Phone Features and Setup

You can enable users to configure these features on line keys:

• Speed dial

• Busy Lamp Field (BLF) to monitor a coworker's line, with the following options:

  • Speed dial to the monitored line

  • Call pickup from the monitored line

Users can select any available line keys to configure features. They can also select a line key that is functioning as a speed-dial key or as a BLF key. The user's configuration will override any existing configuration for the line key. Users cannot select line keys on which you have configured other features. If a user selects a BLF list key, the phone adjusts the positions of the BLF list keys using the next available line keys.

For the BLF feature options, the phone subscribes to the BLF list URI that you specify (XML parameter BLF_List_URI), to be notified of changes in the status of the monitored lines. If you do not specify a BLF list URI, the phone subscribes to $USER@$PROXY.

You can configure speed dial on an idle line of a user phone. The user can then use that line key to speed-dial a number. When you enable the speed dial on the line key, the user sees the speed-dial icon a name for the speed dial line key. The user presses the line key to dial the assigned extension.

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

You can configure speed dial on a key expansion module line. The user can then press the line key to call a frequently dialed number.

You can configure dial assistance so that your users can place calls more quickly. As a user dials, the phone displays a list of closely-matched phone numbers on the screen.

You can configure busy lamp field to work with other features on your key expansion module, such as speed dial, and call pickup. Use the information in the following table as a guide when selecting the correct string format.

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 phone web page, you can configure alphanumeric dialing with speed-dial, blf, and call pickup.

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.

• Paging groups must use even-numbered port numbers.

• 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.

You can set paging priority. You no longer need to register the phone to send or receive a page and this feature is known as "Out of Band Paging" feature. You can configure maximum of five paging groups on the phone.

When a paging is initiated during an active call, your user sees an incoming page or outgoing page icons on the phone.

Priority has no impact during a regular page. Only when the phone receives a call during an active page, priority impacts the active call. Following scenarios explain how priority of an active page impacts an active call:

• PG_PRI_EMERGENT(Priority 0): If the phone receives a page with priority 0 during a call, the call will be put on hold. After the paging is complete, the call resumes.

• PG_PRI_IMPRORTANT(Priority 1): If the phone receives a page with priority 1 during a call, the call and the page audio is mixed.

• PG_PRI_NORMAL (Priority 2): If the device receives a page with priority 2 during a call, the phone does not display any incoming page icon on the phone screen and the user only hears a notification tone. Once the call ends and if the page is still active, the user sees the paging notification on the phone.

• PG_PRI_MINOR (Priority 3): If the phone receives a page with priority 3 during a call, the page is ignored.

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).

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 users sign in, they 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.

Set up the hotel feature on Broadworks and set the phone as a host or a guest.

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

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.

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")

You can generate PRT automatically at specific intervals and can define the PRT file name.

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.");
}

?>

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.

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).

For TR-069 Enhancements, see TR-069 Parameter Comparison.

The Electronic Hookswitch feature enables users to use headsets that electronically connect a wireless headset to a phone. Typically, the headset requires a base that plugs into the phone and communicates with the headset. Here are the supported headsets:

• Plantronics Savi 740

• Jabra PRO920

• Jabra PRO9400

• Sennheiser DW Pro1

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 phone web page and upload them to a remote log server.

You can factory reset the phone from the phone web page. The reset only happens if the phone is idle. If the phone is not idle, the phone web page shows a message that the phone is busy and that you need to try again.

You can configure an extension to only accept secure calls. If the extension is configured to only accept secure calls then any calls the extension makes will be secure.

You can also configure a secured extension with XML services. Enter a string in this format:

<Secure_Call_Serv ua="na">Yes</Secure_Call_Serv>

<Secure_Call_Option_1_ ua="na">Optional</Secure_Call_Option_1_>

For SIP messages, you can either specify the transport protocol of your choice, or, you can let the phone select the appropriate protocol automatically, for each extension.

When you set up automatic selection, the phone determines the transport protocol based on the Name Authority Pointer (NAPTR) records on the DNS server. The phone uses the protocol specified in the record that has the lowest order and preference. When there are multiple records with the same order and preference, the phone looks for a protocol within the records, in the following order of preference: 1. UDP, 2. TCP, and 3. TLS. The phone uses the first protocol that it finds, in that order of preference.

You can disable the ability of the phone to receive incoming SIP messages from a non-proxy server. When you enable this feature, the phone only accepts SIP messages from:

• proxy server

• outbound proxy server

• alternative proxy server

• alternative outbound proxy server

• IN-Dialog message from proxy server and non-proxy server. For example: Call Session dialog and Subscribe dialog

A user privacy header in the SIP message sets user privacy needs from the trusted network.

You can set the user privacy header value for each line extension using the phone web page.

The privacy header options are:

• Disabled (default)

• none—The user requests that a privacy service applies no privacy functions to this SIP message.

• header—The user needs a privacy service to obscure headers which cannot be purged of identifying information.

• session—The user requests that a privacy service provide anonymity for the sessions.

• user—The user requests a privacy level only by intermediaries.

• id—The user requests that the system substitute an id that doesn't reveal the IP address or host name.

You can determine whether to include the P-Early-Media header in the SIP message of outgoing calls. The P-Early-Media header contains the status of the early media stream. If the status indicates that the network is blocking the early media stream, the phone plays the local ringback tone. Otherwise, the phone plays the early media while waiting for the call to be connected.

You can set up the profile account feature from the phone web page. It allows users to collect authentication information. Authentication information is required while the phone tries to resynchronize and download configuration file and gets an HTTP or HTTPS 401 authentication error for the first time. When you enable this feature, the Profile account setup screen is displayed automatically on the phone for the following situations:

• When the HTTP or HTTPs 401 authentication error occurs during provisioning for the first time after the phone reboots

• When the Profile Account username and password are empty

• When there are no username and password in the Profile Rule

There are two ways to trigger the Profile account setup screen. The screen can be triggered when there is an HTTP or HTTPS 401 authentication error for the first time. If the user misses or ignores the popup screen, it can also be triggered through the phone screen menu.

When you disable the feature, the Profile account setup screen isn't displayed on the phone.

Use this information to help you troubleshoot this situation.

Username and password in the Profile Rule have higher priority than profile account.

• When you provide a correct URL in the Profile Rule field without a username and password, the phone requires HTTP or HTTPS basic authentication or digest to resync the profile. With correct profile account, authentication passes. With incorrect profile account authentication fails.

• When you provide a correct URL in the Profile Rule field with correct username and password, the phone requires HTTP or HTTPS basic authentication or digest to resync the profile. Profile account is not used for phone resync. Sign-in is successful.

• When you provide a correct URL in the Profile Rule field with incorrect username and password, the phone requires HTTP or HTTPS basic authentication or digest to resync the profile. Profile account is not used for phone resync. Sign-in always fails.

• When you provide incorrect URL in the Profile Rule field, sign-in always fails.

You can configure a phone to allow a call to seamlessly be moved from one desk phone(location) to another mobile phone or desk phone(location).

When you enable this feature, the Anywhere menu is added into the phone screen. The user can use this menu to add multiple phones as locations to the extension. When there is an incoming call in that extension, all the added phones will ring and the user can answer the incoming call from any location. The locations list also gets saved to the BroadWorks XSI server.

You can sync the Block caller id status on the phone and the Line ID Blocking status on the BroadWorks XSI server. When you enable the synchronization, the changes that the user makes in the Block caller id settings also changes the BroadWorks server settings.

You can configure a phone to display recent call logs from either the BroadWorks server or the local phone. After you enable the feature, the Recents screen has a Display recents from menu and the user can choose the XSI call logs or the local call logs.

You can prioritize voice or video data in limited bandwidth conditions.

You will need to configure the priorities individually on each line of a phone.

You can configure different priorities for different areas of traffic. For example, you can configure different priorities for internal and external traffic by setting up different configurations on internal and external lines. For effective traffic management, specify the same settings on all the phone lines in a group.

The Type of Service (ToS) field of a data packet determines the packet's priority in data traffic. You can configure the desired priorities by specifying appropriate values for the ToS fields of voice and video packets, for each phone line.

For voice data, the phone applies the ToS value that it receives by LLDP. When there is no ToS value available by LLDP, the phone applies the value that you specify for voice packets.

For video data, the phone always applies the ToS value that you specify for video packets.

The default values prioritize voice over video.