Create
Use the following procedure to Create Call Features Profile:
- Select the Call Features tab and click the Create a New Profile (
) icon in the Main Menu bar. The Create Call Features Profile dialog is displayed.

- Enter a Profile Name for this profile, then set the profile values in the Create Call Features Profile dialog, refer to the Call Features Attributes table below and fill in the fields as needed to create the Call Features profile.
Parameter
Description
Default
Profile Name
Each profile must have a unique profile name. This value is user-defined; characters can be alphabetic, numeric, a hyphen (-), or an underscore (_). Blank spaces between characters are allowed. Up to 40 characters may be entered.
Note: This attribute cannot be edited after the profile has been created.
N/A
Description
If a profile is given a description, the description must be unique. This value is user-defined; characters can be alphabetic, numeric, a hyphen (-), or an underscore (_). Blank spaces between characters are allowed.
N/A
Soft Switch
Name
This field identifies the Softswitches identified in "Add/Remove Softswitch Names".
N/A
Abbr
This field identifies the short form name (abbreviation) of the Softswitches identified in Add/Remove Softswitch Names.
N/A
Version
Optional field identifying a software version for the selected softswitch. Range from 0 to 32 characters.
Note: In the future when differences appear in a particular softswitch’s behavior, Tellabs provides the proper setting for this field.
N/A
Digit Plan
Digit Map
A Digit Map can be entered directly into the Digit Map field or created with an ACSII text editor, stored on the server, and loaded into the Digit Map field (refer to "Loading a DigitMap").
Note: Due to the complexity of a Digit Map, it is highly recommended to create the Digit Map offline, store it on the server, and then load the map using the Load button. The default location for Dial Plans is c:TellabsPanoramaPONbbmgrserverdataTConfigSSDialPlan for Windows.
Refer to the Application Note, Configuring Softswitch Voice, ENG-010473; Call Features Profile for all ONTs for examples of dial plan formats and 7/10 digit standard dialing for business enterprises.
When a user goes off hook and starts to dials digits, the line’s digit map is consulted to determine when dialing is complete.
The digit map can also identify:
- the dialing of an emergency call (for example, 911).
- the appropriate locations to apply dial tone or recall dial tone while collecting digits on behalf of the network.
- the dialing of a Vertical Service Code (VSC) for activation of a feature that is implemented locally. This feature must also be enabled through configuring related fields.
- the proper location to insert a character before forwarding digits towards the network. This usually relates to inserting a '#' character when collecting VSCs and subsequent digits on behalf of the network.
A digit map is a regular expression that lists all possible valid dialed digit sequences allowed in the dial plan. It consists of a set of digit pattern alternatives (separated by a vertical pipe ('|')). For example, 411|911, allows the dialing of 411 or 911.
A Digit Map can contain up to 4k characters. For more information on Digit Map syntax and allowable characters.
N/A
Call Waiting Enable CW
This field allows a line to support residential call waiting. An incoming call to an in-use line results in a call waiting tone. This tone repeats once (10 seconds later) if the incoming call is still present, and the user does not hook-flash to switch to the call.
yes - enables the feature no - disables the feature
yes
Business CW
This field allows a line to support business group call waiting.
Business group call waiting differs from residential call waiting in that only one call waiting tone is played on the line when an incoming call occurs (versus two tones played 10 seconds apart), and if the user hangs up with the call waiting call still present, the line only rings for six cycles versus an unlimited number (refer to "enable_cw").
- yes - enables the feature
- no - disables the feature
No
Service Code CCW
This field allows a line to support the cancel call waiting feature. If the user dials the Vertical Service Code (VSC) configured by this field, call waiting service is disabled for the duration of the call.
The field contains up to32 characters consisting of one or more vertical service codes (VSCs) separated by whitespace, typically the pulse dial and DTMF version. If this field is blank, the user cannot invoke the service.
Note: Operation of features implemented locally on the Eqpt that the user invokes via VSCs must have corresponding entries found in the digit map to allow their entry and identification. For more information, refer to the "digit_map" description.
*70 1170
Call Transfer Enable Transfer
This field allows a line to act as the controller of a call transfer. Similar to residential three-way calling (refer to "enable_3wc"), the controller is able to form a three-way conference. The mixing of the three (local or remote) audio paths occurs on the Eqpt. The difference occurs when the controller hangs up. The first party the controller contacted is transferred to the second party (as opposed to all parties being released).
The transfer can take place at various stages of the second call. If the controller hangs up when the second call is ringing, it is a "blind transfer" (or more technically, for SIP, a "consultative transfer while ringing"). If a user answers the second call leg but they have not been made part of the conference, a transfer at this point would be considered a "consultative transfer while answered."The third form of transfer occurs during the three-way conference.
- yes - enables the feature
- no - disables the feature
Note 1: enable_transfer overrides the behavior of residential three-way calling (refer to "enable_3wc").
Note 2: To allow a second call leg to be initiated on a line, it is often necessary to turn this capability ON on the softswitch.
No
Enable Hold
This field allows a line to implement the additional call hold option of call transfer.
Call hold allows the user to put a call on hold by dialing a Vertical Service Code (VSC) and to later retrieve the call using the same VSC. While the call is on hold, the user may make other calls or transfer other calls without affecting the held call.
- yes - enables the feature
- no - disables the feature
Note 1: Define and align the service_code_hold field with the digit_map.
Note 2: To allow a second call leg to be initiated on a line, it is often necessary to turn this capability ON on the softswitch.
No
Service Code Hold
This field allows a line to implement the additional call hold option of call transfer. Call hold allows the user to put a call on hold by dialing a Vertical Service Code (VSC) and to later retrieve the call using the same VSC. While the call is on hold, the user may make other calls or transfer other calls without affecting the held call.
The field contains up to two service codes separated by white space. Typically, these include a DTMF and dial-pulse variation for the service code. If this field is blank, the user cannot invoke the service.
Note 1: The enable_hold field must be enabled for this feature to operate properly.
Note 2: Operation of features implemented locally on the Eqpt that the user invokes via VSCs must have corresponding entries found in the digit_map to allow their entry and identification. For more information, refer to "digit_map".
Note 3: To allow a second call leg to be initiated on a line, it is often necessary to turn this capability ON on the softswitch.
*52 1152
Call Party Identification Enable Caller ID
This field allows calling party identification (number and name) to be displayed. The information displays both regular incoming calls and call waiting calls.
The sending of calling party information to the Eqpt is often controlled through provisioning on the softswitch. This field acts as a second level filter in that, in the SIP protocol, the information can be derived from one of two places: the "From" header or "P-Asserted-Identity" header. The From header is always present. If a softswitch relies on the "From" header to transport calling party information to the Eqpt, this field indicates whether the contents of the "From" header should be displayed on the line.
The typical setting for this field is to disable it for the NORTEL_CS2K as it uses the P-Asserted-Identity header, and it does not send this header if it does not want the data to display.
For other softswitches, enable or disable the field based on whether calling party information should be displayed.
yes - enables the feature
no - disables the featureYes
Three-way Calling Enable 3WC
This field allows a line to act as the controller of a residential three-way call. The mixing of the three (local or remote) audio paths occurs on the Eqpt. The controller manages this feature via hook-flash signals to initiate a second call leg and then conference all parties. A further hook-flash drops the last party the controller added to the conference. If the controller hangs up, all parties are released.
yes - enables the feature
no - disables the featureNote 1: If the enable_transfer parameter (call transfer variant of conferencing) is enabled, it overrides the behavior of enable_3wc.
Note 2: To allow a second call leg to be initiated on a line, it is often necessary to turn this capability ON at the softswitch.
No Message Waiting Indication Message Event URI
Message Waiting Indications (MWI) notify a line that a message is waiting in its network-provided voice mail account. If the line has to subscribe via the SIP protocol to learn of notifications, this field provides the Uniform Resource Identifier (URI) for the subscription.
At this time, all softswitch choices that support voice mail autonomously send notifications to the line without prior subscription. The softswitch configuration decides whether the line supports MWI or not; therefore, this field is usually not set.
A standard SIP URI (refer to section 19.1 of IETF RFC3261) (sip:<user>@<host>)
Example: "sip:2125550102@provider.com"
The Message Event URI can consist of 0 to 256 characters.
N/A
Message Waiting Mode
Message Waiting Indications (MWI) notify a line that a message is waiting in its network-provided voice mail account. This field determines the means of alerting the user.
Options include none or any combination of dial-tone, visual, and reminder-ring is acceptable. If the field is not set but message_event_uri is set, dial-tone is used.
none
dial-tone - provides a stutter dial tone (10 stutters) in place of the regular dial tone when the user goes off-hook and a message is present.
visual - lights up a phone’s visual message waiting indicator when a message is present (refer to "vmwi_abbreviated_ring" and "vmwi_refresh_interval").
reminder-ring - provides a periodic reminder ring (actually ring splash) on the line to indicate that a message is present.
dial-tone
VMWI Refresh Interval
When the line is configured to use a "visual" indication (refer to "message_waiting_mode") to notify the user of a waiting message, this field indicates how often the visual message waiting status should refresh on the line. If a phone is unplugged when a message waiting indicator is sent on the line, this mechanism allows the phone’s visual message waiting indicator to eventually refresh to the correct network state.
The value indicates the time in minutes, 0 to 65535, between periodic refreshes. If the value is zero, there are no refreshes.
30 VMWI Abbreviated Ring
When the line is configured to use a "visual" indication (refer to "message_waiting_mode") to notify the user of a waiting message, this field indicates if the abbreviated ring option should be used. The abbreviated ring option applies a brief ring-splash to the line when the user goes on-hook if a new message arrived while they were off-hook. This action is in addition to lighting the phone’s visual message waiting indicator.
- yes - enables the feature
- no - disables the feature
No MWI Reminder Burst
When the line is configured to use a "reminder-ring" (refer to "message_waiting_mode") to notify the user of a waiting message, this field indicates the length of the periodic ring-splash.
The value represents the length of the ring-splash in milliseconds, 0 to 2000.
500 MWI Reminder Interval
When the line is configured to use a "reminder-ring" (refer to "message_waiting_mode") to notify the user of a waiting message, this field indicates how often the periodic ring-splash occurs.
The value represents the period between ring-splashes in minutes, 0 to 65535.
15 Hotline Hotline Mode - Hotline Disabled (default)
- Static Hotline
Hotline Disabled SIP URI
Active if Hotline Mode set to Static Hotline. Allows the line to act as a direct connection (or hotline). When the user goes offhook, a call is directed toward the configured SIP URI, without providing dial tone or collecting digits, unless a Hotline Timeout greater than 0 is set.
Syntax: sip:[user]@[host], e.g., sip:2125557777@provider.com.
N/A
Hotline Timeout
Active if Hotline Mode is set to Static Hotline. If set to a number greater than 0, the system waits for the specified timeout. If digits are dialed prior to the specified timeout, the call proceeds as normal. If no digits are dialed prior to the specified timeout, the system initiates a call to the programmed SIP URI. The value represents the length of the timeout in seconds, 0 to 9. N/A
Home Intercom Intercom Code 1
Home Intercom is a service provided wholly within the Eqpt (without softswitch interaction). A user picks up the line, dials one of the two home intercom Vertical Service Codes (VSCs), and hangs up. The line rings in one of the two distinctive patterns (based on which VSC the user entered) and when the ringing stops, the user goes off-hook as someone else on the same phone line has gone off-hook. The two ends are now connected.
This field represents one of the two VSCs which are associated with a particular distinctive ring.
The field contains up to two service codes that are separated by white space. Typically, these include a DTMF and dial-pulse variation for the service code. If this field is blank, the user cannot invoke the service.
Note: Operation of features implemented locally on the Eqpt that the user invokes via VSCs must have corresponding entries found in the digit_map to allow their entry and identification. For more information, refer to "digit_map".
*53 1153
Intercom Code 2
Refer to "intercom_code_1".
*54 1154
Call Forwarding Status Indications CFWD State
Nortel Softswitch only.
This field enables a call forwarding status indication of "stutter dial tone" on the line (3 stutters) when a user goes off-hook and call forwarding is active on the line.
- Yes - Enables the feature
- No - Disables the feature
No
CFWD Reminder
Nortel Softswitch only.
This field enables a call forwarding status indication of a "splash ring" on the line when the network forwards an incoming call.
- Yes - Enables the feature
- No - Disables the feature
No
CFWD Event URI
Nortel Softswitch only.
This application supports two call forwarding status indications. One provides a "stutter dial tone" when a user goes off-hook on a line with call forwarding activated (refer to "cfwd_state"). The other provides a "splash ring" on the line when an incoming call is forwarded (refer to "cfwd_reminder").
This field provides a SIP URI to which the port subscribes via SIP to learn changes in call forwarding activation (refer to "cfwd_state") or of redirected calls (refer to "cfwd_reminder"). If this field (typical case) is not configured, but cfwd_state and/or cfwd_reminder is enabled, the port’s Address of Record (refer to "address_of_record_uri") is used in place of this field.
A standard SIP URI (refer to section 19.1 of IETF RFC 3261)
Example: sip: 2125550102@provider.com
N/A
Suppressed Ringing/Telemetry Enable Supp Ring
Nortel Softswitch only.
Suppressed ringing allows incoming calls to be made to a line. The line does not ring, rather the incoming call activates automated equipment on the line to go off-hook to communicate with the calling equipment. Telemetry, used by utilities to read their meters, is one application of this service.
This field enables the suppressed ringing service on the line.
- Yes - Enables the feature
- No - Disables the feature
No
Supp Ring Offhook Timer
Nortel Softswitch only.
Suppressed ringing allows incoming calls to be made to a line. The line does not ring, rather, the incoming call activates automated equipment on the line to go off-hook to communicate with the calling equipment. Telemetry, used by utilities to read their meters, is one application of this service.
When a suppressed ringing call is presented to the line, this timer starts, and the line waits for the automated equipment to go off-hook. The call is rejected if this timer expires before the equipment goes off-hook.
This value is the time interval in seconds to wait for the automated equipment to go off-hook.
5 Supp Ring Interrupt Timer
Nortel Softswitch only.
Suppressed ringing allows incoming calls to be made to a line. The line does not ring, rather, the incoming call activates automated equipment on the line to go off-hook to communicate with the calling equipment. Telemetry, used by utilities to read their meters, is one application of this service.
If the suppressed ringing call is interruptible (as specified in the incoming call’s SIP protocol), a new incoming call triggers the automated equipment to go on-hook. This field determines the length of time that the Eqpt waits for the automated equipment to go on-hook before it must reject the new incoming call.
This value is the time interval in seconds to wait for the automated equipment to go on-hook.
2 Emergency Call Enable Forced Hold
Nortel Softswitch only.
A line’s digit map (refer to "digit_map"), if correctly configured, has the ability to determine if the user has made an emergency call (for example, 911). If the Eqpt knows the line is involved in an emergency call, a number of call feature behaviors occur (For example, a user cannot use hook-flash to put the 911 operator on hold).
This field controls when a user hangs up an emergency call when the operator is still present, the Eqpt puts the line on hold as opposed to releasing the network connection. If the user goes back off-hook, they remain connected to the operator as opposed to having to reconnect to the operator (a side-effect of SIP providing local dial tone and collecting digits on standard calls).
Whether this feature is appropriate depends on the capabilities of the particular softswitch and whether the emergency operator can maintain the connection.
- Yes - Enables the feature
- No - Disables the feature
No
Bridge Lines Bridge Lines URI
Nortel Softswitch only.
A bridged line group allows two or more ports (on one or more devices) to act as if they share the same phone line. Incoming calls alert all ports in the group, any one of which may answer. If one port in the group is involved in a call, the other ports may join the call by going off-hook. Tellabs customized this field to the implementation of the NORTEL_CS2K softswitch. Other softswitches may support a similar feature in a different manner.
If this field is configured, the port is a member of a bridged line group. The value of the field acts as the SIP Address of Record (AOR - public identity) for the bridged line group. The AOR typically contains the Directory Number (DN) for the bridged line group, but it does not have to be a DN depending on the softswitch configuration.
A standard SIP URI (refer to section 19.1 of IETF RFC 3261)
Example: sip:2125550102@provider.com
N/A
Bridge Line Dialog URI
Nortel Softswitch only.
A bridged line group allows two or more ports (on one or more devices) to act as if they share the same phone line. Incoming calls alert all ports in the group, any one of which may answer. If one port in the group is involved in a call, other ports may join the call by going off-hook. Tellabs customized this field to the implementation of the NORTEL_CS2K softswitch. Other softswitches may support a similar feature in a different manner.
If the related bridged_line_uri field is configured, the port is a member of a bridged line group. The port must use a SIP subscription mechanism to be told the current state of the bridged line group. The bridged_line_dialog_uri field determines the address where the subscription is sent.
If this field is not configured, the related bridged_line_uri is used (the typical case). A standard SIP URI (refer to section 19.1 of IETF RFC 3261)
Example: sip:2125550111@provider.com
N/A
Multi-way Call Enable Multiway
Nortel Softswitch only.
A multi-way call is one where the local port, acting as the conference controller, forms a conference hosted on an off-board conference server. When calling a potential participant, the controller may add or drop the participant by initiating a hook-flash and dialing either an add or release Vertical Service Code (a feature code, refer to "service_code_multiway" and "service_code_multiway_rls"). The network limits the conference size (usually to six participants).
This field enables the line to act as a multi-way call controller.
- Yes - enables the feature
- No - disables the feature
Note 1: The conference_uri, service_code_multiway and service_code_multiway_rls fields must be defined and aligned with the digit_map.
Note 2: To allow a second call leg to be initiated on a line, it is often necessary to turn this capability ON at the softswitch.
No
Service Code Multiway RLS
Nortel Softswitch only.
A multi-way call is one where the local port, acting as the conference controller, forms a conference hosted on an off-board conference server. When calling a potential participant, the controller may add or drop the participant by initiating a hook-flash and dialing either an add or release Vertical Service Code (VSC) (refer to "service_code_multiway" and "service_code_multiway_rls"). The network limits the conference size (usually to six participants).
This field configures the multi-way "release" VSC to allow the controller to release a consultative call without joining the called party into the conference.
The field contains up to two service codes separated by white space. Typically, these include a DTMF and dial-pulse variation for the service code. If this field is blank, the user cannot invoke the service.
Note 1: The enable_multiway, conference_uri, and service_code_multiway_rls fields must be enabled for the feature to operate properly.
Note 2: Operation of features implemented locally on the Eqpt that the user invokes via VSCs must have corresponding entries found in the digit_map to allow their entry and identification. For more information, refer to "digit_map".
Note 3: To allow a second call leg to be initiated on a line, it is often necessary to turn this capability ON at the softswitch.
*43 1143
Service Code Multiway
Nortel Softswitch only.
A multi-way call is one where the local port, acting as the conference controller, forms a conference hosted on an off-board conference server. When calling a potential participant, the controller may add or drop the participant by initiating a hook-flash and dialing either an add or release Vertical Service Code (VSC) (refer to "service_code_multiway" and "service_code_multiway_rls"). The network limits the conference size (usually to six participants). This field configures the multi-way "add" VSC to allow the user to add a participant to the conference. The field contains up to two service codes separated by white space. Typically, these include a DTMF and dial-pulse variation for the service code. If this field is blank, the user cannot invoke the service.
The enable_multiway, conference_uri, and service_code_multiway_rls fields must be enabled for the feature to operate properly.
Operation of features implemented locally on the Eqpt that the user invokes via VSCs must have corresponding entries found in the digit_map to allow their entry and identification.
For more information, refer to "digit_map".
To allow a second call leg to be initiated on a line, it is often necessary to turn this capability ON at the softswitch.
*41.1141
Conference URI
Nortel Softswitch only.
A multi-way call is one where the local port, acting as the conference controller, forms a conference hosted on an off-board conference server. When calling a potential participant, the controller may add or drop the participant by initiating a hook-flash and dialing either an add or release Vertical Service Code (that is, feature code, refer to "service_code_multiway" and "service_code_multiway_rls"). The network limits the conference size (usually to six participants).
This field represents the SIP Uniform Resource Identifier (URI) to which participants are redirected to join the off-board conference (the conference server).
Note: Enable_multiway, service_code_multiway, and service_code_multiway_rls fields must be enabled for the feature to operate properly.
A standard SIP URI (refer to section 19.1 of IETF RFC 3261)
Example: sip:conference@softswitch1.provider.com
N/A
- When all the fields have been filled in, click on the Apply button.
- The Create Call Features Profiles dialog remains open. If no additional profiles are to be created, click the Cancel button. Otherwise, repeat the same procedure to create another Call Features profile.
-
If no further profiles are to be created, click on the Cancel button to close the Create Call Features Profile dialog.
- Click on the Close button again to exit the Profiles display dialog.