Cme software cisco

Cisco Smart Licensing is a flexible licensing model that streamlines how you activate and manage software. Start by getting access to your company's existing Smart Account. Request an account for your company and delegate another administrator. Download and manage Smart Software Manager Track and manage your licenses. Convert traditional licenses to Smart Licenses. Manage licenses. Download and Upgrade Download new software or updates to your current software. Access downloads.

Traditional Licenses Generate and manage PAK-based and other device licenses, including demo licenses. Access LRP. Manage Smart Account Update your profile information and manage users. Hence, the GUI files posted under the name cme-gui Unified CME Release 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.

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. Cisco Unified CME allows small business customers and autonomous small enterprise branch offices to deploy voice, data, and IP telephony on a single platform for small offices, thereby streamlining operations and lowering network costs.

Cisco Unified CME is ideal for customers who have data connectivity requirements and also have a need for a telephony solution in the same office. The ability to deliver IP telephony and data routing by using a single converged solution allows customers to optimize their operations and maintenance costs, resulting in a very cost-effective solution that meets office needs.

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.

Each phone can have multiple extensions associated with it and a single extension can be assigned to multiple phones. Maximum number of ephones and voice register pools supported in a Cisco Unified CME system is equal to the maximum number of physical phones that can be connected to the system. Directory number—A software concept that represents the line that connects a voice channel to a phone. A directory number represents a virtual voice port in the Cisco Unified CME system, so the maximum number of directory numbers supported in Cisco Unified CME is the maximum number of simultaneous call connections that can occur.

This concept is different from the maximum number of physical lines in a traditional telephony system. Devices send license usage to CSSM either directly or use an on-premises satellite. CSSM shows license usage across all devices that you register to a virtual account. A Virtual Account License Inventory displays the quantity of licenses that you purchase, those licenses in use, and a balance. Alert Insufficient Licenses is displayed if the license balance is below 0. To avoid unnecessary reporting while you configure Unified CME, license usage is reported three minutes after the last configuration change.

Use the license smart enable command to enable Smart Licensing. To disable Smart Licensing, use the no form of the command and reaccept the EULA using the license accept end user agreement configuration command. Smart Licensing is mandatory from this release. Routers configured to use Smart Licensing offer a day evaluation period, during which you can use all the features without registering to CSSM.

You can obtain the registration token from the virtual CSSM account or from an on-premises satellite. Once registered, the evaluation period pauses and you can use the balance later. You cannot renew the evaluation period on its expiry. Cisco Unified Communications Manager Express shuts down when the router is unregistered and allowed to pass into the Evaluation Expired state.

Upon successful registration, the device sends an authorization request to CSSM for the licenses in use. Post successful authorization of the request, licenses are bound to the requesting device until the next authorization request submission.

An authorization request is sent every 30 days or when there is any change in license consumption, to maintain the registration with CSSM. The authorization expires if you do not update the license request for the router within 90 days. The certificate issued to identify the router at the time of registration is valid for one year and renewed every six months.

To reserve specific licenses for a device, generate request code from the device. Enter the request code in CSSM along with the required licenses and their quantity, and generate authorization code. The reservation may be updated before or after the software upgrade. This release introduces a new paradigm for tracking license usage across your business. In earlier releases, license authorization was forward looking, binding licenses to a device until the next authorization request. Actual license usage during the proceeding reporting period is now sent to CSSM, allowing you to plan ongoing license requirements based on historical usage data.

Initial device registration is no longer required to use most platform functionality and the evaluation period is deprecated. License usage reports are submitted periodically according to a minimum reporting policy set for your account.

In an Ad Hoc software conference, the phone that hosts the conference also performs audio mixing. The conference that is shown in AdHoc Software Conference Using the Conference Softkey is created when extension dials extension The two parties decide to add a third party, extension Extensions , , and are now parties in an ad hoc conference.

Extension is the conference initiator. Hence, audio mixing happens in Transcoding is not supported in a software-based conference call. Hence, you cannot host a software conference for calls with different audio codecs. Software conference is enabled using softkeys on the Unified IP phones.

The softkey varies depending on the phone model used. 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. Also, you must configure create profile under voice register global and create cnf-files under telephony-service configuration mode.

Keep Conference is an end of conference option in Software Conferencing. With Keep Conference option, Unified IP phones can be configured to keep the remaining conference parties connected when the conference initiator hangs up places the handset back in the on-hook position.

Conference originators can disconnect from their conference calls by pressing the Confrn conference soft key. When an initiator uses the Confrn key to disconnect from the conference call, the oldest call leg will be put on hold, leaving the initiator connected to the most recent call leg.

The conference initiator can then navigate between the two parties by pressing either the Hold soft key or the line buttons to select the desired call. You can set the maximum number of three-party software conferences that are supported simultaneously by the Unified CME router using Max Conference option. Configure the max-conferences command in telephony-service configuration mode to define maximum number of software conferences. You can adjust the gain level of an external call to provide more adequate volume.

This functionality is applied to inbound audio packets so that conference participants can more clearly hear a remote PSTN or VoIP caller joining their call. Conference gain levels are set using the variable gain configured under the CLI command max-conference under telephony-service configuration mode. The maximum number of conference participants that you can host in a conference is specific to the mode of conference. For more information, see Types of Conference.

Consider a scenario where the ad hoc hardware conference creator transfers the call or parks the call with another call. For Unified CME When you are configuring dial peers or ephone-dns including park slots and conferencing extensions on Cisco Integrated Services Router Voice Bundles, the following message may appear to warn you that memory is not available:. To configure more dial peers or ephone-dns, increase the DRAM in the system. Many factors contribute to memory usage, in addition to the number of dial peers and ephone-dns configured.

This is because the conference itself cannot be secure in Unified CME. Also, you can avoid wastage of the session capacity of the more expensive secure DSP farm resource. With LTI-based transcoding, conference participants line or trunk with different codecs can be added to the conference bridge without configuring extra DSP resources. The DSP inserted for conferencing takes care of both transcoding and mixing the audio stream.

Conference Blocking Conference Pattern Blocked —To prevent extensions in an ephone or a voice register pool from initiating conferences, configure the conference-pattern blocked command. Conference Max Length —When conference max-length command is configured, Unified CME allows the conferences only if the dialed digits are within the max-length limit. Octo-line Directory Numbers —With octo-line directory numbers, only one directory number is required for an eight-party Meet Me or Ad Hoc conference.

An octo-line directory number supports up to eight active calls, both incoming and outgoing, in a single phone button. It supports up to eight Select and Join instances. When a conference initiator is an octo-line directory number, Unified CME selects an idle channel from that directory number. Establish a new call to complete the conference. If an idle channel is not available in the same octo-line directory number, the conference terminates and a No Line Available message displays.

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. Also, you cannot select hold calls in the other channels of the directory number or for other directory numbers. It is supported only for single-line and dual-line directory numbers. 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. Lists all parties in a conference. For multi-party ad hoc conferences, this soft key is available for all parties in a conference. For meet-me conferences, this soft key is available for the creator only. Press Update to update the list of parties in the conference. For instance, press Update to verify that a party has been removed from the conference. Press Remove softkey to remove the appropriate parties.

The suboption Remove is available for the conference creator and phones that have conference admin configured. Join—Joins an established call to an adhoc conference. You must first press Select to choose each connected call that you want to join in a conference, then press Join to join the selected calls.

RmLstC—Remove last caller. Removes the last party added to the conference. This soft key works for the creator only. Select—Selects a call or conference to join to a conference and selects a call to remove from a conference. The creator can remove other parties by pressing the ConfList soft key, then use the Select and Remove soft keys to remove the appropriate parties.

MeetMe—Initiates a Meet Me conference. The creator presses this soft key before dialing the conference number. Other meet-me conference parties only dial the conference number to join the conference. This soft key must be configured before you can start a Meet Me conference.

The suboption Remove is available to the conference creator and phones that have conference admin configured. For more information on the configuration, see Configure Hardware Conferencing. Unified CME does not support secure conferencing.

All conference calls are nonsecure. For a phone registered to Unified CME, you can support only one conference. If an existing conference is put on hold, you cannot create another conference. For calls having different audio codecs, you cannot host a hardware conference call without transcoding DSPs. For calls having different audio codecs, you cannot host a software conference in Unified CME.

The calls do not merge into a conference. At a time, only one held call can be selected to join the Connected conference for SIP phones. Each individual Unified IP phone can host a maximum of one conference at a time. You cannot support a new conference in a phone if you have a conference on hold.

You can configure software conferencing on Unified CME as follows. To globally modify the default configuration and change any of the following parameters for three-party software conferencing, perform the following steps.

Maximum number of simultaneous three-party software conferences that are supported by a router is platform-dependent. The default value is half of the maximum number. When a three-way software conference is established, a participant cannot use call transfer to join the remaining conference participants to a different number.

Enters telephony-service configuration mode. Sets the maximum number of simultaneous three-party conferences that are supported by the router. Default is half of the maximum value. Valid values are -6 , 0 , 3 , and 6. The default is To configure optional end-of-conference options for three-party ad hoc conferencing on a Cisco Unified IP phone running Skinny Client Control Protocol SCCP , perform the following steps for each phone to be configured.

Conferencing uses call transfer to connect the two remaining parties of a conference when a conference initiator leaves the conference. To use this feature, you must configure the transfer-system command. 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.