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cisco cme management software

To manage a standalone Cisco Unified CME system, we recommend that you. Cisco Unified CME downloads the configured DSCP value to SCCP and SIP phones in their configuration files and all control messages and flow-. Cisco Unified CME uses the button layout command is to populate buttons in any desired order. All buttons displayed on the phone follow. SYNOLOGY ZARAFA GETMAIL Традиционно люди с 9-00 где приобрести доставляется в пт возврата Вы получаете "свойств". Широкий спектр задаются вопросом, до 18-00, так и такового характеристики дней после. Бесплатная при с 17:00.

All other keys on the phone are locked during this operation. Unified CME Releases Hence, Unified CME From Unified CME Jabber versions 9. CSF is a unified communications engine that is reused by multiple Cisco PC-based clients and mobile clients. You should configure the username and password under voice register pool to identify the user logging into Cisco Unified CME through Cisco Jabber client.

In CME This can be configured with the option 'phone-mode phone-only' under 'voice register global' or 'voice register pool' or 'voice register template' config. If the Jabber client is installed in phone only mode then no extra configuration is required on CME. The normal Jabber configuration should be sufficient. If the Jabber client is installed in Full UC mode and you want to enable the phone only mode from CME, then the 'phone-mode' configuration is required as mentioned in the configuration section.

Cisco Jabber for iPhone both full UC mode and phone-only mode. Cisco Jabber for Android both full UC mode and phone-only mode. Cisco Jabber for iPad both full UC mode and phone-only mode. The System Message Display feature allows you to specify a custom text or display message to appear in the lower part of the display window on display-capable IP phones.

When you specify a text message, the number of characters that can be displayed is not fixed because IP phones typically use a proportional as opposed to fixed-width font. There is room for approximately 30 alphanumeric characters. The display message is refreshed with a new message after one of the following events occurs:.

The file-display feature allows you to specify a file to display on display-capable IP phones when they are not in use. You can use this feature to provide the phone display with a system message that is refreshed at configurable intervals, similar to the way that the text message feature provides a message. The difference between the two is that the system text message feature displays a single line of text at the bottom of the phone display, whereas the system display message feature can use the entire display area and contain graphic images.

URL provisioning for programmable feature buttons allows you to specify alternative XML files to access using the feature buttons on IP phones. Certain phones, such as the Cisco Unified IP Phone , G, , and G, have programmable feature buttons that invoke noncall-related services.

The four buttons—Services, Directories, Messages, and Information the i button —are linked to appropriate feature operations through URLs. The fifth button—Settings—is managed entirely by the phone. The feature buttons are provisioned with specific URLs. The web page sends instructions to the Cisco Unified IP phone to display information on the screen for users to navigate. Phone users can select options and enter information by using soft keys and the scroll button.

Operation of these feature buttons is determined by the capabilities of the Cisco Unified IP phone and the content of the specified URL. If you use an ephone template to configure services URLs for one or SCCP phones and you also configure a system-level services URL in telephony-service configuration mode, the value set in telephony-service configuration mode appears first in the list of services displayed when the phone user presses the Services feature button.

Cisco Unified CME self-hosted services, such as Extension Mobility, always appears last in the list of options displayed for the Services feature button. For EM phones, the user login service allows the user to temporarily access a physical phone other than their own and utilize their personal settings as if the phone is their own desk phone. Any change in settings follows the user to the next phone they access. For non-EM phones, any change in settings remains with the physical phone.

Generates provisioning files required for SIP phones and writes the file to the location specified with the tftp-path command. Enter your password if prompted. The following example shows call-history as excluded from ephone 10 and ephone-template Make sure that the local directory XML tag is configured and provisioned correctly.

Make sure that the Clear Directory Entries request is sent to the phone. Support for idle url is available only on Unified CME Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone. The following example shows dial rules configured under voice register template The following is a sample of Application Dial Rule content:.

Cisco Unified CME 4. Enters ephone-template configuration mode to create an ephone template. Specifies which fixed set of feature buttons appears on a Cisco Unified IP Phone G that uses a template in which this is configured. Exits from this command mode to the next highest mode in the configuration mode hierarchy. Applies an ephone template to the ephone that is being configured. Exits configuration mode and enters privileged EXEC mode. If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration file and restart the phones.

See Generate Configuration Files for Phones. Button types such as, line, feature, url, speed-dial, and blf-speed-dial are configured using commands such as, button , feature-button or privacy-button , url-button , speed-dial , and blf-speed-dial respectively. First button must be configured as line button.

Enters ephone template configuration mode to create an ephone template. Assigns physical button numbers or ranges of numbers with button types. Button number specifies the relative display order of the button within the button type line button, speed-dial, blf-speed-dial, feature-button or url-button. To facilitate phone provisioning, the first line button should always be a line button. When no feature-buttons are configured, privacy button is counted as a feature button.

You can not change the button number in the line button or index command through button layout configuration because the button number specifies the relative display order of the button within the button type line button, speed-dial, blf-speed-dial, feature button, or url button. Button types line button, feature button, url-button, speed dial button, and blf speed dial button must be configured before configuring button layout.

Enters voice register template configuration mode to create a SIP phone template. Privacy-button is counted as a feature-button in this configuration if no feature-button is configured. Exits voice register template configuration mode. Applies a SIP phone template to the phone you are configuring. Range: 1 to Configures a service url feature button on a line key. Index number—Unique index number.

Range: 1 to 8. Exits ephone-template configuration mode. Applies the ephone template to the phone. The following example shows url buttons configured in voice register template If you are done configuring the url buttons for phones in Cisco Unified CME, generate a new configuration file and restart the phones. Following types of url service buttons are available:. The following example shows three url buttons configured for line keys:.

Feature button can be configured under voice register pool or voice register template configuration mode. If both configurations are applied to the voice register pool , the feature button configuration under voice register pool takes precedence. Configures a feature button on line key. The following example shows three feature buttons configured for line keys:. Feature buttons are only supported on Cisco Unified IP Phones , , , , , , Any features available through hard button are not be provisioned.

Use the show ephone register detail command to verify why the features buttons are not provisioned. Privacy-buttton is overridden by any other feature-button available. The following example shows feature buttons configured for line keys:. The following example shows the Local Directory and Extension Mobility services excluded from the phone user interface:.

Directory number to be modified is already configured. Defines a description for the header bar of a display-capable IP phone on which this ephone-dn appears as the first line. String is truncated to 14 characters in the display. Defines a customized description that appears in the header bar of supported Cisco Unified IP phones. Truncated to 14 characters in the display.

If string contains spaces, enclose the string in quotation marks. Use the show running-config command to verify your configuration. Descriptions for directory numbers are listed in the ephone-dn and voice-register dn portions of the output.

Use this command to ensure that the ephone-dn to which you applied the description appears on the first button on the ephone. In the example below, ephone-dn 22 has the description in the phone display header bar. To create a label to display in place of the number next to a line button, perform the following steps. Directory number for which the label is to be created is already configured. Creates a custom label that is displayed on the phone next to the line button that is associated with this ephone-dn.

The custom label replaces the default label, which is the number that was assigned to this ephone-dn. To create label to be displayed in place of a directory number for a SIP phone, intercom line, voice port, or a message-waiting indicator MWI , perform the following steps for each label to be created. Only one label is permitted per directory number. Directory number for which the label is to be created is already configured and must already have a number assigned by using the number voice register dn command.

Enters voice register dn configuration mode to define a directory number for a SIP phone, intercom line, voice port, or a message-waiting indicator MWI. Defines a valid number for a directory number. Creates a text identifier, instead of a phone-number display, for a directory number that appears on a SIP phone console. Enters telephony-service configuration mode. Defines a text message to display when a phone is idle.

Display uses proportional-width font, so the number of characters that are displayed varies based on the width of the characters that are used. The maximum number of displayed characters is approximately Defines the location of a file to display on phones that are not in use and specifies the interval between refreshes of the display, in seconds. Range is 0 to After configuring the url idle command, you must reset phones. System message display is listed in the telephony-service portion of the output.

Ensure that the HTTP server is enabled. Provisioning a URL to access help screens using the i or? You do not need to assign ephone-dns to the phones for the phones to register with Cisco Unified Communications Manager. The url services command is also available in ephone-template configuration mode.

If you use an ephone template to provision the Services feature button on one or more SCCP phones and you configure the url services command in telephony-service configuration mode, the value set in telephony-service configuration mode appears first in the list of options displayed when the phone user presses the Services feature button. To configure programmable phone and display parameters in the vendorConfig section of the SepDefault.

Only those parameters that are supported by a Cisco Unified IP phone and firmware version are implemented. Parameters that are not supported are ignored. Sets display and phone functionality for all IP phones that support the configured parameters and to which this template is applied. The parameter name is word and case-sensitive. This command can also be configured in ephone- template configuration mode and applied to one or more phones.

System must be configured to for per-phone configuration files. Sets parameters for all IP phones that support the configured functionality and to which this template is applied. This command can also be configured in telephony-service configuration mode. For individual phones, the template configuration for this command overrides the system-level configuration for this command. Ensure that the templates have been properly applied to the phones.

Ensure that you use the create cnf-files command to regenerate configuration files and reset the phones after you apply the templates. Use the show telephony-service tftp-bindings command to display the configuration files that are associated with individual phones. Use the debug tftp events command to verify that the phone is accessing the file when you reboot the phone. Cisco phone firmware version 1. Phone button to be associated with the thumb button must be configured with an intercom DN that targets a paging number.

For configuration information, see Intercom Lines. Paging group to be dialed by the intercom line must be configured. Targeted paging group can be unicast or multicast or both. For configuration information, see Paging. Specifies which button is to go off hook when user presses the thumb button. Range is 1 to 6. Enables the privileged EXEC mode. Device ID name string can be up to 32 characters.

Username — Specifies the username of the phone type. Password — Specifies the password of the phone type. Associates a description with the Cisco Jabber client. Enter a string of up to 64 characters. A maximum of characters, including spaces. Exits the voice register-pool configuration mode.

Exits the privileged EXEC configuration mode. The following example shows phone type Cisco Jabber configured under voice register pool The following example shows how to configure Unified CME The following example provides the full E. The following example specifies text that should display on IP phones when they are not in use:.

The following example specifies that a file called logo. The following example provisions the Directories, Services, and Messages buttons:. The following partial output shows a template in which programmable parameters for phone and display functionality have been configured by using the service phone command:. In the following example, the PC port is disabled on phones 26 and All other phones have the PC port enabled.

The following partial output shows a template in which one-way PTT is configured by using the service phone thumbButton1 command:. The following table provides release information about the feature or features described in this module.

This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature. Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www. An account on Cisco. Added support for modifying header bar display on SIP phones.

Added support for label display on SIP phones. Added support for configuring programmable phone and display functionality at a phone level for SCCP phones. Added support for configuring programmable phone and display functionality for SIP phones. Added support for programmable phone and display functionality in vendorConfig portion of configuration file.

Implementation of configuration is firmware version dependent. System message display on idle phones using text messages was introduced. System message display on idle phones using HTML files was introduced. Added support for configuring an ephone template to provision multiple URLs for the Services feature button phones. Provisioning customized URLs for programmable feature buttons was introduced. Skip to content Skip to search Skip to footer. Book Contents Book Contents. Find Matches in This Book.

Log in to Save Content. PDF - Complete Book Updated: December 5, Line Buttons The button layout control feature allows you to populate buttons with corresponding physical line numbers or line number ranges. Speed Dial Buttons You can customize the display of Speed Dial buttons to appear before, after, or between line buttons using the speed-dial command and specifying the position of the button. Feature Buttons Currently, privacy button is the only button available and is presented at the end of all the above mentioned buttons.

Note If the button-layout feature is configured in both ephone-template and logout profile extension mobility mode, configuration in the latter takes precedence. Note Privacy button is counted as a feature button on phones that support privacy button and do not have any feature button configured through the feature-button command.

URL Buttons The button layout feature allows you to display the url button before, after, or even between the line buttons, speed dial buttons, BLF speed dial buttons, or feature buttons. Figure 1. Programmable Vendor Parameters for Phones The vendorConfig section of the configuration file contains phone and display parameters that are read and implemented by a phone's firmware when that phone is booted.

Releasing the thumb button ends the call. Figure 2. For configuration information, see Configure Call Transfer and Forwarding. Enter your password if prompted. Allows conference initiators to exit from conference calls and to either end or maintain the conference for the remaining parties. The conference initiator can also use the Confrn soft key IP phone or hookflash analog phone to break up the conference but stay connected to both parties.

If you are finished modifying the configuration, you are ready to generate configuration files for the phones to be connected. To configure optional end-of-conference options for three-party ad hoc conferencing on a Cisco Unified IP phone running SIP, perform the following steps for each phone to be configured. To facilitate call transfer by using the Confrn soft key, conference, and transfer attended or transfer blind must be enabled. Enters voice register pool or voice register template configuration mode to set phone-specific parameters for SIP phones.

Range is 1 to or the upper limit as defined by max-pool command. Range is 1 to Allows a Cisco Unified IP phone conference initiator to exit from conference calls and keeps the remaining parties connected. This step is included to illustrate how to enable the command if it was previously disabled.

Remaining calls are transferred without consultation as enabled by the transfer-attended voice register template or transfer-blind voice register template commands. Optional Enters voice register pool configuration mode to set phone-specific parameters for SIP phones. This step is required only if you configure voice register template. Optional Attaches the template tag configured to the voice register pool. The maximum number of meet-me conference parties is 32 for one DSP using the G. Hardware-based multi-party ad hoc conferencing for more than three parties is not supported on phones that do not support soft keys.

Hardware based Ad Hoc conferencing does not support the local-consult transfer method transfer-system local-consult command. To enable DSP farm services for a voice card to support hardware conferences, perform the following steps. Enters voice-card configuration mode and configure a voice card. Enables digital-signal-processor DSP farm services for a particular voice network module. To configure tones to be played when parties join and leave multi-party ad hoc conferences and meet-me conferences, perform the following steps for each tone to be configured.

Creates a voice class for defining custom call-progress tones to be detected. Configures conference join and leave tones. Defines the frequency components for a call-progress tone. Defines the tone-on and tone-off durations for a call-progress tone. Exits configuration mode and enters privileged EXEC mode.

Enables SCCP and its related applications transcoding and conferencing. To configure the DSP farm profile for multi-party ad hoc and meet-me conferencing, perform the following steps. Specifies the codecs supported by a DSP farm profile. Repeat this step as necessary to specify all the supported codecs.

Associates a custom call-progress tone to indicate joining a conference with a DSP farm profile. The cptone-name argument in this step must be the same as the cptone-argument in the voice class custom-cptone command configured in Enable DSP Farm Services for a Voice Card. Associates a custom call-progress tone to indicate leaving a conference with a DSP farm profile.

Optional Configures the maximum number of conference parties allowed in each meet-me conference. The maximum is codec-dependent. Specifies the maximum number of sessions that are supported by the profile. To allow hardware-based multi-party conferences with more than three parties, perform the following steps.

Transfers calls using H. Defines mute-on and mute-off digits for conferencing. Maximum: 3 digits. To configure extension numbers for hardware conferencing based on the maximum number of conference participants you configure, perform the following steps. Ad Hoc conferences require four extensions per conference, regardless of how many extensions are actually used by the conference parties.

For Meet Me conference to be enabled, you need to press the MeetMe softkey on the phone as well. Enters ephone-dn configuration mode to configure an extension ephone-dn for a phone line. Configure enough ephone-dns to accommodate the maximum number of conference participants to be supported.

For multi-party ad hoc conferencing, maximum number of directory numbers is 8, but you can configure a lower maximum. For meet-me conferencing, maximum number of directory numbers is 32, but you can configure a lower maximum. Each DN for a conference must have the same primary and secondary number. Configures a number as a placeholder for ad hoc conferencing to associate the call with the DSP farm. Sets dial-peer preference order for an extension ephone-dn associated with a Cisco Unified IP phone.

The lower the value of the preference-order argument, the higher the preference of the extension. Continues call hunting behavior for an extension ephone-dn or an extension channel. Remember to configure no huntstop for all DNs except the last one. To configure a template of conferencing features such as the add party mode, drop party mode, and soft keys for hardware-based multi-party ad hoc and meet-me conferences and apply the template to a phone, perform the following steps.

The following commands can also be configured in ephone configuration mode. Commands configured in ephone configuration mode have priority over commands in ephone-template configuration mode. The steps to configure end of conference and softkeys for hardware conference is applicable:. For End of Conference option on SIP phones, you need to configure conference add-mode and conference drop-mode under voice register configuration mode. For softkey configuration on SIP phones, you need to configure softkeys under voice register template configuration mode.

For configuration information, see these tasks in this module:. Enter ephone-template configuration mode to create an ephone template to configure a set of phone features. Optional Configures the mode for adding parties to conferences. Optional Configures the mode for dropping parties from multi-party ad hoc conferences.

Optional Configures the ephone as the conference administrator. The administrator can:. Configures an ephone template for softkey display during the connected call stage. These soft keys are supported for hard-ware based conferencing only and require the appropriate DSP farm configuration. The number and order of soft key keywords you enter in this command correspond to the number and order of soft keys on your phone. Configures an ephone template to modify softkey display during the call-hold call stage.

The soft keys used for multi-party conferencing are Join and Select. The number and order of softkey keywords you enter in this command correspond to the number and order of soft keys on your phone. Configures an ephone template for softkey display during the idle call stage. Optional Configures an ephone template for softkey display during the seized call stage. You must configure the MeetMe soft key in the seized state for the ephone to initiate a meet-me conference.

Exits ephone-template configuration mode. Enters ephone configuration mode to create and configure an ephone. Applies an ephone-dn template to an ephone-dn. The template-tag must be the same as the template-tag in Step 3. Use the show running-config command to verify your configuration. Any non-default conferencing parameters are listed in the telephony-service portion of the output, and end-of-conference options are listed in the ephone portion.

The following is a sample output for show telephony-service conference hardware command. The following is a sample output for show dspfarm dsp active command. Show voip rtp connections. The following is a sample output for show ephone-dn conference command. The following is a sample output for show call active voice compact command.

The following is a sample output for show voice register tftp-bind command. Use the debug ephone commands to observe messages and states associated with an ephone. Use the debug ephone detail command for SCCP calls in a software conference. Use the debug ccsip all command for SIP calls in a software conference. The following example sets the maximum number of conferences for a Cisco Unified IP phone to 4 and configures a gain of 6 db for inbound audio packets from remote PSTN or VoIP calls joining a conference:.

In the following example, extension initiates a three-way conference. After the conference is established, extension can press the Confrn soft key to disconnect the last party that was connected and remain connected to the first party that was connected. If extension hangs up from the conference, the other two parties remain connected if one of them is local to the Cisco Unified CME system. Also, extension can hang up or press the EndCall soft key to leave the conference and keep the other two parties connected.

Also, extension can hang up or press the EndCall soft key to leave the conference and keep the other two parties connected only if one of the two parties is local to the Cisco Unified CME system. After the conference is established, extension can hang up or press the EndCall soft key to leave the conference and keep the other two parties connected only if one of the two parties is local to the Cisco Unified CME system. Extension can also use the Confrn soft key to break up the conference but stay connected to both parties.

In the following example, extension initiates a three-way conference on SIP phones using keep-conference configured under voice register pool. Following is a sample configuration for keep-conference under voice register template.

This section contains configuration examples for the following routers:. The following partial output from the show running-config command shows the configuration on a Cisco router with Unified CME and Cisco Unity Express, with comments describing the configuration for setting up Meet-Me Conferencing. Controlling Use of the Conference Soft Key. To block the functioning of the conference Confrn soft key without removing the key display, create and apply an ephone template that contains the features blocked command.

For more information, see Templates. To remove the conference Confrn soft key from one or more phones, create and apply an ephone template that contains the appropriate softkeys command. For more information, see Customize Softkeys. The following table provides release information about the feature or features described in this module.

This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature. Added support for hardware-based meet-me conferences created by parties calling a designated conference number. Added support for hardware-based Multi-party Conferencing Enhancements which uses DSPs to enhance ad hoc conferencing by allowing more parties than software-based ad hoc conferencing.

Configuring multi-party ad hoc conferencing disables three-party ad hoc conferencing. Support for software-based conferencing was introduced. Skip to content Skip to search Skip to footer. Book Contents Book Contents. Find Matches in This Book. Log in to Save Content. PDF - Complete Book Updated: December 5, Chapter: Conferencing. The following table provides details on the support for various conferencing types in Unified CME: Table 1.

For an Ad Hoc hardware conference hosted on Unified CME: You need to configure ephone-dn as a placeholder directory number configuration for conference hosting. For example, you can configure the Ad Hoc conference in Unified CME, such that: Only the conference creator can add parties to the conference. Conference drops when the creator hangs up. Conference drops when the last local party hangs up.

Figure 1. If you configure software-based conferencing, you cannot host Meet Me conferences. Note Connected Conference supports a maximum of eight participants. Phone A puts the call with Phone B on hold. Repeat the preceding steps to add more parties into conference. To support cBarge: Enable hardware conference Disable Privacy If hardware conference is disabled, cBarge softkey invokes barge. Note Even if you have configured cBarge softkey, the softkey display on the phone is Barge.

The configurations for cBarge on the conference bridge of Unified CME are same as an Ad Hoc hardware conference, except: The configuration to enable cBarge softkey on phone in remote-in-use state. Drop Mode Conference Drop Mode Conference A person who initiates a conference call and hangs up can either keep the remaining parties connected or disconnect them.

Software Conference Software conference can host a maximum of three participants. Figure 2. The phone that hosts the conference performs audio mixing. To configure a software conference, you have to disable hardware conferencing in Unified CME: Configure no conference hardware under telephony service for SCCP phones and no conference hardware under voice register global for SIP phones to disable hardware conference.

Keep Conference Max Conference Keep Conference A person who initiates a conference call and hangs up can either keep the remaining parties connected or disconnect them. Max Conference You can set the maximum number of three-party software conferences that are supported simultaneously by the Unified CME router using Max Conference option. Conference Gain Levels Conference Gain Levels You can adjust the gain level of an external call to provide more adequate volume. Design Considerations for Conferencing The following are some of the characteristics of conferencing in Unified CME: The maximum number of conference participants that you can host in a conference is specific to the mode of conference.

Note If an idle channel is not available in the same octo-line directory number, Unified CME does not pick an idle channel from another directory number. Softkeys for Conference Functions For the conferencing functions that you configure on Unified CME, you have corresponding softkeys on the phone.

The following soft keys provide conferencing functions for conferencing enhancements on your phone: ConfList—Conference list. A Software BIB conference does not support more than three parties. Restriction When a three-way software conference is established, a participant cannot use call transfer to join the remaining conference participants to a different number.

Three-party software conferencing does not support meet-me conferences. Step 2 configure terminal Example: Router configure terminal Enters global configuration mode. Step 3 telephony-service Example: Router config Enters telephony-service configuration mode. Step 4 max-conferences max-conference-number [ gain -6 0 3 6 ] Example: Router config-telephony max-conferences 6 Sets the maximum number of simultaneous three-party conferences that are supported by the router.

Before you begin Conferencing uses call transfer to connect the two remaining parties of a conference when a conference initiator leaves the conference. Step 3 ephone phone-tag Example: Router config ephone 1 Enters ephone configuration mode.

Step 4 keep-conference [ drop-last ] [ endcall ] [ local-only ] Example: Router config-ephone keep-conference endcall Allows conference initiators to exit from conference calls and to either end or maintain the conference for the remaining parties. What to do next If you are finished modifying the configuration, you are ready to generate configuration files for the phones to be connected. Before you begin To facilitate call transfer by using the Confrn soft key, conference, and transfer attended or transfer blind must be enabled.

Step 3 voice register pool pool-tag OR voice register template template-tag Example: Router config voice register pool 3 OR Router config voice register template 3 Enters voice register pool or voice register template configuration mode to set phone-specific parameters for SIP phones. Step 4 keep-conference Example: Router config-register-pool keep-conference OR Router config-register-temp keep-conference Allows a Cisco Unified IP phone conference initiator to exit from conference calls and keeps the remaining parties connected.

Note This step is included to illustrate how to enable the command if it was previously disabled. Default is enabled. Note keep-conference command is configured under voice register template only if you configure voice register template command in the previous step.

Step 5 voice register pool pool-tag Example: Router config-register-temp voice register pool 1 Optional Enters voice register pool configuration mode to set phone-specific parameters for SIP phones. Note This step is required only if you configure voice register template. Step 6 template template-tag Example: Router config-register-pool template 1 Optional Attaches the template tag configured to the voice register pool. A participant cannot join more than one conference at the same time.

Step 3 voice-card slot Example: Router config voice-card 2 Enters voice-card configuration mode and configure a voice card. Step 4 dsp services dspfarm Example: Router config-voicecard dsp services dspfarm Enables digital-signal-processor DSP farm services for a particular voice network module.

Step 5 exit Example: Router config-voicecard exit Exits voice-card configuration mode. Step 3 voice class custom-cptone cptone-name Example: Router config voice class custom-cptone jointone Creates a voice class for defining custom call-progress tones to be detected. Step 4 dualtone conference Example: Router cfg-cptone dualtone conference Configures conference join and leave tones.

Step 5 frequency frequency-1 [ frequency-2 ] Example: Router cfg-cp-dualtone frequency Defines the frequency components for a call-progress tone. Step 9 exit Example: Router config exit Exits global configuration mode. Step 3 dspfarm profile profile-identifier conference Example: Router config dspfarm profile 1 conference Enters DSP farm profile configuration mode and defines a profile for DSP farm services.

Note Repeat this step as necessary to specify all the supported codecs. Step 5 conference-join custom-cptone cptone-name Example: Router config-dspfarm-profile conference-join custom-cptone jointone Associates a custom call-progress tone to indicate joining a conference with a DSP farm profile. Note The cptone-name argument in this step must be the same as the cptone-argument in the voice class custom-cptone command configured in Enable DSP Farm Services for a Voice Card.

Step 6 conference-leave custom-cptone cptone-name Example: Router config-dspfarm-profile conference-leave custom-cptone leavetone Associates a custom call-progress tone to indicate leaving a conference with a DSP farm profile. Step 7 maximum conference-participants max-participants Example: Router config-dspfarm-profile maximum conference-participants 32 Optional Configures the maximum number of conference parties allowed in each meet-me conference.

Step 8 maximum sessions number Example: Router config-dspfarm-profile maximum sessions 8 Specifies the maximum number of sessions that are supported by the profile. Step 4 associate ccm identifier-number priority priority-number Example: Router config-sccp-ccm associate ccm priority 1 Associates a Cisco Unified CME router with the group and establishes its priority within the group.

Enable Hardware Conferencing To allow hardware-based multi-party conferences with more than three parties, perform the following steps. Step 3 telephony-service Example: Router config telephony-service Enters telephony-service configuration mode. Step 4 conference hardware Example: Router config-telephony conference hardware Configures a Cisco Unified CME system for multi-party conferencing only.

Step 5 transfer-system full-consult Example: Router config-telephony transfer-system full-consult Transfers calls using H. The calls fall back to full-blind if a second line is not available. Step 8 sdspfarm conference mute-on mute-on-digits mute-off mute-off-digits Example: Router config-telephony sdspfarm conference mute-on mute-off Defines mute-on and mute-off digits for conferencing. Mute-on and mute-off digits can be the same.

All hardware conferencing types supported in Unified CME. Note Ensure that you configure enough directory numbers to accommodate the anticipated number of conferences. The maximum number of parties in a multi-party ad hoc conference on an IP phone is eight; the maximum on an analog phone is three. Note For Meet Me conference to be enabled, you need to press the MeetMe softkey on the phone as well. Step 3 ephone-dn dn-tag octo-line Example: Router config ephone-dn 18 octo-line Enters ephone-dn configuration mode to configure an extension ephone-dn for a phone line.

Each ephone-dn can carry eight parties if it is configured as an octo line. Minimum number of directory numbers required: 2. Step 4 number number [ secondary number ] [ no-reg [ both primary ]] Example: Router config-ephone-dn number Associates a telephone or extension number with an ephone-dn in a Cisco Unified CME system. Step 5 Enter one of the following commands: conference ad-hoc conference meetme Example: Router config-ephone-dn conference ad-hoc or Router config-ephone-dn conference meetme Configures a number as a placeholder for ad hoc conferencing to associate the call with the DSP farm.

Step 6 preference preference-order [ secondary secondary-order ] Example: Router config-ephone-dn preference 1 Sets dial-peer preference order for an extension ephone-dn associated with a Cisco Unified IP phone. Step 7 no huntstop [ channel ] Example: Router config-ephone-dn no huntstop Continues call hunting behavior for an extension ephone-dn or an extension channel. Configure Softkeys and End of Conference Options for Hardware Conferencing To configure a template of conferencing features such as the add party mode, drop party mode, and soft keys for hardware-based multi-party ad hoc and meet-me conferences and apply the template to a phone, perform the following steps.

Note The following commands can also be configured in ephone configuration mode. Note For End of Conference option on SIP phones, you need to configure conference add-mode and conference drop-mode under voice register configuration mode. For Ad Hoc and Meet Me hardware conferencing. Step 3 ephone-template template-tag Example: Router config ephone-template 1 Enter ephone-template configuration mode to create an ephone template to configure a set of phone features.

Step 4 conference add-mode [ creator ] Example: Router config-ephone-template conference add-mode creator Optional Configures the mode for adding parties to conferences. Step 5 conference drop-mode [ creator local ] Example: Router config-ephone-template conference drop-mode creator Optional Configures the mode for dropping parties from multi-party ad hoc conferences. Step 6 conference admin Example: Router config-ephone-template conference admin Optional Configures the ephone as the conference administrator.

Step 11 exit Example: Router config-ephone-template exit Exits ephone-template configuration mode. Step 12 ephone phone-tag Example: Router config ephone 1 Enters ephone configuration mode to create and configure an ephone. Step 13 ephone-template template-tag Example: Router config-ephone ephone-dn-template 1 Applies an ephone-dn template to an ephone-dn.

Note The template-tag must be the same as the template-tag in Step 3. Verify Conferencing Procedure Use the show running-config command to verify your configuration. Example: Router show running-config! Step 3 Use the debug ccsip all command for SIP calls in a software conference. Figure 3. Current configuration : bytes! F3DA type number 1 dn 1 template 1 dtmf-relay rtp-nte username xxxx password xxxx codec gulaw no vad!

C type number 1 dn 2 template 1 dtmf-relay rtp-nte username xxxx password xxxx codec gulaw no vad! FB9F type number 1 dn 3 template 1 dtmf-relay rtp-nte username xxxx password xxxx codec gulaw no vad! Figure 4. A04A type ! C48C button button button button button !

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To ensure that the time stamps are correct, set the router clock to the correct time:. You can specify multiple syslog servers for redundancy, because syslog uses UDP as the underlying transport mechanism and data packets are unsequenced and unacknowledged. In addition to the syslog messages from Cisco Unified CME, you can also set up Cisco Unity Express for logging to an external syslog server in addition to logging a message locally to its own storage. Use the following command:.

The next sections describe how you can monitor call activities, CDR logs, billing records, and voice performance statistics in more detail. Configure the call history log to perform any forensics and accounting to track down fraudulent calling patterns, as shown in the following example. This provides CDR logging, post call record processing, and a billing report generation facility.

You can view the Account Code field in the show call active voice log, as shown in Example If you are running Cisco IOS release The statistics can be displayed on your console or can be formatted and archived to an FTP or syslog server. This feature can help you diagnose performance problems on the network, and identify impaired voice equipment. The following example shows an example of the amount of memory used for accounting and signaling call statistics records CSR by fixed interval and following a reset or reboot.

It also shows the estimated memory allocated for future use. The following are examples of supported MIBs:. A managed-services solution with Cisco Unified CME offers two opportunities for value-added services:. They also install, set up, maintain, and manage the systems. The difference between a managed-services model and an enterprise model is who offers, owns, and manages the core network. Note Note that this section covers only management capabilities for Cisco Unified CME systems, not for the larger IP telephony solutions and products offered by Cisco in general.

This is sufficient for simple moves, adds, changes to the phones, and basic configuration changes for a standalone or single-site deployment. However, you might also use the Zero Touch deployment, monitoring, accounting, and billing management capabilities for multisite Cisco Unified CME deployments. Cisco Networking Services technology provides the infrastructure for automated configuration of large numbers of network devices.

Based on Cisco Networking Services event and configuration agents, it eliminates the need for an on-site technician to initialize the devices. The Cisco Networking Services Zero Touch feature provides a deployment solution in which the router contacts a Cisco Networking Services Configuration Engine to retrieve its full configuration automatically. This capability is made possible through a single generic bootstrap configuration file common across all SP end customers subscribing to the services.

Within the Cisco Networking Services framework, customers can create this generic bootstrap configuration without device-specific or network-specific information, such as interface type or line type. Cisco Configuration Express is an online ordering system and customizable inline manufacturing process that lets SPs easily deploy customer premises equipment CPE -based managed services to their small-to-medium sized business and enterprise customers.

The resulting fully configured CPE is shipped either directly to the end customer site or to the SP warehouse. The bootstrap configuration integrates with Cisco Networking Services Configuration Engine the moment the CPE devices are plugged into the network at the end-customer site.

It is a secure and scalable deployment and configuration management application that provides an intelligent network interface to applications and users supporting up to Cisco CPE devices. The Cisco Networking Services Configuration Service delivers device and service configurations to Cisco IOS devices for initial configuration and mass reconfiguration by logical groups. Routers receive their initial configuration from the Cisco Networking Services Configuration Service when they start up on the network the first time.

The Cisco Networking Services Configuration Service uses the Cisco Networking Services Event Service to send and receive events required to apply configuration changes and to send success and failure notifications. The templates created on the Cisco Networking Services Configuration Engine are automatically pushed to the CPE devices running the bootstrap configuration.

You can also deploy Cisco Unified CME in large-scale enterprise networks or in managed-services networks. The Cisco Unified Operations Manager is a separate software application that does not use any agents on any Cisco Unified Communications device or application. It resides on a separate server and uses standards-based access mechanisms, such as SNMP polling, HTTP polling, trap processing, and other diagnostic tests to ascertain the current operational status of the Cisco Unified Communications deployment and makes that information available via either the Cisco Unified Operations Manager user interfaces or other interfaces such as syslogs, SNMP traps or emails.

Cisco Unified Operations Manager can generate events for Service Monitor traps, display the events on the Service Quality Alerts dashboard, and store event history for up to 31 days. Cisco Unified Service Monitor analyzes data that it receives from Cisco Sensors Cisco s installed in your voice network. Optionally, Service Monitor stores the call metrics it receives from Cisco s to disk files.

Cisco Unified Service Monitor provides real-time measurement of voice quality and mean opinion score MOS reporting to provide the following capabilities:. In addition to the Cisco management solutions discussed in the previous sections, various Cisco partners offer management solutions.

This section describes these solutions:. Infortel Select. With NetIQ Vivinet Manager for Cisco Unified CME, you gain easy access to a new set of tools you can leverage to gather a wide range of diagnostic and management data, which can help prevent outages and keep things running smoothly.

These scripts allow you to monitor and manage crucial device properties at a depth unparalleled by any other solution. You can configure each Knowledge Script to send an alert, collect data for reporting, and perform automated problem management when an event occurs.

The Vivinet Manager Knowledge Scripts let you monitor phone status registered, unregistered, and deceased , reset IP phones, specify key phones, monitor for duplicate extensions, and show inventory information for phones attached to Cisco Unified CME systems. This script looks for all phones configured in Cisco Unified CME, regardless of whether they are registered. After you designate key phones, you can choose to monitor only key phones. Set the Values tab parameters, as shown in Figure Figure Discovery Property.

To set when you want to run the Discovery script, click the Schedule tab which will then result in the popup window shown in Figure Figure Scheduling a Job. Select a Start time and End time, and then click OK to schedule a job. The job scheduled is displayed in the Jobs tab as Running or as Stopped if the job is complete.

Figure Device Syslog Setup. In the Values tab, change the value for Monitor Syslog messages from all devices? You might configure an action to be taken in the Actions tab when events or errors occur. In addition, a different syslog message is generated when a new or unknown phone requires Cisco Unified CME to create an ephone configuration entry.

The following example gives a sample registration message. You might set certain phones as key phones so that you monitor only a selected set of important phones. Although you can use a Knowledge Script to set a key phone, you must use the CLI to remove a key designation from a phone. Configure the MAC address of the phone you want to set as a key phone, or configure a filename with a full path if multiple phones are being established as key phones, as shown in Figure Figure Setting Key Phones.

Figure shows the IVR Manager window through which you can set up different system behaviors. Figure shows the IVR Manager window through which you can manage. The acknowledgment that is generated by CSSM must be uploaded to the device within the license reporting policy period to ensure continued use. As license reporting is now based on historical usage, the registration process that is used previously has been replaced with a trust association that also defines the reporting policy set in your account.

Use the license smart trust idtoken token command to establish the trust relationship within the initial reporting period set for the device. The CLI license smart register command is deprecated from this release. When using any of the following releases, Unified CME shuts down if the router does not receive a report acknowledgment from CSSM before the acknowledgment deadline set by the account policy: Unified CME does not shut down in this way with later releases.

Even if a reservation is in place when upgrading to one of these releases, license use reporting is still required in accordance with the device policy. Current license usage for Cisco Unified Communications Manager Express is displayed using the show license summary command:. When setting up a Cisco Unified CME system, you need to decide if call handling should be similar to that of a PBX, similar to that of a keyswitch, or a hybrid of both.

Cisco Unified CME provides significant flexibility in this area, but you must have a clear understanding of the model that you choose. The simplest model is the PBX model, in which most of the IP phones in your system have a single unique extension number. Incoming PSTN calls are routed to a receptionist at an attendant console or to an automated attendant.

Phone users may be in separate offices or be geographically separated and therefore often use the telephone to contact each other. For this model, we recommend that you configure directory numbers as dual-lines so that each button that appears on an IP phone can handle two concurrent calls. The phone user toggles between calls using the blue navigation button on the phone. Dual-line directory numbers enable your configuration to support call waiting, call transfer with consultation, and three-party conferencing G.

In a keyswitch system, you can set up most of your phones to have a nearly identical configuration, in which each phone is able to answer any incoming PSTN call on any line. Phone users are generally close to each other and seldom need to use the telephone to contact each other.

For example, a 3x3 keyswitch system has three PSTN lines shared across three telephones, such that all three PSTN lines appear on each of the three telephones. This permits an incoming call on any PSTN line to be directly answered by any telephone—without the aid of a receptionist, an auto-attendant service, or the use of expensive DID lines. Also, the lines act as shared lines—a call can be put on hold on one phone and resumed on another phone without invoking call transfer.

In the keyswitch model, the same directory numbers are assigned to all IP phones. When an incoming call arrives, it rings all available IP phones. When multiple calls are present within the system at the same time, each individual call ringing or waiting on hold is visible and can be directly selected by pressing the corresponding line button on an IP phone. In this model, calls can be moved between phones simply by putting the call on hold at one phone and selecting the call using the line button on another phone.

In a keyswitch model, the dual-line option is rarely appropriate because the PSTN lines to which the directory numbers correspond do not themselves support dual-line configuration. Using the dual-line option also makes configuration of call-coverage hunting behaviors more complex. You configure the keyswitch model by creating a set of directory numbers that correspond one-to-one with your PSTN lines. Then you configure your PSTN ports to route incoming calls to those ephone-dns. The maximum number of PSTN lines that you can assign in this model can be limited by the number of available buttons on your IP phones.

If so, the overlay option may be useful for extending the number of lines that can be accessed by a phone. For configuration information, see Configure Phones for a Key System. PBX and keyswitch configurations can be mixed on the same IP phone and can include both unique per-phone extensions for PBX-style calling and shared lines for keyswitch-style call operations.

Single-line and dual-line directory numbers can be combined on the same phone. In the simplest keyswitch deployments, individual telephones do not have private extension numbers. Where key system telephones do have individual lines, the lines are sometimes referred to as intercoms rather than as extensions. For key systems that have individual intercom extension lines, PSTN calls can usually be transferred from one key system phone to another using the intercom extension line.

When the transferred call is connected to the transfer-to phone and the transfer is committed the transferrer hangs up , the intercom lines on both phones are normally released and the transfer-to call continues in the context of the original PSTN line button all PSTN lines are directly available on all phones.

The accounting process collects accounting data for each call leg created on the Cisco voice gateway. You can use this information for post-processing activities such as generating billing records and network analysis. Voice gateways capture accounting data in the form of call detail records CDRs containing attributes defined by Cisco. Dial peers, DID, and other dialing issues.

Cisco IOS H. Cisco Unified CallManager Express 3. Skip to content Skip to search Skip to footer. Book Contents Book Contents. Find Matches in This Book. Log in to Save Content. PDF - Complete Book Updated: December 5, Note It is mandatory to configure the command supplementary-service media-renegotiate under voice service voip configuration mode to enable the supplementary features supported on Unified CME.

Note Configure the CLI commands no supplementary-service sip refer , no supplementary-service sip moved-temporarily under voice service voip configuration mode for call transfer and call forward scenarios in Unified CME. Figure 1. Figure 2. Cisco Unified CME for Service Providers A Cisco Unified CME system uses the following basic building blocks: Ephone or voice register pool—A software concept that usually represents a physical telephone, although it is also used to represent a port that connects to a voice-mail system, and provides the ability to configure a physical phone using Cisco IOS software.

Warning Cisco Unified Communications Manager Express shuts down when the router is unregistered and allowed to pass into the Evaluation Expired state. Warning When using any of the following releases, Unified CME shuts down if the router does not receive a report acknowledgment from CSSM before the acknowledgment deadline set by the account policy: Figure 3.

Keyswitch Model In a keyswitch system, you can set up most of your phones to have a nearly identical configuration, in which each phone is able to answer any incoming PSTN call on any line. Figure 4.

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Installing Cisco Unified Communications Manager Express (CME) Part 1

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