Advanced options: Polycom phones or templates

Option

Description

Default  

Call Offering Timeout

Call Ringback Timeout

This option specifies the time in seconds to allow an outgoing call to remain in the ringback state before dropping the call.

0

Call Dialtone Timeout

This option specifies the time in seconds to allow the dialtone to be played before dropping the call. If set to 0, the call is not dropped. If set to <NULL> (or no value), the call is dropped after 60 seconds.

15

Configuration Time Zone Overrides DHCP

This option determines whether the time zone that is set for the location that is associated with the IP phone overrides the DHCP time zone settings.

No

Configuration NTP Server Overrides DHCP

This option determines whether the SNTP server settings that are associated with the IP phone override the DHCP server settings.

No

DTMF On Time

This option sets the length of time (in milliseconds) that the tones are generated when a sequence of DTMF tones is played automatically.

This option also sets the minimum time the tone is played when an agent dials manually, regardless of the duration of the key press.

80

DTMF Off Time

This option sets the length of time (in milliseconds) that the IP phone pauses between digits when a sequence of DTMF tones is played automatically.

This option also sets the minimum inter-digit time when an agent dials manually.

80

Phone Limits Calls Per Line Key

This option indicates whether the number of calls per line key is limited by the specific model of Polycom IP phone settings.

No

Always Reboot on Reload

This option indicates whether the phone is automatically rebooted when a user clicks Reload Now.

Note: When you enable this option, another port/service is open on the phone. Consider the security implications before you enable this option.

No

Enable Configuration Web Page

This option allows you to enable or disable the webpage on Polycom phones.

Yes

Phone Offer Cryptosuites

This option determines which cryptosuite(s) the phone offers in SDP. The choices are:

• AES_CM_128_HMAC_SHA1_32 only

• AES_CM_128_MHAC_SHA1_80 only

• Both

Note: If you do not set this option, the phone automatically offers both cryptosuites.

Both

Phone Require Secure Media

This option indicates whether the phone is allowed to use only secure media streams.

• If set to "No,"the phone can accept non-secure media streams.

• If set to "Yes,"the phone accepts only secure media streams.

• • Any offered SIP INVITEs must include a secure media description in the SDP or the call is rejected.

  • For outgoing calls, only a secure media stream description is included in the SDP of the SIP INVITE. The non-secure media description is not included.

No

Enable Power Saving

Select this option to enable power saving mode. When the phone is in power saving mode, the phone's LCD display is automatically turned off after 1 minute.

This option is available for Polycom VVX models only:

• By default, this option is disabled on the VVX 300, 301, 310, 311, 400, 401, 410 and 411 models.

• By default, this option is enabled on the VVX 500, 501, 600 and 601 models.

Off

Use 486 For Reject

This option indicates whether CIC sends a busy signal as the rejection reason to a SIP request.

No

Phone Warning Level

This option determines when the phone’s warning icon and warning pop-message appear on the Polycom phones. The choices are:

• All warnings

• Critical warnings

• None

The default value is All warnings. You must opt-in for settings other than the default value.

Note: The warning, “Default admin password is in use. Please contact your administrator,” reminds you to change the default administrator password on the phone. Genesys recommends that you change this password.

Alternatively, you can change the Phone Warning Level to 1 and the warning will no longer appear. However, changing the phone warning level does not actually fix the root problem involving the default admin password.

On the phone, the warnings always appear under Settings>Status>Diagnostics>Warnings.

TCP Session Timeout (seconds)

This option sets the number of seconds CIC waits before it attempts a TCP connection. CIC attempts to connect until it meets the TCP session retries count.

To control the timeout behavior in the event of a TCP connection failure, set this parameter and the TCP Session Retries parameter.

<Null> or 0 (The IP phone's default settings are used.)

TCP Session Retries

This option sets the number of times CIC attempts to make a TCP connection. If it cannot connect within this number of times, CIC moves to the next server. A retry is made if the session reaches in seconds the timeout as set in the TCP Session Timeout (above).

To control the timeout behavior in the event of a TCP connection failure, change this parameter and the TCP Session Timeout parameter.

<Null> or 0 (The IP phone's default settings are used.)

Boot Server Type

This option determines whether the phone controls boot server options by the menu or the provisioning server controls boot server options. If controlled by the phone, the CIC provisioning subsystem does not write configuration for boot server options. If controlled by the provisioning server, the server writes configuration to the provisioning files.

Note: If the Boot Server Option, Boot Server Option Type, or Provisioning URL contain incorrect values, the provisioning server does not write the configuration for boot server options. If incorrect values exist, the provisioning logs contain errors such as:

• Boot Server Option not in range of 128-254.

• Provisioning URL is not a compliant RFC 2396/RFC 3986 URI.

This option is available for Polycom phones capable of 4.0 or newer firmware.

Select:

Phone to use the phone menu to control boot server options.

Custom to use the DHCP option you set by using the Boot Server Option option and the Boot Server Option Type option.

Custom + Opt.66 to use the DHCP option you set by using the Boot Server Option option and Boot Server Option Type option. If the phone cannot determine the information from the Boot Server Option option, the phone uses DHCP option 66.

Opt.66 to use DHCP option 66 to determine boot server parameters.

Static to use the Provisioning URL option to determine the boot server.

Phone

Boot Server Option

This option is the DHCP option that the phone uses when trying to determine the boot server parameters. Set this option if you selected Custom or Custom + Opt.66 in the Boot Server Type option. Valid entries include empty string or an integer in range from 128 to 254. This option is available for Polycom phones capable of 4.0 or newer firmware.

This option indicates the DHCP option type to use with the Boot Server Option option. You can select IP Address or String. Set this option if you selected Custom or Custom+Opt.66 in the Boot Server Type option. This option is available for Polycom phones capable of 4.0 or newer firmware.

String

Provisioning URL

This option indicates the URL that the phone points to. Set this option if you selected Static in the Boot Server Type option. Enter a compliant RFC 2396/RFC 3986 URI and use a scheme of FTP/TFTP/FTPS/HTTP/HTTPS. If you do not enter the scheme, the phone uses HTTP. If you do not enter the port, the phone uses the default value for the scheme. For HTTP provisioning, the URL must specify port 8088 since the Provisioning subsystem on the CIC server listens on port 8088 and not port 80. Polycom phones assume port 80 for an HTTP URL.
Example: http://provision.mydomain.com:8088

This option is available for Polycom phones capable of 4.0 or newer firmware.

Option

Description

Default  

Presence

This option is related to the Presence feature of Microsoft Windows Messenger 5.1 on the IP phone.

Note: The Presence feature is reserved for custom integrations outside of CIC. Enabling this option in Interaction Administrator has no effect in the CIC system.

No

Messaging

This option is related to the Instant Messaging feature of Microsoft Windows Messenger 5.1 on the IP phone.

Note: The Messaging option is reserved for custom integrations outside of CIC. Enabling these features in Interaction Administrator has no effect in the CIC system.

No

Directory

This option specifies whether the user can edit the local contacts.

Yes

All Call Lists

This option specifies whether the user can view all call lists, including the  Received Call List, the Placed Call List, and the Missed Call List.

Yes

Received Call List

This option specifies whether the user can view the list of received calls.

Yes

Placed Call List

This option specifies whether the user can view the list of placed calls.

Yes  

Missed Call List

This option specifies whether the user can view the list of missed calls.

No

URL Dialing

This option specifies whether URL/name dialing is available from a private line. (This feature is never available from a shared line.)

No

Call Park

This option specifies whether active calls can be parked and retrieved.

Yes

Group Call Pickup

This option specifies whether calls to another phone within a predefined group can be picked up without dialing the extension of the other phone.

Yes  

VVX D60 Profile

This option indicates whether to enable the VVX D60 feature.

No

Option

Description

Default  

Idle Screen Logo

This option sets the background image that the Polycom IP phone displays when the phone is not on a call.

Enter the name of the file excluding the extension.  For example, if logo_ip600.bmp is the complete file name, enter the value of "logo_ip600."

Supported formats for VVX phones are PNG, JPG, and BMP. Supported formats for SoundPoint IP phones are JPG and BMP.

Note: If you do not specify an extension, CIC assumes that BMP is the extension.

Place the source image file in the \\ic\provision\polycom directory on the CIC server.

If you leave this value, which is the default setting, the system looks for the following file names, based on the model:

Note: The resolution is dependent on the model. JPG, PNG, and BMP images may require special file format parameters. See the "Adding a Background Logo" section in the Polycom Administration Guide for information on resolutions.

Idle Screen Background Color

This option sets a custom background color for the Polycom IP670 phone. This option is used to improve the appearance of the Idle Screen Logo for the IP670. Enter the color in either hexidecimal (for example, "0x0077FF" - must being with "0x"), or decimal value.

Not specified

Headset Echo/Noise Suppression

This option indicates whether echo and noise suppression is used with the agents' headsets.

Note: When you enable the Headset Echo/Noise Suppression option, there is a short transmit delay of 30 milliseconds.

No

Headset Microphone Gain

This option specifies the gain value for the headset microphone:

• IP330 group (320, 321, 330, 331, 335)=21, and acceptable values are 0 to 21

• IP430=21, and acceptable values are 0 to 39

• IP450=21, and acceptable values are 0 to 21

• IP550 and IP560=21, and acceptable values are 0 to 21

• IP650 and 670=21, and acceptable values are 0 to 21

Headset Speaker Gain

This option specifies the gain value for the headset speaker:

• IP330 group (320, 321, 330, 331, 335)=4, and acceptable values are 0 to 25

• IP430 and IP450=1, and acceptable values are 0 to 25

• IP550 and IP560=1, and acceptable values are 0 to 25

• IP650 and IP670=1, and acceptable values are 0 to 25

Headset Sidetone Gain

This option specifies the gain value for the headset sidetone:

• IP330 group (320, 321, 330, 331, 335)=-3, and acceptable values are -25 to -3

• IP430 and IP450=-3, and acceptable values are -25 to -3

• IP550 and IP560=-3, and acceptable values are -25 to -3

• IP650 and IP670=-3, and acceptable values are -25 to -3

Electronic Hookswitch Mode

This option indicates whether optional external hardware is available for use with a headset attached to the IP phone's analog headset jack.

The options are:

• Regular (None)

• Jabra (DHSG)

• Plantronics

Regular (None)

Auto Dial on Off-hook

This option indicates whether contacts can be automatically dialed when the phone is off-hook. Contacts must be configured with the Auto Dial on Off-hook Number option.

No

Auto Dial on Off-hook Number

This option specifies a number to auto dial when the Auto Dial on Off-hook Number option is enabled.

VVX D60 Allowing Paring

This option indicates whether a user can pair or unpair a VVX D60 base station with a VVX business media phone. You can select:

• None - To not allow users to pair or unpair a base station from the VVX phone.

• Pairing - To allow users to pair the base station with the VVX phone, but not allow users to unpair a base station.

• Unpairing - To allow users to unpair the base station from the VVX phone, but not allow users to pairing a base station.

• Both - To allow users  to pair and unpair the base station with the VVX phone.

None

Option

Description

Default  

Digitmap

This option allows you to edit digitmap properties.

The digitmap feature eliminates the need for using the Dial or Send soft key when making outgoing calls when a matching digit pattern is detected. As soon as a digit pattern match is found, the call setup process completes automatically.

Acceptable values are strings compatible with the digit map feature of MGCP described in 2.1.5 of RFC 3435. String is limited to 768 characters and 30 segments; a comma is also allowed; when reached in the digit map, a comma will turn dial tone back on;’+’ is allowed as a valid digit; extension letter ‘R’ is used.

For complete information, see the Polycom Technical Bulletin 11572 and Polycom's Administrator’s Guide for the SoundPoint® IP/SoundStation® IP Family.

 x.T|*T|*905|*90[1-4]x.T

Digitmap Timeout

This option sets the timeout in seconds for each segment of the digitmap.

Note: If there are more digitmaps than timeout values, the default value of 3 is used. If there are more timeout values than digitmaps, the extra timeout values are ignored.

3|1|3|3

Option

Description

Default  

NAT IP Address

This option sets the IP address to use in NAT traversal.

Note: This setting only changes the IP address that is displayed in SIP signaling. All IP addresses in managed phones configuration are pass-through strings to support IPv6.

Not specified

NAT Signal Port (1024-65535)

This option sets the signal port to use in NAT traversal. This setting changes the port setting used by the IP phone.

Not specified

NAT Audio Port Start (1024-65535)

This option sets the audio port to start in NAT traversal. This setting changes the RTP port that is initially allocated to the IP phone.

Not specified

NAT Keep Alive Interval (0-3600 seconds)

This option sets the interval time in seconds in which IP phones send a keep-alive packet to the NAT device to keep the communication port open so that NAT can continue to function as initially configured.

Not specified

Option

Description

Default  

Admin Password

This setting changes the phone’s local administrator password. Any string is an acceptable value.

Note: When this password is changed, the value is passed to the phone one time, and then is removed from DS (Directory Services). The Polycom phone remembers the password change, and the provisioning server does not send this value to the phone configuration in any subsequent phone reboots.

Not specified

802.1Q VLAN Identifier

This setting is used by the IP phone to set the phone’s 802.1Q VLAN identifier if VLAN ID is not obtained by CDP or DHCP. Acceptable values are 0 to 4094

Not specified

(no VLAN tagging)

LAN Port Mode

This option sets the network speed over the Ethernet for the IP phone through the LAN port.

Acceptable values are:

• Auto

• 10 Full Duplex

• 10 Half Duplex

• 100 Full Duplex

• 100 Half Duplex

Auto

Option

Description

Default  

Syslog Server

This option sets the syslog server IP address or host name.

When a Syslog Server is specified, the Syslog Transport and Syslog Render Level settings are used. When a Syslog Server is not specified, the Syslog Transport and Syslog Render Level settings are ignored. All IP addresses in managed phones configuration are pass-through strings to support IPv6. 

Note: When you specify a Syslog Server value, the IP phone's application log which is sent to the provisioning server, is no longer used, or disabled. This means the log file no longer appears in the IC, and all information is present in the syslog instead, which protects Polycom's flash memory abilities.  

Not specified (syslogging is enabled, and the log file no longer appears in the CIC system)

Syslog Transport

This option sets the syslog transport protocol.

Acceptable values include TCP and TLS.

UDP

Syslog Render Level

This option sets the lowest class of event that will be rendered to the syslog.  

The acceptable range of values is 0 through 6.

0

Disable App Logs When Using Syslog

When Syslog Tracing is enabled, application logs are disabled (set to Yes) by default. Change this setting to No, to enable the creation of application logs by Polycom phones.

The information generated and collected in the application log (which is stored in the phones folder in the IC server's logging directory, usually \\IC\Logs\[date]\..., is the same information as the information generated and collected in the system log. Tracing bandwidth can be reduced by turning off the application log collection when the system log collection is enabled.

Application

This option sets the tracing level for the Application syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level when you no longer need it.

For more information, see Tracing levels for the Polycom Syslog.

4

Browser

This option sets the tracing level for the Browser syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level as soon as possible.

For more information, see Tracing levels for the Polycom Syslog.

4

Configuration

This option sets the tracing level for the Configuration syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level as soon as possible.

For more information, see Tracing levels for the Polycom Syslog.

4

Copy Utilities

This option sets the tracing level for the Copy Utilities syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level as soon as possible.

For more information, see Tracing levels for the Polycom Syslog.

4

CURL

This option sets the tracing level for the CURL syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level as soon as possible.

For more information, see Tracing levels for the Polycom Syslog.

4

Key Observer

This option sets the tracing level for the Key Observer syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level as soon as possible.

For more information, see Tracing levels for the Polycom Syslog.

4

SIP

This option sets the tracing level for the SIP syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level as soon as possible.

For more information, see Tracing levels for the Polycom Syslog.

4

Support Objects

This option sets the tracing level for the Support Objects syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level as soon as possible.

For more information, see Tracing levels for the Polycom Syslog.

4

TLS

This option sets the tracing level for the TLS syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level as soon as possible.

For more information, see Tracing levels for the Polycom Syslog.

4

Wapp Mgr

This option sets the tracing level for the Wapp Mgr syslog topic.

Note: When you increase the tracing level, phone performance may be adversely affected. This is especially true if you increase multiple tracing levels simultaneously Be sure to reduce the tracing level as soon as possible.

For more information, see Tracing levels for the Polycom Syslog.

4

Option

Description

Default  

Enable RTCP-XR Reports

This setting indicates whether Polycom phones to generate reports in RTCP XR packet metrics on listening and conversational quality.

No  

Enable Collector Session Reports

This setting indicates whether Polycom phones to create session type reports, which are generated at the end of a call.

No  

Enable Collector Periodic Reports

This setting indicates whether  Polycom phones to create periodic type reports, which are generated throughout a call.

No

Report Collector Address

This option sets the IP address or host name of the SIP server (report collector) that accepts voice quality reports contained in SIP PUBLISH messages. All IP addresses in managed phones configuration are pass-through strings to support IPv6.

Report Collector Port

This option sets the port of the SIP server (report collector) that accepts voice quality reports contained in SIP PUBLISH messages. If port is 0 or <Null>, port 5060 will be used.

Collector Period (5 to 20)

This option sets the time interval between successive periodic quality reports.

20

Option

Description

Default  

Enable Triggered Collector Periodic Reports

This option indicates whether the generation of periodic quality reports triggered by alert states.

The options are:

• Critical: periodic reports will be generated when an alert state is critical

• Warning or Critical: periodic reports will be generated when an alert state is either warning or critical

Disabled

MOS-LQ Warning Threshold (15 to 40)

This option sets the threshold value of listening MOS score that causes the IP phone to send a warning alert quality report.

Acceptable values are 15 to 40.

Not specified

(warning alerts are not generated)

MOS-LQ Critical Threshold (15 to 40)

This option sets the threshold value of listening MOS score that causes the IP phone to send a critical alert quality report.

Acceptable values are 15 to 40.

Not specified

(warning alerts are not generated)

Delay Warning Threshold (10 to 2000)

This option sets the threshold value of one way delay (in ms) that causes the IP phone to send a warning alert quality report.

Acceptable values are 10 to 2000.

Not specified

(warning alerts are not generated due to one way delay; includes both network and end system delays)

Delay Critical Threshold (10 to 2000)

This option sets the threshold value of one way delay (in ms) that causes the IP phone to send a critical alert quality report.

Acceptable values are 10 to 2000.

Not specified

(critical alerts are not generated due to one way delay; includes both network and end system delays)

Option

Description

Default  

Inbound SIP Security Challenge

This option sets the level of security the Polycom IP phone uses to validate the SIP inbound (to the phone versus inbound to CIC) traffic.  

The possible values are:

• Source: The IP phone checks the source IP address of the SIP request against the IP addresses of its SIP Registrations. If addresses do not match, the SIP request is rejected. If there is a match, the request is accepted.

• Digest: The IP phone sends a 401 Unauthorized challenge when a SIP request comes in, and the sender of the request must respond with the proper credentials for the SIP transaction to proceed.

• Source and Digest: The IP phone first applies the Source method, then the Digest method.

None (the phone does not take any steps to authenticate inbound SIP traffic)

Phone side only SRTP (Offloading)

This setting has the same effect as the as the audio protocol setting in the general configuration, however it only changes the phone setting.

No