Resource configuration parameters
This resource allows and administrator to configure a Polycom IP Phone.
- BLF (Busy Lamp Field): Removed since v5.0.0; use speedial function instead (See "Personal Directory") . If set to 'enabled', the phone will request to be notified of the status (busy, idle, ringing, ...) of each extension in the 'Speeddial' list and render the result on the screen. Please note that if not needed, this should be turned off to remove unnecessary load on the SOP.
- Speeddial: See "Personal Directory"
- Dialplan: Dialplan used to have an automatic number composition when the user dial a number matching the dialplan. A dialplan is defined either by a 'string' or by a list of strings. Each string in the list is an alternative numbering scheme, specified either as a set of digits or timers, or as an expression over which the phone will attempt to find a shortest possible match. Refer to chapter 2.1.5 of RFC3435 for more information
- If set to 'From SOP', the contact directory will contain the up to 99 contacts from the SOP internal directory. Hidden extensions are never part of this directory.
- If set to 'From SOP, filtered', the contact directory is the same as with 'from SOP', but only contacts where the selected 'Key' matches the specified 'Value' (in the next fields) are included.
- If set to 'Clear', the directory will only contain the speeddials. (ESCAUX recommends to use this for PUM logout).
- If set to 'Local', the contact directory will be manageable by the user through the phone. When the phone reboots, the contacts configured by the user will persist in memory. This option is incompatible with the use of speeddials .
- (Deprecated since v5.0.0) If set to 'Extensions to supervise', the contact directory is cleared except for the speeddials. With version 5.0.0 and higher, speeddials are always part of the directory and this option defaults to 'Clear'.
- Editable directory: (Deprecated since v5.0.0; The contact directory is only editable when 'Storage' is set to 'Local') The "Contact Directory" will be editable on the phone if the parameter is set to "enabled". It will be freezed if the parameter is set to "disabled" (by default it will take the value given in the module). Changes made directly on the phone will be lost on next reboot if the contact directory 'Storage' is not set to 'Local'.
- Personal Directory (previously "Speeddial"):
- A list of speeddial buttons (extension which are frequently called). The phone will associate the first speed dial to the first available line key (Line 2, Line 3, ...). The remaining contacts will be available through the phone menu:'Features > Speed Dials'. The number of line keys depends of the phone model. For example, the IP330 has 2 line keys (1 available), the IP 501 has 3 line keys (2 available) and a IP 650 with 3 Expansion module has 48 line keys (47 available).
- Starting with version 5.0.0, BLF activation (also called Supervision/Presence) can be performed on a per-extension basis. The label of each extension is also configurable.
- On the Polycom VVX300, VVX310, VVX400, VVX410, VVX500 and VVX600, the BLF or speeddial will be allocated on the specified linekey. It is therefore possible to have linekey2 not configured on the phone while linekey3 is configured with a BLF or a speeddial. This is not possible with the SoundPoints and the VVX1500
- It is not possible to do a supervision of an extension located on another SOP
- It is not possible to assign the same extension to both BLF and SpeedDial
- Application URL:
- Proxy: Address of the desired HTTP proxy to be used by the Microbrowser. If blank, normal unproxied HTTP is used by the Microbrowser.
- Possible value:(Null or domain name or IP address in the format <address>:<port>)
- Application URL: URL used for Microbrowser home page. If blank, the browser will notify the user that a blank home-page was used.
- Application URL Status Bar: Flag to determine whether or not to turn off display of status messages.
- Application URL Timeout (sec): Timeout for the interactive browser. If the interactive browser remains idle for a defined period of time, the phone should return to the idle browser.
- Idle URL: URL used for Microbrowser idle display home page. If it doesn't start with http, it is interpreted as a page on the SOP and the IP of the SOP is added at the beginning (e.g. /page => http://sop_ip/page).
- Auto append extension: If activated and the URL doesnt start with http (points to a page on the SOP), the default behavior is to add '&ext=ext' to the URL (where ext is the extension that has this phone as primary phone). This can be disabled by setting this to no
- Idle URL Refresh (sec): The period in seconds between refreshes of the idle display Microbrowser's content. If set to 0, the idle display Microbrowser is not refreshed.
- Allow Push Message: Message type which are to be accepted by the phone. Set to 'Default' to keep the current phone settings. Set to 'None' to deactivate the Message Push feature. 'All' is only relevant for Polycom Phone Support 4.0+ (very special case). Otherwise select the type of message to be accepted.
- Push Url: Root URL to be used by the phone on reception of a Push Message
- Login: Username to be used when sending push message
- Password: Password to be used when sending push message
- SIP server feature: Note: These technical settings if improperly configured can have impact on the availability of the phone
- SIP re-invite: allow the voice media to be transfered directly between SIP endpoints through the SIP re-invite procedure. Possible value are:
- Default: Keep default settings as defined in the Telephony server module (for example Asterisk 1.2.x)
- Yes: Allow the server to perform the re-invite procedure.
- No: Do not allow the server to perform the re-invite procedure.
- Qualify SIP devices registration: In order to monitor the SIP device availability, the Telephony server can send SIP OPTIONS messages (every 60s). If the device does not respond within the defined delay, it will be considered as off-line. Possible values are:
- Default: Keep the default setting defined in the Telephony server module (for example Asterisk 1.2.x)
- No: Disable Qualify SIP registration for this device
- Yes: Enable Qualify SIP registration for this device, with a default timeout value of 2000ms
- Suggested number: Enable Qualify SIP registration for this device, with the chosen timeout value (expressed in milliseconds).
- Force specific codec: If this parameter is set to 'Yes', the SIP server will only accept the 3 codecs defined in this resource with the specified preference.
- First Codec, Second Codec, Third Codec: Ordered list of codec. The ptime defines the size of the payload to be transmitted on the network. For more information about RTP packetization, see below.
- Use authentication: Method of authentication.
- Password: If using custom password as method of authentication, the custom password has to be set here.
- Use SIP TLS: Use encrypted signaling between phone and SOP.
- Use SRTP: Use encrypted voice packets between phone and SOP.
- Voice codecs Jitter Buffer Minimum : The smallest jitter buffer depth that must be achieved before play out begins for the first time. After this depth, play out will continue even if the depth falls below this point. This impacts all voice codecs.
- Voice codecs Jitter Buffer Maximum : The maximum jitter buffer depth supported. Jitter above this size will always cause lost packets. This parameter should be set to the smallest possible value that will support the expected network jitter. This impacts all voice codecs.
- Voice codecs Jitter Buffer Shrink : The absolute minimum duration time (in milliseconds) of RTP packet Rx with no packet loss between jitter buffer size shrinks. Use smaller values (1000 ms) to minimize the delay on known good networks. This impacts all voice codecs.
- Retry TimeOut : If set to 0, use standard RFC 3261 signaling retry behavior. Otherwise retryTimeOut determines how often retries will be sent. Units = milliseconds.
- Retry Max Count : retryMaxCount retries will be attempted before moving on to the next available server.
- Advanced Options:
- Language: Language of the phone interface. If 'Default' is chosen, the user will be able to set his own language through the phone menu. When set to 'Global', the language of the user is used.
- Automatic Off-hook Call Placement: If set to enabled, the phone will automatically compose the Contact to call when the handset is offhook.
- Calllist-received, Calllist-placed, Calllist-missed: Enable or Disable, for example for confidentiality reasons, the phone call diaries (Call list).
- Url-dialing: Controls whether URL/name dialing is available from a line. If this option is enabled, the phone will display, when the CallerID is unknown, "Anonymous" <sip:firstname.lastname@example.org>. When disabled, it will just display Anonymous
- Date format/Hour format: choose the date and hour format as it should be displayed on the phone. (ex: Fri, Mar 18 15:59)
- Local call forwarding on the phone: Enable or disable the functionality of local call forwarding (forward on the phone instead of on server level)
- Call Waiting Notification: When set to "Disabled", the phone will not play an auditive beep in the headset nor show a new incoming call on the display when the user is already on the line. The caller who generates the second call will always hear a normal ringtone. If this behavior is unwanted, any additional call can be avoided entirely by using a callflow action.
- Enable custom softkey: (Deprecated since v5.0.0; Custom softkeys are enabled on a per case basis) Enable/disable the custom softkeys (black buttons under the screen)
- Enable Mystat softkey: (Deprecated since v5.0.0; always disabled) On bigger Polycom devices, a 'MyStat' softkey is enabled by default. This is not supported and should be disabled
- Enable Mystat buddies: (Deprecated since v5.0.0; always disabled) On bigger Polycom devices, a 'Buddies' softkey is enabled by default. This is not supported and should be disabled
- NewCall button: By default, a 'NewCall' button is displayed. This can be disabled if needed
- Backlight intensity: Choose the intensity of the phone display backlight, when using it and after using it (only on IP550,560,650,670).
- Softkey X:
- Enabled: Show/don't show a custom softkey (needs "enable custom softkeys: Enabled!)
- Prioritair over existing softkey: By default, some softkeys will exist on the phone. If you enable priority, the softkey will be added before default softkeys, when disabled, it will be added after the default softkeys. If you define more softkeys than possible to display at once, a 'more' button will be added automatically to access the remaining softkeys
- Enabled when phone is idle: If enabled, the softkey will be visible when the phone is idle
- Enabled when phone is in conversation: If enabled, the softkey will be visible when the phone is in conversation
- Enabled when phone gives the dial tone: If enabled, the softkey will be visible when the phone is giving the dialtone (of hook but not yet calling)
- Enabled when phone is ringing: If enabled, the softkey will be visible when the phone is ringing
- Enabled when phone is trying to make a call: If enabled, the softkey will be visible when the phone is trying to make a call (ex: you dialed a number but the call is still being set up(the moment between pressing 'dial' and the moment it rings))
- Enabled during the phone setup menu: If enabled, the softkey will be visible while browsing the menu
- Enabled when the phone put the conversation on hold: If enabled, the softkey will be visible when putting a caller on hold
- Action: Defines the action that occurs when the softkey is pressed and can be either a number to dial or a custom URL to browse (see "Intentional Status Change and Call Forwarding" section)
- Headset Configuration:
- Hookswitch mode: The Electronic Hookswitch mode for the phones analog headset jack, according to the headset you want to use.
- Headset memory mode: If disabled, handsfree mode will be used by default instead of the handset. If enabled, the headset will be used as the preferred audio mode after the headset key is pressed for the first time, until the headset key is pressed again.
- Internet access options:
- Custom NTP Server: Allow to specify another NTP server than the SOP itself which is the default.
- Custom RTP range id: An id corresponding to one of the RTP port ranges defined in Communication Server. By default (empty), range 0 is used.
- 802.1x authentication:
- This feature requires the Polycom Firmware 4.0 or higher. Set to 'Enable' to activate the 802.1x authentication, 'Default' to keep the settings currently on the phone and 'Disable' to deactivate the authentication.
- Method: The authentication method to be used
- Username: The username used for the authentication
- Password: the password to be used for the authentication
Conditional Idle is a service that can be polled or pushed to the supported phones to display the status for that particular extension.
It also allows you to display a certain variable under certain conditions. For example: you can use this service, to check if you are in forward mode, and if so, display the forward number.
Navigate to: Resources > IP Phone > Select your Polycom phone > Edit
For Polycom Phones, add the following url to your idle url, and set append extension to true.
As of Polycom Phone Support 4.3.4, we allow to show if the extension is logged into a queue. You can use up to 3 queues. You will need SOP Base 1.4.6 installed.
To set up the queues to monitor you have to set the following parameters in the URL:
q1v is the first queue to monitor.
q2v is the second queue to monitor.
q3v is the third queue to monitor.
If you are not logged into any queue, no indicator will be displayed.
Some devices don't have enough space to display the queues if your status and your extension fill up the whole screen.
Queue names are fetched from the profile parameters, you cannot set the name directly. You will have to indicate a var number, as explained below.
Acessing Profile Parameters
Possible values for if, thenshow, q1v, q2v and q3v are var_1 to var_42, they match the parameter order defined in your profile
Navigate to: Communication Flow Studio > Profile
| Parameter XY matches
This service is supported up to 50 phones.
RTP packetization (ptime)
The following table lists the minimum and maximum values that are valid per codec, as well as the increment value used for each. Please note that the maximum values here are only recommended maximums, and should not exceed the RTP MTU.
|| Minimum (ms)
|| Maximum (ms)
|| Default (ms)
|| Increment (ms)
Invalid framing options are handled based on the following rules:
If the specified framing is less than the codec's minimum, then the minimum value is used.
If the specific framing is greater than the codec's maximum, then the maximum value is used
If the specificed framing does not meet the increment requirement, the specified framing is rounded down to the closest valid framing options.
PUM, Configuration Update and Phone Restart
When doing a PUM
login or PUM
logout, a 'Configuration Update' is requested to the Polycom Phone. Depending of the configuration change, the phone might decide to force a restart of the phone. This happens in the following cases:
- When the Directory Storage option is changed to or from the 'Local' option
- On the SoundPoints when 'speeddial' or 'supervision' are added or removed
- On the VVX when 'supervision' are added or removed. For the VVX speeddial change does not cause any restart
Some other configuration settings might also causes a restart of the phone. To avoid it, the configuration settings on the Polycom Virtual Phone and the Physical one should be the same as much as possible.
Intentional Status Change and Call Forwarding
The combination of "Polycom Phone Support" module 5.0.0+ and any of the "Polycom VVX" resources 5.1.0+ provides a convenient graphical way to allow a user to change his intentional status.
To activate this service:
- In the phone resource, enable one of the Softkeys and add a label such as 'Status'
- In the corresponding action field add:
Where <extension> should be replaced by the actual extension, or by the ETL
If you want to control the call forwarding and forward number from the phone, since "Polycom Phone Support" module 5.1.0+ and any of the "Polycom VVX" resources 5.1.1+, the url supports two new arguments:
Where <forward_parameter> should be replaced with the parameter name that informs if the call forward is enabled. For a green user in Fusion template, it is UnifyUserGreenCFU.
Where <number_parameter> should be replaced with the parameter name that stores the forward number. For a green user in Fusion template, it is UnifyUserGreenForwardAllNumber