The Implementing Cisco Collaboration Core Technologies (350-801 CLCOR) (350-801)

Cisco IP Phone 7800 Series devices support remote command-line access through SSH , but this access is disabled by default in Cisco Unified Communications Manager (CUCM). When an engineer is unable to reach the phone’s command line while the device is otherwise reachable via ICMP/HTTP and registered to CUCM, the most common cause is that SSH is not enabled or the required credentials have not been configured .

Within the Phone Configuration → Product Specific Configuration Layout , CUCM provides administrative controls for remote access capabilities. The “SSH Access” parameter in this menu explicitly controls whether the phone will accept SSH connections; this must be enabled for any CLI access to function. In addition, CUCM requires the configuration of SSH username and password under the Secure Shell Information section. These credentials are then provisioned to the phone through its configuration file. If SSH is enabled but no credentials are defined, the phone will reject the connection even if TCP/22 is open.

Other listed options (such as disabling Web Access, enabling Settings Access, or turning on FIPS mode) do not impact SSH functionality. Web Access controls HTTP/HTTPS availability, Settings Access controls local phone menu access, and FIPS mode enforces cryptographic compliance but does not enable SSH by itself.

Therefore, the two actions that resolve the issue are:

Enabling SSH Access in the Product Specific Configuration Layout

Configuring SSH username and password under Secure Shell Information

These steps ensure the phone downloads updated configuration files, activates SSH, and allows authenticated remote CLI access—fully aligned with Cisco’s standard operational behavior for 7800 Series endpoints.

The line partition must be inside the inbound CSS assigned to the CUC SIP trunk. This ensures that the SIP MWI messages are sent to the correct destination.

The other options are incorrect because:

Assigning an MWI extension on the mailbox is not required for SIP MWI to work.

The line partition does not need to be inside the rerouting CSS assigned to the Cisco Unity Connection SIP trunk.

Setting the " Enable message waiting indicator " on the port group is not required for SIP MWI to work.

Call forwarding is a feature that allows users to forward incoming calls to another number. This can be useful in a number of situations, such as when a user is not available to take a call, or when a user wants to forward calls to a different number during certain times of the day.

Call forwarding can be used to improve business continuity by ensuring that calls are always answered, even if the user is not available. For example, if a user is out of the office, they can forward their calls to their voicemail or to another employee. This ensures that customers and clients can always reach someone, even if the user is not available.

The Cisco Directory Connector is a software application that is installed on a dedicated on-premises server. It synchronizes user identities between the on-premises directory and the Cisco Webex cloud.

The engineer must set the Enable Activation Code enterprise parameter to True. This will enable the use of activation codes for onboarding on-premises devices to a Cisco UCM server. The other options are not necessary to complete the configuration.

Here are the steps to complete the configuration:

Log in to the Cisco Unified Communications Manager (CUCM) Administration interface.

Go to System > Enterprise Parameters.

Set the Enable Activation Code enterprise parameter to True.

Click Save.

The activation code onboarding feature is now enabled. You can use it to onboard new phones to the CUCM server.

To access a Cisco Unity Connection voice mailbox from a secondary device, you must configure an alternate extension for the mailbox. This is a phone number that is different from the mailbox ' s primary extension. When you call the alternate extension, you will be prompted to enter the mailbox ' s PIN. Once you have entered the PIN, you will be able to access the mailbox just as you would if you were calling from the primary device.

A Cisco Unity Connection Call Handler is a software application that answers calls, plays greetings, and routes calls to users based on caller input. Call handlers can be used to create automated attendants, voice menus, and other interactive voice response (IVR) applications.

Call handlers are created and managed using the Cisco Unity Connection Administration interface. When creating a call handler, you can specify a variety of settings, including the greeting that is played, the caller input options that are available, and the destination that calls are routed to.

Call handlers are a powerful tool that can be used to create a variety of IVR applications. By using call handlers, you can improve the efficiency of your organization ' s communications and provide a better experience for your callers.

Here are some additional tips for using call handlers:

Use call handlers to create automated attendants that can answer calls and route them to the appropriate person or department.

Use call handlers to create voice menus that can provide callers with information or options.

Use call handlers to create interactive voice response (IVR) applications that can collect information from callers and process their requests.

The SDP offer includes the following line of text:

M=audio 498181 RTP/AVP 0 8 97

This line of text indicates that the following codecs are available:

0: G.711ulaw

8: G.711alaw

97: iLBC

The Cisco IP Phone 7841 is registered to a Cisco Unified Communications Manager with default configuration. This means that the phone will negotiate the G.711ulaw codec.

The G.711ulaw codec is a standard codec that is used for voice communication. It is a low-bandwidth codec that provides good quality.

The iLBC codec is a newer codec that is designed for use in low-bandwidth environments. It provides good quality, but it is not as widely supported as the G.711ulaw codec.

The G.722 codec is a high-quality codec that is used for voice communication. It provides excellent quality, but it requires more bandwidth than the G.711ulaw codec.

The following commands are used to configure the codec on a Cisco Unified Border Element to prefer the G.711 ulaw and use G.711 alaw as the next codec:

Code snippet

voice class codec 100

codec preference 1 g711ulaw

codec preference 2 g711alaw

The voice class codec 100 command creates a new voice class with the ID of 100. The codec preference 1 g711ulaw command sets the preference for the G.711 ulaw codec to 1. The codec preference 2 g711alaw command sets the preference for the G.711 alaw codec to 2.