Cisco IP Conference Phone Features and Setup

Phone Features and Setup Overview

After you install Cisco IP Phones in your network, configure their network settings, and add them to Third-Party Call Control System, you must use the Third-Party Call Control System to configure telephony features, optionally modify phone templates, set up services, and assign users.

You can modify additional settings for the Cisco IP Phone from Third-Party Call Control Configuration Utility. Use this web-based application to set up phone registration criteria and calling search spaces, to configure corporate directories and services, and to modify phone button templates, among other tasks.

Cisco IP Phone User Support

If you are a system administrator, you are likely the primary source of information for Cisco IP Phone users in your network or company. It is important to provide current and thorough information to end users.

To successfully use some of the features on the Cisco IP Phone (including Services and voice message system options), users must receive information from you or from your network team or must be able to contact you for assistance. Make sure to provide users with the names of people to contact for assistance and with instructions for contacting those people.

We recommend that you create a web page on your internal support site that provides end users with important information about their Cisco IP Phones.

Consider including the following types of information on this site:

  • User guides for all Cisco IP Phone models that you support

  • List of features supported

  • User guide or quick reference for your voicemail system

Telephony Features for Cisco IP Phone

After you add Cisco IP Phones to Third-Party Call Control system, you can add functionality to the phones. The following table includes a list of supported telephony features, many of which you can configure by using Third-Party Call Control system.


Note

The Third-Party Call Control system also provides several service parameters that you can use to configure various telephony functions.

Feature Description and More Information

AES 256 Encryption Support for Phones

Enhances security by supporting TLS 1.2 and new ciphers.

Alphanumeric Dialing

Allows users to place a call with alphanumeric characters. You can use these characters for alphanumeric dialing: a-z, A-Z, 0-9, -, _, ., and +.

Any Call Pickup

Allows users to pick up a call on any line in their call pickup group, regardless of how the call was routed to the phone.

Auto Answer

Connects incoming calls automatically after a ring or two.

Auto Answer works with either the speakerphone or the headset.

Blind Transfer

Blind Transfer: This transfer joins two established calls (call is in hold or in connected state) into one call and drops the feature initiator from the call. Blind Transfer does not initiate a consultation call and does not put the active call on hold.

Some JTAPI/TAPI applications are not compatible with the Join and Blind Transfer feature implementation on the Cisco IP Phone and you may need to configure the Join and Direct Transfer Policy to disable join and direct transfer on the same line or possibly across lines.

Call Back

Provides users with an audio and visual alert on the phone when a busy or unavailable party becomes available.

Call Display Restrictions

Determines the information that will display for calling or connected lines, depending on the parties who are involved in the call. RPID and PAID caller id handling are supported.

Call Forward

Allows users to redirect incoming calls to another number. Call Forward options include Call Forward All, Call Forward Busy, Call Forward No Answer.

Call Forward Notification

Allows you to configure the information that the user sees when receiving a forwarded call.

Call History for Shared Line

Allows you to view shared line activity in the phone Call History. This feature will:

  • Log missed calls for a shared line

  • Log all answered and placed calls for a shared line

Call Park

Allows users to park (temporarily store) a call and then retrieve the call by using another phone.

Call Pickup

Allows users to redirect a call that is ringing on another phone within their pickup group to their phone.

You can configure an audio and visual alert for the primary line on the phone. This alert notifies the users that a call is ringing in their pickup group.

Call Waiting

Indicates (and allows users to answer) an incoming call that rings while on another call. Incoming call information appears on the phone display.

Caller ID

Caller identification such as a phone number, name, or other descriptive text appear on the phone display.

Caller ID Blocking

Allows a user to block their phone number or name from phones that have caller identification enabled.

Calling Party Normalization

Calling party normalization presents phone calls to the user with a dialable phone number. Any escape codes are added to the number so that the user can easily connect to the caller again. The dialable number is saved in the call history and can be saved in the Personal Address Book.

Conference

Allows a user to talk simultaneously with multiple parties by calling each participant individually.

Allows a noninitiator in a standard (ad hoc) conference to add or remove participants; also allows any conference participant to join together two standard conferences on the same line.

Note   

Be sure to inform your users whether these features are activated.

Configurable RTP/sRTP Port Range

Provides a configurable port range (2048 to 65535) for Real-Time Transport Protocol (RTP) and secure Real-Time Transport Protocol (sRTP).

The default RTP and sRTP port range is 16384 to 16538.

You configure the RTP and sRTP port range in the SIP Profile.

Directed Call Pickup

Allows a user to pick up a ringing call on a DN directly by pressing the GPickUp softkey and entering the directory number of the device that is ringing.

Divert

Allows a user to transfer a ringing, connected, or held call directly to a voice-messaging system. When a call is diverted, the line becomes available to make or receive new calls.

Do Not Disturb (DND)

When DND is turned on, either no audible rings occur during the ringing-in state of a call, or no audible or visual notifications of any type occur.

Headset Sidetone Control

Allows an administrator to set the sidetone level of a wired headset.

Group Call Pickup

Allows a user to answer a call that is ringing on a directory number in another group.

Hold Status

Enables phones with a shared line to distinguish between the local and remote lines that placed a call on hold.

Hold/Resume

Allows the user to move a connected call from an active state to a held state.

  • No configuration required unless you want to use Music On Hold. See "Music On Hold" in this table for information.

  • See "Hold Reversion" in this table.

HTTP Download

Enhances the file download process to the phone to use HTTP by default. If the HTTP download fails, the phone reverts to using the TFTP download.

HTTPS for Phone Services

Increases security by requiring communication using HTTPS.

Note   

When the web is in HTTPS mode, the phone is a HTTPS servers.

Improve Caller Name and Number Display

Improves the display of caller names and numbers. If the Caller Name is known then the Caller Number is displayed instead of unknown.

Jitter Buffer

The Jitter Buffer feature handles jitter from 10 milliseconds (ms) to 1000 ms for both audio and video streams.

Join Across Lines

Allows users to combine calls that are on multiple phone lines to create a conference call.

Some JTAPI/TAPI applications are not compatible with the Join and Direct Transfer feature implementation on the Cisco IP Phone and you may need to configure the Join and Direct Transfer Policy to disable join and direct transfer on the same line or possibly across lines.

Join

Allows users to combine two calls that are on one line to create a conference call and remain on the call.

Message Waiting

Defines directory numbers for message waiting on and off indicators. A directly-connected voice-message system uses the specified directory number to set or to clear a message waiting indication for a particular Cisco IP Phone.

Message Waiting Indicator

A light on the handset that indicates that a user has one or more new voice messages.

Minimum Ring Volume

Sets a minimum ringer volume level for an IP phone.

Missed Call Logging

Allows a user to specify whether missed calls will be logged in the missed calls directory for a given line appearance.

Multicasting Paging

Enables users to page some or all phones. If the phone is on an active call while a group page starts, the incoming page is ignored.

Multiple Calls Per Line Appearance

Each line can support multiple calls. By default, the phone supports two active calls per line, and a maximum of ten active calls per line. Only one call can be connected at any time; other calls are automatically placed on hold.

The system allows you to configure maximum calls/busy trigger not more than 10/6. Any configuration more than 10/6 is not officially supported.

Music On Hold

Plays music while callers are on hold.

Mute

Mutes the handset or headset microphone.

No Alert Name

Makes it easier for end users to identify transferred calls by displaying the original caller’s phone number. The call appears as an Alert Call followed by the caller’s telephone number.

Onhook Dialing

Allows a user to dial a number without going off hook. The user can then either pick up the handset or press Dial.

Pause in Speed Dial

Users can set up the speed-dial feature to reach destinations that require Forced Authorization Code (FAC) or Client Matter Code (CMC), dialing pauses, and additional digits (such as a user extension, a meeting access code, or a voicemail password) without manual intervention. When the user presses the speed dial, the phone establishes the call to the specified DN and sends the specified FAC, CMC, and DTMF digits to the destination and inserts the necessary dialing pauses.

Plus Dialing

Allows the user to dial E.164 numbers prefixed with a plus (+)  sign.

To dial the + sign, the user needs to press and hold the star (*) key for at least 1 second. This applies to dialing the first digit for an on-hook (including edit mode) or off-hook call.

Power Negotiation over LLDP

Allows the phone to negotiate power using Link Level Endpoint Discovery Protocol (LLDP) and Cisco Discovery Protocol (CDP).

Problem Reporting Tool

Submits phone logs or reports problems to an administrator.

Programmable Feature Buttons

You can assign features, such as New Call, Call Back, and Forward All to line buttons.

Redial

Allows users to call the most recently dialed phone number by pressing a button or the Redial softkey.

Remote Customization (RC)

Allows a service provider to customize the phone remotely. There is no need for either the service provider to physically touch the phone or a user to configure the phone. The service provider can work with a sales engineer at the time of ordering to set this up.

Ringtone Setting

Identifies ring type used for a line when a phone has another active call.

RTCP Hold For SIP

Ensures that held calls are not dropped by the gateway. The gateway checks the status of the RTCP port to determine if a call is active or not. By keeping the phone port open, the gateway will not end held calls.

Serviceability for SIP Endpoints

Enables administrators to quickly and easily gather debug information from phones.

This feature uses SSH to remotely access each IP phone. SSH must be enabled on each phone for this feature to function.

Shared Line

Allows a user with multiple phones to share the same phone number or allows a user to share a phone number with a coworker.

Show Calling ID and Calling Number

The phones can display both the calling ID and calling number for incoming calls. The IP phone LCD display size limits the length of the calling ID and the calling number that display.

The Show Calling ID and Calling Number feature applies to the incoming call alert only and does not change the function of the Call Forward and Hunt Group features.

See "Caller ID" in this table.

Show Duration for Call History

Displays the time duration of placed and received calls in the Call History details.

If the duration is greater than or equal to one hour, the time is displayed in the Hour, Minute, Second (HH:MM:SS) format.

If the duration is less than one hour, the time is displayed in the Minute, Second (MM:SS) format.

If the duration is less than one minute, the time is displayed in the Second (SS) format.

Speed Dial

Dials a specified number that has been previously stored.

Time Zone Update

Updates the Cisco IP Phone with time zone changes.

Transfer

Allows users to redirect connected calls from their phones to another number.

Some JTAPI/TAPI applications are not compatible with the Join and Direct Transfer feature implementation on the Cisco IP Phone and you may need to configure the Join and Direct Transfer Policy to disable join and direct transfer on the same line or possibly across lines.

Voice Message System

Enables callers to leave messages if calls are unanswered.

Web Access Enable by Default

Web services are enabled by default.

Feature Buttons and Softkeys

The following table provides information about features that are available on softkeys, features that are available on dedicated feature buttons, and features that you need to configure as programmable feature buttons. An "X" in the table indicates that the feature is supported for the corresponding button type or softkey. Of the two button types and softkeys, only programmable feature buttons require configuration in Cisco IP Phone administration.

Table 1 Features with Corresponding Buttons and Softkeys 

Feature Name

Dedicated Feature Button

Programmable Feature Button

Softkey

Answer

X

X

Call Forward All

X

X

Call Park

X

X

Call Park Line Status

X

Call Pickup (Pick Up)

X

X

Call Pickup Line Status

X

Conference

X

X (only displayed during connected call conference scenario)

Divert

X

Do Not Disturb

X

X

Hold

X

X

Mute

X

PLK Support for Queue Status

X

X

Redial

X

X

Speed Dial

X

X

Speed Dial Line Status

X

Transfer

X

X (only displayed during connected call transfer scenario)

Configure a Speed Dial with the Configuration Utility Page

You can configure speed dials on the phone with the web interface.

Procedure
    Step 1   On the Configuration Utility page, select Admin Login > Voice > User.
    Step 2   In the Speed Dial section, enter a name and number that corresponds to the speed dial entry.
    Step 3   Click Submit All Changes.

    Speed Dial

    Parameter

    Description

    Speed Dial Name

    Indicates the name given to the speed dial.

    Speed Dial Number

    Indicates the number allocated to the speed dial.

    Enable Conference Button with a Star Code

    You can add a star code to the Conference button so that your user can press the button only once to add many active calls to a conference. You can enable this feature from the phone web page.
    Before You BeginThe phone server must suppport this feature.
    Procedure
      Step 1   On the phone web page, select Admin Login > Advanced > Voice > Ext(n), where n is an extension number.
      Step 2   In the Call Features Settings section, select Yes for the Conference Single Hardkey field, enter a star code in the Conference Bridge URL, and press Submit All Changes. For example, you can enter *55 to represent the conference bridge URL of a telecom service provider.

      You can also enable conference button with a xml file. Enter a string in this format: 

      <Conference_Bridge_URL_1_ ua="na">*55</Conference_Bridge_URL_1_>
      <Conference_Single_Hardkey_1_ ua="na">Yes</Conference_Single_Hardkey_1_>

      Configure Alphanumeric Dialing

      You can configure a phone so that the user of the phone can make a call by dialing alphanumeric characters instead of dialing only digits. In the configuration utility page, you can configure alphanumeric dialing with speed-dial, blf, and call pickup.
      Procedure
        Step 1   On the Configuration Utility page, select Admin Login > Advanced > Voice > Ext.
        Step 2   In the Enable URI Dialing 1, select Yes to enable alphanumeric dialing.

        In the phone page, you can add a string on a line key in this format to enable speed dial with alphanumeric dialing capability:

        fnc=sd;ext=xxxx.yyyy@$PROXY;nme=yyyy,xxxx

        For example:

        fnc=sd;ext=first.last@$PROXY;nme=Last,First

        The above example will enable the user to dial "first.dial" to make a call.

        Note   

        The supported characters that you can use for alphanumeric dialing are a-z, A-Z, 0-9, -, _, ., and +.

        .
        Step 3   Click Submit All Changes.

        Configure a Paging Group (Multicast Paging)

        You can configure multicast paging so that users can page all the phones at once or page a group of phones without involving a server. On the Configuration Utility page, you configure a phone as a part of a paging group and can subscribe them to the same multicast address. This enables users to direct pages to specific groups of phones. When you assign each paging group with a unique number, the user dials the paging group number to start paging. All phones that are subscribed to the same multicast address (also configured on the Configuration Utility page) receive the page. The user hears a paging tone of three short beeps when there is an incoming paging call.

        Keep these things in mind:

        • Your network must support multicasting so that all devices in the same paging group are able to join the corresponding multicast group.

        • If the phone is on an active call when a group page starts, the incoming page is ignored.

        • Group paging is one way and uses the G711 codec. The paged phone can only listen to the call from the originator.

        • Incoming pages are ignored when DND is enabled.

        • When paging occurs, the speaker on the paged phones automatically powers on unless the handset or the headset is in use.

        • If the phone is on an active call when a group page starts, the incoming page is ignored. When the call ends, the page is answered, if the page is active.

        • When multiple pages occur, the pages are answered in chronological order. Until the active page ends, the next page is not answered.

        Procedure
          Step 1   On the Configuration Utility page, select Admin Login > Advanced > Voice > Phone.
          Step 2   In the Multiple Paging Group Parameters section, enter a string in the Group Paging Script field in this format:

          pggrp=multicast-address:port;[name=xxxx;]num=yyy;[listen={yes|no}]];

          where:

          • multicast-address = Multicast IP address of the phone that listens for and receives pages.

          • port = Port on which to page; you must use different ports for each paging group.

          • name (optional) = xxxx is the name of the paging group. Replace xxxx with a name. The name can consist maximum of 64 characters.

          • num= yyy is a unique number that the user dials to access the paging group. Replace yyy with a number. The number can consist maximum of 64 characters and the allowed range is 1024 to 32767.

          • listen = Indicates whether the phone listens on the page group. Only the first two groups with listen set to yes listen to group pages. If the field is not defined, the default value is no, so you must set this field to listen to the group pages.

          You can add more paging groups by appending to the configuration string. Here is an example.

          pggrp=224.168.168.168:34560;name=All;num=500;listen=yes;
pggrp=224.168.168.168:34562;name=GroupA;num=501;listen=yes;
pggrp=224.168.168.168:34564;name=GroupB;num=502;
pggrp=224.168.168.168:34566;name=GroupC;num=503;

          This example creates four paging groups: All, GroupA, GroupB, and GroupC. Users dial 500 to send pages to all phones, 501 to send pages to phones configured as part of the GroupA group, 502 to send pages to phones configured as part of the GroupB group, and 503 to send pages to phones configured as part of the GroupC group. The configured phone receives pages directed to the groups All and GroupA.

          Step 3   Click Submit All Changes.

          Configuring Programmable Softkeys

          You can customize the softkeys displayed on the phone. The default softkeys (when the phone is in an idle state) are Redial, Directory, Call Forward, and Do Not Disturb. Other softkeys are available during specific call states (for example, if a call is on hold, the Resume softkey displays).

          Procedure
            Step 1   Click Admin Login > advanced > Voice > Phone
            Step 2   Under Programmable Softkeys, edit the softkeys depending on the call state that you want the softkey to display. For more information, see Programmable Softkeys.

            In the Programmable Softkeys section, each phone state is displayed and the softkeys that are available to display during that state are listed. Each softkey is separated by a semicolon. Softkeys are shown in the format: 

            softkeyname  |[  position  ]

            where softkeyname is the name of the key and position is where the key is displayed on the IP phone screen. Positions are numbered, with position one displayed on the lower left of the IP phone screen, followed by positions two through four. Additional positions (over four) are accessed by pressing the right arrow key on the phone. If no position is given for a softkey, the key will float and appears in the first available empty position on the IP phone screen.

            Step 3   Click Submit All Changes.

            Customize a Programmable Softkey

            The phone provides sixteen programmable softkeys (fields PSK1 through PSK16). You can define the fields by a speed-dial script.

            Procedure
              Step 1   On the Configuration Utility page, select Admin Login > advanced > Voice > Phone.
              Step 2   In the Programmable Softkeys section, set the Programmable Softkey Enable to Yes.
              Step 3   Select a programmable softkey number field on which to configure a phone feature.
              Step 4   Enter the string for the programmable soft key. See the different types of programmable softkeys described in Configure Speed Dial on a Programmable Softkey.
              Step 5   Click Submit All Changes.

              Configure Speed Dial on a Programmable Softkey

              You can configure programmable softkeys as speed dials. The speed dials can be extensions or phone numbers. You can also configure programmable softkeys with speed dials that perform an action that a vertical service activation code (or a star [*] code) defines. For example, if you configure a programmable softkey with a speed dial for *67, the call is placed on hold.

              Procedure
                Step 1   On the Configuration Utility page, select Admin Login > advanced > Voice > Phone.
                Step 2   In the Programmable Softkeys section, set the Programmable Softkey Enable to Yes.
                Step 3   To configure a speed dial PSK, enter the following in the PSK number field:

                fnc=sd;ext=extensionname/starcode@$PROXY;vid=n;nme=name

                Where:

                • fnc= function of the key (speed dial)

                • extensionname=extension being dialed or the star code action to perform

                • vid= n is the extension that the speed dial will dial out

                • name is the name of the speed dial being configured

                Note   

                The name field displays on the softkey on the IP phone screen. We recommend a maximum of 10 characters for a phone. If more characters are used, the label might be truncated on the phone screen.

                Step 4   Edit the following:

                • Idle Key List: Edit the field as described in the following example:

                  redial|1;newcall|2;dnd;psk1

                  If the user incorrectly configures the programmable softkey list features on the phone, the key list on the phone LCD does not update. For example:

                  • If a user enters rdeial;newcall;cfwd (redial has been misspelt), the key list is not updated and the user does not see any change on the LCD.

                  • If a user enters redial;newcall;cfwd;delchar, the user will not see a change on the LCD, as the delchar softkey is not allowed in the Idle Key List. Hence, this is an incorrect configuration of the programmable softkey list.

                • PSK1:

                  fnc=sd;ext=5014@$PROXY;nme=sktest1

                Note   

                In this example, we are configuring a softkey on a phone as a speed dial number for extension 5014 (sktest1).

                You can also configure an XML service on the programmable soft key. Enter the string in this format:

                fnc=xml;url=http://xml.service.url;nme=name

                Step 5   Click Submit All Changes.

                Programmable Softkeys for Braavos

                The following softkey lists have been updated for Braavos.

                Keyword

                Key Label

                Definition

                Available Phone States

                acd_login

                Agt signin

                Logs user in to Automatic Call Distribution (ACD).

                Idle

                acd_logout

                AgtSignOut

                Logs user out of ACD.

                Idle

                answer

                Answer

                Answers an incoming call.

                Ringing

                astate

                Agt Status

                Checks the ACD status.

                Idle

                avail

                Avail

                Denotes that a user who is logged in to an ACD server has set his status as available.

                Idle

                barge

                Barge

                Allows another user to interrupt a shared call.

                Shared-Active, Shared-Held

                bargesilent

                BargeSilent

                Allows another user to interrupt a shared call with the mic disabled.

                Shared-Active

                bxfer

                BlindXfer

                Performs a blind call transfer (transfers a call without speaking to the party to whom the call is transferred). Requires that Blind Xfer Serv is enabled.

                Connected

                call (or dial)

                Call

                Calls the selected item in a list.

                Dialing Input

                cancel

                Cancel

                Cancels a call (for example, when conferencing a call and the second party is not answering.

                Off-Hook

                cfwd

                Forward / Clr fwd

                Forwards all calls to a specified number.

                Idle, Off-Hook, Shared-Active, Hold, Shared-Held

                conf

                Conference

                Initiates a conference call. Requires that Conf Server is enabled and there are two or more calls that are active or on hold.

                Connected

                confLx

                Conf line

                Conferences active lines on the phone. Requires that Conf Serv is enabled and there are two or more calls that are active or on hold.

                Connected

                delchar

                delChar - backspace Icon

                Deletes a character when entering text.

                Dialing Input

                dir

                Contacts

                Provides access to phone directories.

                Idle, Miss, Off-Hook (no input), Connected, Start-Xfer, Start-Conf, Conferencing, Hold, Ringing, Shared-Active, Shared-Held

                dnd

                DND / Clr Dnd

                Sets Do Not Disturb to prevent calls from ringing the phone.

                Idle, Off-Hook, Hold, Shared-Active, Shared-Held, Conferencing, Start-Conf, Start-Xfer

                em_login (or signin)

                Sign in

                Logs user in to Extension Mobility.

                Idle

                em_logout (or signout)

                Sign out

                Logs user out of Extension Mobility.

                Idle

                endcall

                End call

                Ends a call.

                Connected, Start-Xfer, Start-Conf, Conferencing, Hold

                favorites

                Favorites

                Provides access to "Speed Dials".

                Idle, Miss, Off-Hook (no input), Connected, Start-Xfer, Start-Conf, Conferencing, Hold, Ringing, Shared-Active, Shared-Held

                gpickup

                GrPickup

                Allows user to answer a call ringing on an extension by discovering the number of the ringing extension.

                Idle, Off-Hook

                hold

                Hold

                Put a call on Hold.

                Connected, Start-Xfer, Start-Conf, Conferencing

                ignore

                Decline

                Ignores an incoming call.

                Ringing

                join

                Join

                Connects a conference call. If the conference host is user A and users B & C are participants, when A presses "Join", A will drop off and users B & C will be connected.

                Conferencing

                lcr

                Call Rtn/lcr

                Returns the last missed call.

                Idle, Missed-Call,Off-Hook (no input)

                left

                Left arrow icon

                Moves the cursor to the left.

                Dialing Input

                messages

                Messages

                Provides access to voicemail.

                Idle, Miss, Off-Hook (no input), Connected, Start-Xfer, Start-Conf, Conferencing, Hold, Ringing, Shared-Active, Shared-Held

                miss

                Miss

                Displays the list of missed calls.

                Missed-Call

                newcall

                New Call

                Begins a new call.

                Idle, Hold, Shared-Active, Shared-Held

                option

                Option

                Opens a menu of input options.

                Off-Hook

                park

                Park

                Puts a call on hold at a designated "park" number.

                Connected

                phold

                PrivHold

                Puts a call on hold on an active shared line.

                Connected

                pickup

                PickUp

                Allows a user to answer a call ringing on another extension by entering the extension number.

                Idle, Off-Hook

                recents

                Recents

                Displays the All calls list from call history.

                Idle, Off-Hook, Hold, Shared-Active, Shared-Held

                redial

                Redial

                Displays the redial list.

                Idle, Connected, Start-Conf, Start-Xfer, Off-Hook (no input), Hold

                resume

                Resume

                Resumes a call that is on hold.

                Hold, Shared-Held

                right

                Right arrow icon

                Moves the cursor to the right.

                Dialing (input)

                settings

                Settings

                Provides access to "Information and Settings".

                All

                starcode

                Input Star Code/*code

                Displays a list of star codes that can be selected.

                Off-Hook, Dialing (input)

                unavail

                Unavail

                Denotes that a user who is logged in to an ACD server has set his status as unavailable.

                Idle

                unpark

                Unpark

                Resumes a parked call.

                Idle, Off-Hook, Connected, Shared-Active

                xfer

                Transfer

                Performs a call transfer. Requires that Attn Xfer Serv is enabled and there is at least one connected call and one idle call.

                Connected, Start-Xfer, Start-Conf

                xferlx

                Xfer line

                Transfers an active line on the phone to a called number. Requires that Attn Xfer Serv is enabled and there are two or more calls that are active or on hold.

                Connected

                Configure Provisioning Authority

                You can set up provisioning authority so that users can access their personalized phone settings from other phones. For example, people who work different shifts or who work at different desks during the week can share an extension, yet have their own personalized settings.

                The Sign in softkey appears on the phone when you enable provisioning authority on the phone. Users enter their usernames and passwords to access their personal phone settings. Users can also ignore the sign-in and use the phone as a guest. After signing on, users have access to their personal directory numbers on the phone. When the user signs out, the phone reverts to a basic profile with limited features.

                Procedure
                  Step 1   On the Configuration Utility page, select Admin Login > advanced > Voice > Provisioning.
                  Step 2   In the Configuration Profile section, set the Profile Rule field to the phone configuration file's URL.

                  Example:http://192.0.2.1:80/dms/CP-8851-3PCC/8851System.xml The EM Enable and EM User Domain fields are filled in, based on the information provided in the phone configuration file.
                  Step 3   In the Extension Mobility section, set the amount of time (in minutes) that the phone can be inactive before it automatically signs out from the provisioning authority in Inactivity timer(m).

                  To access Extension Mobility section, select Admin Login > advanced > Voice > Phone.

                  Step 4   Set the amount of time (in seconds) that the user has to cancel the sign-out in Countdown Timer(s).
                  Step 5   (Optional)If the Programmable Softkey Enable field in the Programmable Softkeys section is set to Yes, add signin to Idle Key List.

                  Example:newcall|1;signin|2
                  Step 6   Click Submit All Changes.

                  Configure Provisioning Authority in the Phone Configuration File

                  You can enable provisioning authority in the default configuration file for your phones, so that you don't need to set up the feature manually for each phone.
                  Procedure
                    Step 1   In the phone configuration file, set the following parameters:
                    1. Set the Provisioning Authority profile rules in the Profile_Rule parameters.

                      Example: 
                      <Profile_Rule ua="na">("$EMS" eq "mobile" and "$MUID" ne "" and "$MPWD" ne "")?[--uid $MUID$PDOM --pwd $MPWD] http://10.74.121.51:80/dms/CP-8851-3PCC/8851System.xml|http://10.74.121.51:80/dms/CP-8851-3PCC/8851System.xml</Profile_Rule>
                    2. Set the EM_Enable parameter to Yes.

                      Example: 
                      <EM_Enable ua="na">Yes</EM_Enable>
                    3. Enter the enter the domain for the phone, or the authentication server in the EM_User_Domain parameter.

                      Example: 
                      <EM_User_Domain ua="na">@10.74.121.51</EM_User_Domain>
                    Step 2   Save the configuration file and upload it to your provisioning server.
                    Step 3   On the Configuration Utility page, select Admin Login > advanced > Voice > Provisioning.
                    Step 4   Enter the filepath to the configuration file in one of the Profile Rule fields.

                    Example:http://<SERVER IP ADDRESS>:80/dms/td_8861/8861System.xml
                    Step 5   Click Submit All Changes.

                    Enable Hoteling on a Phone

                    You can allow users to sign into phones as guests. This is called hoteling.
                    Procedure
                      Step 1   On the Configuration Utility page, select Admin Login > advanced > Voice > Ext [n] (where [n] is the extension number).
                      Step 2   In the Call Feature Settings section, set Enable Broadsoft Hoteling to Yes.
                      Step 3   Set the amount of time (in seconds) that the user can be signed in as a guest on the phone in Hoteling Subscription Expires.
                      Step 4   Click Submit All Changes.

                      Set User Password

                      Users can set their own password on their phones, or you can set a password for them.

                      Procedure
                        Step 1   On the Configuration Utility page, select Admin Login > advanced > Voice > System.
                        Step 2   Set a password in the User Password field.
                        Step 3   Click Submit All Changes.

                        Download Problem Reporting Tool Logs

                        Users submit problem reports to you with the Problem Reporting Tool.

                        If you are working with Cisco TAC to troubleshoot a problem, they typically require the logs from the Problem Reporting Tool to help resolve the issue.

                        To issue a problem report, users access the Problem Reporting Tool and provide the date and time that the problem occurred, and a description of the problem. You need to download the problem report from the Configuration Utility page.

                        Procedure
                          Step 1   On the Configuration Utility page, select Admin Login > Info > Debug Info > Device Logs.
                          Step 2   In the Problem Reports area, click the problem report file to download.
                          Step 3   Save the file to your local system and open the file to access the problem reporting logs.

                          Configure PRT Upload

                          You must use a server with an upload script to receive the problem reports that the user sends from the phone.

                          • If the URL specified in the PRT Upload Rule field is valid, users get a notification alert on the phone UI saying that they have successfully submitted the problem report.

                          • If the PRT Upload Rule field is empty or has an invalid URL, users get a notification alert on the phone UI saying that the data upload failed.

                          The phone uses an HTTP/HTTPS POST mechanism, with parameters similar to an HTTP form-based upload. The following parameters are included in the upload (utilizing multipart MIME encoding):

                          • devicename (example: "SEP001122334455")

                          • serialno (example: "FCH12345ABC")

                          • username (The user name is either the Station Display Name or the User ID of the extension. The Station Display Name is first considered. If this field is empty, then the User ID is chosen.)

                          • prt_file (example: "probrep-20141021-162840.tar.gz")

                          A sample script is shown below. This script is provided for reference only. Cisco does not provide support for the upload script installed on a customer's server.

                          <?php

// NOTE: you may need to edit your php.ini file to allow larger
// size file uploads to work.
// Modify the setting for upload_max_filesize
// I used:  upload_max_filesize = 20M

// Retrieve the name of the uploaded file 
$filename = basename($_FILES['prt_file']['name']);

// Get rid of quotes around the device name, serial number and username if they exist
$devicename = $_POST['devicename'];
$devicename = trim($devicename, "'\"");

$serialno = $_POST['serialno'];
$serialno = trim($serialno, "'\"");

$username = $_POST['username'];
$username = trim($username, "'\"");

// where to put the file
$fullfilename = "/var/prtuploads/".$filename;

// If the file upload is unsuccessful, return a 500 error and
// inform the user to try again

if(!move_uploaded_file($_FILES['prt_file']['tmp_name'], $fullfilename)) {
        header("HTTP/1.0 500 Internal Server Error");
        die("Error: You must select a file to upload.");
}

?>
                          Procedure
                            Step 1   On the Configuration Utility page, select Admin Login > advanced > Voice > Provisioning.
                            Step 2   In the Problem Report Tool section, enter the path to the PRT upload script in the PRT Upload Rule field.

                            Example:

                            https://proxy.example.com/prt_upload.php

                            or

                            http://proxy.example.com/prt_upload.php

                            Step 3   Use the PRT Upload Method drop-down list box to choose the upload method:

                            • POST

                            • PUT

                            Step 4   Click Submit All Changes.

                            Configure a Phone to Accept Pages Automatically

                            The Single Paging or Intercom feature enables a user to directly contact another user by phone. If the phone of the person being paged has been configured to accept pages automatically, the phone does not ring. Instead, a direct connection between the two phones is automatically established when paging is initiated.

                            Procedure
                              Step 1   On the Configuration Utility page, select Admin Login > advanced > Voice > User.
                              Step 2   In the Supplementary Services section, choose Yes for the Auto Answer Page field.
                              Step 3   Click Submit All Changes.

                              Server-Configured Paging

                              You can configure a paging group on a server so that users can page a group of phones. For more details, refer to your server documentation.

                              Manage Phones with TR-069

                              You can use the protocols and standards defined in Technical Report 069 (TR-069) to manage phones. TR-069 explains the common platform for management of all phones and other customer-premises equipment (CPE) in large-scale deployments. The platform is independent of phone types and manufacturers.

                              As a bidirectional SOAP/HTTP-based protocol, TR-069 provides the communication between CPEs and Auto Configuration Servers (ACS).

                              Procedure
                                Step 1   On the Configuration Utility page, select Admin Login > advanced > Voice > TR-069.
                                Step 2   Set up the fields as described in theTR-069 table.
                                Step 3   Click Submit All Changes.

                                View TR-069 Status

                                When you enable TR-069 on a user phone, you can view status of TR-069 parameters on the Configuration page.
                                Procedure
                                On the Configuration Utility page, select Admin Login > advanced > Info > Status > TR-069 Status.

                                You can view status of TR-069 parameters in the TR-069 Status table.


                                Report All Phone Issues with the Configuration Utlility

                                If you are working with Cisco TAC to troubleshoot a problem, they typically require the logs from the Problem Reporting Tool to help resolve the issue. You can generate PRT logs using the Configuration Utlility and upload them to a remote log server.

                                Procedure
                                  Step 1   On the Configuration Utility page, select Admin Login > advanced > Info > Debug Info.
                                  Step 2   In the Problem Reports section, click Generate PRT. The Report Problem dialog appears.
                                  Step 3   Enter the following information in the Report Problem dialog:
                                  1. Enter the date that you experienced the problem in the Date field. The current date appears in this field by default.
                                  2. Enter the time that you experienced the problem in the Time field. The current time appears in this field by default.
                                  3. In the Select Problem drop-down list box, choose the description of the problem from the available options.
                                  Step 4   Click Submit in the Report Problem dialog.

                                  The Submit button is enabled only if you select a value in the Select Problem drop-down list box.

                                  You get a notification alert on the Configuration Utility page that indicates if the PRT upload was successful or not.

                                  Identify Phone Issues with a URL in the Phone Web Page

                                  When the phone is not working or doesn't register, a network error or any misconfiguration might be the cause. To identify the cause, add a specific IP address or a domain name to the phone admin page. Then, try to access so that the phone can ping the destination and display the cause.

                                  Procedure
                                  In a supported web browser, enter a URL that consist of your phone IP address and the destination IP that you want to ping.

                                  Enter a URL in the format:

                                  http:/<Phone IP>/admin/ping?<ping destination>

                                  where:

                                  Phone IP = actual IP address of your phone.

                                  /admin = path to access admin page of your phone.

                                  ping destination = any IP address or domain name that you want to ping. Only alphanumeric characters, ‘-’, and “_” are allowed as the ping destination. Otherwise the phone shows an error on the web page. If the <ping destination> includes spaces, only the first part of the address is used as the pinging destination. For example, “http://<Phone IP>/admin/ping?192.168.1.1 cisco.com” will actually ping 192.168.1.1.


                                  Factory Reset the Phone from Phone Web Page

                                  You can restore your phone to its original manufacturer settings so that the phone can be reconfigured, do it from the phone web page.

                                  Procedure
                                  Enter the URL in a supported web browser and click Confirm Factory Reset.

                                  You can enter URL in the format:

                                  http://<Phone IP>/admin/factory-reset

                                  where:

                                  Phone IP = actual IP address of your phone.

                                  /admin = path to access admin page of your phone.

                                  factory-reset = command that you need to enter in the phone web page to factory-reset your phone.