Concorde C-1020 User Manual

1

2

ATA Configuration

1. Features ......................................................................................................................................................................5

1.1 VOIP SIP Protocols (RFC3261)/Interfaces......................................................................................................5

1.2 LAN Protocols /Interfaces................................................................................................................................ 6

1.3 Soft Switch Interoperability.............................................................................................................................6

2. A TA Overview............................................................................................................................................................7

2.1 Ports and Buttons ............................................................................................................................................. 7

2.2 LED Description .............................................................................................................................................. 8

3. Installing ATA.............................................................................................................................................................9

3.1 Configure the Obtain and IP Address automatically for LAN Card ..............................................................10

3.2 Easy Setup......................................................................................................................................................16

3.2.1 DHCP Client Mode.............................................................................................................................17

3.2.2 PPPoE Client Mode.............................................................................................................................19

3.2.3 Static IP Settings Mode.......................................................................................................................21

3.3 Basic VoIP Configuration............................................................................................................................... 23

3.3.1 Access to the web configuration of ATA.............................................................................................23

3.3.2 VoIP Configuration.............................................................................................................................. 25

3.3.3 WAN Configuration ............................................................................................................................33

3.3.3.1 Static IP Configuration............................................................................................................. 33

3.3.3.2 DHCP Client Mode Configuration........................................................................................... 39

3.3.3.3 PPPoE Client Mode Configuration..........................................................................................44

4. Advanced VoIP Configuration .............................................................................................................................49

4.1 ATA Configuration Page ...................................................................................................................... 51

4.1.1 Country Specific Ring & Tones Configuration...................................................................................55

4.1.2 General Configuration Page................................................................................................................58

4.1.3 SIP Service Provider Configuration Page ....................................................................................62

4.1.4 User Login Account Configuration Page.....................................................................................65

4.1.5 Timer Parameters Configuration Page......................................................................................... 67

4.1.6 Call Feature Configuration Page..................................................................................................69

4.1.7 Address Book Configuration Page...............................................................................................71

4.1.8 Advanced Telephony Settings Page.............................................................................................73

4.1.9 Advanced System Telephony Parameters Page............................................................................77

4.1.10 FXO Configuration Parameters Page...........................................................................................79

4.2 Building Dial Plan String..................................................................................................................... 81

4.2.1 Syntactic Format for Dial Plan Strings.........................................................................................81

4.2.2 Dial Plan String and Serial Codes................................................................................................82

4.2.3 Framing Dial Plan Rules.....................................................................................................................83

4.2.4 Default Dial Plan String...................................................................................................................... 84

4.2.4.1 Default Dial Plan Rule 0>#t4...................................................................................................85

3

4.2.4.2 Default Dial Plan Rule Nx.5t8xt2>#........................................................................................85

4.2.4.3 Default Dial Plan Rule 1Nx.2Nx.5tfxt2>#............................................................................... 86

4.2.4.4 Default Dial Plan Rule 011x>#x.et8xt2................................................................................... 86

4.2.4.5 Default Dial Plan Rule 1:*72;>#x.etfxt2 ................................................................................. 87

4.2.4.6 Default Dial Plan Rule 3:*74;>#x.etfxt2 ................................................................................. 88

4.2.4.7 Default Dial Plan Rule 4:*75;>#x.etfxt2 ................................................................................. 89

4.2.4.8 Default Dial Plan Rule 2:*73;>#t4........................................................................................... 89

4.2.4.9 Default Dial Plan Rule 11:*70;>#t4......................................................................................... 90

4.2.4.10 Default Dial Plan Rule 12:*69;>#t4....................................................................................... 90

4.2.4.11 Default Dial Plan Rule 16:*90;x>#x.dtfxt2 ...........................................................................91

4.2.4.12 Default Dial Plan Rule 18:*47;x>#[0-9*].f[0-9*].ft8[0-9*].ft4.............................................91

4.2.4.13 Default Dial Plan Rule 19:*78;x>#t4..................................................................................... 92

4.2.4.14 Default Dial Plan Rule 20:#;x.3tf>#x.atfxt2.......................................................................... 92

4.2.4.15 Default Dial Plan Rule 22:*83;>#t4....................................................................................... 93

4.2.4.16 Default Dial Plan Rule 23:*76;>#t4....................................................................................... 93

4.2.4.17 Default Dial Plan Rule 24:*77;>#t4....................................................................................... 94

4.2.4.18 Default Dial Plan Rule 25:N1t41;>#......................................................................................95

4.2.4.19 Default Dial Plan Rule 26:*67;>#t4....................................................................................... 96

4.2.4.20 Default Dial Plan Rule [0-9*]>#[0-9*].e[0-9*].ft4................................................................96

4.2.5 Empty Dial Plan String .......................................................................................................................97

4.2.6 Incorrect Dial Plan Rule......................................................................................................................97

5. Using Conexant ATA................................................................................................................................................ 98

5.1 Setting up ATA for VoIP Calls........................................................................................................................98

5.2 Making Basic Calls ........................................................................................................................................98

5.3 Advanced Call Features..................................................................................................................................99

5.3.1 Caller ID..............................................................................................................................................99

5.3.2 Call-Waiting Caller ID ......................................................................................................................100

5.3.3 Consultation Hold ............................................................................................................................. 100

5.3.4 Blind Transfer....................................................................................................................................100

5.3.5 Attended Transfer.............................................................................................................................. 101

5.3.6 3-Way Conferencing..........................................................................................................................101

5.3.7 Call Waiting in USA..........................................................................................................................102

5.3.8 Call Forwarding in USA ................................................................................................................... 103

5.3.8.1 Call Forwarding Unconditionally...........................................................................................103

5.3.8.2 Call Forwarding On Busy ......................................................................................................103

5.3.8.3 Call Forwarding On No Answer.............................................................................................104

5.3.8.4 Canceling Call Forwarding ....................................................................................................104

5.3.9 Call Return in USA...........................................................................................................................105

5.3.10 IP Dialing ........................................................................................................................................105

5.3.11 Speed Dialing for Address Book Entries......................................................................................... 106

5.3.12 Caller ID Transmission Block......................................................................................................... 106

5.4 PSTN Backup (Failsafe Relay Mode)..........................................................................................................107

4

5.5 FXO Support................................................................................................................................................107

5.5.1 Incoming PSTN Call:........................................................................................................................108

5.5.2 Initiate PSTN Call:............................................................................................................................108

5.5.3 Call Waiting - Receiving a Second Call:...........................................................................................108

6. Admin Privilege...................................................................................................................................................... 109

6.1 Miscellaneous Configuration .............................................................................................................109

6.2 System Log......................................................................................................................................... 111

6.3 Admin Level Username / Password Configuration............................................................................112

6.4 Local Code Image Update..................................................................................................................113

Appendix A Glossary ................................................................................................................................................. 114

A.1 Acronyms..................................................................................................................................................... 114

A.2 Keyword and Definitions ............................................................................................................................ 114

Appendix B Dial Plan for Pulver Service Provider.................................................................................................... 115

B.1 Basic Dial Plan ............................................................................................................................................ 115

B.1.1 Pulver - USA Dial Plan.....................................................................................................................115

B.1.2 Explanation of the Rules................................................................................................................... 116

B.2 Calling Other Service Provider Numbers through Pulver........................................................................... 117

Appendix C Configuring *.ini Files...........................................................................................................................118

C.1 vophwcfg.ini - Configuring PSTN Backup Support ................................................................................... 118

C.1.1 Configuration for CX82310 Based System...................................................................................... 118

C.1.2 Configuration for 1-Line Reference Hardware ................................................................................ 119

C.1.3 Configuration (2FXS-1FXO) Based Reference Hardware...............................................................120

Appendix D Ring Tone Descriptions .........................................................................................................................121

For Hungarian user manual please visit our website:

www.concorde.hu/drvpdf/02-05-405.pdf

5

1. Features

1.1 VOIP SIP Protocols (RFC3261)/Interfaces

· Single voice port that supports legacy (analog) touch-tone telephones

· Connects legacy telephones to IP-based networks

· Advanced pre-processing to optimize full-duplex voice compression

· High performance line-echo cancellation eliminates noise and echo

· Dynamic network monitoring to reduce jitter artifacts such as packet loss

· Clear, natural-sounding voice quality VoIP Specifications

· Radvision Sip stack version 3.1.1.30

· Digest Authentication MD5

· Record-Route Headers
Voice codecs

· G.729A

· G.726

· G.711 A-Law

· G.711 U-Law
Voice QoS
. Diff serv
. VLAN tags
STUN (Simple Traversal UDP Through NAT)
. RFC3489
Dual-tone multi-frequency (DTMF)

· DTMF tone detection and generation

· Mid-Call DTMF for IVR support

· RFC 2833 AVT tones

· Configurable for two sets of frequencies and single set of on/off cadence (US, UK)
Line-echo cancellation

· G.165, G.168 compliant

· Echo canceller for each port

· 8 to128 msec configurable echo length

· Nonlinear echo suppression

· Convergence time = 250 ms

· ERLE = 10 to 20 dB

· 1 PSTN port interface (FXO) (Optional)

· 1 VOIP port interface (FXS)

· 2 VOIP port interface (FXS) (Optional)

6

Modem & Fax modes

· G.711 fax/modem pass-through with fax/modem detection

· T.38 support

1.2 LAN Protocols /Interfaces

. Ethernet interface
. Support Ethernet Interface (10/100Mbps) which compliant with IEEE 802.3x
standards.
. 10/100 Mbps auto selection
. Ethernet ports support Automatic MDI/MDI-X configuration transceiver.

1.3 Soft Switch Interoperability

. Cisco Proxy/Gateway
. Nortel Proxy/Gateway
. Nokia Proxy/UA
. Avaya Proxy/UA
. Mitel UA
. GL Communications UA/Proxy Simulator
. Marconi Proxy/UA
. Naveltel Proxy/UA
. Texas Instruments UA
. Metaswitch Proxy/Gateway
. BATM UA
. Alcatel B2BUA
. CCL/ITRI Proxy/UA
. 3COM Proxy/SIP IP Phone
. BroadSoft

7

2. ATA Overview

ATA has many ports, switches and LEDs. ATA may have some or all of the features listed
below

2.1 Ports and Buttons

1WAN + 4 LAN + 2 FXS+1 FXO

POWER: Connect the power adapter that came with the ATA. Using a power supply with a
different voltage rating will damage this product. Make sure to observe the proper power
requirements. The power requirement is 12 volts.

WAN port: Connect to Broadband devices , such as a ADSL or Cable modem.
ETH port: Connect to Ethernet network devices, such as a PC, hub, switch, or router.

Depending on the connection, you may need a cross over cable or a strait through cable.
RESET: The RESET button will set the ATA to its factory default setting and reset the ATA.
You may need to place the ATA into its factory defaults if the configuration is changed, you
loose the ability to enter the ATA via the web interface, or following a software upgrade, and
you loose the ability to enter the ATA. To reset the ATA, simply press the reset button for
more than 10 seconds. The ATA will be reset to its factory defaults and after about 30
seconds the ATA will become operational again.

PSTN Jack: Connect a telephone cable between the ATA line jack and a wall jack.
PHONE 1/PHONE 2 Jack: Connect a standard telephone handset to the ATA phone jack

using a telephone cable.

8

2.2 LED Description

Power LED: The LED stays lighted to indicate the system is power on properly.
SIP1/SIP2 LED: This LED is lighted when the ATA is REGISTERED successfully to the SIP

Server.
WAN LED: The LED is lighted when a connection is established to WAN port and flashes
when WAN port is sending/receiving data.
LAN LED: The LED is lighted when a connection is established to LAN port and flashes
when LAN port is sending/receiving data.

9

3. Installing ATA

1. Locate an optimum location for the ATA.

2. For connections to the Ethernet interfaces, refer to figure below.

3. Connect the AC Power Adapter. Depending upon the type of network, you may want to
put the power supply on an uninterruptible supply. Only use the power adapter supplied
with the ATA. A different adapter may damage the product.
Now that the hardware installation is complete, proceed to reset Chapters to set up ATA

10

3.1 Configure the Obtain and IP Address

automatically for LAN Card

Step 1:
Click " Start -> Control Panel "

11

Step 2:
Double click " Network Connections "

12

Step 3:
Right click " Local Area Connection " and then click " Properties "

13

Step 4:
Click " Internet Protocol [TCP/IP] " and then click " Properties "

14

Step 5:
Select " Obtain and IP Address automatically " and then click " OK "

15

Step 6:
Click " Close "

16

3.2 Easy Setup

For easy advanced configuration, insert the CD into your CD-ROM drive.
The CD should auto-start and then click “Easy Setup”. If it does not start, click on Start ->
Run and type in CD:\Easysetup\vbpES.exe (where CD is the drive letter of your CD-ROM
drive.)

There are Three options of Protocol Modes: DHCP Client, PPPoE Client and Static IP
Setting Mode.

17

3.2.1 DHCP Client Mode

1. After selecting the Protocol mode: DHCP Client Mode

2. Enter Registrar Address and Port / Proxy Address and Port / Outbound Proxy
Address and Port / Auth User ID / Password in VoIP Line 0 and VoIP Line 1 which was
given by Telecom or by your Internet Service Provider (ISP).

3. Click Setup, it will start to configure the ATA for a while. Follow the instructions of the
Easy Setup utility which will guide you to complete the configuration.

4. Easy setup configuration completed.

5. Please connect the Ethernet cable between the ATA and Broadband Device.

6. Please check the SIP LED is lighted or not. If the SIP LED is lighted, the ATA is

18

REGISTERED successfully to the SIP Server. If not, please press reset button and
reconfigure configuration again.

7. Now you are ready to use the ATA !!!

19

3.2.2 PPPoE Client Mode

1. After selecting the Protocol mode: PPPoE Client Mode

2. Enter Username / Password / Registrar Address and Port / Proxy Address and Port
/ Outbound Proxy Address and Port / Auth User ID / Password in VoIP Line 0 and VoIP
Line 1 which was given by Telecom or by your Internet Service Provider (ISP).

3. Click Setup, it will start to configure the ATA for a while. Follow the instructions of the
Easy Setup utility which will guide you to complete the configuration.

4. Easy setup configuration completed.

5. Please connect the Ethernet cable between the ATA and Broadband Device.

6. Please check the SIP LED is lighted or not. If the SIP LED is lighted, the ATA is

20

REGISTERED successfully to the SIP Server. If not, please press reset button and
reconfigure again.

7. Now you are ready to use the ATA !!!

21

3.2.3 Static IP Settings Mode

1. After selecting the Protocol mode: Static IP Settings Mode

2. Enter IP Address / Subnet Mask / Gateway / DNS Server Url Name / DNS Server IP /

Registrar Address and Port / Proxy Address and Port / Outbound Proxy Address and
Port / Auth User ID / Password in VoIP Line 0 and VoIP Line 1 which was given by

Telecom or by your Internet Service Provider (ISP).

3. Click Setup, it will start to configure the ATA for a while. Follow the instructions of the
Easy Setup utility which will guide you to complete the configuration.

4. Easy setup configuration completed.

5. Please connect the Ethernet cable between the ATA and Broadband Device.

6. Please check the SIP LED is lighted or not. If the SIP LED is lighted, the ATA is

22

REGISTERED successfully to the SIP Server. If not, please press reset button and
reconfigure again.

7. Now you are ready to use the ATA !!!

23

3.3 Basic VoIP Configuration

3.3.1 Access to the web configuration of ATA

Step 1:

1. Launch the Web browser (Internet Explorer, Netscape, etc.).

2. Enter the LAN port default IP address (default gateway) http://12.0.0.2 in the address
bar.

3.

Entry of the username and password will be prompted. Enter the default login User Name and
Password:

The default login User Name of the administrator is admin, and the default login Password is epicrouter.

Remember my password checkbox: By default, this box is not checked. Users can check this box so that

Internet Explorer will remember the User name and Password for future logins. It is recommended to leave

this box unchecked for security purposes.

24

Step 2:

Now you could configure the ATA in detail.

25

3.3.2 VoIP Configuration

Step 1:
Click " VoIP " click Line to Confiure (if you have this option) and then click " Update
Service Provider "

26

Step 2:
Enter the information of "New Service Provider / Registrar Address / Registrar Port /
Proxy Address / Proxy Port / OutboundProxy Address / OutboundProxy Port " ,
select the Service Provider Action to "ADD NEW SP" and then click "Submit Changes ->
VoIP ".

27

Step 3:
Select the Service Provider which you configured and then click " Update User Login
Account".

28

Step 4:

Enter the "New Account Name / User ID / Password / Auth User ID / Display Name"
which you applied to the SIP Server, select the Login Action to "ADD " User and then click
"Submit Changes -> VoIP ".

29

Step 5:
Select the Login Account which you configured and then click " Submit Changes ->
Save Configuration".

30

Step 6:
Click " Save & Reboot ".

31

Step 7:

Your settings are being saved and the modem being rebooted. Save-reboot in progress,
please wait…

32

Step 8:

Your settings are being saved and the modem being rebooted. Done.

33

3.3.3 WAN Configuration

3.3.3.1 Static IP Configuration

Step 1:
Click " WAN " and then enter the " IP Address / Subnet Mask / Gateway " in Static IP
Settings Mode and then click " Submit "

34

Step 2:
Click " DNS " and then check User Configuration, enter " DNS Server -> Add / DNS
Server -> Enabled / Url Name / Host IP -> Add -> Apply "

35

Step 3:
Click " Save Configuration "

36

Step 4:
Click " Save & Reboot ".

37

Step 5:

Your settings are being saved and the modem being rebooted. Save-reboot in progress,
please wait…

38

Step 6:

Your settings are being saved and the modem being rebooted. Done.

Please check the SIP LED is lighted or not. If the SIP LED is lighted, the ATA is
REGISTERED successfully to the SIP Server. If not, please press reset button and
reconfigure configuration again.

39

3.3.3.2 DHCP Client Mode Configuration

Step 1:

1. Launch the Web browser (Internet Explorer, Netscape, etc.).

2. Enter the LAN port default IP address (default gateway) http://12.0.0.2 in the address
bar.

3.

Entry of the username and password will be prompted. Enter the default login User Name and
Password:

The default login User Name of the administrator is admin, and the default login Password is epicrouter.

Remember my password checkbox: By default, this box is not checked. Users can check this box so that

Internet Explorer will remember the User name and Password for future logins. It is recommended to leave

this box unchecked for security purposes.

40

Step 2:

Click " WAN " and then Select " DHCP Client Enable " in DHCP Client Mode and then
click " Submit -> Save Configuration"

41

Step 3:
Click " Save & Reboot ".

42

Step 4:

Your settings are being saved and the modem being rebooted. Save-reboot in progress,
please wait…

43

Step 5:

Your settings are being saved and the modem being rebooted. Done.

Please check the SIP LED is lighted or not. If the SIP LED is lighted, the ATA is
REGISTERED successfully to the SIP Server. If not, please press reset button and
reconfigure configuration again.

44

3.3.3.3 PPPoE Client Mode Configuration

Step 1:

1. Launch the Web browser (Internet Explorer, Netscape, etc.).

2. Enter the LAN port default IP address (default gateway) http://12.0.0.2 in the address
bar.

3.

Entry of the username and password will be prompted. Enter the default login User Name and
Password:

The default login User Name of the administrator is admin, and the default login Password is epicrouter.

Remember my password checkbox: By default, this box is not checked. Users can check this box so that

Internet Explorer will remember the User name and Password for future logins. It is recommended to leave

this box unchecked for security purposes.

45

Step 2:
Click " WAN ", check Enable, enter Username and Password, check Automatic
Reconnect in PPPoE Client Mode and then click " Submit -> Save Configuration"

46

Step 3:
Click " Save & Reboot ".

47

Step 4:

Your settings are being saved and the modem being rebooted. Save-reboot in progress,
please wait…

48

Step 5:

Your settings are being saved and the modem being rebooted. Done.

Please check the SIP LED is lighted or not. If the SIP LED is lighted, the ATA is
REGISTERED successfully to the SIP Server. If not, please press reset button and
reconfigure configuration again.

49

4. Advanced VoIP Configuration

The ATA is configured using the web interface. The ATA Configuration page can be
reached as follows:

1. Launch the Web browser (Internet Explorer, Netscape, etc.).

2. Enter the LAN port default IP address (default gateway) http://12.0.0.2

in the address

bar.

3. Entry of the username and password will be prompted. Enter the default login User
Name and Password:
The default login User Name of the administrator is admin, and the default login Password
is epicrouter.

Remember my password checkbox: By default, this box is not checked. Users can check
this box so that Internet Explorer will remember the User name and Password for future
logins. It is recommended to leave this box unchecked for security purposes.

50

4. On the router Home Page, click the VoIP link on the left frame to view the ATA
Configuration page.

In general, configuration changes made using the web interface will be activated only upon
clicking Save & Reboot button on the Save Savings / Reboot page.

Note: Certain Voice Parameters do not require a Save & Reboot to take effect. These
Voice Parameters will take effect on the next voice call after the Voice Parameter is entered
and submitted. If Save & Reboot is not done, then these Voice Parameters will not be
saved over a power cycle. The Voice Parameters that can be changed “on the fly” are
noted in the respective sections.

51

4.1 ATA Configuration Page

The ATA Configuration page sets parameters for the VoIP application.

The ATA Configuration page is divided into three general categories: Version Details,

Line Based Config, and Non-Line Config.
Version Details: This section displays the current versions of the ATAA and PTM software.
Line Based Config: This section configures parameters for the selected line.

Line to Configure: Select 0 (default) to configure Line 1 or select 1 to configure Line 2.

When a line is selected, the other fields in this section are refreshed with the currently
configured values. For example, if 0 (Line 1) is selected, the web page is refreshed to
reflect the current value for Enable SIP Registration, Service Provider To Use, Login

Account To Use, and the Current Registration Status.

Enable SIP Registration: Click Yes to enable SIP Registration (default) or click No to

disable SIP Registration for the selected line.

52

Note: Disabling SIP Registration means incoming calls cannot be received and outgoing
calls through the proxy also cannot be made; this is done to enable back-to-back direct
User Agent (UA) calling.

Transport Type: Select the transport to use for SIP signaling, UDP or TCP. 
Service Provider To Use: Select the service provider to work with the ATA for

the selected line. Different service provider-specific details can be configured by clicking
the Update Service Provider link to display the SIP Service Provider configuration page
and by following the instructions in Section 4.1.3.

Note: When a different service provider is chosen from the drop-down list, the Login
Account To Use drop-down list is updated to reflect the login details available and
configured for the selected service provider.

Login Account To Use: Select the Login Account to work with the ATA for the selected line
and the selected service provider. Different Login Account details can be configured by
clicking the Update User Login Account link to display the User Login Account page
and by following the instructions in Section 4.1.4. This permits multiple logins to be created
for each service provider.
Current Registration Status: This field displays the current registration status of the
selected line as defined in the following table:

Current Registration Status Shown Condition

VOIP SERVICES DISABLED SIP is disabled
REGISTRATION DISABLED Registration is disabled on web page

REGISTERING ........ WAN is up and Registration is being

attempted (assuming Registration is

enabled)
REGISTERED Registration succeeded
REGISTRATION FAILED Registration failed
UNREGISTERED Board is powered on, but WAN is still down.

53

Current Voice Call Status: This field displays the current registration status ofthe selected
line as defined in the following table:

Current Voice Call Status Shown Condition

NO VOICE CALL IN PROGRESS No voice calls on the selected endpoint

currently
VOICE CALL IN PROGRESS... CODEC
USED IS G711U

Voice call in progress with PCMU or

G711U as the selected codec
VOICE CALL IN PROGRESS... CODEC
USED IS G711A

Voice call in progress with PCMA or

G711A as the selected codec
VOICE CALL IN PROGRESS... CODEC
USED IS G729

Voice call in progress with G729 as the

selected codec
VOICE CALL HELD BY REMOTE END Remote endpoint has held the voice call
VOICE CALL ON HOLD BY LOCAL END Local endpoint has put the voice call on

hold

Timer Parameters: Click this link to open the Timeout page to configure timer values for
the selected line. See configuration details in Section 4.1.5.

Call Feature Parameters: Click this link to open the Call Feature page to configure
parameters related to call features for the selected line. See configuration details in Section

4.1.6.

Address Book Configuration: Click this link to open the Address Book Configuration
webpage to configure address book entries. See configuration details in Section 4.1.7.

Advanced Telephony Settings: Click this link to open the Advanced Telephony Settings
page to configure advanced telephony parameters for the selected line. See configuration
details in Section 4.1.8

Submit Changes: Click Submit Changes to save the configured settings in this section to
system RAM for the selected line.

Note: After clicking Submit Changes to save section settings to system RAM, you must
permanently save the configuration and reinitialize the system as follows.

Save Configuration: Click this link to go to the Save Settings / Reboot page. On the
Save Settings / Reboot page, click Save & Reboot to permanently save the settings to

system flash memory and to reinitialize the system to the new settings for the selected line.

Non-Line Config: This section configures system-level parameters independently of the
selected line.

54

Country Specific Ring & Tone Parameters: Click this link to display the Country Specific
Ring & Tones configuration page to define parameters for various tones. See configuration
details in Section 4.1.1.

General Parameters: Click this link to display the General configuration page for non-line
system-level parameters. See configuration details in Section 4.1.2.

Advanced System Telephony Parameters: Click this link to display the webpage for
configuring system level telephony settings. See configuration details in Section 4.1.9.
PSTN Parameters: Click this link to display the PSTN Configuration page to configure
FXO settings. See configuration details in Section 4.1.10.

55

4.1.1 Country Specific Ring & Tones Configuration

The Country Specific Ring & Tones configuration page defines parameters for the
various tones (ring, dial, busy, ring back, etc.) generated by the ATA application. ATA
provides default ring and tone parameters configured for operation in the USA. Flexibility is
provided to change the existing ring-tone parameters, and to add new countries as well as
to edit/delete existing countries.

Working Country: Select the country for which the ring-tone parameters will apply. The
default country is USA. When a country is selected from this drop-down list, the ring- tone
parameters entered for that country are automatically displayed. The maximum number of
countries that can be configured is 32. New countries can be defined and added manually.

New Country: Enter the name of the country to be added if the field is blank, or change the
name displayed in the Working Country field as desired.
Notes

1. For the following tone parameters, time is specified in milliseconds and frequency is
specified in Hertz.

2. Amplitude values, specified in dBm multiplied by a factor of 10.

56

Ring Parameters

Ring: Enter five consecutive fields separated by commas: Frequency, OnTime1,
OffTime1, OnTime2, and OffTime2.

Tone Parameters

PSTN Dial Tone: Enter six consecutive fields separated by commas: Freq1, Amp1, Freq2,
Amp2, OnTime1, OffTime1, OnTime2, and OffTime2. The PSTN is used when WAN is

disconnected or not-operative.

Peer2Peer (IP Dialing) Dial Tone: Enter six consecutive fields separated by commas:
Freq1, Amp1, Freq2, Amp2, OnTime1, OffTime1, OnTime2, and OffTime2. These

parameters are used when Registration is disabled.

Normal Dial Tone: Enter six consecutive fields separated by commas: Freq1, Amp1,
Freq2, Amp2, OnTime1, OffTime1, OnTime2, and OffTime2. These parameters are used

when Registration is enabled and up.

Busy Tone: Enter six consecutive fields separated by commas: Freq1, Amp1, Freq2,
Amp2, OnTime1, OffTime1, OnTime2, and OffTime2.

RingBack Tone: Enter six consecutive fields separated by commas: Freq1, Amp1, Freq2,
Amp2, OnTime1, OffTime1, OnTime2, and OffTime2.

Call Waiting Tone: Enter six consecutive fields separated by commas: Freq1, Amp1,
Freq2, Amp2, OnTime1, OffTime1, OnTime2, and OffTime2.

Alerting Tone: Enter six consecutive fields separated by commas: Freq1, Amp1, Freq2,
Amp2, OnTime1, OffTime1, OnTime2, and OffTime2.

Congestion Tone: Enter six consecutive fields separated by commas: Freq1, Amp1,
Freq2, Amp2, OnTime1, OffTime1, OnTime2, and OffTime2.

Fast Busy Tone: Enter six consecutive fields separated by commas: Freq1, Amp1,
Freq2, Amp2, OnTime1, OffTime1, OnTime2, and OffTime2.

Confirm Tone: Enter six consecutive fields separated by commas: Freq1, Amp1, Freq2,
Amp2, OnTime1, OffTime1, OnTime2, and OffTime2.

Warble Tone: Enter six consecutive fields separated by commas: Freq1, Amp1, Freq2,
Amp2, OnTime1, OffTime1, OnTime2, and OffTime2.

57

Unobtainable Tone: Enter six consecutive fields separated by commas: Freq1, Amp1,
Freq2, Amp2, OnTime1, OffTime1, OnTime2, and OffTime2.

Recall Tone: Enter five consecutive fields separated by commas: Freq1, Amp1, Freq2,
Amp2, and Stutter Duration.

Stutter Dial Tone: Enter five consecutive fields separated by commas: Freq1, Amp1,
Freq2, Amp2, and Stutter Duration.

VMI Dial Tone: Enter five consecutive fields separated by commas: Freq1, Amp1, Freq2,
Amp2, and Stutter Duration. This plays a distinctive stutter dial tone when Off Hook and

there is Voice Mail waiting.

RingTone Action: Select the drop-down option (DISPLAY, ADD, EDIT or DELETE)
to display or manipulate the ring-tone parameters for the selected working country.

•DISPLAY: Select DISPLAY to enable display of the ring-tone parameters when a country
is selected in the Working Country field. This is the default selection.

•ADD: Select ADD to add the updated ring-tone parameters for the new country appearing
in the New Country field, upon clicking Submit Changes. The New Country field must not
be empty. After Submit Changes is clicked, DISPLAY will be displayed.

• EDIT: Select EDIT to overwrite the ring-tone parameters for the selected country in
Working Country field with the updated ring-tone parameters and to overwrite the name in
Working Country field if a name has been entered in the New Country field, upon clicking
Submit Changes. After Submit Changes is clicked, DISPLAY will be displayed.

•DELETE: Select DELETE to delete the selected Working Country from the country list,

upon clicking Submit Changes. After Submit Changes is clicked, DISPLAY will be
displayed.

Submit Changes: Click Submit Changes to save the settings on this page to system
RAM.

Note: After clicking Submit Changes to save page settings to system RAM, you must
permanently save the configuration and reinitialize the system as follows.

Save Configuration: Click this link to go to the Save Settings / Reboot page. On the
Save Settings / Reboot page, click Save & Reboot to permanently save the settings to

system flash memory and to reinitialize the system to the new settings.

58

4.1.2 General Configuration Page

The General configuration page configures system-level parameters not related to the
selected line. This has five sections: SIP Device, VoIP General, BLAM Server, STUN

Parameters, and Default Dial Plan Parameters.

SIP Device: This section configures the following information:

•• Enable SIP: Click Yes to specify SIP is to be used for signaling (default), or No to specify
SIP is not to be used for signaling.

•• Local SIP Port: Enter the local SIP Port number on which ATA should listen for
messages. The range is 1 to 65535. The default port is 5060.

• •Media Base Port: Enter the Media Base Port (also known as RTP port) number.This
parameter provides the base value from the media (RTP) ports that are assigned for
various lines and the different call-sessions that may exist within an end-point. Odd port
values are not recommended. If an Odd Value is entered, the next higher even value is
used as the Media Base Port. This is to conform to the RFC specifications. The range is 1
to 65500. The default port is 5000.

59

VoIP General: This section configures the following information:

Enable Auto Login: If enabled, the system will obtain the login information from the

service provider using the MAC address.
Note: When enabled and successful in obtaining the login information, no VoIP web pages
will be displayed.

User Agent string: Enter the User Agent string to be sent in the User Agent/Server Header
of the SIP requests. Usually customer software version is provided.

Append User ID: Enable this to append the local registration number or PIN Number
entered in the following field to the TO header in the SIP requests. If PIN Number field is left
empty, the User ID (see Section 2.1.4) is appended to the TO Header. If filled, the PIN
Number is appended to the TO Header.

Pin Number: Enter PIN number to be appended to the TO Header after enabling Append
User ID field.

Interface To Use: Select the VoIP interface to be used for calls. The default is
the first available and enabled WAN interface.
Note: The user can make calls only when the specified or default (in case specified
interface is not available) interface has come up successfully.
For a non-ADSL system, the available interface can be Ethernet 1.
Ethernet 1 is the default MAC interface available on the 1-Line
Reference Voice system.
For an ADSL system, the available interfaces can be Pvc 0, Pvc 1, etc.
Any of the available PVCs can be chosen from the drop down list. (PVC refers to the
Permanent Virtual Circuit typically available on an ADSL+Voice systems.)

RegFail Retry Timer: Enter the registration retry timer value in seconds. This is the time
period for which the system will wait after a registration failure before attempting to retry.

BLAM Server: This section allows the user to start and stop the BLAM server. The fields
are RTrace fifo size, RDump fifo size, and Port number.

To start the BLAM server, enter a valid set of values into the RTrace fifo size, RDump fifo
size, and Port fields, then click Submit Changes.

To stop the BLAM server, set both the Rtrace fifo size and Rdump fifo size values to 0,
then click Submit Changes.

60

• RTrace fifo size: Enter the Rtrace FIFO size in KB. A non-zero value is required for
starting the BLAM server. The maximum value is 100. Enter 0 for stopping the BLAM
server.

•RDump fifo size: Enter the Rdump FIFO size in KB. A non-zero value is required for
starting the BLAM server. The maximum value is 100. Enter 0 for stopping the BLAM
server.

• Port: Enter the port number on which the BLAM server listens for connection from the
BLAMski application. The range is 1 to 32767.

•BLAM Current State: This display-only field indicates the current state of the BLAM
server. This status is refreshed upon clicking Submit Changes.

NAT Traversal Parameters: This section configures for the NAT Traversal technique
support in ATA.
NAT Traversal Technique: Select USE STUN to enable STUN (default) if the ATA is
behind a NAT enabled router and the router has no ALG for SIP, or NONE to disable STUN
(ATA is not to use STUN for NAT traversal). ATA also supports a proprietary implementation
of NAT traversal where the Service provider is expected to provide some relay support. If
NONE is selected, then based on the responses received, the ATA will dynamically
determine if the SIP Server supports the proprietary implementation.
Note: Even when STUN is enabled, the ATA does an automatic detection of the

presence of SIP ALG and disables the use of STUN. This is to avoid some media
problems arising out of the behavior of some ALGs when STUN is used at the user
end.

STUN Parameters: This section configures STUN (Simple Transversal of UDP through

NAT) support in ATA.

STUN Server: Enter the IP address or Domain Name of the STUN Server. The default is

66.7.238.210. This field is applicable only if USE STUN is selected as the NAT traversal
technique.

STUN Port: Enter the port number on which the STUN server listens for requests from the
STUN Client on ATA. The range is 1 to 65535. The default is 3478. This field is applicable
only if USE STUN is selected as the NAT traversal technique.

Force Keep Alive: Only valid when STUN is not used. If STUN is not enabled, and keep
alive is still expected to be sent then select Yes otherwise select No.

Keep Alive Period: The keep alive interval in seconds to be used when STUN is not
enabled.

Default Dial Plan Parameters: This field provides the default dial plan string that can be

61

configured for the system (see Section 4.2.4 for more details on the default value for this
field). Edit the dial plan as required.

When a new service provider is added, the initial dial plan string for the service provider is
taken from this default dial plan string. If this field is left empty, the system attaches an
empty dial plan string to the new service provider, in which case it applies the default rule
upon save and reboot (see Section 4.2.4). Refer to Section 2.2 for more details on how to
build the dial plan string.

Submit Changes:

For SIP Device, VoIP General, STUN Parameters, and Default Dial Plan Parameters,
click Submit Changes to save the settings to the system RAM. In order to save changes
permanently to the firmware and to make them effective, the settings must be saved by
going to the Save Settings / Reboot web page.

Save Configuration: Click this link to go to the Save Settings / Reboot page. On the
Save Settings / Reboot page, click Save & Reboot to permanently save the settings to

system flash memory and to reinitialize the system to the new settings.

62

4.1.3 SIP Service Provider Configuration Page

The SIP Service Provider configuration page sets the configuration related to the SIP
service provider.

Service Provider List: Enter the name of the service provider to be configured. When a
service provider is selected from this drop-down list, the respective parameters are
automatically displayed.

A DEFAULT service provider is provided with a default set of parameters. This can be
edited. New service providers can be manually defined and added. An existing service
provider can be edited or deleted.

New Service Provider: Enter the name of the service provider to be added if the Service
Provider List field is blank, or a new string to rename the service provider displayed in the
Service Provider List field.

Restrict Packets from Service Provider: Check this box, if you want the system to accept

packets from only this service provider and drop packets received from other IP addresses.

63

Registration Interval (in secs): Enter the re-registration interval in seconds. The range is
0 to 2147483347 seconds. The default is 3600 seconds.
Authentication Method: Select the authentication method. Only MD5 is supported.

• AUTH_NONE: Disable any authentication method

• AUTH_MD5: Use MD5 authentication method.

Registrar Address: Enter the IP address or Domain Name of the registrar with which the

ATA must register in order to receive or send calls.

Registrar Port: Enter the port number of the registrar on which it will listen for Register
requests from the ATA. The range is 1 to 65535. The default port is 5060.

Proxy Address: Enter the IP address or Domain Name of the SIP proxy server.

Proxy Port: Enter the port number on which the SIP proxy server will listen for messages.

The range is 1 to 65535. The default port is 5060.

OutboundProxy Address: Enter the IP address or Domain Name of the Outbound proxy
server. This is useful in cases where the ATA is behind a NAT.

OutboundProxy Port: Enter the port number on which the Outbound proxy server listens
for messages from the ATA. The range is 1 to 65535. The default port is 5060.

Note: Refer to RFC 3261 [ 2] for more SIP definitions.

Dial Plan String: This parameter provides the dial plan string as required by the service
provider. Modifying this field while adding a new service provider will not take effect after
ADD has been selected and Submit Changes has been clicked. While adding a new
service provider, the dial plan string takes the value from the default dial plan string
specified in the VoIP General Parameters web page only. To modify this field, complete
adding the service provider, and then edit it, select EDIT in the drop-down box of Service
Provider Action and click Submit Changes. Please refer to Section 4.2 for more details
on how to build the dial plan string.

Service Provider Action: Select the drop-down option (DISPLAY SP RULES, ADD NEW
SP, DELETE SEL SP, or EDIT SEL SP) to display and manipulate the SIP and dial plan

parameters for the service provider selected in the Service Provider List.

• DISPLAY SP RULES: Select DISPLAY SP RULES to enable display of the selected
service provider parameters when a service provider is selected in the Service Provider

64

List field. This is the default selection.

• ADD NEW SP: Select ADD NEW SP to add a new service provider after clicking Submit
Changes according to the value that appears in the New Service Provider field. This field
must not be empty.

Note: The maximum number of service providers is 4.

• DELETE SEL SP: Select DELETE SEL SP to delete the selected service provider from
the Service Provider List.

• EDIT SEL SP: Select EDIT SEL SP to overwrite the selected service provider’s (in the
Service Provider List field) parameters with the current parameters displayed on the web
page. The New Service Provider field is optional and needs to be filled only when the
service provider name also has to be changed.

Submit Changes: Click Submit Changes to save the settings on this page to system
RAM.

Note: After clicking Submit Changes to save page settings to system RAM, you must
permanently save the configuration and reinitialize the system as follows.

Save Configuration: Click this link to go to the Save Settings and Reboot page. On the
Save Settings and Reboot page, click Save & Reboot to permanently save the settings to

system flash memory and to reinitialize the system to the new settings.

65

4.1.4 User Login Account Configuration Page

The User Login Account configuration page sets and configures login accounts for the
service provider chosen in the index web page, i.e., for the currently selected service
provider in the main web page.

Service Provider Name: This field displays the service provider for which the login
information is being configured. The service provider is selected on the ATA Configuration
page.

Login Account List: Select the login account to be configured. When a login account is
selected from this drop-down list, the respective parameters are automatically displayed. A
default set of parameters is provided for every new login account added. These parameters
can be edited. New login accounts can be defined and added. An existing login account
can also be edited or deleted.

New Account Name: Enter the name of the new login account to be added, or a new string
to rename an existing login account. The login account edited will be the one selected from
the Login Account List field.

66

User ID: Enter the registration ID of the user with the registrar.

Password: Enter the password used for authentication with the registrar.

Auth User ID: Enter the Authorization User ID for authentication with the registrar. If not

specified explicitly by the service provider, this is same as the User ID.

Display Name: Enter the Display Name as it should appear on Caller ID.
Login Action: Select a drop-down option (DISPLAY, ADD, EDIT, or DELETE) to
manipulate the various login parameters for the login account selected in the Login
Account List.

• DISPLAY: Select DISPLAY for the selected login account details to be displayed after
clicking Submit Changes. This is the default selection.

• ADD: Select ADD to add a new login account after clicking Submit Changes according
to the value that appears in the New Account Name field. This field must not be empty.

• EDIT: Select EDIT to overwrite the selected login account (in the Login Account List
field) parameters with the displayed parameters. The New Account Name field is optional
and needs to be filled only when the login account name is to be changed.

• DELETE: Select DELETE to delete the selected login account from the Login Account
List.

Note: The above parameters are specific to the service provider selected in the ATA
Configuration page.

Note: Up to four login accounts can be added per service provider.

Submit Changes: Click Submit Changes to save the settings on this page to system
RAM.

Note: After clicking Submit Changes to save page settings to system RAM, you must
permanently save the configuration and reinitialize the system as follows.

Save Configuration: Click this link to go to the

Save Settings and Reboot page. On the

Save Settings and Reboot page, click Save & Reboot to permanently save the settings to

system flash memory and to reinitialize the system to the new settings.

67

4.1.5 Timer Parameters Configuration Page

The Timer Parameters configuration page displays and configures timers used at the
system level. This section explains the various timers available for configuration. These
timers are not applicable for PSTN calls.

Selected Endpoint: The endpoint for which these timer parameters are applicable. This
corresponds to the current selected endpoint on the ATA Configuration webpage.

Predial Timeout (Sec): Enter the length of time in seconds the dial tone will be generated
once the phone has been lifted off hook. At the end of this period, if no digits have been
pressed, the ATA will start playing the Fast-Busy tone.

Call Progress Timeout (Sec): Enter the length of time in seconds the ATA will wait for the
initial response from the other end point once an outgoing call has been made.

Alert Timeout (Sec): Enter the length of time in seconds the ATA will play the Ring tone
when an incoming call has arrived and the phone is on-hook. At the end of this period, the
ATA will automatically stop the ring and reject the call.

68

Call Waiting Timeout (Sec): Enter the length of time in seconds the Call Waiting tone will
be played when an incoming call arrives in the connected/held state. The Call Waiting tone
is played at an interval of 10 sec. for USA. It is configurable using the Call Waiting tone
parameters.

Disconnect Timeout (Sec): Enter the length of time in seconds the Fast-Busy tone will be
played once a call has been disconnected by the remote-end. At the end of this period, the
Warble tone will be played until the user hangs up the phone.

Call Forward No Ans Timeout (Sec): Enter the length of time in seconds after which the
call will be forwarded when it is not answered. This timer is applicable when Call
Forwarding on No Answer is enabled from either the Call Feature configuration page
Section 4.1.6) configuration page or the dial-pad.

RingBack Timeout (Sec): Enter the length of time in seconds ATA will wait while the
RingBack tone is being played for the final response from the other end point once an
outgoing call has been made and the initial response has been received.

Submit Changes: Click Submit Changes to save the settings on this page to system
RAM.

Note: After clicking Submit Changes to save page settings to system RAM, you must
permanently save the configuration and reinitialize the system as follows.

Save Configuration: Click this link to go to the Save Settings and Reboot page. On the
Save Settings and Reboot page, click Save & Reboot to permanently save the settings to

system flash memory and to reinitialize the system to the new settings.

69

4.1.6 Call Feature Configuration Page

The Call Feature configuration page displays and modifies call features. Enabling any of
these options allows the user to apply the appropriate configuration as specified in the dial
plan by using the dial-pad.

Selected Endpoint: The endpoint for which these call features parameters are applicable.
This corresponds to the current selected endpoint on the ATA Configuration webpage.

Call Forwarding:

• Enable Call Forwarding Unconditionally: Click this check box to enable Unconditional

Call Forwarding. When enabled, an incoming call will be forwarded to the entered Call
Forwarding Number when using the dial-pad as described in Section 5.3.8.

• Enable Call Forwarding On Busy: Click this check box to enable Call Forwarding on
Busy. When enabled and the telephone is connected, an incoming call will be forwarded to
the entered Call Forwarding Number when using the dial-pad as described in Section 5.3.8.

• Enable Call Forwarding On No Ans: Click this check box to enable Call Forwarding on

70

No Answer. When enabled and the incoming call is not answered within a time limit, the
incoming call will be forwarded to the entered Call 5.3.8. The Forwarding Number when
using the dial-pad as described in Section Call Forward No Ans Timeout timer is
configured as described in Section 4.1.5.

• Call Forwarding Number: Enter the telephone number to be used when Call Forwarding
is enabled as described above. The user can over-ride this number from the dial-pad if
desired as described in Section 5.3.8.

Call Waiting: Click Yes to enable Call Waiting or No to disable Call Waiting.
The user can temporarily disable Call Waiting from the dial-pad (see Section 5.3.7) only
when this feature is enabled.

Call Return: Click Yes to enable Call Return or No to disable Call Return.
The user can do Call Return from the dial-pad (Section 5.3.9) only when this feature is
enabled.

Caller ID Transmit Block: Click Yes to allow Caller ID Transmit Blocking or No to disable
Caller ID Transmit Blocking.
The user can use Caller ID Transmit Block from the dial-pad (see Section 5.3.12) only
when this feature is enabled.

Privacy Display Name: The name to display when using Caller ID Transmit Blocking.

Conferencing: Click Yes to enable 3-way conferencing or No to disable 3-way

conferencing.
The user can temporarily disable 3-way conferencing from the dial-pad (see Section 5.3.6)
only when this feature is enabled. When disabled, the user can switch between the calls
using hookflash.

Submit Changes: Click Submit Changes to save the settings on this page to system
RAM.
Note: After clicking Submit Changes to save page settings to system RAM, you must
permanently save the configuration and reinitialize the system as follows.

Save Configuration: Click this link to go to the Save Settings and Reboot page. On the
Save Settings and Reboot page, click Save & Reboot to permanently save the settings to

system flash memory and to reinitialize the system to the new settings.

71

4.1.7 Address Book Configuration Page

The Address Book Configuration Webpage allows for configuration of address book
entries which can be used for speed dial execution of calls.

The top half of the webpage displays the current address book table for the selected
EndPoint.

The bottom half of the webpage can be used for editing the address book. This includes
adding new entries, deleting existing entries, modifying existing entries and displaying the
values corresponding to an entry index or speed dial index in the address book.

Selected Endpoint: The endpoint for which this address book is applicable. This
corresponds to the current selected endpoint on the ATA Configuration webpage.

Address Book Table: This table displays the current address book as configured for this
endpoint.

Display Name: Enter the Display Name for this address book entry.

72

Number: The user phone number or name for this entry. This field is optional, if the IP
Address is specified. A phone number that can be reached through the current configured
proxy server can also be added as an entry in which case the IP address/Domain name
and the Port number fields are not necessary.

IP Address/Domain Name: Enter the IP address or the domain name that corresponds to
this address book entry. If this field is left empty, then the User number or name must be
specified, in which case the current configured proxy server for this endpoint will be used as
the domain name.

Port Number: Specify the SIP port number on which the remote end will receive our call.
This is useful when you want to specify a non-phone number entry, where the call can be
made directly without going through the configured proxy server. When this field is not
specified, the default SIP port of 5060 will be assumed.

Speed dial code: This refers to the index in the address book as well as the speed dial
entry code. This needs to be specified following *78 (or the configured speed dial service
code – see section 4.2.2) to dial out the number corresponding to this address book entry.

Address Book Action: Select a drop-down option (DISPLAY, ADD, EDIT, or DELETE) to
manipulate the various address book parameters for the entry index selected from the
speed dial code drop down box.

• DISPLAY: Select DISPLAY for the selected speed dial code details to be displayed after
clicking Submit Changes. This is the default selection.

• ADD: Select ADD to add a new address book entry after clicking Submit Changes.

• EDIT: Select EDIT to overwrite the selected address book entry. 

• DELETE: Select DELETE to delete the selected address book entry

Note: The above parameters are specific to the endpoint selected in the ATA
Configuration page.

Note: Up to ten address book entries can be added per endpoint.

Submit Changes: Click Submit Changes

to save the settings on this page to system
RAM and Flash also.
Note: Address book entries addition/deletion/editing DONOT need a save and reboot. The
changes are reflected immediately.

73

4.1.8 Advanced Telephony Settings Page

The Advanced Telephony Settings page allows configuration of the various endpoint
level telephony parameters.

Selected Endpoint: The endpoint for which these call advanaced telephony settings are
applicable. This corresponds to the current selected endpoint on the ATA Configuration
webpage.

VAD: This section configures Voice Activity Detection.

• Disable VAD: Click this option button to disable VAD, i.e., voice activity will not be
detected.

• Enable VAD with COMFORT NOISE: Click this option button to enable VAD with comfort
noise generation.

• Enable VAD with STANDARD SID: Click this option button to enable VAD with SID
according to G.729 and PT13 of G.711.

74

• Enable VAD and SUPPRESS SID: Click this option button to enable VAD with
discontinuous transmission. When silence is detected, transmission of packets will be
paused until voice is again detected.

Echo Cancellation: This section configures Echo Cancellation with NLP or with CNG_NLP
options.

• Disable Echo Cancellation: Click this button to disable Echo Cancellation.

• Enable Echo Cancellation: Click this button to enable Echo Cancellation with no other
options.

• Enable Echo Cancellation with NLP: Click this button to enable Echo Cancellation
using NLP (Non Linear Processor). With this option, the Start Attenuation and Max
Attenuation can be set.

• Enable Echo Cancellation with CNG_NLP: Click this button to enable Echo
Cancellation using NLP and CNG (Comfort Noise Generation).

• Start Attenuation: Enter the minimum or start attenuation, in dB, for Echo Cancellation
when NLP is enabled. The default value is 8192.

• Max Attenuation: Enter the maximum converged attenuation, in dB, for Echo
Cancellation when NLP is enabled. The default value is 16384.

• Echo Cancellation Tail Length: Select the Echo Cancellation Tail Length, in ms. The
default value is 24 ms.

DTMF Relay: This section selects how packets containing Mid-Call DTMF tones are sent.

• NONE (IN AUDIO): Click this button to send Mid-Call DTMF tones in RTP packets with
the same payload as voice, i.e., dynamic payload negotiation for DTMF digits will not be
done.

• RFC 2833: Click this button to send Mid-Call DTMF tones in RTP packets separately
using RFC2833, i.e., dynamic negotiation of RTP payload for DTMF digits will be done.

• DTMF Payload Number: This field is configurable when RFC 2833 is selected as the
DTMF Relay mechanism. Specify the payload number that needs to be used for DTMF
information negotiated in SDP during SIP signaling.

Hook Flash Time: This section configures the Hook Flash time.

• HookFlash Max Time (ms): Enter the maximum time, in ms, of Hook Flash. For a Hook
Flash to be detected, there should be an on-hook occurrence followed by an off-hook
occurrence within the entered HookFlash Max Time period. If HookFlash Max Time expires
before an off-hook, an on-hook event will be sent.

If debounce time is configured, OffOnDebounce must be less than HookFlash MaxTime.

75

Debounce Times: This section configures two debounce delays.

• Debounce On-Off Time (ms): Enter the delay, in ms, before informing on an off-hook
event to ensure it is not a spike. If an on-hook event is received during this delay time, the
two events will be ignored. If 0 is entered, no debounce will be done.

• Debounce Off-On Time (ms): Enter the delay, in ms, before informing on an on-hook
event to ensure it is not a spike. If an off-hook event is received during this delay time, the
two events will be ignored. If a HookFlash Max Time is configured, it must be more than
OffOnDebounce. If 0 is entered, no debounce will be done.

Codec Preference: This section determines the order in which supported codecs will be
placed in a call setup message sent to any destination line. It also helps determine the
selected codec when a message indication for an incoming call is received from the remote
end with codec preference information.

• G711U: Select priority 1, 2, or 3 to be assigned to the G711U codec, or NONEif the
G711U codec is not to be used.

• G711A: Select priority 1, 2, or 3 to be assigned to the G711A codec, or NONEif the G711A
codec is not to be used.

• G729A: Select priority 1, 2, or 3 to be assigned to the G729A codec, or NONEif the
G729A codec is not to be used.

• G726-16: Select the priority 1 through 7 to be assigned to the G726-16 codec, or NONE if
the G726-16 codec is not to be used.

• G726-24: Select the priority 1 through 7 to be assigned to the G726-24 codec, or NONE if
the G726-24 codec is not to be used.

• G726-32: Select the priority 1through 7 to be assigned to the G726-32 codec, or NONE if
the G726-32 codec is not to be used.

• G726-40: Select the priority 1through 7 to be assigned to the G726-40 codec, or NONE if
the G726- 40 codec is not to be used.

Note: If two codec types are assigned the same priority, then the priority is assigned in the
order as G711U > G711A > G729> G726-16> G726-24> G726-32> G726-40 in the
decreasing order of priority. For example, if 1, 2, and 2 is selected for G711U, G711A, and
G729, respectively, priority will be assigned as 1, 2, and 3 for G711U, G711A, and G729,
respectively.
Note: There is no dynamic payload negotiation support for G726 bit rates.

Submit Changes: Click Submit Changes to save the settings on this page to system
RAM.

Note: All

Advanced Telephony setting parameters do not require a Save & Reboot to

take effect.

76

Note: After clicking Submit Changes to save page settings to system RAM, you must
permanently save the configuration and reinitialize the system as follows.

Save Configuration: Click this link to go to the Save Settings and Reboot page. On the
Save Settings and Reboot page, click Save & Reboot to permanently save the settings to

system flash memory and to reinitialize the system to the new settings.

77

4.1.9 Advanced System Telephony Parameters Page

The Advanced System Level Telephony Parameters page allows configuration of the
system level telephony parameters for Caller ID and Jitter Buffer.

Codec Prefereces: This section configures the frames per RTP packet for the selected
codec.

• User can select 1, 2, or 3 in the “Number frames per packet” text field.

• 1, 2, 3 corresponds to 20, 40, 60 ms voice size with respect to G711u, G711a and G729

• 1, 2, 3 corresponds to 30, 60, 90 ms voice size with respect to G723

Jitter Buffer Configuration: This section configures the jitter buffer information for
selected codec.

•Max Reorder delay (ms): When the Reorder buffer reaches max reorder delay value
specified, it is rapidly emptied.

78

• Max accept late seq num: If the new packet sequence number is low by “Max Accept
late seq num” from the reorder buffer tail sequence number, then a new sequence has

started and the reorder buffer is initialized.

• Initial delay (ms): Sets the initial delay for the jitter buffer.

• Excessive frame deletion mode (0/1): Once the reorder buffer grows past

peak_transit_delay, the ATA limits further growth in one of the two user selectable modes:
Soft (1): Every time a frame is read from the reorder buffer for decoding, a second frame is
read, decoded and deleted. This is continued until the reorder buffer is below the deletion
threshold.

Hard (0): Frames past the deletion threshold are immediately deleted, before decoding.

• Max Transit delay (ms): The maximal delay in the delay buffer, in millisec.

• Peak Transit delay (ms): The deletion threshold, in millisec. When the delay exceeds this

limit, frames are deleted according to excessive frame deletion mode.

• Delay Buffer Increment (ms): This field specifies the value by which the size of delay
buffer is increased when the delay buffer underruns.

• Transit delay decrease threshold (ms): The minimal interval of decreased jitter before
the delay buffer adapts downwards.

Caller ID Info: This section configures Caller ID (CID).

• Modulation: Select CID_BELL202 to use Bell 202 modulation or CID_V23 to use V.23

modulation.

• Delay to CID SLIC State (ms): Enter the time delay, in milliseconds, to change the SLIC
state, i.e., the delay after the end of the ring and before sending the CID.

• Delay to CID Transmission (ms): Enter the time delay, in milliseconds, between the end
of the first ring and the beginning of CID transmission.

• FSK AMP 0: Enter the amplitude of logic 0 frequency, in Hz.

• FSK AMP 1: Enter the amplitude of logic 1 frequency, in Hz.

Submit Changes: Click Submit Changes to save the settings on this page to system

RAM.

Note: All Advanced Telephony setting parameters do not require a Save & Reboot to
take effect.
Note: After clicking Submit Changes to save page settings to system RAM, you must
permanently save the configuration and reinitialize the system as follows.

Save Configuration: Click this link to go to the Save Settings and Reboot page. On the
Save Settings and Reboot page, click Save & Reboot to permanently save the settings to

system flash memory and to reinitialize the system to the new settings.

79

4.1.10 FXO Configuration Parameters Page

The FXO Configuration page allows configuration of system level FXO parameters.
See Section 5.5 for more details on FXO support.
If FXO support is enabled the following parameters are displayed.

Attach FXO to FXS: This field determines which FXS is to be attached to the FXO line.
Outgoing and incoming PSTN/FXO calls can be made and received on the attached FXS
only.
CID Enable: Click Yes to enable PSTN Caller ID display or No to disable PSTN Caller ID
display.
Tone Detection Wait Timer (Secs): Enter the length of time in seconds the ATA will wait
for the PSTN dial tone to be detected (to make sure CO supports call on hold) after the user
has pressed hook flash in a connected PSTN call. In case of tone not being detected or a
timeout, a fast busy tone will be played. The user can press hook flash again to come back
to the first PSTN call.
PSTN Backup: Click Yes to enable PSTN Backup or No to disable PSTN Backup.

80

Note: PSTN Parameters are only displayed when the vophwcfg.ini is appropriately
configured (see Appendix C.1).

If FXO support is disabled and the hardware does not have a FXO, no parameters will be
displayed.

Submit Changes: Click Submit Changes to save the settings on this page to system
RAM.

Note: After clicking Submit Changes to save page settings to system RAM, you must
permanently save the configuration and reinitialize the system as follows.

Save Configuration: Click this link to go to the Save Settings and Reboot page. On the
Save Settings and Reboot page, click Save & Reboot to permanently save the settings

to system flash memory and to reinitialize the system to the new settings.

81

4.2 Building Dial Plan String

4.2.1 Syntactic Format for Dial Plan Strings

The dial plan string consists of a set of dial plan rules specified according to the following
syntax:

1. DTMF digits 0–9, A–F, *, and #.

2. Range and sub-range of digits (“[ ]”). For example, [135] specifies digits 1, 3, or 5,
[5–9] specifies digits 5 to 9, and [125–8] specifies digits 1, 2, and 5 to 8.

3. Zero or more repetitions of the preceding event (“.”). For example, X.7 specifies 7
digits where each digit is 0–9, N.5 specifies 5 digits where each digit is 2–9, and [1–5].5
specifies 5 digits where each digit is 1–5.

4. Inter-digit timeout (IDT) in seconds (“t”) followed by a character that specifies the
timeout interval in seconds for the preceding event. For example, X.5t5 specifies 5 digits
where each digit is 0–9 with an IDT of 5 seconds.

5. Dial plan rule separator (“|”).

6. A suffix character (“>“) that defines the end of the required dialing digits; optional
dialing digits can follow.

7. “X” or “x” signifies digits 0–9.

8. “N” or “n” signifies digits 2–9.

9. Star service serial code separator (“:”) indicating the end of the serial code.

10. Star service dial-pad code separator (“;”) indicating the end of the dial-pad code.

11. Digit not allowed in a given position (“!”).

82

4.2.2 Dial Plan String and Serial Codes

This data defines the dial plan string as entered or configured by the user or the service
provider.
The user or service provider enters the dial string consisting of supported call-related dial
patterns and the star service dial patterns.
The call patterns can generally be entered in the order preferred by the user or service
provider. The normal call patterns do not have any serial codes associated with them.
The serial codes used for specifying the dial-pad code related to each star service are fixed
and the user must follow them.

Notes:

1. The dial plan interpreter analyzes the collected digits from the user on a first-come
first-served basis. If, at the end of the collection and analysis, the interpreter determines
that there are two or more possible candidates for a match, it chooses the first available
match in the order in which the rules appear in the dial plan string.

2. The maximum storage space allocated for the dial plan string is 512 characters.

The serial codes with associated star service features and default dial-pad codes are
defined in Table 2-1.

Table 2-1. Serial Codes, Star Service Features, and Default Dial-Pad Codes

Serial Code Star Service Feature Default Dial-Pad Code
1 Enable Unconditional Call Forwarding *72
2 Disable Unconditional Call Forwarding *73
3 Enable Call Forwarding on Busy *74
4 Enable Call Forwarding on No Answer *75
11 Temporary Disable Call Waiting *70
12 Call Return *69
16 Blind Transfer *90
18 IP Dialing *47
19 Speed Dialing from Address Book *78
20 PSTN Call #
22 Temporary Disable 3-Way Conferencing Call *83
23 Disable Call Forwarding on Busy *76
24 Disable Call Forwarding on No Answer *77
25 Emergency Call N11
26 Temporary Enable Caller ID Transmit Block *67
Note: Missing number codes (5-10, 17, 21, and 26-31) are not supported.

83

4.2.3 Framing Dial Plan Rules

Dial plan interpretation and parsing is based on the incoming character and a set of
expected characters as follows:

0–9, A–D, #, or * A DTMF digit is recognized as valid if it is one of the following:

between 0–9 or A or B or C or D or # or *.

X Recognized as any valid DTMF digit in the range 0–9. The expected

characters following this are “.” or “t” or any other valid syntactic
character.

N Recognized as any valid DTMF digit in the range 2–9. The expected

characters following this are “.” or “t” or any other valid syntactic
character.

t The associated inter-digit timer (IDT) value in seconds. The expected

character following this must be between 0–9 or a–z to represent the
timer value. For a–z, “a” corresponds to 10 seconds, “b” to 11 seconds,
and so on. The maximum timer value is “z” which corresponds to 35
seconds.

. This character represents zero or more repetitions of the previous

character (which must be a valid DTMF digit). This must be followed by
a value between 0–9 and a–f to specify the number of repetitions.
Letters a–f correspond to numbers 10–15, respectively.

> This character is used for two purposes:

a) To specify the suffix associated with the dial plan rule. A single valid
DTMF digit must follow this character.
b) To specify the minimum number of digits required to be dialed for the
corresponding dial plan rule. The > character must be included at the
minimum position required for meaningful match.
For the example pattern, 0>#t411t8x.etfxt2, only digit 0 is required for a
meaningful match.
For the example pattern, 0t411x>#t8x.etfxt2, a 4-digit dial string of 011X
is required for a meaningful match.

! This character, in association with a following DTMF digit, represents

what a digit in the specified position should not be. A DTMF digit
representation/range/sub-range must follow this character.

[X-X] Sub-range starting from first value to next value. For example, [2-8]

means a digit in the range 2 through 8.

[XX] Specifies an array of digits allowed in the position mentioned. For

example, [128] means digit 1, 2, or 8.

84

4.2.4 Default Dial Plan String

The default dial plan string corresponds to the North American Numbering Plan (NANP).
This default dial plan string is (where each rule is separated by the separator
“|”):0>#t4|Nx.5t8xt2>#|011x>#x.et8xt2|1Nx.2Nx.5t8xt2>#|1:*72;>#x.etfxt2|2:*73;>#t4|3:*74;
>#x.etfxt2|4:*75;>#x.etfxt2|11:*70;>#t4|12:*69;>#t4|16:*90;x>#x.dtfxt2|
18:*47;x>#[0-9*].f[0-9*].ft8
[0-9*].ft4|19:*78;x>#t4|20:#;x.3>#x.atfxt2|22:*83;x>#x.dtfxt2|23:*76;>#t4|24:*77;>#t4|25:N1
t41;># |26:*67;>#t4| [0-9*]>#[0-9*].e[0-9*].ft4

The individual default dial plan rules are summarized in Table 2-2.

Table 2-2. Default Dial Plan Rules

Default Dial Plan Rule Dial Plan Rule Function Reference

0>#t4 Dial local operator 4.2.4.1
Nx.5t8xt2># Dial a local call 4.2.4.2
011x>#x.et8xt2 Dial a long distance call 4.2.4.4
1Nx.2Nx.5t8xt2># Dial an international long distance call 4.2.4.3
1:*72;>#x.etfxt2 Star service for Enable Unconditional Call

Forwarding

4.2.4.5

2:*73;>#t4 Star service for Disable Unconditional

Call Forwarding

4.2.4.8

3:*74;>#x.etfxt2 Star service for Enable Unconditional Call

Forwarding on Busy

4.2.4.6

4:*75;>#x.etfxt2 Star service for Enable Unconditional Call

Forwarding on No Answer

4.2.4.7

11:*70;>#t4 Star service for Temporary Disable Call

Waiting

4.2.4.9

12:*69;>#t4 Star service for Call Return 4.2.4.10
16:*90;x>#x.dtfxt2 Star service for Blind Transfer 4.2.4.11
18:*47;x>#[0-9*].f[0-9*].ft8[0-9*].ft4 Star service for IP Dialing call 4.2.4.12
19:*78;t4x>#t2 Star service for Speed Dialing from

Address Book

4.2.4.13

20:#;x.3>#x.atfxt2 Dial PSTN call 4.2.4.13
22:*83;x>#x.dtfxt2 Star service for Temporary Disable 3-Way

Conferencing call

4.2.4.15

23:*76;>#t4 Star service for disabling Call Forwarding

on Busy

4.2.4.16

24:*77;>#t4 Star service for disabling Call Forwarding

on No Answer

4.2.4.17

25:N1t41;># Dial an emergency call 4.2.4.18
26:*67;>#t4 Star Service for Temporary blocking of 4.2.4.19

85

Caller ID Transmission

[0-9*]>#[0-9*].e[0-9*].ft4 Default number call 4.2.4.20

There are three major components of any star service rule:

1. The star service rule must begin with the serial code identified in Table 2-1. This serial
code identifies the feature/dial plan action to be executed. This serial code is followed by
the separator “:”.

2. Following this is the dial-pad code (digits the user must dial first) to execute the feature.
The separator “;” follows this dial-pad code.

3. Then the rule for the digits or number needed to execute the star service is specified.

Note: If the IDT elapses during the minimum number of digits to be dialed, the dialed string
does not match the plan, in which case a match to another plan is attempted. If the IDT
elapses during the optional digits to be dialed, the dialed number is sent.

4.2.4.1 Default Dial Plan Rule 0>#t4

This is the dial plan rule for dialing the local operator. Digit 0 must be dialed, optionally
followed by pressing #. After the 0 digit is dialed, press # to send the digit immediately,
otherwise there is a 4-second delay before the digit is sent.

Dial Plan Rule Explanation of the Rules

0 One digit 0
># Position of >specifies 1 digit minimum; suffix is #
t4 IDT of 4 seconds

4.2.4.2 Default Dial Plan Rule Nx.5t8xt2>#

This is the dial plan rule for dialing a local call within the geographical region covered by the
NANP. Seven digits must be dialed, each digit within 8 seconds of the preceding digit,
optionally followed by pressing #. After the seven digits are dialed, press # to send the
digits immediately, otherwise there is a 2-second delay before the digits are sent.

Dial Plan Rule Explanation of the Rules

N One digit in range 2–9
x.5 Five digits in range 0–9
t8 IDT of 8 seconds
x One digit in range 0–9
t2 IDT of 2 seconds
># Position of >specifies 7 digits minimum; suffix is #

86

4.2.4.3 Default Dial Plan Rule 1Nx.2Nx.5tfxt2>#

This is the dial plan rule for dialing a long distance call.
Eleven digits must be dialed starting with 1, each digit within 15 seconds of the preceding
digit, optionally followed by pressing #.
After the 11 digits are dialed, press # to send the digits immediately, otherwise there is a
2-second delay before the digits are sent.

Dial Plan Rule Explanation of the Rules

1 One digit 1
N One digit in range 2–9
x.2 Two digits in range 0–9
N One digit in range 2–9
x.5 Five digits in range 0–9
tf IDT of 15 seconds
x One digit in range 0–9
t2 IDT of 2 seconds
># Position of >specifies 11 digits minimum; suffix is #

4.2.4.4 Default Dial Plan Rule 011x>#x.et8xt2

This is the dial plan rule for dialing an international long distance call.
Four digits must be dialed starting with 011, followed by 15 optional digits and optionally
followed by pressing #. Each digit must be dialed within 8 seconds of the preceding digit.
After the 011x digits are dialed, press # to send the digits immediately, otherwise the digits
will be sent 8 seconds after the last dialed digit for digits 4-18 is dialed, or 2 seconds after
digit 19 is dialed.

Dial Plan Rule Explanation of the Rules

011 Three digits with 011 pattern
x One digit in range 0–9
># Position of >specifies 4 digits minimum; suffix is #
x.e Fourteen digits in range 0–9
t8 IDT of 8 seconds
x One digit in range 0–9
t2 IDT of 2 seconds

87

4.2.4.5 Default Dial Plan Rule 1:*72;>#x.etfxt2

This is the dial plan rule for Enable Unconditional Call Forwarding star service. Digits *72
must be dialed, followed by 15 optional digits and the optional # suffix. Each digit must be
dialed within 15 seconds of the preceding digit.
If only the *72# digits are dialed or the IDT elapses before dialing any digits after *72, the
default number configured on the Call Feature configuration page (Section 4.1.6) is sent.
After the *72 digits are dialed, press # to send the digits immediately, otherwise the digits
will be sent 15 seconds after the last dialed digit for digits 4-17 is sent, or 2 seconds after
digit 18 is dialed.

Dial Plan Rule Explanation of the Rules

1: Serial code for Enable Unconditional Call Forwarding followed by “:”

separator

*72; Dial-pad code for Enable Unconditional Call Forwarding followed by “;”

separator
># Position of >specifies 3 digits minimum; suffix is #
x.e Fourteen digits in range 0–9
tf IDT of 15 seconds
x One digit in range 0–9
t2 IDT of 2 seconds

Note: The default above rule states the number specified can have digits in the range 0–9
and can be a maximum length of 15 digits. This is flexible; for example, a Call Forwarding
rule could be specified as 1:*72;>#[78]x.6t4. In this case, Call Forwarding is done only to a
number beginning with either 7 or 8.

88

4.2.4.6 Default Dial Plan Rule 3:*74;>#x.etfxt2

This is the dial plan rule for Enable Unconditional Call Forwarding on Busy star service.
Digits *74 must be dialed, followed by 15 optional digits and the optional # suffix. Each digit
must be dialed within 15 seconds of the preceding digit.
If only the *74# digits are dialed or the IDT elapses before dialing any digits after *74, the
default number configured on the Call Feature configuration page (Section 4.1.6) is sent.
After the *74 digits are dialed, press # to send the digits immediately, otherwise the digits
will be sent 15 seconds after the last dialed digit for digits 4-17 is sent, or 2 seconds after
digit 18 is dialed.

Dial Plan Rule Explanation of the Rules

3: Serial code for Enable Unconditional Call Forwarding on Busy followed

by “:” separator
*74; Dial-pad code for Enable Unconditional Call Forwarding on Busy

followed by “;” separator
># Position of >specifies 3 digits minimum; suffix is #
x.e Fourteen digits in range 0–9
tf IDT of 15 seconds
x One digit in range 0–9
t2 IDT of 2 seconds

89

4.2.4.7 Default Dial Plan Rule 4:*75;>#x.etfxt2

This is the dial plan rule for Enable Unconditional Call Forwarding on No Answer star
service.
Digits *75 must be dialed, followed by 15 optional digits and the optional # suffix. Each digit
must be dialed within 15 seconds of the preceding digit. If only the *75# digits are dialed or
the IDT elapses before dialing any digits after *75, the default number configured on the
Call Feature configuration page (Section 4.1.6) is sent.
After the *75 digits are dialed, press # to send the digits immediately, otherwise the digits
will be sent 15 seconds after the last dialed digit for digits 4-17 is sent, or 2 seconds after
digit 18 is dialed.

Dial Plan Rule Explanation of the Rules

4: Serial code for Enable Unconditional Call Forwarding on No Answer

followed by “:” separator
*75; Dial-pad code for Enable Unconditional Call Forwarding on No Answer

followed by “;” separator
># Position of >specifies 3 digits minimum; suffix is #
x.e Fourteen digits in range 0–9
tf IDT of 15 seconds
x One digit in range 0–9
t2 IDT of 2 seconds

4.2.4.8 Default Dial Plan Rule 2:*73;>#t4

This is the dial plan rule for Disable Unconditional Call Forwarding star service.
Digits *73 must be dialed, each digit within 4 seconds of the preceding digit, optionally
followed by pressing #.

Note: The *73 code will disable all enabled Call Forwarding star services previously
enabled by the *72, *74, or *75 codes.
After the*73 digits are dialed, press # to send the digits immediately, otherwise there is a
4-second delay before the digits are sent.

Dial Plan Rule Explanation of the Rules

2: Serial code for Enable Unconditional Call Forwarding followed by “:”

separator
*73; Dial-pad code for Enable Unconditional Call Forwarding followed by “;”

separator
># Position of >specifies 3 digits minimum; suffix is #
t4 IDT of 4 seconds

90

4.2.4.9 Default Dial Plan Rule 11:*70;>#t4

This is the dial plan rule for Temporary Disable Call Waiting star service.
Digits *70 must be dialed, each digit within 4 seconds of the preceding digit, optionally
followed by pressing #.
After the*70 digits are dialed, press # to send the digits immediately, otherwise there is a
4-second delay before the digits are sent.

Dial Plan Rule Explanation of the Rules

11: Serial code for Temporary Disable Call Waiting followed by “:” separator
*70; Dial-pad code for Temporary Disable Call Waiting followed by “;”

separator
># Position of >specifies 3 digits minimum; suffix is #
t4 IDT of 4 seconds

4.2.4.10 Default Dial Plan Rule 12:*69;>#t4

This is the dial plan rule for Call Return star service.
The digits *69 must be dialed, each digit within 4 seconds of the preceding digit, optionally
followed by pressing #.
After the*69 digits are dialed, press # to send the digits immediately, otherwise there is a
4-second delay before the digits are sent.

Dial Plan Rule Explanation of the Rules

12: Serial code for Call Return followed by “:” separator
*69; Dial-pad code for Call Return followed by “;” separator
># Position of >specifies 3 digits minimum; suffix is #
t4 IDT of 4 seconds

91

4.2.4.11 Default Dial Plan Rule 16:*90;x>#x.dtfxt2

This is the dial plan rule for Blind Transfer star service.
Digits *90 followed by a single-digit number must be dialed, each digit within 15 seconds of
the preceding digit, followed by 14 optional digits and the optional # suffix.
After the *90x digits are dialed, press # to send the digits immediately, otherwise the digits
will be sent 15 seconds after the last digit for digits 4-17 is dialed, or 2 seconds after digit 18
is dialed.

Dial Plan Rule Explanation of the Rules

16: Serial code for Call Return followed by “:” separator
*90; Dial-pad code for Call Return followed by “;” separator
x One digit in range 0–9
># Position of >specifies 4 digits minimum; suffix is #
x.d Thirteen digits in range 0–9
tf IDT of 15 seconds
x One digit in range 0–9
t2 IDT of 2 seconds

4.2.4.12 Default Dial Plan Rule 18:*47;x>#[0-9*].f[0-9*].ft8[0-9*].ft4

This is the dial plan rule for IP Dialing star service.
Digits *47 followed by at least a single digit must be dialed, each digit within 8 seconds of
the preceding digit (up to the first 31 digits), optionally followed by pressing #.
After the *47<User URL> digits are dialed, press # to send the digits immediately, otherwise
there is a 4-second delay before the digits are sent. See Section 5.3.10 for more clarity on
how to enter user numbers, IP address and Port using dial pad.

Dial Plan Rule Explanation of the Rules

18: Serial code for IP Dialing followed by “:” separator
*47; Dial-pad code for IP Dialing followed by “;” separator
x One digit in range 0–9
># Position of >specifies 1 digit minimum; suffix is #
[0-9*].f Fifteen digits in range 0–9 or *
[0-9*].f Fifteen digits in range 0–9 or *
t8 IDT of 8 seconds
[0-9*].f Fifteen digits in range 0–9 or *
t4 IDT of 4 seconds

92

4.2.4.13 Default Dial Plan Rule 19:*78;x>#t4

This is the dial plan rule for speed dialing an entry in the address book.
Digits *78 followed by a single digit must be dialed, each digit within 4 seconds of the
preceding digit, optionally followed by pressing #.

Note: The single digit is an index into the address book maintained by the system. The
required address book entries must be configured according to section.
After the *78x digits are dialed, press # to send the digits immediately, otherwise there is a
2-second delay before the digits are sent.

Dial Plan Rule Explanation of the Rules

19: Serial code for Speed dial followed by “:” separator
*78; Dial-pad code for Speed Dialing followed by “;” separator
x One digit in range 0–9
># Position of >specifies 4 digits minimum; suffix is #
t4 IDT of 4 seconds

4.2.4.14 Default Dial Plan Rule 20:#;x.3tf>#x.atfxt2

This is the rule for dialing a PSTN call.
Digit # followed by three single-digit numbers, each digit within 15 seconds of the preceding
digit, optionally followed up to 11 optional digits, and optionally followed by pressing #.
After the four required digits are dialed, press # to send the digits immediately, otherwise
the digits will be sent 15 seconds after the last digit for digits 4-13 is dialed, or 2 seconds
after digit 14 is dialed.

Dial Plan Rule Explanation of the Rules

20: Serial code for PSTN call followed by “:” separator
#; Dial-pad code for PSTN call followed by “;” separator
x.3 Three digits in range 0–9
tf IDT of 15 seconds
># Position of >specifies 4 digits minimum; suffix is #
x.a Ten digits in range 0–9
tf IDT of 15 seconds
x One digit in range 0–9
t2 IDT of 2 seconds

93

4.2.4.15 Default Dial Plan Rule 22:*83;>#t4

This is the dial plan rule for Disabling 3-Way Conferencing temporarily.
Digits *83 must be dialed, each digit within 4 seconds of the preceding digit, optionally
followed by pressing #. No number needs to be dialed.
After the *83 is dialed, press # to send the rule immediately, otherwise there is a 4-second
delay before the rule is executed.

Dial Plan Rule Explanation of the Rules

22: Serial code for Temporary Disable of 3-Way Conferencing followed by “:”

separator
*83; Dial-pad code for Disabling 3-Way Conferencing followed by “;”

separator
># Position of >specifies 4 digits minimum; suffix is #
t4 IDT of 4 seconds

4.2.4.16 Default Dial Plan Rule 23:*76;>#t4

This is the dial plan rule for Disable Call Forwarding on Busy star service.
Digits *76 must be dialed, each digit within 4 seconds of the preceding digit, optionally
followed by pressing #. No number needs to be dialed.
After the *76 is dialed, press # to send the rule immediately, otherwise there is a 4-second
delay before the rule is executed.

Dial Plan Rule Explanation of the Rules

23: Serial code for Disable Call Forwarding on Busy followed by “:” separator
*76; Dial-pad code for Disable Call Forwarding on Busy followed by “;”

separator
># Position of >specifies 3 digits minimum; suffix is #
t4 IDT of 4 seconds

94

4.2.4.17 Default Dial Plan Rule 24:*77;>#t4

This is the dial plan rule for Disable Call Forwarding on No Answer star service.
Digits *77 must be dialed, each digit within 4 seconds of the preceding digit, optionally
followed by pressing #. No number needs to be dialed.
After the *77 is dialed, press # to send the rule immediately, otherwise there is a 4-second
delay before the rule is executed.

Dial Plan Rule Explanation of the Rules

24: Serial code for Disable Call Forwarding on No Answer followed by “:”

separator
*77; Dial-pad code for Disable Call Forwarding on No Answer followed by “;”

separator
># Position of >specifies 3 digits minimum; suffix is #
t4 IDT of 4 seconds

95

4.2.4.18 Default Dial Plan Rule 25:N1t41;>#

This is the dial plan rule for Emergency Call. However, this rule also handles other N11
services.
One digit in range 2–9 followed by the digit 1 and the digit 1 must be dialed, each digit
within 4 seconds of the preceding digit.
After digits N11 are dialed, the digits are immediately sent.

Dial Plan Rule Explanation of the Rules

25: Serial code for Emergency Call followed by “:” separator
N1t41; Dial-pad code for Emergency Call followed by “;” separator

N One digit in range 2–9

1 One digit 1

t4 IDT of 4 seconds

1 One digit 1
># Position of >specifies 3 digits minimum; suffix is # (# is included because

a character must be provided after >, however, pressing # is not an

option because the dialed digits are sent immediately)

Notes:

1. Digits N11 match the [0-9*]>#[0-9*].e[0-9*].ft4 dial plan, except the 25:N1t41;># dial plan
rule allows a 911 call will be sent without delay.

2. This emergency rule is not part of the default service provider’s dial plan string as Pulver
does not support 911 calls (see Appendix B).

3. Any number of such emergency rules can be specified in the dial plan string. In such
emergency call match cases, the dial plan interpreter will not wait for any timeout and will
send out the matched digits immediately.

96

4.2.4.19 Default Dial Plan Rule 26:*67;>#t4

This is the dial plan rule for Temporary Enable Caller ID Transmission Block.
Digits *67 must be dialed, each digit within 4 seconds of the preceding digit, optionally
followed by pressing #. After the*67 digits are dialed, press # to send the digits immediately,
otherwise there is a 4-second delay before the digits are sent.

Dial Plan Rule Dial Plan Rule Function
26: Serial code for Emergency Call followed by “:” separator
*67; Dial-pad code for Temporary Enable Caller ID Transmission Block

followed by “;” separator
># Position of >specifies 1 digit minimum; suffix is #
t4 IDT of 4 seconds

4.2.4.20 Default Dial Plan Rule [0-9*]>#[0-9*].e[0-9*].ft4

This is the dial plan rule for Default Number Call, which applies if no previous rule matches.
This rule takes a minimum of 1 digit and a maximum of 30 digits. All dialed digits will be sent.
The service provider decides what to do with the dialed digits. All digits dialed can be
between 0 through 9, or *.
After the first digit is dialed, press # to send the digits immediately, otherwise the digits will
be sent 4 seconds after the last dialed digit.

Dial Plan Rule Dial Plan Rule Function
[0-9*] One digit in range 0–9 or *
># Position of >specifies 1 digit minimum; suffix is #
[0-9*].e Fourteen digits in range 0–9 or *
[0-9*].f Fifteen digits in range 0–9 or *
t4 IDT of 4 seconds

Notes:

1. If this default call rule is removed from the dial plan string, then any sequence of digits
which does not match any of the other rules will be blocked by the ATA i.e., the dialed digits
will not be sent to the service provider.

2. The default call rule is a superset of all the other dial plan rules. The dial plan interpreter
works on a first-come first-serve basis. Ensure that this rule is included and is always
specified at the end of the dial plan string.

97

4.2.5 Empty Dial Plan String

If a default dial plan string is not required, the Default Dial Plan String field on the General
configuration page (Section 4.1.2) can be left empty, in which case the default dial pattern
to accept all dialed digits will be incorporated.
The default dial pattern, [0-9*]>#[0-9*].e[0-9*].ft4, is transparent to the user and will not be
displayed on the General configuration page (see Section 4.1.2).
Please note that when the dial plan string is left empty, there will no star services support
available locally at ATA. Since the default dial plan rule sends all collected digits to the
service provider, it is expected that the service provider handles the star services using his
dial plan.

4.2.6 Incorrect Dial Plan Rule

If the user enters a syntactically incorrect dial plan rule, the rule is rejected while parsing
but the rule is not removed from the dial plan string. In other words, syntactic analysis is not
performed during the parsing stage after the Save and Reboot, but after parsing the dial
plan string, the invalid dial plan rule is skipped.
If all the dial plan rules are invalid, the default dial pattern is used. This default dial pattern,
[0-9*]>#[0-9*].e[0-9*].ft4, is transparent to the user and will not be displayed on the General
configuration page (see Section 4.1.2).

98

5. Using Conexant ATA

5.1 Setting up ATA for VoIP Calls

This section describes the setup and configuration needed before the ATA can be used.

1. Ensure the ATA software has been flashed onto the VoIP reference board.

2. When power is applied to the board, the system initializes with the default configuration
which is not initially set up to make VoIP calls.

3. The configured default service provider is Pulver. You can either use Pulver or configure
a new service provider. See Section 4.1.3 for adding or selecting a new service provider.

4. If you choose to use Pulver, then a free login account must be obtained for each line from
the website http://www.fwd.pulver.com

.

5. After the login accounts have been obtained, configure the login account information for
the service provider you intend to use. See Section 4.1.4 for adding and selecting the login
account to use.

6. The other default parameters are sufficient enough to start making and receiving calls.

7. After you have saved your changes, Registration status on the ATA Configuration
page (see Section 4.1) will be displayed as REGISTERED if you have chosen to go through
a service provider.

Note: If your system has PSTN backup hardware, you must have configured vophwcfg.ini
for PSTN backup (see Appendix C.2).

5.2 Making Basic Calls

Follow these steps to make an outgoing phone call:

1. Pick up the handset.

2. When you hear a dial tone, dial the number you want to reach.

3. Press # to send the number immediately, or wait for the IDT to expire and the number will
be sent automatically.

Note: The ATA supports Mid-Call DTMF in order to call toll-free numbers and other
interactive voice response (IVR) systems. ATA is configured automatically during call
establishment to support RFC 2833-based dynamic payload negotiation and to send DTMF
tones to the other end point.

99

5.3 Advanced Call Features

The supplementary services described in this section, and their configuration and
implementation, depend on the system of the country in which the service is activated.
This section describes the following topics:

• Caller ID

• Call-Waiting Caller ID

• Consultation Hold

• Blind Transfer

• Attended Transfer

• 3-Way Conferencing

• Call Waiting

• Call Forwarding

• Call Return

• IP Dialing

• Speed Dialing

• Caller ID Transmit Blocking

Notes:

1. Web configuration overrides any dial-pad based configuration, and will be reflected only
upon Save and Reboot.

2. The digit sequences to be dialed for different actions to be executed are based on the
default dial plan rules specified in this document.

3. Some features require that a hook flash be performed. To perform a hook flash, quickly
press and release the switchhook (push button) on the phone cradle or quickly press and
release the flash button or other dedicated key/button on the telephone base unit or
handset, depending on the telephone.

Be careful not to hold the switchhook or flash button down too long, e.g., for three seconds,
which will disconnect your call. In this case, when you release the switchhook or flash
button, you will hear a regular dial tone.

5.3.1 Caller ID

When the telephone rings, the ATA sends a Caller ID signal to the telephone between the
first and second ring (with name, telephone number, time, and date information, if these are
available).

100

5.3.2 Call-Waiting Caller ID

The ATA plays the first Call Waiting tone, and then sends the name, telephone number,
time, and date information, if these are available, before the second Call Waiting tone is
played.

5.3.3 Consultation Hold

This feature allows a user to put the existing call on hold and call another number.

How to Use:

1. During an existing call, perform a hook flash to put the other party on hold and to get a
dial tone.

2. When you hear the dial tone, dial another number.

3. When you are finished with the second call, perform a hook flash to revert to the first
party. You can also alternate between calls by performing a hook flash.

5.3.4 Blind Transfer

This feature allows a user (transferor) to transfer an existing call to another telephone
number (transfer target) without connecting to the transfer target number.

How to Use:

1. During an existing call, perform a hook flash to put the other party on hold and get a dial
tone.

2. When you hear the dial tone, press *90 on your telephone dial-pad.

3. When you hear the alert tone indicating that the ATA is expecting a number, dial the
phone number to which you want to transfer the other party, then press # (optional).

4. When you hear the Fast Busy Tone, hang up your phone. Note: This dial-pad code can
be changed as part of the dial plan rules which can be modified for the service provider on
the SIP Service Provider configuration page as described in Section 4.1.3.