details, refer to Section 4.7.1 Played to the User and Section Played to Remote Side.

Recording Voice Messages
The VOIP-500 Series Phone has the capability of recording up to six different voice messages (each up to one minute in length).
To record a voice message on the phone from the remote side:
  1. Call the VOIP-500 Series Phone.
  2. Once the call is answered, enter the DTMF authentication code. The default DTMF authentication code is *4**.
  3. To initiate voice message recording, press *91*<message num>* where message num is from 0-5. The voice message programmed for message num 0 is used for the “Key Press to Answer” feature.
  4. After the single beep, begin recording the voice message.
  5. Press any key to stop the voice recording.
Note The maximum voice message length is one minute. If no key is pressed within one minute or the call is disconnected, the voice message recording stops automatically.

Previewing Voice Messages
To preview your recorded voice messages from the remote side:
  1. Call the VOIP-500 Series Phone.
  2. Once the call is answered, enter the DTMF authentication code. The default DTMF authentication code is *4**.
  3. To initiate voice message previewing, press *92*<message num>* where message num is from 0-5. The voice message programmed for message num 0 is used for the “Key Press to Answer” feature.
  4. After the single beep, the voice message will begin playing.
  5. Press any key to stop the preview.

Feedback Tones
Feedback tones/beeps are used to indicate the status of the requested operation from the local keypad or from the remote side.
  1. One beep is played after successful translation of local/remote key codes. For example, to activate Auxiliary Output #1 the local key code is 123. When the User presses 123, the phone plays a single beep on the built-in speaker or handset before activating Auxiliary Output #1. Similarly, when the remote side enters the correct code the remote user hears a single beep before the phone performs the requested operation.
  2. Two beeps are played indicating an incorrect key code. For example, to activate an Auxiliary Output #1 the local key code is 123. If the User presses 124 the phone plays two beeps. Similarly, when the remote side enters the incorrect code the remote user hears a double beep indicating an error.
  3. Three beeps are played to the remote side in a case where DTMF authentication is required for a particular operation. For example, key code 123 requires DTMF authentication to activate Auxiliary Output #1. After pressing 123 the phone plays three beeps to the remote side indicating DTMF authentication is required in order to perform the requested operation.

Paging Operation
The phone can receive one-way paging messages from the WEBS Contact® server either on the line level output or on the built-in hands-free speaker. The phone must be registered with the WEBS Contact® server in order to receive incoming pages.

The system administrator can prioritize (Refer to Phone Settings: assign the priority) the paging operation over the phone operations to facilitate high priority paging messages.

Line Level Recording
This feature is used to locally record communication between the calling party and the called party once the call is established. This feature is enabled only when the paging output mode is set to the built-in hands-free speaker. Configuring the paging output to line level output automatically disables this feature.

Silent Monitoring
Silent Monitoring is used to monitor locations as required. This feature is only supported by phones with a built-in hands-free speaker and microphone. To use this feature:
  1. Disable the speaker (Refer to Audio Settings: Enable or disable the Speaker and/or Microphone) and enable the auto answer mode of operation.
  2. Place an incoming call to the VOIP-500 Series Phone. The phone will answer the call silently. The remote phone can be used to monitor the nearby situation without any audible or visual indication on the phone.
Please be certain to follow all federal, state, and local laws when using this feature.

Rebooting the Phone
The phone can be rebooted using the Reset button (Refer to Internal View: RESET BUTTON) on the phone. Pressing the Reset button once will reboot the phone. The phone can also be rebooted via the Web GUI (Refer to Reboot).

Factory Default Settings
The phone can be reset to factory settings using the Reset button (Refer to Internal View: RESET BUTTON). Press and hold the Reset button for 10 seconds. The phone will reboot with factory default settings.

Note: All data including IP address and username/password will be reset to factory defaults.



Basic Troubleshooting


Problem
Possible Causes & Corrective Measures
Unable to access the phone's WEB GUI
- The IP address being used in the Web browser is incorrect. Connect the serial console and check the phone’s IP address.
- IP address of the phone is conflicting with another network device. Reassign a new IP address to the phone.
- The phone’s Web server is not responding. Reboot the phone.
Unable to login to the phone’s Web GUI
- The VOIP-500/VOIP-600's database is possibly corrupt. Reset the VOIP-500/VOIP-600 to factory default settings using the Reset button.
- The IP address of the phone is conflicting with another network device. Reassign a new IP address to the phone.
“Call Placed” and “Call Received” LED’s are continuously blinking.
- The phone’s Ethernet link is down. Check the Ethernet connection.
- The phone is unable to register to a SIP Registrar. Check SIP settings.
- A firmware upgrade is in progress. Wait until the process is complete.
Unable to make outgoing calls
- The Phone is not on the network. Check network settings.
- The destination is not reachable. Check SIP settings.
Unable to receive incoming calls
- The Phone is not on the network. Check network settings.
- The Phone is off-hook. Check the hook switch status.
Not able to recognize the DTMF codes from remote side
- The phone’s DTMF settings are not compatible with remote side. Check the DTMF settings of the remote side. It should be RFC 2833 compliant.
- The DTMF digit duration of the phone is not compatible with the remote phone. Check the DTMF settings at the remote side. It should be RFC 2833 compliant.


Frequently Asked Questions (FAQ's)


Q: I am experiencing audio problems on the phone such as echo, distorted sound, or choppiness. How do I fix this?
A: Verify whether proper bandwidth is allocated for VoIP traffic. You can utilize various mechanisms (RTP header compression and QoS) on the network. OR Check the network priority configuration on the Buttons configuration page (Refer to Buttons: Network Priority).

Q: How do I determine the IP address of the phone?
A: To determine the IP address:
  1. Connect a computer to the phone in question using a serial port and serial console.
  2. Open HyperTerminal or similar console application on the connected computer.
  3. At the command prompt, enter the ifconfig command. The IP address of the phone will be displayed.
  4. Using the WEB GUI, The IP Address field will display the IP Address of the phone. (SEE Network Setting in Appendix)
Q: The phone is not receiving paging requests from the WEBS Contact® server. How do I fix this?
A: Check the following: (Refer to Paging Settings for more on this)
  • Check the Registration status with the WEBS Contact® server.
  • If the status is Unregistered then you will need to contact the Administrator of the WEBS Contact® server in order to re-register the phone for paging requests.
  • If the status is registered at <WEBS Contact® server> then refer to Section Phone Settings.
  • Check whether Phone Mode has priority over Paging Mode or Paging Mode has priority over Phone Mode.
  • If Phone Mode has priority over Paging Mode, then check for any phone activity – Is the phone dialing? Is the handset off-hook? In order to receive pages, the phone should be idle.



APPENDIX

Network Requirements

The network for VOIP-500 Series phones should allow the following:
  • IPv4 enabled
  • Allow the following protocols:
    • SIP
    • RTP
    • HTTP/HTTPS
    • SMTP
  • Routed network
  • DHCP Server (Optional)
Network elements:
  • SIP Proxy/Registrar
  • SIP end points
  • Layer-2 Switch
  • WEBS Contact Server

Ports:
The following ports needs to be enabled on the VOIP-500 Series phones and allowed across the firewall and routers in the network.
Service Module
Port
Transport Type
Mode
HTTP
80
TCP/UDP
Inbound
HTTPS
443
TCP/UDP
Inbound
FTP
21
TCP
Inbound
SSH
22
TCP
Inbound
Paging
5000
UDP
Inbound
SIP Signaling
5060
UDP
Inbound/Outbound
SIP Secure
5061
UDP
Inbound/Outbound
SIP Start RTP
8000-65535
UDP
Inbound/Outbound
STUN
3478
UDP
Inbound/Outbound
WEBS Contact Port
3001
UDP
Inbound/Outbound
SMTP Server Port
25
TCP/UDP
Inbound/Outbound


Remote Operation Key Codes – Quick Reference

During a call, the following operations can be performed on the phone by dialing a key code sequence from the remote side phone which supports the sending of DTMF . If configured, these operations may require DTMF authentication codes (Refer to Authentication: DTMF Authentication Code).

For example, to adjust the volume of speaker from a remote phone, follow these steps:
  1. Authenticate using key sequence *4*<authentication code>*. Successful authentication is indicated by a single beep. This step is required only when DTMF authentication is enabled for volume adjustments (Refer to 4.8 Authentication: enable/disable the DTMF authentication)
  2. Dial the key sequence *31*8*. This sets the volume level of the speaker to 8. Successful operation is indicated by a single beep.
Note The DTMF Authentication Code is only required once during a call and is valid until the call is terminated. Some of the codes below are default codes and can be changed using the WEB GUI.

Remote Operation Key Codes Table:
Operation
Key Code
Function
DTMF Authentication Code Required? (If Configured)
Enter Configuration Mode
*4*<AUTHENTICATION_CODE>*
Code is used to control access to the device.
N/A
Microphone Gain Adjustment
*31*<LEVEL>*

(Valid LEVEL range: 1-20)
Code is used to control microphone gain levels. This applies to both handset and hands-free models. The code takes effect for the current call only.
Yes
Speaker Gain Adjustment
*32*<LEVEL>*

(Valid LEVEL range: 1-20)
Code is used to control speaker gain levels. This applies to both handset and hands-free models. The code takes effect for the current call only.
Yes
Speaker Enable
*5*
Code is used to enable/disable the speaker. This applies to both handset and hands-free models. The code takes effect for the current call only.
Yes
Speaker Disable
*6*
Code is used to enable/disable the speaker. This applies to both handset and hands-free models. The code takes effect for the current call only.
Yes
Microphone Enable
*7*
Code is used to enable/disable the microphone. This applies to both handset and hands-free models. The code takes effect for the current call only.
Yes
Microphone Disable
*8*
Code is used to enable/disable the microphone. This applies to both handset and hands-free models. The code takes effect for the current call only.
Yes
AUX Output #1 Activation
*11*
Code is used for activation/deactivation of the Auxiliary Output by the remote side phone during the conversation.
Yes
AUX Output #2 Activation
*12*
Code is used for activation/deactivation of the Auxiliary Output by the remote side phone during the conversation.
Yes
AUX Output #3 Activation
*13*
Code is used for activation/deactivation of the Auxiliary Output by the remote side phone during the conversation.
Yes
AUX Output #1 Deactivation
*21*
Code is used for activation/deactivation of the Auxiliary Output by the remote side phone during the conversation.
Yes
AUX Output #2 Deactivation
*22*
Code is used for activation/deactivation of the Auxiliary Output by the remote side phone during the conversation.
Yes
AUX Output #3 Deactivation
*23*
Code is used for activation/deactivation of the Auxiliary Output by the remote side phone during the conversation.
Yes
Voice Message Recording
*91*<Message Number>*

Message Number : 0-5

Note: Voice message programmed for Message Number 0 is used for the “Key Press to Answer” feature.
Code is used for recording the message after the beep. Pressing a key stops the recording.
Yes
Voice Message Previewing
*91*<Message Number>*

Message Number : 0-5

Note: Voice message programmed for Message Number 0 is used for the “Key Press to Answer” feature.
Code is used for previewing a recorded voice message. Pressing a key stops the previewing.
Yes
“Help on the Way” LED - TURN ON
*98*
Code is used to enable/disable the “Help on the Way” LED.
Yes
"Help on the Way" LED - TURN OFF
*97*
Code is used to enable/disable the “Help on the Way” LED.
Yes
Call Termination (Disconnect the Call)
#
Single digit key code to be used to disconnect the call.
No


VOIP-500 and VOIP-600 Device Modes

This section provides detail about the various device modes of the VOIP-500 and VOIP-600 Series Call Stations. The device mode is specified under the Home page of the Web GUI (Refer Web GUI Layout above).
NOTE: Where X in the model number in the table below represents for the 5 for the VOIP-500 and the 6 for the VOIP-600.
IF there is a specific model that does not cover both models it will be specified.
For example, a VOIP-X00H can be either the VOIP-500H or VOIP-600H.

Model Number
Device Mode
Number of Buttons
Handset
Hands-free Speaker/Microphone
Keypad
VOIP-X00HNS
0BH
0
Yes
No
No
VOIP-X00KHNS
0BKH
0
Yes
No
Yes
VOIP-X00H
1BH
1
Yes
No
No
VOIP-X00
1BS
1
No
Yes
No
VOIP-X00H2
1BSH
1
Yes
Yes
No
VOIP-X00KS
1BSK
1
No
Yes
Yes
VOIP-X00KSH2
1BSKH
1
Yes
Yes
Yes
VOIP-X00KSH
1BKH
1
Yes
No
Yes
VOIP-X00DH
2BH
2
Yes
No
No
VOIP-X00D
2BS
2
No
Yes
No
VOIP-X00DH2
2BSH
2
Yes
Yes
No
VOIP-X00K
2BSK
2
No
Yes
Yes
VOIP-X00KH2
SBSKH
2
Yes
Yes
Yes
VOIP-X00KH
2BKH
2
Yes
No
Yes
VOIP-X00-OEM
4BSKH
4
Yes
Yes
Yes
WEBS-CM-2
0BS
0
No
Yes (Speaker Only)
No



DETAILED WEB GUI Screen Layouts

TAP or CLICK on the images to see a larger version

WEB Application Menu (Left Navigation)

This section provides a preview of the phone’s Web Menu hierarchy/organization.


HOME

The Home page displays phone information and contact information for Talkaphone.


Field Name
Description
Device Mode
This indicates the type of device. For example, 2BHSK indicates a device having two buttons, a handset, a keypad, and a speaker. Refer to VOIP-500 and VOIP-600 Device Modes for details on other device modes.
Firmware Version
Indicates the version of the call station's firmware.
Bootloader Version

MAC Address





If you need additional assistance, please call 773-539-1100 and select Technical Support option. You can also email us at support@talkaphone.com.