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X6v VoIP
User Manual
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Zoom X6v VoIP User Manual

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Page 1
X6v VoIP
Features
TECHNICAL REFERENCE
Page 2
Page 3
Contents
1 INTRODUCTION.....................................................................................................................................................5
2 CHANGING CONFIGURATION PARAMETERS...............................................................................................7
3 SYSTEM PARAMETERS ......................................................................................................................................8
4 VOIP ACCOUNTS.................................................................................................................................................13
5 VOIP PARAMETERS...........................................................................................................................................19
6 SIP PARAMETERS ..............................................................................................................................................21
7 REGIONALIZATION.............................................................................................................................................24
Call Progress Tone Parameters.....................................................................................................................25
Standard Ringing Patterns Parameters.........................................................................................................27
Distinctive Ringing Patterns Parameters.......................................................................................................28
Distinctive Call Waiting Patterns Parameters...............................................................................................28
Voice, Tone and DTMF Parameters..............................................................................................................29
SLAC Configuration Parameters....................................................................................................................30
Values for Port Impedance (SLAC & CODEC).............................................................................................30
SLAC Command Strings..................................................................................................................................31
Table of Contents 3
Page 4
8 SUBSCRIPTION SERVICES ..............................................................................................................................34
Examples of Dial Strings..................................................................................................................................39
North American Number Plan Area (NANPA) Dialing Examples...............................................................40
Dial String Tips..................................................................................................................................................40
Entering Easily-Confused Patterns................................................................................................................41
9 USER CONFIGURATION....................................................................................................................................44
10 FEATURE CODES .............................................................................................................................................49
4 X6v VoIP Features Technical Reference
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1
Introduction
This document describes the ADSL X6v modem's VoIP features. It provides information about the VoIP configuration parameters and explains how to view and modify them using the Configuration Manager interface or by downloading configuration files via the VoIP Subsystem's update mechanism.

Using the Configuration Manager

The Configuration Manager is the interface to the ADSL X6v modem. To access the interface:
http://192.168.0.1 in your browser's address field.
Type
1 2 When prompted, log on in administrator mode, using the following Username and Password:
Username: admin Password: zoomadsl
Note to service providers: If you are going to lock units to your service, we strongly recommend that
you change the password before shipping any product to the field. See
6 for instructions.
page
Changing the admin Password on
When the ADSL Setup page opens, click the VoIP icon on the Zoom menu bar to access the VoIP
3
Subsystem.
Click the Advanced VoIP Setup icon, then select VoIP System from the left pane's menu to access the
4
configuration parameter categories.
Select items from the VoIP System menu to view or modify the parameters within these groups:
5
System Parameters
•
VoIP Accounts
•
VoIP Parameters
•
SIP Parameters
•
Regionalization
•
Subscription Services
•
User Configuration
•
Feature Codes
•
Chapter 1: Introduction 5
Page 6

Downloading Configuration Files

Configuration files are prepared and stored on the service provider’s update server. At power up, reboot, or configurable periodic intervals, the VoIP Subsystem can contact an update server. When it contacts the update server, the VoIP Subsystem provides unique identification. The update server then checks a database to determine whether there is new firmware and/or a configuration file for the VoIP Subsystem. If there is, the update server instructs the VoIP Subsystem to download the relevant file or files. The configuration server can use the VoIP Subsystem’s device identification to prepare a specific configuration file that might include, for example, detailed account information.

Changing the admin Password

To change the admin password:
http://192.168.0.1 in your browser's address field.
Type
1 2
When prompted, log on in administrator mode:
Username: admin Password: zoomadsl
3
When the ADSL Setup page opens, click the Router Setup icon on the Zoom menu bar.
4
On the Router Setup page, click Admin Password.
5
On the Admin Password Configuration page, type the old and new passwords, then confirm the change.
6
Click Save.
7
When the authentication dialog opens, type the new password in the Password field, then click OK.
8
Click Write Settings to Flash.
Important!
If you change the admin password, and then forget the new password, you cannot retrieve it. You will need to reset the unit to the factory default settings which will erase any previously saved (changed) settings.
6
X6v VoIP Features Technical Reference
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2
Changing Configuration
Parameters
As administrator (admin), you can view and modify the VoIP configuration parameter values described in
this Technical Reference and set user access privileges for each parameter. See Chapters 3 through 10
for a description of the available menus and configurable parameters.

Setting User Access Privileges

When you are logged on as admin, the VoIP interface displays a pull-down menu labeled User to the right
of each configurable parameter. The pull-down menu values are E, P, V and
choose defines user access privileges for each field.
- (dash). The value that you
Note: Each account page has only one pull-down menu that controls access for all fields on that
page. On some pages, there are additional pull-down menus to the right of the User fields. These menus are labeled Phone and they control access to features (setting up speed dials, call blocking based on caller ID, etc.) that can be activated using a handset. For the Phone pull-downs, only the
symbols E and
Value
E
P
V
-
- (dash) are available.
Description
Full Edit capabilities. Read, write, delete.
Full Edit with Priority. Cannot be overwritten by config download via update server
View. Read only.
No access. (This value is not seen by the user.)
Chapter 2: Changing Configuration Parameters 7
Page 8
3
System Parameters
You can use the VoIP -> Advanced VoIP Setup -> VoIP System menu to configure overall system settings. The
menu items include:

VoIP System Identification

•

Date/Time

•
VoIP Subsystem Network Configuration
•
Static IP/DNS Configuration
•
HTTP / Telnet / FTP Server
•
STUN Settings
•
Firmware and Configuration Update Settings
•
VoIP System Maintenance
•
VoIP System Identification
Parameter Description Default
Boot ROM Revision
Firmware Revision
Configuration Revision
MAC Address

Note: Default revisions will vary according to the release date of your product. Configuration suffixes vary by region

Boot code revision 6.3.1
Run-time code revision 6.3.1
Configuration file revision 6.3.1 – 00/70/72
Ethernet MAC address assigned during manufacture
(as assigned)
Date/Time
Parameter Description Default
Date (yyyy/mm/dd)
Time (23:59:59)
Time Zone (rel. GMT; -12 to 13)
Daylight Savings
Obtain Time from NTP Timeserver
Current date
Current time
Number of hours to subtract from GMT to form local time
Enable or disable local application of daylight savings time
Enable or disable use of network timeserver
-5
Enabled
Enabled
8 X6v VoIP Features Technical Reference
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Parameter Description Default
NTP IP Address
Note: When the date and time are set independently of NTP (that is, if a timeserver is unavailable or the
use of a timeserver is disabled), adjustments must be made to a time at least one hour ahead or behind the currently displayed time, to prevent errors related to the internal workings of the time system.
Fully qualified domain name (FQDN) (including an optional port number) for the NTP/SNTP timeserver server
time-a.nist.gov

VoIP Subsystem Network Configuration

Parameter Description Default
Manually configured VoIP subsystem startup delay. This parameter configures the VoIP subsystem to delay the indicated time before booting up. Normally there is
VoIP Startup Delay (ms)
VoIP Name
VoIP Host Name
VoIP Domain Name
MTU
no need to set it. If the VoIP subsystem has trouble registering at power up, you might set this delay to allow your X6v sufficient time to establish a DSL connection before the VoIP subsystem attempts to register.
Manually configured VoIP subsystem device name.
Manually configured host device name (or name automatically assigned and saved).
Manually configured domain name.
Manually configured maximum transmit unit size (range of 576 to 1500).
Note: the MTU setting is prepared for the
use of PPPoE. Some system configurations require an MTU setting of
1500.
0
ZOOM_VoIP
ZOOM_VoIP
1492

Static IP/DNS Configuration

Parameter Description Default
Static IP Address
Subnet Mask
Gateway IP Address
Primary DNS Address
Chapter 3: System Parameters 9
Manually configured IP address (or address automatically assigned and saved)
Manually configured local network mask (or netmask automatically assigned and saved)
Manually configured gateway IP address (or address automatically assigned and saved)
Manually configured IP address of primary domain name server (DNS)
192.168.0.234
255.255.255.0
192.168.0.1
192.168.0.1
Page 10
Note: You must change the VoIP Subsystem IP/DNS configuration settings first to the new subnet if you
change the LAN subnet of your X6v.

HTTP / Telnet / FTP Server

Parameter Description Default
HTTP Server Access Enable
HTTP Server Port
Telnet Server Enable
Telnet Server Port
FTP Server Enable
FTP Server Port
Note: External access may be blocked by your X6v firewall.
Enable or disable access to Configuration Manager
Assigned port number for HTTP server 8080
Enable or disable remote access via telnet
Assigned port number for Telnet server 8023
Enable or disable remove access via ftp Enabled
Assigned port number for FTP server 8021
Enabled
Enabled

STUN Settings

Parameter Description Default STUN Enable
STUN Server Address
Enables or disables use of STUN for
discovery of Network Address Translation (NAT) mapping
Fully qualified domain name (including optional port number) for the STUN server
Enabled

Firmware and Configuration Update Settings

Parameter Description Default
Update Server Domain Name
Automatic Configuration Update Enable
Automatic Configuration Update on Reboot
Automatic Configuration Update (SIP)
Configuration Update Message on Request
Configuration Update Message on Success
Configuration Update Message on Failure
Fully qualified domain name (including an optional port number) for the update server
Control to enable automatic updating of configuration
Control to enable automatic update of configuration on reset
Control to enable automatic update on receipt of SIP message
SYSLOG message body sent when requesting a configuration update
SYSLOG message body sent when configuration update completed successfully
SYSLOG message body sent when configuration update completed unsuccessfully
zoom.voipconfigure.com: 5080
Enabled
Enabled
Disabled
Configuration update requested
Configuration update successful
Configuration update failed
10 X6v VoIP Features Technical Reference
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Parameter Description Default
Configuration Update Periodic Delay(s)
Configuration Update Random Delay(s)
Configuration Update Error Retry Delay(s)
Automatic Firmware Update Enable
Automatic Firmware Update on Reboot
Firmware Update Message on Request
Firmware Update Message on Success
Firmware Update Message on Failure
Firmware Update Periodic Delay(s)
Firmware Update Random Delay(s)
Firmware Update Error Retry Delay(s)
Periodic delay between configuration update checks (in seconds - limit 4,294,967,296)
Uniform random delay applied when contact with the update server fails
Fixed delay applied when the configuration update operation fails
Control to enable automatic updating of firmware
Control to enable automatic update of firmware on reset
SYSLOG message body sent when requesting a firmware update
SYSLOG message body sent when firmware update completed successfully
SYSLOG message body sent when firmware update completed unsuccessfully
Periodic delay between firmware update checks (in seconds - limit 4,294,967,296)
Uniform random delay applied when contact with the update server fails (in seconds)
Fixed delay applied when the firmware update operation fails (in seconds)
76400
240
120
Enabled
Enabled
Firmware update successful
Firmware update failed
86400
240
120
Note: The configuration and/or firmware update periodic delay is by default about a day. This can be
changed to a week by specifying 604,800 seconds, or a month by specifying 2,620,800 seconds.

VoIP System Maintenance

Parameter Description Default
Syslog Enable
Syslog Server Address
Debug Enable
Debug Server Address
Debug Level ATA
Chapter 3: System Parameters 11
Enable or disable transmission of SYSLOG messages
Fully qualified domain name (including an optional port number) for the SYSLOG server
Enable or disable transmission of Debug messages
Fully qualified domain name (including an optional port number) for the Debug server
VoIP Subsystem debug 0
Disabled
Disabled
Page 12
Parameter Description Default Debug Level SIP
Debug Level Net
Debug Level PMP
Session Initiation Protocol debug
0
Network debug 0
Port Mapping Protocol debug 0
12 X6v VoIP Features Technical Reference
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4
VoIP Accounts
You can use the VoIP > Advanced VoIP Setup > VoIP Accounts menu to configure user accounts for up to four
providers. The menu items include:
•
My VoIP Account Accounts 2, 3, and 4
•

Logging in to the Configuration Manager

To access the VoIP menus, you must log in to the Configuration Manager.
http://192.168.0.1 in your browser's address field.
Type
1 2
When prompted, log on in administrator mode, using the following Username and Password:
Username: admin Password: zoomadsl
3
When the ADSL Setup page opens, click the VoIP icon on the Zoom menu bar to access the VoIP Subsystem.
4
Click the Advanced VoIP Setup icon, then select VoIP Accounts to view or modify parameters.
Notes to service providers:
If you are going to lock units to your service, we strongly recommend that you change the admin
password before shipping any product to the field. See Changing the admin Password on page 6 for
instructions.
As an added precaution, we recommend that you also change the VoIP subsystem password. Please refer to the deployment package for details.

Setting User Privileges

You may set access to account information for the user level login (see Setting User Access Privileges, on page My VoIP Account, and allow full access (privilege E) to accounts 2, 3 and 4. Alternatively, you may want to hide access to all four accounts.
On each of the account pages there is a column of priority settings on the right-hand side. The top setting determines access for that page as a whole. The remaining settings determine the privileges of the individual parameters that they control.
There is a limitation in the implementation of the privileges of the individual parameters. These must all be the same for all four accounts. Thus, you should set the individual parameters to support the level of access you wish to grant for the account(s) with the most open access. You may restrict the access to other accounts by choosing an appropriate value for the top level setting that controls those pages.
7). For example, you may wish to hide (privilege -) or to make read-only (privilege V) access to
Chapter 4: VoIP Accounts 13
Page 14
The VoIP Express Setup page is affected by settings on the My VoIP Account page. Six parameters on this page are drawn from the My VoIP Account page, Turn My VoIP Service (On/Off), and the five parameters beginning My …. The user is granted the same access to these parameters through the Express page as through the My VoIP Account page. (The VoIP Express Setup page offers control or view of a subset of settings that are appropriate for many users).
Note: In some fields you might see default values that were used in Zoom’s manufacturing test
procedures. You can safely ignore or delete these values.

My VoIP Account

Parameter Description Default
Turn My VoIP Service
My VoIP Providers Name
My Caller ID When I Call Someone
My VoIP Phone Number (SIP User ID)
My VoIP Service Authorization ID
My VoIP Service Authorization Password
SIP Server
Enables (On) or disables (Off) this account
Name of VoIP provider
Holds an identifier (name or number) that can be displayed at the receiving party’s phone when someone makes a call from the VoIP Subsystem to another SIP phone. When someone makes a call from the VoIP Subsystem that terminates on the PSTN, this ID will generally not display on the receiving party’s phone.
Specifies the name to be used when logging in to the service provider’s server. Commonly implemented in the form of an E.164 number. (E.164 is the ITU recommendation for standard telephone number format.) This ID/number will often appear on the receiving party’s phone as the Caller ID when someone places a call from the VoIP Subsystem.
User name for authentication
User password for authentication
Identifies the SIP Server (Format: FQDN)
On
14 X6v VoIP Features Technical Reference
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Parameter Description Default
Auth Domain
Specifies the authentication domain name corresponding to the
Authentication User Name. This
field must match the authentication realm URL assigned by the service provider. It must NOT be translated into any dotted-decimal address equivalent. For many service providers, this can be left blank as SIP messages in the registration process will convey the authentication domain name. (Format: FQDN)
Outbound Proxy
Identifies the outbound proxy server and port, or if the provider doesn’t use an outbound proxy server, the default SIP proxy server and port to be used when making outgoing calls. (Format: FQDN)
Register Domain
Identifies the default SIP registration server name and port used to identify the VoIP Subsystem device providing the service end-point for the assigned subscription service. (Format: FQDN)
ReReg Interval (s)
Sets the default registration update
120 period in seconds. The VoIP Subsystem must re-register before this period expires to prevent service interruption.
Subscribe Domain
Fully qualified domain name (with optional port number) for the SIP registration server. (Format: FQDN)
ReSub Interval (s)
Use Outbound Proxy for REFER
Re-subscription interval in seconds 1800
Enables or disables the use of an
Disabled outbound proxy for SIP service remote call transfers
DNS Server Lookup for SIP Server
Enables or disables DNS Server
Disabled lookup services for the SIP server
Ring Type
Selects a distinctive ring type for the
1 account.
Chapter 4: VoIP Accounts 15
Page 16
Parameter Description Default
Dial Prefix
Preferred Codecs
Contains the dial string pattern matching used to distinguish and route calls to a VoIP service provider.
The default for My VoIP Account is null (that is, all calls are routed via this account, unless preceded by a prefix defined for accounts two through four).
Accounts 2 through 4 can be configured with prefixes that are used to invoke these accounts. The dial string pattern match is in the standard form. Prefix strings of #8, #9, 8 and 9, if specified, are automatically removed from the dialed number. Other prefixes can be altered through the substitution flexibilities of the pattern matching strings.
Allows listing, in order of preference, the Codec code points preferred for use with the service provider.
Menu options are: G.711u, G.711A, G.729B, and iLBC.
The codecs listed here must also
be included in the list under Audio Settings on the VoIP Parameters page.
If any codecs are listed here, then only those codecs will be negotiated. If no codecs are listed here, then all Preferred Codecs options will be negotiated.
Preferred Codecs

Accounts 2, 3, and 4

Parameter Description Default
Turn My VoIP Service
My VoIP Provider Name
16 X6v VoIP Features Technical Reference
Enables or disables this account Disabled
Name of VoIP provider
Page 17
Parameter Description Default
My Caller ID When I Call Someone
Holds the number that can be displayed at the receiving party’s phone when the user makes a call from the VoIP Subsystem to another SIP phone. When the user makes a call from the VoIP Subsystem that terminates on the PSTN, this name will generally not display on the receiving party’s phone.
My VoIP Phone Number (SIP User ID)
Specifies the name to be used when logging in to the service provider’s server. Commonly implemented in the form of an E.164 number. This ID/number will often appear on the receiving party’s phone as the Caller ID when someone places a call from the VoIP Subsystem.
My VoIP Service Authorization ID
My VoIP Service Authorization
User name for authentication
User password for authentication
Password
SIP Server
Identifies the SIP Server. (Format: FQDN)
Auth Domain
Specifies the authentication domain name corresponding to the user's
Authorization ID. This field must
match the authentication realm URL assigned by the service provider. It must NOT be translated into any dotted-decimal address equivalent. For many service providers, this can be left blank as SIP messages in the registration process will convey the authentication domain name. (Format: FQDN)
Outbound Proxy
Identifies the outbound proxy server and port, or if the provider doesn’t use an outbound proxy server, the default SIP proxy server and port to be used when making outgoing calls. (Format: FQDN)
Register Domain
Identifies the default SIP registration server name and port used to identify the VoIP Subsystem device providing the service end-point for the assigned subscription service. (Format: FQDN)
Chapter 4: VoIP Accounts 17
Page 18
Parameter Description Default
ReReg Interval (s)
Sets the default registration update
120 period in seconds. Once the period has expired, the VoIP Subsystem must re-register to prevent service interruption.
Subscribe Domain
Fully qualified domain name (with optional port number) for the SIP registration server. (Format: FQDN)
ReSub Interval (s)
Ring Type
Re-subscription interval in seconds 120
Selects a distinctive ring type for the account.
2 for Account 2
3 for Account 3
4 for Account 4
Dial Prefix
Contains the dial string pattern matching used to distinguish and route calls to a VoIP service provider.
The default is null (i.e., all calls are routed via this account, unless preceded by a prefix defined for accounts two through four).
Accounts 2, 3, and 4 can be configured with prefixes that are used to invoke these accounts. The dial string pattern match is in the standard form. Prefix strings of #8, #9, 8 and 9, if specified, are automatically removed from the dialed number. Other prefixes can be altered through the substitution flexibilities of the pattern matching strings.
Preferred Codecs
Allows listing, in order of preference, the Codec code points preferred for use with the service provider.
Menu options are: G.711u, G.711A, G.729B, and iLBC.
The codecs listed here must also
be included in the
Preferred Codecs
list under Audio Settings on the VoIP
Parameters page.
If any codecs are listed here, then only those codecs will be negotiated. If no codecs are listed here, then all Preferred Codecs options will be negotiated.
18 X6v VoIP Features Technical Reference
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5
VoIP Parameters
You can use the VoIP > Advanced VoIP Setup > VoIP Parameters menu to configure various common aspects of the VoIP Subsystem device. The menu items include:

Audio Settings

•
RTP Protocol Parameters
•
SDP Protocol Parameters
•
SDP Audio Codec Names
•
Audio Settings
Parameter Description Default
Preferred Codecs
Silence Suppression Enable
Echo Canceller Enable
Echo Canceller Mode
Echo Canceller Tail Length (ms)
Fax Transmission Mode
DTMF Transmission Method
iLBC High Rate Enable
Lets you arrange the Codec names in order of preference. These entries must agree with the on the My VoIP Account page.
Prevents audio frames from being sent during periods of silence, thus reducing the network traffic necessary for making calls. (Note: This feature is useful only with audio codecs that support silence suppression.)
If enabled, the G.168 echo canceller is applied to all calls.
Sets the echo canceller operating mode.
Specifies length of echo canceller in msec 16
Control for FAX processing method: Off, or Passthrough (μLaw or ALaw)
Control for DTMF processing method: Off, Audio Passthrough, RTP Out-of-band, SIP Out-of-band
Enables 15.2 kbps / 20 ms frames. When disabled, 13.33 kbps / 30 ms frames. Many implementations negotiate 13.33 kbps / 30 ms only.
Preferred Codecs specified
G.711u, iLBC, G.729B, G.711A
Disabled
Enabled
Do not change the setting, which is 2.
Off
RTP Out-of-band
Disabled
Chapter 5: VoIP Parameters 19
Page 20

RTP Protocol Parameters

Parameter Description Default
Base RTP port (1024-65535)
Maximum RTP port (1024-65535)
RTP Public External IP Address
RTP Public External Port
RTP TOS Value (0x00-0xff)
RTP Packet Duration (ms)
RTP Stream Duration (ms)
RTP Session Timeout Interval (s)
RTP Jitter Buffer Start Depth (ms)
RTP Jitter Buffer Minimum Depth (ms)
The minimum IP port number for RTP traffic. Can be used in conjunction with firewall mappings.
The maximum IP port number for RTP traffic.
Forces a specific external IP address as the source address for SDP messages that the VoIP Subsystem sends.
Specifies the RTP port associated with the minimum RTP port number in a NAT firewall that performs fixed port mapping.
Type of service (TOS) value or DIFFServ DSCP used for RTP (audio) packets.
The duration (in milliseconds) for frame­based codecs
The duration (in milliseconds) for sample stream-based codecs
The session timeout interval (in seconds) 120
The start depth (in milliseconds) of the buffer
The minimum depth (in milliseconds) of the buffer
1234
1253
0.0.0.0
0 (Disabled)
68 (Assured Forwarding)
30
20
20
20

SDP Protocol Parameters

Parameter Description Default
SDP Session Name
SDP Session Owner
Identifies the session name. -
Identifies the session owner. Zoom

SDP Audio Codec Names

These parameters are passed to the remote end-point for outgoing calls only.
Parameter Description Default
G711u Codec (PCMU/8000)
G711A Codec (PCMA/8000)
G729b Codec (G729B/8000)
iLBC/Codec (iLBC/8000)
The string passed during outgoing calls to negotiate the payload type for G.711 μLaw
The string passed during outgoing calls to negotiate the payload type for G.711 ALaw
The string passed during outgoing calls to negotiate the payload type for G.729B
The string passed during outgoing calls to negotiate the payload type for iLBC
PCMU/8000
PCMA/8000
G729B/8000
iLBC/8000
20 X6v VoIP Features Technical Reference
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SIP Parameters
You can use the VoIP > Advanced VoIP Setup > SIP Parameters menu to configure particular aspects of the Session Initiation Protocol (SIP) implementation. The menu items include:
SIP Protocol Parameters
•
SIP Response Codes
•
SIP Distinctive Ring Names
•
SIP Protocol Timers
•

SIP Protocol Parameters

Parameter Description Default
SIP Require User Name
SIP Local Port (1024-65535)
SIP Public External IP Address
SIP Public External Port
TOS Value (0x00 – 0xff)
SIP Accept Language String
SIP Send Response to SRC Port
SIP Max Forwards
SIP Ringing Retransmit
SIP Use NAT Discovery
SIP Use Received Via Info
Enables or disables a requirement that an incoming INVITE include a SIP user name assigned to the VoIP subsystem in an active account.
Local UDP port used for sending/ receiving SIP call control messages. This port can be mapped by a firewall.
Forces a specific external IP address for SIP messages sent
Forces a specific external UDP port for SIP messages sent
Type of service (TOS) value or DIFFServ DSFIELD used for SIP message
Specifies the language for user-viewable messages used in the SIP accept message
Respond to the sender’s IP address/UDP port used by SIP request message
Maximum forward value 15
Enables or disables retransmission Enabled
Enable use of NAT discovery procedures to obtain an external IP address/UDP port mapping for SIP messages
Use VIA header IP address/UDP port parameters in received messages as external IP address/UDP port
Disabled
5060
0.0.0.0 (Disabled)
0 (Disabled)
68 (DIFFSRV Expedited Forwarding)
English
Enabled
Enabled
Disabled
6
Chapter 6: SIP Parameters 21
Page 22
Parameter Description Default
NAT Keep Alive Enable
NAT Keep Alive Interval (s)
NAT Keep Alive Domain Name
NAT Keep Alive Message
Send periodic SIP messages to keep port mapping active
Periodic interval for SIP keep alive messages (in seconds)
Fully qualified domain name (including an optional port number) for the destination of SIP keep alive message (sends to the proxy server if blank)
Type of message to be sent as SIP keep alive: empty, notify or register
Enabled
15

SIP Response Codes

Parameter Description Default
SIP Response Code SIT1
SIP Response Code SIT2
SIP Response Code SIT3
SIP Response Code SIT4
SIP Response Code Try Backup
SIP Response Code Retry Registration
SIP response code which plays the SIT1 tone sequence
SIP response code which plays the SIT2 tone sequence
SIP response code which plays the SIT3 tone sequence
SIP response code which plays the SIT4 tone sequence
SIP response code to use backup server 0
SIP response code to retry the registration
0
0
0
0
30
Note: The range for the SIP Response Codes is 0 through 65535. However, the SIP Response Codes are not
implemented.

SIP Distinctive Ring Names

Parameter Description Default
01
02
03
04
05
06
07
08
Telephone event name to produce distinctive ring pattern 1 Belcore-r1
Telephone event name to produce distinctive ring pattern 2 Belcore-r2
Telephone event name to produce distinctive ring pattern 3 Belcore-r3
Telephone event name to produce distinctive ring pattern 4 Belcore-r4
Telephone event name to produce distinctive ring pattern 5 Belcore-r5
Telephone event name to produce distinctive ring pattern 6 Belcore-r6
Telephone event name to produce distinctive ring pattern 7 Belcore-r7
Telephone event name to produce distinctive ring pattern 8 Belcore-r8
22 X6v VoIP Features Technical Reference
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SIP Protocol Timers

Parameter Description Default
SIP Timer INVITE Expires (s)
SIP Timer Re -INVITE Expires (s)
SIP Timer Registration Min (s)
SIP Timer Registration Max (s)
SIP Timer Registration Retry (s)
SIP Timer No Answer Duration (s)
SIP Timer Re-Register Interval (s)
SIP Session Timer (s)
Note: The range for the SIP Protocol Timers is 0 through 65535. However, the SIP Protocol Timers are not
implemented.
The time (in seconds) after which an INVITE request expires.
The time (in seconds) after which a retransmitted INVITE request expires.
The minimum Registration Period (in seconds).
The maximum Registration Period (in seconds).
The time interval (in seconds) for retrying a (failed) REGISTER request.
The length of time (in seconds) before terminating a session request.
The elapsed time (in seconds) between an initial and repeat REGISTER request.
The time interval (in seconds) for the session timer.
180
180
1
7200
30
60
20
0
Chapter 6: SIP Parameters 23
Page 24
7
Regionalization
You can use the VoIP > Advanced VoIP Setup > Regionalization menu to configure the VoIP Subsystem for
local operating conventions. The menu options include:

Call Progress Tones

•
Standard Ringing Patterns
•
Distinctive Ringing Patterns
•
Distinctive Call Waiting Patterns
•
Voice and Tone Parameters
•
SLAC Configuration
•
SLAC Command Strings
•
CODEC Configuration
•
CODEC Command Strings
•
Other
•
Note: In some fields below you might see default values that are valid for the United States only. If you
are reviewing or configuring VoIP settings for other regions, those default values do not apply.
Call Progress Tones
Call progress tones are specified by a list of values indicating the number of tones, number of on/off transitions, frequency/signal level pairs, and tone on/off times. The format is:
no_of_tones, no_of_times, duration, {tone_element1_freq, tone_element1_db, tone_element2_freq, tone_element2_db, …}, {tone_on_time1, tone_off_time1, tone_on_time2, …}
where:
no_of_tones is the number of tone elements that are combined to form a tone. Each tone element
has an associated frequency and amplitude. Up to four tone elements can be combined – to form a
chord, or played in sequence – as a tune (see no_of_times). A negative no_of_tones indicates
that the tones will be synchronized to a two-second timer (relevant for multi-port ATAs only).
no_of_times is the total of both on-to-off and off-to-on transitions in the tone pattern. If this value is
positive, it produces a composite tone. If it is negative, the tones are played in sequence. Zero produces a continuous composite tone
duration is the length of time in seconds that the call progress tone will be played. A value of zero
means that the tone will be played until instructed otherwise.
tone_elementX_freq and tone_elementX_db represent the frequency (Hz) and signal level (dB) of
each tone. A negative frequency is used to modulate the prior tone components summed together.
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A negative dBm level can be offset by ipbx_tone_gain. Allowed values for freq are from 0 to 3000Hz. Allowed values for db levels are from –1 to –40 dB.
tone_on_timeX and tone_off_timeX are interleaved Tone On and Tone Off durations in msec. A
value of zero for a Tone On time indicates a continuous tone. A value of zero for a Tone Off time produces silence, while a negative value (-1) terminates the tone pattern, removing the silencing. (With silencing, the voice channel is blocked until the tone pattern is stopped.) The maximum number of tones is four. The maximum number of on-to-off and off-to-on times counted individually is nine.
For example, the default setting for initial North American dial tone is:
{2, 0, 0, {350, -19, 440, -19}, {0}}
where:
2 is the number of frequency/dB pairs (350, -19, and 440, -19) The first 0 is the number of on/off transitions in the tone pattern, in this case a constant tone. The second 0 indicates that the tone will be played until otherwise instructed. The first pair of frequency/dB (350, -19) specifies that the first tone is at 350Hz with a level of -19dB. The second pair of frequency/dB (440, -19) specifies that the second tone is at 440Hz with a level of
-19dB.
The final {0} specifies that there are no on/off times and that the tone is constant.

Call Progress Tone Parameters

Parameter Description Default (North America)
Initial Dial Tone
Alternate Dial Tone
Secondary Dial Tone
Stutter Dial Tone
Message Waiting Dial Tone
Call Forward Dial Tone
Pre-Ringback Tone
Ringback Tone
Call Waiting Tone Default
PSTN Call Waiting Tone Default
The default tone used when a person begins any dialing operation
The alternate tone used when a person begins any dialing operation
The tone used in cases where a person can dial a number to access a designated type of line
Indicates a message waiting
Indicates a message waiting
Indicates that calls are being forwarded
Played while a call is being signaled before a confirmation is received from the SIP server
Played while a call is connecting
Played when an incoming call arrives and the phone is in use
Played when a call is on hold longer than the timeout hold duration
2 0 0 350 -19 440 -19
1 0 0 400 -16
2 0 0 420 -19 520 -19
2 7 0 350 -19 440 -19 100 110 100 110 100 110 0
2 2 0 350 -19 440 -19 160 160
2 3 0 350 -19 440 -19 250 400 0
0 0 0 (Silence)
2 2 0 440 -19 480 -19 2000 4000
1 2 0 440 -16 300 9700
1 2 0 440 -16 300 9700
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Parameter Description Default (North America)
Station Call Waiting Tone Default
Call Holding Tone
Call waiting pattern for station to station calls. Applies to multi-port units only.
Reminder tone that a call is on hold
1 2 0 440 -16 300 9700
1 4 0 1200 -16 100 200 100 ­1
Call Disconnect Tone
Played when a call on hold has
1 4 0 350 -16 50 100 50 -1
disconnected
Call Conference Tone
Played when a conference is in
1 2 0 350 -16 100 15000
progress
Busy Tone
Reorder Tone
Off Hook Warning Tone
SIT1 Tone
Sent back to the caller when the recipient's line is busy
A fast, busy, or congestion tone sent to the caller when a call cannot go through
Sounds when the telephone is off-hook for longer than the timeout alert duration
Sent to the user when a telephone number is invalid or has been
2 2 0 480 -19 620 -19 500 500
2 2 0 480 -19 620 -19 250 250
4 2 0 1400 11 2050 11 2450 11 2600 11 100 100
3 -6 0 985 -16 1428 -16 1777
-16 330 5 330 5 330 1000
disconnected
SIT2 Tone
Sent to the user when a telephone number is invalid or has been
3 -6 0 914 -16 1371 -16 1777
-16 330 5 330 5 330 1000
disconnected
SIT3 Tone
Sent to the user when a telephone number is invalid or has been
3 -6 0 985 -16 1428 -16 1777
-16 380 5 380 5 380 1000
disconnected
SIT4 Tone
Sent to the user when a telephone number is invalid or has been
3 -6 0 914 -16 1371 -16 1777
-16 380 5 380 5 380 1000
disconnected
Prompt Tone
Played when the user has completed a
2 0 0 520 -19 620 -19
segment of input
Confirm Tone
Played when the user has entered an
1 2 0 600 -16 400 0
acceptable value
Input Error Tone
Number Error Tone
Played when the user has made an invalid entry
Played when the user has entered an invalid dial string
2 2 0 480 -19 620 -19 250 250
2 2 0 480 -19 620 -19 250 250
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Standard Ringing Patterns

Ring patterns are specified by a list of values indicating the frequency, number of on/off transitions, and Ring On/Ring Off times. The format is:
ring_frequency, no_of_times, duration, {ring_on_time1, ring_off_time1, ring_on_time2, ring_off_time2, …}
where:
ring_frequency specifies the frequency of the ringing tone in Hz for sinusoidal and trapezoidal ringing. This value is only used if the default ringer parameter slac_ring_frequency is zero.
no_of_times is the total of both on and off transitions in the ring pattern. This can be zero for a
continuous ring signal (which may not be desirable and may exceed the rated power capacity of the ATA).
duration is the length of time in seconds to ring. A value of zero means until instructed otherwise. ring_on_timeX and ring_off_timeX are interleaved Ring On and Ring Off durations in msec. A
value of zero for a Ring On time indicates a continuous tone. A value of zero for a Ring Off time produces continuous silence.
Possible values for frequency are between 0Hz and 60Hz. The maximum total of on and off times summed together is nine.

Standard Ringing Patterns Parameters

Parameter Description Default (North America)
Ring Default
PSTN Ring Default
Station Ring Default
Call Hold Re-Ring
Call Back Ring
Call Back Ring Splash
Call Forward Ring Splash
Message Waiting Ring Splash
Default ring pattern 20 2 0 2000 4000
Default PSTN call ring pattern 20 2 0 2000 4000
Default station call ring pattern 20 2 0 1000 3000
Call on hold reminder re-ring pattern
Call back success ring pattern 20 2 0 1500 0
Call back in progress ring pattern 20 2 0 700 0
Call forward reminder ring pattern 20 2 0 500 0
Audible message waiting ring pattern. This parameter is for analog telephone adapter products only.
20 2 0 500 0
20 2 0 500 0

Distinctive Ringing Patterns

The distinctive ring feature allows different ring patterns to be sent to the telephone according to Distinctive Ring parameters 1 - 8. Distinctive ringing patterns are specified in the same way as standard ringing
patterns.
The user can assign distinctive ringing patterns to particular callers under User Configuration…Ringing Based on Caller ID.
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Distinctive Ringing Patterns Parameters

Parameter Description Default (All Regions)
Distinctive Ring 1
Distinctive Ring 2
Distinctive Ring 3
Distinctive Ring 4
Distinctive Ring 5
Distinctive Ring 6
Distinctive Ring 7
Distinctive Ring 8
Specifies the pattern for Ring 1 20 2 0 2000 4000
Specifies the pattern for Ring 2 20 4 0 1000 1000 1000 3000
Specifies the pattern for Ring 3
Specifies the pattern for Ring 4 20 4 0 800 400 800 4000
Specifies the pattern for Ring 5 20 4 0 400 200 400 2000
Specifies the pattern for Ring 6 20 2 0 1000 3000
Specifies the pattern for Ring 7 20 4 0 300 200 1500 2000
Specifies the pattern for Ring 8 20 4 0 800 400 800 2000
20 6 0 300 200 1000 200 300 4000

Distinctive Call Waiting Patterns

A call waiting tone is played when an incoming call arrives while the phone is in use. Support for up to eight distinctive call waiting tone patterns is available. Distinctive call waiting patterns are specified in the same way as standard ringing patterns.
When the user assigns a distinctive ringing pattern to a particular Caller ID, the corresponding distinctive call waiting pattern is also assigned to that Caller ID.

Distinctive Call Waiting Patterns Parameters

Parameter Description Default (North America)
Call Waiting Tone 1
Call Waiting Tone 2
Call Waiting Tone 3
Call Waiting Tone 4
Call Waiting Tone 5
Call Waiting Tone 6
Call Waiting Tone 7
Call Waiting Tone 8
Specifies the pattern for Tone 1 1 2 0 440 -16 300 9700
Specifies the pattern for Tone 2
Specifies the pattern for Tone 3
Specifies the pattern for Tone 4
Specifies the pattern for Tone 5 1 2 0 620 -16 300 9700
Specifies the pattern for Tone 6
Specifies the pattern for Tone 7
Specifies the pattern for Tone 8
1 6 0 440 -16 100 20 100 20 100 9660
1 4 0 440 -16 100 100 100 9700
1 6 0 440 -16 100 100 100 100 100 9500
1 6 0 620 -16 100 20 100 20 100 9660
1 4 0 620 -16 100 100 100 9700
1 6 0 620 -16 100 100 100 100 100 9500
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Voice and Tone Parameters

The parameters in the following sections control the connection to the local phone (FXS) port on the VoIP
Subsystem. This includes control of both the Subscriber Line Interface Circuit (SLIC) and Subscriber Line Audio Circuit (SLAC) that together make up the FXS port.

Voice, Tone and DTMF Parameters

The following table lists parameters that control voice and tone signals, transmit and receive levels, and
Dual-Tone Multi-Frequency (DTMF) signaling tone characteristics.
Parameter Description Default
Voice RX Gain (-20 to +10 dB)
Voice TX Gain (-20 to +10 dB)
Tone Gain (-20 to +10 dB)
Tone Max (-20 to 0 dBm)
DTMF Low Tone Gain (-20 to
-5 dBm)
DTMF High Tone Gain (-20 to
-5 dBm)
DTMF Tone On Time (ms)
DTMF Tone Off Time (ms)
DTMF Detect ABCD
DTMF Generate ABCD
DTMF Pad Duration (ms)
DTMF Wait Duration (ms)
DTMF Playout Min Duration (ms)
Voice receive gain in dB 0
Voice transmit gain in dB 0
Tone signal gain in dB (applied to locally generated tones such as call paging tones).
When two tones of equal amplitude are added together, the signal level is 3dB higher than the individual components. When four tones of equal amplitude are added together, the signal level is 6dB higher than the individual components. This limit prevents inadvertent saturation and user hearing damage.
Low frequency group DTMF tone level in dBm
High frequency group DTMF tone level in dBm
DTMF generation On time (50 to 200 ms) 80
DTMF generation Off time (50 to 200 ms) 50
DTMF detection enable for ABCD dual tone pairs
DTMF generation enable for ABCD dual tone pairs
DTMF out-of-band On time in milliseconds (0 to 10,000 ms)
DTMF out-of-band Off time in milliseconds (0 to 10,000 ms)
DTMF out-of-band minimum on time in milliseconds (0 to 10,000 ms)
0
-12
-9
-7
Enabled
Enabled
100
50
100
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SLAC Configuration Parameters

Parameter Description Default
Port Impedance
for Port Impedance table.)
Port RX Gain (GR) (-12 to 6dB)
Port TX Gain (GX) (+12 to 6dB)
Audio Clamp Duration (ms)
Caller ID Type 1 Mode
Caller ID Type 2 Mode
Message Waiting Mode
Ring Type
Ring Frequency (0 to 60 Hz)
Ring Transition (ms)
Ring Amplitude (v)
Ring Bias (v)
Message Waiting Type
Message Waiting Frequency (Hz)
Message Waiting Transition (ms)
Message Waiting Amplitude (v)
Message Waiting Bias (v)
(See the Values
Synthetic impedance matching network control for a choice of one of 10 common world-wide configurations
SLAC receive gain in dB units -1
SLAC transmit gain in dB units (Note: A
value of 6 dB of attenuation is automatically applied by the GX gain block prior to the specified transmit gain.)
Audio clamp On time (0 to 65535 ms) 100
Caller ID type 1 (on-hook) mode (None, Belcore MDMF, Belcore SDMF, ETSI Wink, ETSI Ring, DTMF)
Caller ID type 2 (off-hook) mode (None, Belcore MDMF, Belcore SDMF, ETSI Wink, ETSI Ring, DTMF)
Message waiting mode (None, Belcore MDMF Belcore SDMF, ETSI)
Selects ring waveform type: Sinusoidal or Trapezoidal
Ringer frequency in Hz (zero to use ring pattern frequency specification)
Trapezoidal transition time (0 to 1000ms) 15ms
Ringer voltage in volts (-155v to +1.55v) 85v
Ringer bias in volts DC (-155v to +1.55v) 0
Selects visual message waiting waveform type: Sinusoidal or Trapezoidal
Visual message waiting frequency in Hz (0 to 60Hz)
Trapezoidal transition time in msec (0 to 1000ms)
Visual message waiting voltage in volts (-155v to +155.v)
Visual message waiting bias in volts (-155v to +155v)
Varies by region
5
Belcore MDMF
Belcore MDMF
Belcore MDMF
Sinusoidal
0
Sinusoidal
25Hz
15ms
50v
0

Values for Port Impedance (SLAC & CODEC)

Index Impedance Country
0 600 (default)
1 900
2 600 + 1.0 μF
3 900 + 2.16 μF
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US
US/Canada
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Index Default Country
4 270 + 750 || 150 nF
5 220 + 820 || 120 nF
6 220 + 820 || 115 nF
7 370 + 620 || 310 nF
8 200 + 680 || 100 nF
9 800 || 50 nF
Sweden/CTR21
Germany/Austria/Australia/New Zealand #1
Bulgaria/South Africa/Slovakia
UK #1/India/New Zealand #2
China

SLAC Command Strings

The SLAC initialization commands provide a method to set up the device for unusual conditions. Do not change the default value unless the factory has suggested you do so.
Parameter Description Default
Initialization Commands
Specifies device setup for unusual conditions.
100

CODEC Configuration

This section describes the TELCO (FXO) port connection on the VoIP Subsystem and the CODEC (COde DECode) configuration that provides the signal interface to the FXO port.
Parameter Description Default Port Impedance (See the Values
for Port Impedance table on
30)
page
Port RX Gain (GR) (-12 to +6 dB)
Port TX Gain (GX) (-12 to +12 dB)
Audio Clamp Duration (ms)
Line in Use Detect Method
Line in Use Inhibit
Parallel in Use Debounce
Parallel in Use Detect Method
Parallel in Use Disconnect
Synthetic impedance matching network control for a choice of one in ten common world-wide configurations
SLAC receive gain in dB units 0
SLAC transmit gain in dB units
(Note: 6dB of attenuation is automatically
applied by the GX gain block prior to the specified transmit gain.)
Audio clamp On time in milliseconds (0 to 65535ms)
Defines the method to use for detecting the TELCO line's status.
Enables or disables use of the TELCO line.
Specifies the number of lines that can be used in parallel. 0 to 65535 lines are the possible min/max values; however, the physical limit is 5.
Defines the method to use for detecting the availability of a parallel line.
Enables or disables disconnection of a parallel line.
Default
-2
300
Default
Disabled
4
Default
Disabled
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Parameter Description Default
Caller ID Type 1 Mode
Caller ID type 1 (on-hook) mode (None,
Belcore MDMF Belcore MDMF, Belcore SDMF, ETSI WINK, ETSI RING, DTMF)
Caller ID Type 2 Mode
Caller ID type 2 (off-hook) mode (None,
Belcore MDMF Belcore MDMF, Belcore SDMF, ETSI WINK, ETSI RING, DTMF)
Message Waiting Mode
Message waiting mode (None, Belcore
Belcore VMWI VMWI, ETSI, Low Voltage Ring)
Ring Detect Duration (ms)
Ring Detect Period Minimum (ms)
Ring Detect Period Maximum (ms)
Ring Detect Threshold
Ring Silence Period
Ring Minimum period (ms)
Disconnect Voltage Enable
Disconnect Voltage Duration (ms)
Disconnect Polarity Enable
Disconnect Reversals Answer
Disconnect Reversals Originate
Disconnect Silence Enable
Disconnect Silence Duration (s)
Disconnect Silence Threshold
Disconnect Tone1 Mode
Disconnect Tone 1 Definition
Disconnect Tone 1 Duration (ms)
Disconnect Tone 1 Bandwidth (Hz)
Disconnect Tone 2 Mode
Disconnect Tone 2 Definition
The range is 0 to 65535 ms 100 ms
The range is 0 to 65535 ms 18 ms
The range is 0 to 65535 ms 64 ms
The range is 0 to 65535 ms 0
The range is 0 to 10,000 ms 5200 ms
The range is 0 to 10,000 ms 1500 ms
Disconnect on on-hook voltage Enabled
The range is 0 to 10,000 ms 100 ms
Disconnect on TIP/RING reversal Enabled
The range is 0 to 10 1
The range is 0 to 10 2
Interpret silence on line as disconnect Disabled
The range is 0 to 10,000 s 15 s
The range is -32768 to +32767 dB m0 -40
Select Mode (Dial Tone, Busy, or other) Dial Tone
Definition as per Call Progress tones 2 0 0 350 - 19 440 - 19
The range is 0 to 10,000 ms 5000 ms
The range is 0 to 100 Hz 30 Hz
The range is 0 to 100 Hz Busy Tone
The range is 0 to 100 Hz
2 2 0 480 - 19 620 - 19
500 500
Disconnect Tone 2 Duration (ms)
Disconnect Tone 2 Bandwidth (Hz)
Disconnect Tone 3 Mode
Disconnect Tone 3 Definition
Disconnect Tone 3 Duration (ms)
Disconnect Tone 3 Bandwidth (Hz)
The range is 0 to 10,000 ms 3000 ms
The range is 0 to 100 Hz 30 Hz
The range is 0 to 100 Hz User Defined Tone
The range is 0 to 100 Hz 0 2 0 450 450
The range is 0 to 10,000 ms 3000 ms
The range is 0 to 100 Hz 30 Hz
32 X6v VoIP Features Technical Reference
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CODEC Command Strings

The CODEC initialization commands provide a method to set up the device for unusual conditions. Do not change the default value unless the factory has instructed you to do so.
Parameter Description Default
Initialization Commands
Specifies device setup for unusual conditions.
100

Other Parameters

Parameter Description Default
Hook Debounce (units of 10 ms)
Ring Debounce
Disconnect Debounce
Reconnect Debounce
The range is 0 to 65535 ms 10 (that is, 100 ms)
The range is 0 to 65535 ms 20 ms
The range is 0 to 65535 ms 40 ms
The range is 0 to 65535 ms 20 ms
Chapter 7: Regionalization 33
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8
Subscription Services
You can use the VoIP > Advanced VoIP Setup > Subscription Services menu to configure the VoIP Subsystem for the specific advanced services permitted and/or supported. The menu items include:

Subscription Service Settings

•
Dialing Parameters
•
VoIP and PSTN Dial Patterns
•
Subscription Service Settings
Parameter Description Default
Call Waiting
Caller ID Inbound
Caller ID Outbound
Call Waiting Caller ID Service
Call Back
Call Return
Speed Dial
Do Not Disturb
Block Anonymous
Call Forward Always
Call Forward on Busy
Call Forward on No Answer
Call Forward Priority
Enables customer use of call waiting service
Enables customer use of incoming caller ID service
Enables customer use of outgoing caller ID service (i.e. always send caller ID information)
Enables customer use of incoming caller ID during call waiting service
Enables customer use of call back service Enabled
Enables customer use of call return service
Enables customer use of speed dial service
Enables customer use of do not disturb service
Enables customer use of anonymous call block service
Enables customer use of call forward service
Enables customer use of call forward when busy service
Enables customer use of no answer call forward service
Enables customer use of priority call service
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
34 X6v VoIP Features Technical Reference
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Parameter Description Default
Distinctive Ring
Enables customer use of distinctive ring
Enabled service
Disturb Accept
Enables customer use of do not disturb
Enabled accept service
Blocked Number
Enables customer use of blocked number
Enabled service
Outgoing Block
Forward Last Call
Enables outgoing blocked number Enabled
Enables customer use of forward to last
Enabled caller service
Distinctive Ring Last Call
Enables customer use of distinctive ring
Enabled for last caller service
Disturb Accept Last Call
Enables customer use of do not disturb
Enabled accept last caller service
Block Last Call
Enables customer use of block last caller
Enabled service
Three-Way Calling
Enables customer use of three way
Enabled calling service
Three-Way Conference
Enables customer use of three way
Enabled conference service
Attended Transfer
Enables customer use of attended call
Enabled transfer service
Unattended Transfer
Enables customer use of unattended call
Enabled transfer service
Message Waiting
If voice mail is enabled, the VoIP
Enabled Subsystem can send a distinctive dial tone to indicate that there are unplayed messages in the user’s voice mailbox.
Visual Message Waiting
Enables customer use of visual message
Enabled waiting service
Remote Feature Code
Enables sending all features codes to
Disabled remote service provider
Default Feature Code
Enables sending all unprocessed feature
Disabled codes to remote service provider
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Dialing Parameters

Parameter Description Default Mode
My VoIP Account Unavailable
No VoIP Accounts Available
PSTN Not Available
Dial Direct
Dial After #8
Dial after #9
Speed Dial
VoIP Dial Pattern (See VoIP and
PSTN Dial Patterns on page 37.)
PSTN Dial Pattern (See VoIP and
PSTN Dial Patterns on page 37.)
Configure VoIP Dial Pattern
(See
VoIP and PSTN Dial
Patterns on page 37.)
Mode allows selection of treatment of * and # as the leading digit of a dial string.
These characters may be processed locally, or they may be passed through to the service provider. If there is a requirement that the service provider
process commands that start with #, or for sequences such as “* *”, these characters must be passed through. Select Normal for local processing of these digits; Pass- through to pass these digits to the service provider. Note that when Pass- through mode is selected, feature codes
and speed dials cannot be handled locally on the VoIP Subsystem.
Standard Dial Tone, Alternate Dial Tone, No Dial Tone
Standard Dial Tone, Alternate Dial Tone, No Dial Tone
Standard Dial Tone, Alternate Dial Tone, No Dial Tone
Direct dial processing mode (Disallowed, VoIP only, PSTN only, BOTH or DIRECT)
Processing mode after a #8 prefix (Disallowed, VoIP only, PSTN only, BOTH or DIRECT)
Processing mode after a #9 prefix (Disallowed, VoIP only, PSTN only, BOTH or DIRECT)
Processing mode for speed dial (Disallowed, VoIP only, PSTN only, BOTH or DIRECT)
Pattern match for VoIP dialing
Pattern match for PSTN dialing 100|11x|911|999
Used to configure how the VoIP Subsystem handles VoIP dial strings.
Normal
Interpret * and #
DTMF tones locally.)
Alternate Dial Tone
Alternate Dial Tone
No Dial Tone
BOTH
DIRECT
VoIP only
VoIP only
[3469]11|*xx|**|[1-9]e#
r5xp3r*x|p8[1-9]e#r5xp
3r*x|3[1-9]e#r5xp3r*x|
1010Se#p2r*x|0Se#r5
xp2r*x
[3469]11|*xx|**|[1-9]e#
r5xp3r*x|p8[1-9]e#r5xp
3r*x|3[1-9]e#r5xp3r*x|
1010Se#p2r*x|0Se#r5
xp2r*x
36 X6v VoIP Features Technical Reference
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Parameter Description Default
Configure PSTN Dial Pattern
(See
VoIP and PSTN Dial
Patterns, below.)
Hot Line Dialing
Warm Line Dialing
Hotwarm Dial String
Auto-Add This Area Code ...
Polarity Dialing
Number of Digits I Will Dial For Local Calls
Polarity Dialing
Polarity Dial Tone
Polarity Connect
Polarity Answer
Polarity Idle
Used to configure how the VoIP Subsystem handles PSTN dial strings.
If enabled, the VoIP Subsystem automatically dials the hot/warm dial string as soon as the telephone receiver is picked up.
If enabled, when the telephone receiver is picked up, the VoIP Subsystem automatically dials the hot/warm dial string after a short wait (default is four seconds).
Used in hot and warm dialing when one or the other is enabled.
Sets the area code to add automatically.
Sets the SLAC line polarity during dialing (Forward or Reverse).
Specifies the default number of digits to be dialed for local calls.
Sets the SLAC line polarity during dialing (Forward or Reverse)
Sets the SLAC line polarity during dial tone (Forward or Reverse)
Sets the SLAC line polarity during connect (Forward or Reverse)
Sets the SLAC line polarity during answer (Forward or Reverse)
Sets the SLAC line polarity during idle (Forward or Reverse)
Disabled
Disabled
Forward
7
Forward
Forward
Forward
Forward
Forward

VoIP and PSTN Dial Patterns

The VoIP Dial Pattern and the PSTN Dial Pattern together determine how the VoIP Subsystem handles dial strings when someone dials a number from an attached phone. For units without an FXO port, the PSTN Dial Pattern is ignored. In a given location, there are generally only a few types of dialed numbers that need to be defined:
• Dialing for local calls
• Dialing for domestic toll calls,
• Dialing for international toll calls.
In addition, there are specific short strings that are set aside for emergency dialing, and there might be other special strings that invoke telephone features.
By default, the VoIP Subsystem is configured to handle number patterns in every country in the world. For models with an FXO port, emergency calls are by default routed to the PSTN, and all other calls are routed via VoIP. If no telephone line is connected to the Telco port, emergency calls are routed via VoIP.
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You can use the dial patterns to change which calls are sent via VoIP, and which are sent to the PSTN. For example, you might want to send all local calls via the PSTN, because these might be free on your PSTN line.
You might also want to tailor the dial patterns to precisely reflect the format of telephone numbers in your location. For example, the default configuration recognizes that a local number might be from 5 to 10 digits long. If local numbers are always 8 digits, this means that the VoIP Subsystem will wait a few seconds after the 8 string always to expect 8 digits, and to immediately send the number to the service provider once someone had dialed 8 digits.
th
digit has been dialed, to see if any digits follow. You could redefine the local dial

Dial Pattern Parameters

Parameter Description
|
Any DTMF char or chars
x
~
[]
[^]
[0-9]
[a-d]
r
.(period)
+
!
$
Separates patterns.
Literal list of one or more DTMF characters to match in the order shown, and in the position indicated within the pattern.
Match any numerical digit (0-9)
Match any digit (0-9, A-D, *, #) excluding any specified terminators
Selection group of candidate digits. This group can contain any number of DTMF characters, any of which are considered a match.
Exclusion group of digits. If any DTMF character occurs at this point in the dial string which matches the exclusion digits listed after the carat, the dialed string fails the match test with this pattern.
Selection range of candidate numerical digits
Selection range of candidate letter digits
Repeat operator. Syntax r n p, where r is the repeat operator, n is the number of repetitions, and p is the item that is repeated. n can be 1-9 repetitions, letters
a-z for 10 to 35 repetitions or repeat until the person stops dialing.
Repeat the previous digit until the person stops dialing.
Repeat the previous digit until the person stops dialing.
Disallows pattern. This element can prevent users from dialing numbers or classes of numbers.
Indicates secondary dialing to follow - used only by fixed dial strings.
* (asterisk), + (plus sign) or . (period) to mean
<:>
s
e
f
p
38 X6v VoIP Features Technical Reference
Replace group: replace digits to the left of the colon with those to the right.
Seize on string as only candidate if dialed digits match to this point.
Specify ending termination digit which follows (usually * or #). When the user
dials the ending termination digit, the VoIP Subsystem considers the dial string complete, and immediately sends to the service provider the digits up to the termination character.
Pause timeout causes failure instead of dial.
Pause Operator. Syntax p n, where n is the time in seconds to allow between
digits dialed. If this time is exceeded, the dialing is considered to have timed out, and the person to have stopped dialing.
Page 39
Parameter Description
t
- (dash)
(space)
Notes:
Interdigit timeout, or pause: By default, the device allows five (5) seconds between dialed digits. To change this default, you must insert the p parameter before the point in the match string that you want this parameter to change.
For example, if you would like a nine (9) second delay after each digit is pressed, then you would need to enter p9 at the beginning of the pattern matching string. Similarly, if you would like a shorter timeout of three (3) seconds towards the end of a dial string, you would need to enter p3 before the last entry in the pattern matching string: …p3r*x.
Set digit timeout to default for current pattern.
Human-readable spacing which is ignored.
Human-readable spacing which is ignored.

Examples of Dial Strings

Each parameter in a pattern match string represents a single digit. The only exceptions are parameters that include a repeat operator. We will illustrate these features by examining several entries in the default VoIP dial string:
[346]11|*xx|**[1-9]e#r5xp3r*x|p8[1-9]e#r5xp3r*x|#[1-9]e#r5xp3r*x|1010Se#p2r*x|0Se#r5xp2r*x[3469]11
Entries are separated by the pipe “|” character. Each entry represents a possible match to the digits that someone dials.
The following descriptions explain how some of the entries in the default Dial String behave.
[346]11 indicates to recognize the sequences 311,411, 611 and 911, and send them to the service
provider when complete.
*xx is a string that allows the VoIP Subsystem to recognize and forward feature codes to the service provider. However, note that by default, feature codes are handled locally, in the VoIP Subsystem. The VoIP Subsystem refers to this string only if the remote or default feature code parameters are enabled, or if Passthrough mode is enabled. In those cases, this string must be included in the pattern matching string, so that the VoIP Subsystem will forward feature codes to the service provider.
**[1-9]e#r5xp3r*x is a string that pertains to VoIP provider area codes. The ** prefix is a signal for the service provider to forward this call to another VoIP service provider. The three digits following ** constitute the VoIP provider area code. Recognize a string starting with **, and proceeding with any of the digits 1-9. e# defines # as the terminating character. If someone dials # at any point after the 1-
9, the VoIP Subsystem sends out all digits dialed to that point to the service provider. If the person
doesn’t dial a #, collect five more digits (r5x), switch from the default inter-digit timeout of five (5) seconds to a shorter inter-digit timeout of three (3) seconds (p3), and continue collecting digits until a timeout occurs (r*x). This string will be forwarded only if the VoIP Subsystem is in Passthrough mode.
p8[1-9]e#r5xp3r*x is the workhorse string of the default pattern for dialing. It matches dialing for VoIP
calls, and for local dialing in most countries. It also matches dialing for domestic long distance dialing under the North American dial plan. This string is identical to the preceding string, except for the first two characters. Where the preceding string calls for a match to the prefix **, this string redefines the inter-digit timeout. This value has been increased to eight (8) seconds. This timeout value persists until the first digit plus five other digits have been collected, at which time the timeout value is reduced to three (3) seconds. From that point onward, the VoIP Subsystem continues to collect digits until the user pauses three seconds, at which point the VoIP Subsystem sends the dialed string to the service provider.
Chapter 8: Subscription Services 39
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#[1-9]e#r5xp3r*x is a string that is identical to the previous two, except for the first digit. This string
supports cases where service providers use strings that start with # for various special features or
control purposes. This string is forwarded to the service provider only if the mode is set to
Passthrough.
1010Se#p2r*x is a string included to support cases where North-American style dial-around dialing is
available. The S means that if someone dials 1010 as the first four digits of a dial string, this is the
only string the VoIP Subsystem should match to from that point on. e# means that the user can indicate the completion of dialing at any time by entering #. p2 means that after someone dials 1010,
the timeout between subsequent digits is reduced to two (2) seconds. r*x means that the VoIP Subsystem will continue to collect dialed digits until there is a timeout.
0Se#r5xp2r*x is the second workhorse string of the default pattern matching string. International calls in
almost every country, and domestic long distance calls in most countries outside North America, all
match this pattern. Any number that starts with zero (0) matches this string. The user may dial # at
any time to indicate the number dialed is complete. After the user dials the sixth digit, the inter-digit timeout is reduced to two seconds. After that point, the VoIP Subsystem continues to collect digits until the user pauses two seconds. Then the VoIP Subsystem sends the dialed string to the service provider.
[3469]11 means either 3 OR 4 OR 6 OR 9, followed by 11 (that is, 311 OR 411 OR 611 OR 911).

North American Number Plan Area (NANPA) Dialing Examples

[^1]r6x
Recognize a seven (7) digit number, However, do not match to this string if beginning with a 1(one)
This string will allow a user to dial 2XXXXXX - 9XXXXXX. However, if the number entered begins with a 1 (one), do not match to this pattern.
1r3x[^1]r6x
Match a long distance number to this string, as in 1-<area code>-<7 digit dial>.
This string will allow a user to dial a phone number using a toll prefix of 1 (one). It also makes certain that the seven-digit local phone number under NANPA does not begin with a 1 (one).

Dial String Tips

1900r7x!
Disallow 1900XXXXXXX
This tells the system to look at the first four digits of the entered number, and if they match 1900 to drop to a failed tone.
1900 numbers in the US are premium-rate numbers that may incur high per-minute charges.
976r4!
Disallow a 976XXXX number
This tells the system to look at the first three digits of the entered number, and if they match 976 to drop to a failed tone.
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. 976 numbers in the US are premium-rate local numbers that may incur high per-minute charges.
1800r7x
Recognize a 1800XXXXXXX number
This tells the system to look at the first four digits of the entered number, and if they match 1800 to dial
using 1800 plus the remaining seven digits.
<:>
If you want to set up a dial pattern that allows the user to easily select between two services, you can use the <:> symbol. By including <[89]:> in the dial pattern, you tell the system to replace an 8 or 9 with a null value, and continue pattern matching as necessary.
For example, <[89]:>r7x: as long as the first digit is an 8 or 9, the system will accept an 8 or 9 followed by seven digits, remove the first digit (8 or 9), and dial out the remaining seven digits. You can specifiy an 8 as part of the pattern recognition string for one provider, and 9 as part of the pattern recognition for another provider. This will allow users to easily select among providers with similar numbers. Note that this doesn’t work well if any numbers you want to reach start with 8 or 9. In that case, you may want to consider prefixes that start with *8, #8, *9 or #9.

Entering Easily-Confused Patterns

If you enter two different patterns which can easily be confused with each other, the system will choose the first pattern that is matched. For instance, if you have two patterns, one for eleven digits, and one for twelve, the system will not wait for the twelfth digit, because it will match to the eleven-digit pattern first. To prevent this, you should set up the dial pattern (matching similarly to the two examples above) using 0Se#e*p2r*x or 1010Se#e*p2r*x. These patterns will force the system to wait until after the user has entered as many digits as are necessary before it tries to connect to a provider
.

Bridging From VoIP to PSTN

Parameter Description Default
Bridge from VoIP to PSTN
Auto-Answer VoIP Bridge Calls
VoIP Bridge Accept Any Call
VoIP Bridge Accept Anonymous Calls
VoIP Bridge Single Stage Dialing Enable
Caller Password
Password Dial String
VoIP Bridge Accept Only These Numbers (01 to 10)
VoIP Bridge Billing Delay Duration (10 ms)
VoIP Bridge Security Entry Duration (10 ms)
Enable or disable the bridge Disabled
Enable or disable auto-answer Disabled
Enable or disable call acceptance Disabled
Enable or disable anonymous call acceptance
Enable or disable single stage dialing Disabled
Enable or disable caller password Disabled
Specifies the password dial string
When any numbers are listed here, only calls from those numbers will be bridged.
Specifies the duration of billing delay (0 to 65535 ms)
Specifies the duration for the security entry (0 to 65535 ms)
Disabled
100 ms
1000 ms
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Bridging from PSTN to VoIP

Parameter Description Default
Bridge From PSTN to VoIP
Auto Answer PSTN (FXO) Calls
FXO Port Accept Anonymous Calls
FXO Port Only Accept Calls with Caller ID
FXO Port Accept Only These Numbers (01 to 10)
Caller Password
Password Dial String
Enable or disable the bridge Disabled
Enable or disable auto-answer PSTN calls Disabled
Enable or disable anonymous call acceptance on FXO port
Enable or disable acceptance of caller ID calls only on FXO port
When any numbers are listed, only calls to those numbers will be accepted.
Specifies requirement for caller password Disabled
Specifies required caller password string
Disabled
Disabled

Miscellaneous TELCO Parameters

Parameter Description Default
Telco Port Display Caller ID
Telco Port Caller ID Sent After One Ring
PSTN CID Wait Duration (10 ms)
PSTN CID Clear Duration (10 ms)
Billing Delay Duration (10 ms)
PSTN Security Entry Duration (10 ms)
If My Call Starts With These Digits ....
If I Normally Want Auto-Add Area
Code Calls Routed ....
Route VoIP Calls Via My Telco Line If VoIP Service is Unavailable
Enable or disable the caller ID display Disabled
Indicate to device whether Telco CID is sent before or after the first ring
Time after incoming call initiation (first ring or line reversal to continue looking for CID signal). (0 to 65535 ms)
Time after last ring to continue to display CID. (0 to 65535 ms)
Time after auto-answer to send Bong tone
prompt in bridge mode. (0 to 65535 ms)
In bridge mode, time within which the user must enter security code, if enabled. (0 to 65535 ms)
Requests the line to use when dialing numbers that begin with the specified digits.
Enables or disables alternate auto-add routing of Telco line calls
Enables or disables alternative routing of VoIP calls.
Enabled
500 ms
1000 ms
100 ms
1000 ms
Disabled
Enabled
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Emergency Services and eServices Events

The emergency services numbers follow the same rules as those defined for the pattern matching strings
Dialing Parameters on page 36.
in
The VoIP Subsystem allows flexible treatment of emergency numbers. They can be sent either via the Internet or over the PSTN. When you are connected to a SUBSCRIPTION server that supports the Eservices (Emergency Services) event, the server and VoIP Subsystem can coordinate with each other to make sure that the VoIP Subsystem will route emergency calls via the appropriate connection. Make sure to include all emergency numbers in both the default VoIP and PSTN parameters, if you want the VoIP Subsystem to make a flexible selection.
Parameter Description Default
Emergency Numbers Routed via VoIP
Emergency Numbers Routed via the PSTN
Default Emergency Numbers Routed via VoIP
Default Emergency Numbers Routed via the PSTN
Always Route Emergency Numbers via the PSTN
Emergency Numbers via the PSTN Alt (Click Help)
Specifies which emergency numbers to route over VoIP
Specifies which emergency numbers to route over PSTN
Specifies which default emergency numbers to route over VoIP
Specifies which default emergency numbers to route over PSTN
When enabled, this parameter configures the VoIP Subsystem to always send emergency numbers to the PSTN. If the PSTN line is unavailable, then emergency calls are routed via VoIP.
When enabled, this parameter allows the VoIP Subsystem to determine which port to send emergency numbers to, based on negotiation over the event Eservices with the subscription server. If the subscription to the Eservices event fails, then emergency numbers are routed to the PSTN.
If both Always Route Emergency Numbers via
PSTN and Emergency Numbers via the PSTN
Alt are both disabled, then emergency
calls will be routed according to negotiation through the event Eservices. If the subscription fails, then emergency calls are preferentially routed via VoIP.
100, 11x, 911, 999
100, 11x, 911, 999
100|11x|911|999
100|11x|911|999
Disabled
Enabled
Note: If neither the PSTN nor VoIP is available, users will hear no dial tone when they pick up the
handset. In that case, they should understand that they cannot make an emergency call.
Chapter 8: Subscription Services 43
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9
User Configuration
You can use the VoIP > Advanced VoIP Settings > User Configuration menu to configure the VoIP Subsystem's
user-specific settings. The menu items include:
•
Speed Dials Call Forwarding
•
Ringing Based on Caller ID
•
Do Not Disturb
•
Incoming Call Blocking
•
Call Waiting/Caller ID
•
Timers
•

Speed Dials

The Speed Dial List can be modified by the telephone or via the web pages. Up to 28 numbers can be entered into the Speed Dial List. Each number can be up to 40 digits in length. Dialing a speed dial number
is explained in Chapter 4 of the Zoom ADSL X6v User Guide on your X6v CD.
Parameter Description Default
*20 - *39
#0 - #7
Speed dial number corresponding to *20 to *39
Speed dial number corresponding to #0 to #7

Call Forwarding

With Call Forward enabled, any call on this list will be forwarded to the number stored in the Call Forward List
(1-12). Up to thirty 40-digit numbers can be entered.
Parameter Description Default
Call Forward Always
Call Forward on Busy
Call Forward on No Answer
Enable or disable call forwarding in all cases
Enable or disable call forwarding when line is busy
Enable or disable call forwarding when the call is not answered
Disabled
Disabled
Disabled
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Parameter Description Default
Call Forward Priority
Call Forward Always Number
Call Forward on Busy Number
Call Forward on No Ans Number
Call Forward Priority Number
Priority Forward List – 1 to 30 phone numbers
Enables or disables priority call forward Disabled
Specifies the call forward destination
Specifies the call forward destination when the line is busy
Specifies the call forward destination when the line is not answered
Specifies the priority call forward destination
Specifies the list of numbers

Ringing Based on Caller ID

Parameter Description Default
Ringing Based on Caller ID
Distinctive Ring List – 1 to 30 phone numbers
Enables or disables distinctive ring tones linked to caller IDs
Specifies the phone numbers associated with caller IDs
Enabled

Do Not Disturb

Parameter Description Default
Do Not Disturb Mode
Do Not Disturb Exceptions
Do Not Disturb Exceptions List – 1 to 30 phone numbers
Enables or disables the Do Not Disturb Mode, which blocks all non-priority calls.
Priority calls are permitted if further enabled by the Do Not Disturb Exceptions. This value is reset on power up and restart.
Enables or disables the ringing of calls on the Disturb Exceptions List. All other callers will be blocked.
Specifies the list of numbers
Disabled
Disabled

Incoming Call Blocking

Parameter Description Default
Block Anonymous Incoming Calls
Block Listed Incoming Calls
Blocked Call List – 01 to 30 numbers
Enables or disables the blocking of calls that do not give caller ID information
Enables or disables the blocking of incoming calls from specific numbers in the Blocked Call List
Specifies the list of incoming numbers
Disabled
Disabled
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Parameter Description Default
Block Listed Outgoing Calls
Blocked Call List – 01 to 30 numbers
Enables or disables the blocking of outgoing calls from specific numbers in the Blocked Call List
Specifies the list of outgoing numbers
Disabled

Call Waiting/Caller ID

Availability of these features depends on whether they are supported by your VoIP service provider.
Parameter Description Default
Call Waiting
Inbound Caller ID
Outbound Caller ID
Call Waiting Caller ID
Enables or disables call waiting for all calls. When the line is in use and a call is received, a call waiting tone is played. Pressing the flash or the hook button on the phone momentarily switches between the two calls. While there are calls on both lines, additional incoming calls receive busy signals.
Enables or disables caller ID for inbound calls
Enables or disables caller ID for outbound calls
Enables or disables caller ID during call waiting
Enabled
Enabled
Enabled
Enabled

Timers

Parameter Description Default
Brief pause (10 ms)
Initial Dial (10 ms)
Warm Line (10 ms)
Sets the amount of time after picking up the receiver before dial tone is generated. (The range is 0 to 65535 in units of 10 ms)
Specifies amount of time allowed for the user to dial a digit after picking up the telephone receiver. (The range is 0 to 65535 in units of 10 ms)
Specifies the amount of time from when the receiver is picked up to the first dialed digit before Warm Line dialing occurs. (The range is 0 to 65535 in units of 10 ms)
50
(that is, 500 ms)
1500
(15 s)
400
(4 s)
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Parameter Description Default
Interdigit (10 ms)
Specifies the amount of time the VoIP Subsystem waits after the dial string has matched a dial pattern. After this amount
500
(5 s)
of time, the VoIP Subsystem will go ahead and dial that number. (The range is 0 to 65535 in units of 10 ms)
Dialing (10 ms)
Specifies the amount of time between digits before a timeout occurs. This may be overridden by the ‘p’ parameter in a
1000
(10 s)
Dial String. (The range is 0 to 65535 in units of 10 ms)
Hangup Disconnect (10 ms)
Specifies the amount of time to wait (after the disconnect command) before transitioning to the standby state.
85
(850 ms)
(The range is 0 to 65535 in units of 10 ms)
Hangup Silence (10 ms)
Used if Hangup Disconnect is not enabled; that is, does not have a value. (The range is 0 to 65535 in units of 10
1000
(10 s)
ms)
No Answer (s)
Relative to call forwarding -- time after
20 s which a call-waiting call is considered to be a No Answer call. After this time the call will be forwarded if Forward on No Answer is enabled. (The range is 0 to 65535 s)
Pause Wait (10 ms)
Time that device will pause when a pause symbol is entered in a string that will be dialed onto the PSTN via the FXO port.
300
(3 s)
(The range is 0 to 65535 in units of 10 ms)
Timeout Tone (10 ms)
If a timeout occurs during dialing or answering, a busy signal is sent to the telephone. The dialing duration specifies
1000
(10 s)
the amount of time to send the busy signal. (The range is 0 to 65535 in units of 10 ms)
Timeout Pause (10 ms)
Specifies the amount of time between the busy and alert tones. (The range is 0 to 65535 in units of 10 ms)
100
(1 s)
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Parameter Description Default
Timeout Disconnect (10 ms)
The range is 0 to 65535 in units of 10 ms
85
(850 ms)
Timeout Warning (10 s)
When the telephone is off hook for too long, the alert tone is sent to the phone. The amount of time for the alert tone is
1
(10 s)
specified by the alert duration. (The range is 0 to 65535 s)
Timeout Hold (10 ms)
When a call is placed on hold, this parameter specifies the amount of time to wait before the call holding tone is played.
1000
(10 s)
(The range is 0 to 65535 in units of 10 ms)
Timeout Hold Drop (10 ms)
Timeout No Answer Drop (s)
Drop a call on hold after this time. (The range is 0 to 65535 in units of 10 ms)
If forwarding is not enabled, an incoming
6000
(60 s)
120 s call-waiting call is dropped after the
specified amount of time. (The range is 0 to 65535 ms)
Call Back (s)
Call Back Retry (s)
Call Back Ring Wait (s)
Message Waiting Refresh (s)
Hookflash Maximum (ms)
Not implemented.
Not implemented.
Not implemented.
Request updates to voice message status at this interval.
Sets the maximum amount of time for the
1800
(30 min)
900 ms telephone receiver to stay on-hook before it is regarded as simply on-hook. If the receiver is on-hook for more than the minimum hook-flash time and less than the maximum hook-flash time, the system recognizes hook-flash. (The range is 0 to 1600 ms.)
Hookflash Minimum (ms)
Sets the minimum amount of time for the
300 ms telephone receiver to stay on-hook in order to be regarded as hook-flash. If the receiver does not stay on-hook for the hookflash minimum time, the VoIP Subsystem does not recognize hookflash as having occurred. (The range is 0 to 4150 ms.)
Hookflash Delay (ms)
Answer Hangup Delay (ms)
The range is 0 to 1000 ms 200 ms
Sets the minimum amount of time for the
0 ms telephone receiver to stay on-hook before the VoIP Subsystem ends the current call. This applies only to incoming calls. (The range is 0 to 60,000 ms)
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10
Feature Codes
Feature codes are used to access advanced Class 5 telephony features. You can use the VoIP -> Advanced VoIP Settings -> Feature Codes menu to configure the parameters. The menu includes:
Feature Code Assignments (*55 – *99)
•

Feature Code Assignments (*55 – *99)

The IPBX calling features are assigned the ranges *55 to *89 and *92 to *99. The codes can be re­assigned to better match common local conventions, but they must be given codes within the assigned ranges. The default values represent the commonly used assignments.
Parameter Description Default
Call Waiting Enable
Call Waiting Disable
Call Trace
Call Waiting Caller ID Enable
Call Waiting Caller ID Disable
Blocked Number Enable
Distinctive Ring Enable
Caller ID Outbound Disable
Priority Forward Enable
Disturb Accept Enable
Caller ID Inbound Enable
Busy Number Redial
Caller ID Outbound One-time Enable
Caller ID Outbound One-time Disable
Caller Redial
Call Waiting One-time Disable
Call Waiting One-time Enable
Call Forward Enable
Call Forward Disable
One Digit Speed Dial Program
Two Digit Speed Dial Program
Block Anonymous Enable
Enable call waiting on all calls *55
Disable call waiting on all calls *56
Call trace (reserved) *57
Enable call waiting caller ID generation *58
Disable call waiting caller ID generation *59
Enable call blocking feature *60
Enable distinctive ringing feature *61
Block caller ID on all outbound calls *62
Enable priority call forwarding feature *63
Enable do not disturb accept call feature *64
Enable caller ID generation *65
Busy number redial *66
Unblock caller ID for one call *67
Block caller ID for one call *68
Call the last caller *69
Deactivate call waiting for current call *70
Enable call waiting for current call *71
Enable call forwarding to number that follows *72
Cancel call forwarding of non-priority calls *73
Program speed dials 0 - 7 *74
Program speed dials 20 - 39 *75
Block all anonymous calls *77
Chapter 10: Feature Codes 49
Page 50
Parameter Description Default
Do Not Disturb Enable
Do Not Disturb Disable
Blocked Number Disable
Enter do not disturb state *78
Exit do no disturb state *79
Cancel call lock - remove optional number
*80
from blocked call list, or disable call blocking
Distinctive Ring Disable
Caller ID Outbound Enable
Priority Forward Disable
Disturb Accept Disable
Caller ID Inbound Disable
Busy Number Redial Cancel
Disable distinctive ringing *81
Unblock caller ID on all outbound calls *82
Cancel priority call forward *83
Disable do not disturb accept call feature *84
Disable caller ID generation *85
Cancel busy redial *86
Block Anonymous Disable
Caller Redial Cancel
Forward No Answer Enable
Forward No Answer Disable
Forward Busy Enable
Forward Busy Disable
Outgoing Block Enable
Outgoing Block Disable
Unattended Transfer
Unblock anonymous calls *87
Cancel calling last caller *89
Call forward when no answer - number follows *92
Cancel call forward when no answer *93
Call forward when busy - number follows *94
Cancel call forward when busy *95
Enable Block Outgoing VoIP calls feature *96
Disable Block Outgoing VoIP calls feature *97
Execute Hook Flash followed by *98 to initiate unattended transfer
*98
50 X6v VoIP Features Technical Reference
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NOTICE

This document contains proprietary information protected by copyright, and this Manual and all the accompanying hardware, software, and documentation are copyrighted. No part of this document may be photocopied or reproduced by mechanical, electronic, or other means in any form.
The manufacturer does not warrant that the hardware will work properly in all environments and applications, and makes no warranty or representation, either expressed or implied, with respect to the quality, performance, merchantability, or fitness for a particular purpose of the software or documentation. The manufacturer reserves the right to make changes to the hardware, software, and documentation without obligation to notify any person or organization of the revision or change.
All brand and product names are the trademarks of their respective owners.
© Copyright 2009 All rights reserved
0979-B 27592 ©2009
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