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CONTENTS
Pre-Staging BE4000 26
Supported Codecs in BE4000 27
Subnets Not Supported by BE4000 27
Add a BE4000 Customer Site in Cisco Business Edition 4000 Management Portal 43
Replace a Faulty BE4000 Appliance 44
Flag the Faulty BE4000 Appliance for RMA 45
Deploy the RMA Replacement BE4000 Appliance 46
Local Administration 47
Console Based Local Administration 47
Ethernet based Connectivity for BE4000 Local Administration 48
Run Port Check Tool 50
Deploy a Site 50
Deploy a Site—Software Updates Through Internet 50
Deploy a Site—Software Updates Through USB 51
Add Cisco Wireless IP Phone 8821 to the Wireless LAN 52
Provision the Phone 53
March 21, 2018 • Added Modify the Digit to Send Calls to Voicemail Automatically,
on page 114
• Phones - Field Descriptions, on page 84—Added a note for "Type"
filed description under Buttons
• Call Routing Field Descriptions, on page 149—Added a note for
"Pilot Number" field description under Hunt Groups
• Troubleshoot the Status of Line Cards, on page 64—Updated the
images
March 12, 2018 • Added Add Partner Users for Accessing the Cisco Business Edition
4000 Management Portal, on page 31
• Added Wireless Local Area Network Requirements , on page 25
• Reorganized the Deploy a Site, on page 50 section to include
procedure for deploying the BE4000 using a USB stick
March 2, 2018
February 5, 2018 Updated the BE4000 Deployment Overview, on page 21 with the
following information:
• Requirement to enable port forwarding when using Port Address
Translation (PAT)
• Requirement for IP Phones and BE4000 to be on the same network
for features such as Phone Paging, Intercom, and Music On Hold
to be supported.
• Updated the BE4000 Deployment Overview, on page 21 to indicate that the BE4000
must be deployed behind NAT.
• The "Primary DID" fieldname is renamed as "Default Outbound DID" in the Call
Routing Field Descriptions, on page 149 section.
• The Music On Hold field description in the Features Field Descriptions, on page
156 section is updated to include support for .au file format and filename conventions.
• The "Provider Digits" field is moved from the Line Cards (ISDN trunk) page in
the Setup Assistant to Manage Site > Direct Inward Dial Numbers page.
• BE4000 SIP Trunk Connectivity, on page 23—Added the information that a partner
should provide to expedite the new SIP service provider on boarding process
• BE4000 Management Tunnel, on page 19—Updated to include ICMP access to
SEA (208.115.101.160/27) and DFW (4.16.236.34/27) servers as a requirement
July 20, 2017 • Updated Provision the Phone , on page 53 with the following information:
• Password for accessing the administrator settings on the phone
• Support for DHCP Option 66
• Updated Setup Assistant, on page 125 with the following information to align with
the changes in Cisco Business Edition 4000 Partner Portal:
• "Outbound Proxy" field is now optional
• "Provider Template" field is added
July 12, 2017 • Changed the book title from "Cisco Business Edition 4000 Partner Portal Guide" to
"Cisco Business Edition 4000 Partner Guide"
• Reorganized the topics in the document based on end to end tasks performed by the
partner in deploying a customer site
• Added the following sections:
• BE4000 Customer Network Requirements, on page 19
• Reset OTP, on page 122
• BE4000 Customer Network Requirements, on page 19
• Local Administration, on page 47
• Setup Assistant, on page 125- contains field descriptions for Setup Assistant
wizard
The Cisco Planning, Design & Implementation (PDI) Technical Advisor (TA) team is available to help you
with the BE4000. For more details, or to open a case with PDI TA, go to http://www.cisco.com/go/pdi.
For the latest features and enhancements available in the BE4000, see Release Notes.
Step 5 Collect Customer and You must collect the — Read Customer
Site Details customer contact Contact and
details and network Network Details
specifications before section of the Cisco
adding a customer Business Edition
site in the portal. 4000 Partner Guide
for detailed
procedure.
Step 6 Add customer site in Create configuration Have customer contact See Add Customer
the BE4000 Portal for the customer site and network details Site section of the
in the BE4000 portal ready Cisco Business
and bring it to Edition 4000
“Ready to Deploy” Partner Guide for
state. detailed procedure.
Step 9 (Task performed on You must confirm • BE4000 appliance See Access Local
the customer site) that you have all the is connected to the Administration
required connections network devices Screens section of
Access Local
before deploying the and powered on. the Cisco Business
Administration
customer site. You Edition 4000
screens to verify if • RJ45 or USB
can access local Partner Guide for
BE4000 is connected console cable for
administration detailed procedure.
to the network Console based
screens through
Ethernet or Console. connection
• MGMT port (use a
normal Cat5e or
Cat6 cable) cable
for Ethernet based
connection
Step 10 Run Port Checker You must confirm if • Port Check tool Port Checker
the UDP ports 500 must be run on —htps:/portcheck.be4000.cisco.com.
and 4500 are Chrome, Firefox, or
accessible to the Opera browsers
BE4000 portal. only
• Your computer
must be on the
same network as
your BE4000
appliance
Step 11 Deploy the site You must bring the • You have verified • BE4000
site configurations the network deploy
that you created in connections by site—htp:/be4000.cisco.com/
the BE4000 portal on accessing the local deploy.
to the BE4000 administration
appliance. screens • See Deploy a
Customer Site
You can deploy the • You must have run section of the
site in the following the Port Check Tool Cisco Business
ways: and verified UDP Edition 4000
• Scan the QR Ports are available Partner Guide
code on the for detailed
• Provisioned procedure.
underside of the telephony services
BE4000 • See Cisco
appliance and • A minimum Business
follow the internet download Edition 4000
prompts speed of 2 Mbps Quick Start
• Browse to • Customer Name, Guide.
BE4000 deploy location, and serial
site—htp:/be4000.cisco.com/ number of the
deploy BE4000 appliance
is ready
Step 12 Verify that the site is Ensure that the site is Deployment is BE4000
in the online on the up and running. You successfully completed. Portal—https:/be4000.cisco.com
BE4000 Portal cannot provision the
phones if the site is
not in the Online
state.
Step 13 (Task performed on Connect phones to • Have the list of See Provision the
the customer site) the network and extensions Phone section of the
assign extensions that configured in the Cisco Business
Connect the phones
you added in the BE4000 portal and Edition 4000
to the network and
BE4000 Portal while its associated phone Partner Guide for
provision using
adding a customer model detailed procedure.
Extension Assigner
site.
• Have the required
phone devices
ready on the
customer site
• DHCP option 150
or DHCP Option 66
is enabled
• HTTP (port 80) and HTTPS (port 443) access from the BE4000 to the internet
• Outbound access to Domain Naming System (DNS): DNS port 53 must be open
• ICMP (Internet Control Message Protocol) access to SEA (208.115.101.160/27) and DFW (4.16.236.34/27)
servers
Note Each BE4000 needs it's own public IP address for management.
DHCP Server
• By default the BE4000 uses a DHCP address for initial deployment. A static address may be configured
locally if a DHCP server is not available. Connection can be via console, or Ethernet to the MGMT port.
For information on console and Ethernet based connection, see Local Administration, on page 47.
• When deploying Cisco Unified IP Phones, it is necessary for the phones to automatically discover the
BE4000 to download phone configuration files. DHCP Option 66 and/or DHCP Option 150 is required
to provide the TFTP address of the BE4000 for connecting the IP phones. If neither DHCP option 66
nor DHCP Option 150 is available, you must manually configure each IP phone's TFTP configuration
with the IP address of the BE4000.
The BE4000 can be deployed on the voice VLAN or on a different IP subnet. In both the ways, the BE4000
needs to have directed routed access to all phones, and be able to access the internet.
Note The BE4000 proxies the media. Phones only need signaling and media connectivity to the BE4000. They do
not need signaling nor media connectivity to other phones.
Note Port Address Translation (PAT) without port forwarding is not supported in BE4000.
When deploying the BE4000 solution, ensure that IP phones are deployed on the same network as the BE4000.
This ensures certain features such as IP phone Paging, Intercom, and Music On Hold (MOH) work without
the need for additional changes to the network infrastructure. If IP phones are deployed on a different network,
ensure that any Layer 3 devices in between the IP phones and the BE4000 support multicast and are configured
to forward the traffic. See the documentation provided by the network equipment vendor to determine how
to enable multicast routing.
The BE4000 also allows calling via IP trunks to service providers using SIP.
A variety of options are provided that allow services to be delivered via dedicated service connections, or
over the top of a customer's internet access. To simplify configuration, a number of preconfigured templates
for certain service providers are also offered through the management portal.
SIP Service Provider—Due to diverse SIP implementations by service providers, new service providers
must be brought on board and validated before attempted use. Contact the service provider on boarding team
by sending an email to be4000-siptrunk@external.cisco.com to begin the on boarding process. Depending on
the complexity of the service provider's SIP service specification, we recommend beginning the service
provider on boarding process as early as possible.
To expedite the on boarding process, provide the following information about your deployment:
• Partner name
• Deployment address
• Serial number of the BE4000
• SIP service provider name
• Technical information provided by the service provider
The BE4000 offers two primary options for connecting to a SIP VoIP trunking service:
1. Using the main interface (GE 0/0/0) of the BE4000 for the SIP trunk, media and cloud management
connection
2. Using the BE4000 secondary interface (GE 0/0/1) connected to a dedicated internet connection for your
VoIP service
In this scenario, the BE4000 is connected as a privately addressed host in your local area network, so it is
assumed that all traffic sent to the internet is subject to Network Address Translation (NAT). As such, the
following is required to reliably exchange VoIP traffic:
1. Secure a static public internet address from your ISP. The BE4000 uses this address to ensure that signaling
is properly formatted as it is sent.
2. It may also be necessary to forward inbound traffic to the BE4000, especially if your SIP provider does
not require registration. Depending on the capabilities of your WAN router, this may be accomplished in
a number of ways:
1. Add port forwarding rules for SIP signaling and media traffic.
2. Use static NAT rules.
3. Configure the BE4000 IP address as a DMZ host.
Using the BE4000 Secondary Interface for SIP Trunk and External Media
Figure 5: SIP Trunk Connectivity Using Secondary Interface
The secondary interface may be configured with an IP address either manually or dynamically through DHCP.
If a manual address is configured and traffic is passed through NAT, then an external IP address may be
configured in the portal to fix up signaling traffic. This option is not currently offered where an address is
assigned dynamically. In this case, it is assumed that the assigned address is either publically routable, or if
NAT is used, then the VoIP provider offers transparent hosted NAT traversal.
When using the secondary interface, the BE4000 must be assigned an address in a different subnet to that
used for the primary interface.
To ensure that traffic is correctly routed and accepted, all IP addresses used by your provider must be included
in the portal trusted address list for the trunk.
• The access point platform type, antenna type, access point configuration (channel and transmit power).
We recommend you to select an access point with integrated antennas for non-rugged environments (for
example, office, healthcare, education, hospitality) and an access point platform requiring external
antennas for rugged environments (for example, manufacturing, warehouse, retail)
Signal
The cell edge should be designed to -67 dBm where there is a 20-30% overlap of adjacent access points at
that signal level. This ensures that the wireless IP phones always have adequate signal and can hold a signal
long enough in order to roam seamlessly where signal based triggers are utilized verses packet loss triggers.
Also, ensure that the upstream signal from the wireless IP phones meet the access point’s receiver sensitivity
for the transmitted data rate. Ensure that the received signal at the access point is -67 dBm or higher.
We recommend you to design the cell size to ensure that the wireless IP phones can hold a signal for at least
5 seconds.
Chanel Utilization
Channel Utilization levels should be kept under 40%.
Noise
Noise levels should not exceed -92 dBm, which allows for a Signal to Noise Ratio (SNR) of 25 dB where a
-67 dBm signal should be maintained.
Also ensure that the upstream signal from the wireless phone meets the access point’s signal to noise ratio
for the transmitted data rate.
Packet Loss or Delay
As per voice guidelines, packet loss should not exceed 1% packet loss; otherwise voice quality can be degraded
significantly. Jitter should be kept minimal (< 100 ms).
Retries
802.11 retransmissions should be less than 20%.
Multipath
Multipath should be kept minimal as it can create nulls and reduce signal levels.
Pre-Staging BE4000
It is not normally necessary and we do not recommend that you pre-stage the BE4000. If you do want to
pre-stage the BE4000, you must fully simulate the customer’s network environment including internal and
external IP addresses because certain system parameters cannot be modified after they have been initially
configured.
Note The IP address of the BE4000 and certain other parameters cannot be changed after the BE4000 has been
initially configured. If you need to change any of these values, you must follow the directions to factory reset
your appliance and start a new deployment. If you need to do this, remember that you can save template
configurations to avoid typing everything in a second time.
RequestAccesstotheCiscoBusinessEdition4000Management
Portal
Before you begin
• You must have Cisco ID. If you do not have a Cisco ID, Register Now.
• You must be associated with a Cisco Partner. If your company is new to Cisco, complete Partner
Registration.
Procedure
Step 1 Log in to the Partner Self Service (PSS) portal—http://www.cisco.com/go/pss with your Cisco ID.
Step 2 Select Manage My Access.
Step 3 Click Request Additional Access.
What to do next
A Partner Administrator adds you as a user and grants access to the Cisco Business Edition 4000 Management
Portal. For details on how a Partner Administrator can grant access, see Add Partner Users for Accessing the
Cisco Business Edition 4000 Management Portal, on page 31.
Note Only a Partner Administrator can perform the following steps and grant access to the Cisco Business Edition
4000 Management Portal.
Procedure
Step 1 Log in to Partner Self Service (PSS) portal—www.cisco.com go pss with your Cisco ID.
Step 2 Select Manage My Access from the drop-down list.
Step 3 Select Company Access tab.
Step 4 Search and add user.
Step 5 Click Edit, select BE4000 "User" from the list of services, and click Next.
Step 6 Enter description in the comments field, and click Submit.
Procedure
When you log in to the Cisco Business Edition 4000 Management Portal for the first time, a video containing
the information about how to configure, deploy, and manage a BE4000 site is displayed. You must watch this
video completely to proceed further and view the Cisco Business Edition 4000 Management Portal dashboard.
Once the customer is added on the page, you can view customers and add, update, or delete their configuration
details.
The features of the dashboard are explained as follows:
Feature Description
Customer Search You can search for your customers and find key information about each site,
users, or groups anytime.
Feature Description
View more details of the site and appliance. For example, IP address, last call,
version, the last change made, created by, and so on.
By default, you see details of the following columns:
• Customer Name
• Location
• Status
• Serial Number
• Phones
• Phones
• Last Change
• Actions
Add Customer Add a new customer. See Add a BE4000 Customer Site in Cisco Business
Edition 4000 Management Portal, on page 43.
SIP Trunks
Customer preference for SIP trunk connectivity:
1. Use the main interface (GE 0/0/0) of the BE4000 for the SIP trunk and internet connectivity
2. Use the BE4000 secondary interface (GE 0/0/1) connected to a dedicated internet connection for SIP trunk
and use the main interface (GE 0/0/0) of the BE4000 only for internet connectivity
If customer chose option 2, check which one of the following they are using for the SIP trunk:
• Static address
• Dynamic address
If they are using static address, collect IP address, subnet mask, default gateway, service name, and proxy
server and port details, and trusted IP addresses.
If they are using dynamic address, collect the interface type they are using, service name, and proxy server
and port details, and trusted IP addresses.
Line Cards
Check if your customer wants to connect to traditional telephone services and devices. If yes, ensure that the
customer orders the NIM card based on the required services. Refer to Cisco Business Edition 4000 Release
Notes for the list of supported NIM cards.
Collect the information on the type of NIM card its associated details from your customer.
• For FXS Cards, collect Extension name, extension number, class of restriction (COR), u-law/a-law
• For FXO Cards, collect Line name, line number, Inbound only/In and Out, u-law/a-law
• For BRI Cards, collect Service name and Overlap receiving (enabled or disable)
• For PRI Cards, collect Service name, T1/E1, ISDN Switch type
Dial Plan
• Country
• Local Dialing Options
• Telephony port tone
• Time zone
• Language preference
• Number of digits in an extension
• Digits to dial
• An outside line
• An intercom extension
• To send voicemail automatically
• For user-specific phones—First name, last name, display name, extension number, phone mode, COR,
email address if user requires voicemail functionality
• For public phones (used for common services such as a conference room)—Display name, extension,
and phone type
Business Hours
Customer’s working days and hours.
Maintenance Schedule
A 2-hour block of time each day when it is safe for the system to install the software updates. The system
may be offline and unable to make or receive phone calls during the maintenance schedule.
2 Add a domain to Office 365 Add a domain and users to Office 365, on page
41
4 Add SMTP Server details in the BE4000 Portal Add SMTP Server Details , on page 42
Configure a Connector
Procedure
Step 3 Click Admin Centers > Exchange. The Exchange admin center page is displayed.
Step 4 Click mail flow > Connectors on the Office 365 portal. Click + to add a new connector. Choose the following
for From and To fields and click Next.
• From: Your organization’s email server
• To: Office 365
Step 6 Click By verifying that the IP address of the sending server matches one of these IP addresses that
belong to your organization and provide the public IP used for SMTP. Public IP used for SMTP can be got
by checking with http://whatsmyip.com.
Procedure
Log in to Office 365 portal and add a domain and users. For more information, refer Add a domain and users.
Procedure
Procedure
• In the “IP Address or Domain Name” field, enter one of the following:
• If the customer has Office 365, enter the MX FQDN record copied from Office 365 portal
• If the customer has SMTP relay other than Office 365, enter the SMTP server IP Address or Domain
Name
• In the "Sender's Email Address", enter the email address that is used as "From" address while sending
the emails containing the voicemail attachments.
• (Optional) Check the "Authenticate" check box and enter the username and password. Use the credentials
to authenticate to the SMTP server.
Step 5 Enter the details in rest of the pages and click Yes to apply the configuration.
What to do next
Deploy the BE4000 appliance. Refer Deploy a Customer Site for detailed steps.
Procedure
Step 1 Click Add Customer on the upper right corner of the dashboard.
Step 2 Click Get Started to start creating a customer site.
Step 3 Enter the customer and location details on the Add Customer page. See Add Customer Field Descriptions,
on page 125.
You can manually enter the customer details to create a site or select a template. Selecting the template auto
populates the network information and other site details based on the template.
Step 4 Enter the LAN Network Connection, Direct Dial Numbers, SIP Trunks, and Line Card information
collected from the customer on the Connectivity page. See Connectivity Field Descriptions, on page 126.
Step 5 Enter the Region Settings, System Settings, and Dial Plans information collected from the customer on the
Dial Plan page. See Dial Plan Field Descriptions, on page 145.
Step 6 Enter the details of the users, extensions, and other calling features on the Stations page. See Stations Field
Descriptions, on page 147.
You can manually enter the details by adding rows or upload your spreadsheet (based on the template)
containing the user details.
Step 7 Create Hunt Groups and Auto Attendant based on customer requirements on the Call Routing page. See Call
Routing Field Descriptions, on page 149.
Step 8 Upload an audio file for Music on Hold based on the customer requirements on the Features page. See Features
Field Descriptions, on page 156.
Step 9 Click Yes if you want to apply the entered BE4000 configuration changes for your site. Else, click No to
continue with editing the setup assistant configurations.
Step 10 Click Save as template if you want to load your saved configurations at a later stage.
Once you are done with the setup assistant, you are ready to deploy the site you created.
Note If you terminate the Setup Assistant wizard after adding customer details before completing the
configurations, the customer site is created and listed in the dashboard. You can edit the configurations
later.
Step 1 Flag the faulty Flag the customer — Flag the Faulty
appliance for RMA site for RMA so that BE4000 Appliance
the faulty appliance for RMA, on page
can be replaced with 45
the new appliance.
Step 2 Deploy the RMA Deploy the new • Have the serial Deploy the RMA
Replacement appliance that is number of the Replacement
Appliance received for RMA new appliance BE4000 Appliance,
with the same site on page 46
configurations as it • If you had line
was on the faulty cards on faulty
appliance. appliance,
ensure that the
same (or same
type) line cards
are inserted.
Procedure
Procedure
Step 1 Log in to be4000.cisco.com/deploy or use a QR scanner application on your smart phone to scan the QR code
on the underside of the product.
Step 2 Enter the serial number of the new appliance, if prompted.
Step 3 Select the customer and site from the drop-down lists.
Step 4 Select the backup from the Restore from backup drop-down list.
Step 5 Verify Network Interface Module (NIM) cards, if necessary, are inserted in appropriate slots, cable are
connections and BE4000 is powered ON.
Step 6 Click Deploy Configuration.
Local Administration
You can connect to the BE4000 appliance using a Console or Ethernet connection for monitoring the DMVPN
tunnel and troubleshooting the connectivity issues. After you connect the BE4000 appliance through Console
or Ethernet connection, the local administration screen is displayed with a list of menu options. You can
choose an option based on your requirement and get the related information.
Local administration screen can be used only for monitoring and troubleshooting. You cannot configure a
BE4000 site using the command line interface (CLI) unlike other Cisco IOS-based routers.
Note You must be present physically at the BE4000 site to access local administration screen.
Step 1 Connect your computer back to back to MGMT port (use a normal Cat5e or Cat6 cable).
Step 2 Use SSH client to connect to the BE4000 using either the MGMT IP Address (169.254.100.1) or Host Name
(status@be4000).
Step 3 Log in with username status. No password is required.
Step 4 Type h to see the available options.
Procedure
Deploy a Site
Deploying a site involves downloading the software files from the Cisco Business Edition 4000 Management
Portal through the internet. So, we recommend you to have a minimum of 2 Mbps internet download speed.
If you do not have the minimum required internet download speed, you can download the latest software files
from BE4000 Software Download page onto a USB before deployment. Having the software files on the USB
expedites the deployment process.
While deploying the BE4000 site, the following checks are made in sequential order:
1. Availability of the USB. If USB is not detected, the software files are downloaded from the Cisco Business
Edition 4000 Management Portal.
2. If USB is detected, check for the latest software files. If one or more software files on the USB are out of
date, the software files are downloaded from the Cisco Business Edition 4000 Management Portal.
3. If USB is detected and all the files are the latest, the software files are copied locally from USB.
Procedure
Procedure
Step 8 Click Deploy Configuration. After a successful deployment, the site shows “Online” in the Cisco Business
Edition 4000 Management Portal.
Step 9 (Optional) We recommend that you install the BE4000 appliance in a 19-inch rack. You can also wall mount
or place the appliance on any secure flat surface, if preferred. If you are installing in a rack, use the bracket
mounting point (both sides) to attach the mounting brackets with the screws provided. Use suitable fastenings
to secure the product in place.
• If you are not using DHCP, ensure that you have the following details ready:
• IP address
• Subnet mask
• Default router
• DNS server 1
• TFTP server 1
Procedure
Step 6 (Optional) If you are not using DHCP, select DHCP and press Off. Enter the IP address and subnet mask of
the phone, default router, DNS Server 1, and TFTP server 1 address in the respective fields.
Step 7 Select WLAN configuration.
Step 8 Select SSID. Use the keypad to enter the SSID of the wireless access point. Press More and select Save.
Note Ensure that the SSID matches the name of the wireless LAN.
Step 9 Select Security mode based on the security type configured for your wireless LAN.
Step 10 Select 802.11 mode and select the required mode. The mode determines the frequency. If you set the mode
to Auto, the phone can use either the 5 or 2.4-GHz frequency, with 5 GHz as the preferred frequency.
Step 11 Select On call power save and press Select to change the setting. Only for troubleshooting purposes, set this
field to disabled.
Step 12 Press More and select Save.
Note We recommend you to configure DHCP Option 150 to simplify the provisioning
of phones. Configuring DHCP Option 66 is also supported. You can manually
configure the TFTP address but it may take more time.
• If you are provisioning Cisco Wireless IP Phone 8821, ensure that you have configured the phone with
the wireless LAN. For more information, see Add Cisco Wireless IP Phone 8821 to the Wireless LAN,
on page 52.
Procedure
Step 2 Dial the Extension Assigner directory number, 70000. When prompted for password, enter 1234.
Step 3 Enter the pound (#) key.
Step 4 Enter the permanent extension, followed by the pound (#) key. Enter the extension configured on the portal
for this user or phone.
Step 5 Enter 1, followed by the pound (#) key to confirm the extension.
Step 6 End the call.
The phone reboots and the assigned extension is shown on the phone display.
Note If you are prompted for a password while accessing the administrator settings on the phone after
configuring the permanent extension, enter ptwmjg.
Needs Attention There is an error related Errors can be for the following reasons:
to the service.
• More than 15% of the phones are unregistered.
• Any issue with the traditional or VoIP trunks.
• There is impact to the phone services.
Configuration in Site is being set up by the Ensure that you enter all details in the BE4000 navigation
Process user. wizard to make the appliance ready to be deployed.
Deployment Appliance is not You must view the navigation on the appliance for more
Failed deployed. information. Deployment can fail for the following reasons:
• Most likely the ports are blocked.
• Wrong LAN settings.
• Connectivity issues.
Voicemail Voicemail functionality Check if the voicemail appliance is faulty. Contact Cisco TAC
Configuration does not work. for assistance.
Failed
Inventory Failed Your appliance inventory Go to the Deployment status page and click the trace icon to
failed. get the deployment failure log details.
Redeploy the site again.
Contact Cisco TAC for assistance.
Procedure
Procedure
Step 1 Select a deployed customer site and click the dotted lines next in the Actions column.
Step 2 Click Manage Sites to display the configuration summary and editing options.
You can edit the following and then save the changes:
• Auto Attendant—Modify auto attendant settings.
• Business Hours—Modify open and closed hours.
• Extensions—Modify or delete extension numbers.
• Direct Inward Dial Numbers—Add, edit, or delete the Direct Inward Dial (DID) numbers.
• Group Mailbox—Create group mailbox for various departments.
• Hunt Groups—Modify hunt groups.
• Line Cards—Change line card settings.
• Night Service—Enable or disable night service.
• Personal Mailbox—Create personal mailbox for users.
• Phones—Add or delete phones.
Procedure
Step 1 Click Create Template from the Actions menu selecting the desired site.
You are directed to the Connectivity Settings page.
Step 2 Edit relevant pages of the wizard to create a new site using some of the existing configurations. For detailed
descriptions about each page in the setup assistant page, see Add a BE4000 Customer Site in Cisco Business
Edition 4000 Management Portal, on page 43.
Line Cards
BE4000 allows you to connect to traditional telephony services and devices by adding Network Interface
Modules (NIM). You can add a maximum of two NIM cards. For the list of supported NIM cards, refer
Supported Line Cards.
On the portal, you can add a new NIM card or edit an existing NIM card. Currently, removal and replacement
of NIM cards are not supported on the BE4000 portal.
Procedure
Step 6 Enter the configuration details based on the type of card inserted. For field descriptions, refer Line Card -
Field Descriptions, on page 60.
Note If you already have a line card installed into your BE4000 appliance, then it automatically gets
listed. However, you cannot edit the configurations of an installed line card while adding a new line
card. Refer to Edit a Line Card, on page 64 for details on editing a line card configuration.
Step 7 Enter the details in the Inbound Call Mapping and Outbound Caller ID pages based on your requirements.
NIM-2FXS or NIM-4FXS
Field Description
Class of Restriction decides the type of calls that can be placed from the FXS
phone line.
Law Choose the type of algorithm used for modifying an input signal for digitization:
• u-law
• a-law
NIM-2FXO or NIM-4FXO
Field Description
Field Description
Direction Mark the line as incoming only or bidirectional. The system builds the trunk
groups based on what you select.
• In + Out—Allows the phone line to receive and make calls.
• Inbound Only—Allows the phone line to receive the calls.
Law Choose the type of algorithm used for modifying an input signal for digitization:
• u-law
• a-law
NIM-2FXS/4FXO
Field Description
FXS
Class of Restriction decides the type of calls that can be placed from the FXS
phone line.
Law Choose the type of algorithm used for modifying an input signal for digitization:
• u-law
• a-law
FXO
Field Description
Direction Mark the line as incoming only or bidirectional. The system builds the trunk
groups based on what you select.
• In + Out—Allows the phone line to receive and make calls.
• Inbound Only—Allows the phone line to receive the calls.
Law Choose the type of algorithm used for modifying an input signal for digitization:
• u-law
• a-law
NIM-2BRI-NT/TE or NIM-4BRI-NT/TE
Field Description
Service Name Choose a service name from the drop-down list. The drop-down list contains the
list of service providers that you added in the DID page.
Note You cannot choose the same service provider for SIP and Line Cards.
Static TEI If your service provider requires that your line use a static Terminal Endpoint
Identifier, enter the value between 0 and 63. If the field is left blank, the line
attempt to negotiate a TEI.
Overlap Receiving Choose whether you want your call setup to work based on overlap receiving.
You can enable or disable this option. If your service provider does not use
“enbloc” signaling, this option allows BE4000 to wait for additional digits to be
received before the call is routed.
Send Redirecting IE Check the "Send Redirecting IE Number" check box to include the Redirecting
Number Number Information Element in the outbound Setup messages. Leave unchecked
if you are not sure about your service provider supporting this feature.
ISDN SPID Enter the ISDN SPID. Some service providers use service profile identifiers
(SPIDs) to define the services subscribed to by the ISDN device that is accessing
the ISDN service provider. A SPID is usually a seven-digit phone number with
some optional numbers.
Field Description
TEI Negotiation Method Choose a method for TEI negotiation based on your service provider requirements.
Setting a static TEI overrides TEI negotiation.
The default behavior is TEI to be negotiated on power-up. The following options
are provided to preserve or remove a negotiated TEI when the interface is reset:
• Power Up and Remove
• Power Up and Preserve
• First Call and Remove
• First Call and Preserve
Field Description
Service Name Choose a service name from the drop-down list. The drop-down list contains the
list of service providers that you added in the DID page.
Note You cannot choose the same service provider for SIP and Line Cards.
Card Type Choose the card type based on your customer network requirement. E1 PRI is
chosen by default. The available options are:
• T1 PR1
• E1 PRI
ISDN Switch Type Choose one of the following ISDN Service Provider PRI Switch Types:
• primary-4ess
• primary-5ess
• dms100
• primary-net5
• primary-ni
Controller Setup Defines the controller setup for configuring channelized T1 or E1 controllers.
Choose either Full PRI or partial PRI.
Line Code Choose a line code. By default, the line code for E1 PRI is high-density bipolar
3 (hdb3).
Framing Choose the framing from the drop-down list. This option defines the framing
characteristics.
Field Description
Send Redirecting IE Check the "Send Redirecting IE Number" check box to include the Redirecting
Number Number Information Element in the outbound Setup messages. Leave unchecked
if you are not sure about your service provider supporting this feature.
Procedure
Note It takes around 30 seconds to fetch the status of the line cards.
Column Description
Slot Two line cards slots available in the BE4000 appliance. NIM 1 and NIM 2 are
always displayed irrespective of the card inserted on to the BE4000 appliance.
Card Type Type of card inserted on to the BE4000 appliance. If you have not inserted a line
card, “No card found” is displayed.
Status Based on your line card configuration, one of the following is displayed:
Good Line card is in service. DSP resources and all the ports
are in service.
Needs Attention There is an error related to the line card service. Either
DSP resources or one of the ports is not in service.
Click Needs Attention to see the detailed status.
Column Description
DSP DSP (PVDM4) resources available for the line card. DSP resources are required
for configuring the ports.
Port The number of ports available for the line card type is listed with its status. The
status of the ports can be one of the following:
(Needs Attention)
Port is in service.
Click Cycle Port to disconnect and reconnect the port. When you recycle the port, all active calls on the port
are disconnected.
Figure 16: Cycle Port
SIP Trunk
SIP Trunk can be configured on the BE4000 during the initial deployment. However, in case, if the SIP Trunk
is not added during the initial deployment, it can be added post site deployment.
If SIP trunk is added during initial deployment, you can remove it from BE4000 post site deployment.
Note • You cannot modify the SIP Trunk configurations. You can only add or remove the SIP Trunks.
• You can add only one SIP Trunk per site.
Procedure
Step 1 Log in to the Cisco Business Edition 4000 Management Portal (https://be4000.cisco.com/) .
Step 2 Click Manage Site > SIP Trunk for the desired site from the Actions menu.
Step 3 Click Add SIP Trunk.
Step 4 Add DID Numbers on the Direct Dial Numbers page.
Step 5 Add SIP Trunk details on the SIP Trunk page.
Step 6 Choose a default target for each service provider from the Default Target drop-down list on the Inbound
Call Mapping page. Based on your preference, choose a target type (auto attendant, extension, or hunt group)
for each extension from Target Type drop-down and choose a corresponding target number from Target
Number drop-down list
Step 7 Choose a default outbound DID for each service provider from the Default Outbound DID drop-down list
on the Outbound Caller ID page. Based on your preference, choose a DID number as Caller ID for each
extension from the Caller ID drop-down list.
Step 8 Click Yes to save the changes.
Procedure
Step 1 Log in to the Cisco Business Edition 4000 Management Portal (https://be4000.cisco.com/).
Step 2 Click Manage Site > SIP Trunk for the desired site from the Actions menu.
Step 3 Click Delete SIP from the Actions menu.
You get an option to save or purge the DID numbers that are associated with the SIP Trunk. If you choose to
save, the DID numbers are stored in the system and provides an option to load the same while adding a new
SIP Trunk in future. If you choose to purge, all the DID numbers get deleted from the system.
Step 4 Click Save or Purge, based on your preference for saving DID numbers.
Auto Attendant
Auto Attendant service (also referred to as a virtual receptionist), is a phone system that enables your callers
to be automatically transferred to an extension, eliminating the need for a receptionist and avoiding extended
waiting period. BE4000 provides you an automated phone answering facility to communicate effectively with
customers and improve your business operations. An auto attendant answers all incoming calls with an audio
greeting and options menu (different for open and closed hours). A maximum of five submenus with a
maximum depth of 3 levels can be configured. The caller can select a menu option to reach to the desired
extension.
You can define the number of times the menu options is played to the caller before the call reaches the drop
through destination. You can also define where the call lands if no action is performed by the caller even after
the defined number menu repetitions.
Cisco partner, customer administrator, and any end user with PromptAdministrators privileges on their Personal
Mailbox can update auto attendant greetings and prompts.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Auto Attendant.
Step 3 Enter the information in the fields. For field descriptions, refer to Auto Attendant - Field Descriptions, on
page 69.
Step 4 Click the arrows at the bottom of the screen to move to next screen or to go back to the previous screen.
Step 5 Click Yes.
Pilot Number Enter the number that callers dial to reach auto attendant. A minimum
of four digits is required. Range is from 1000 to 9999.
Field Description
Number of repeats through menu Number of times the audio file is played to the caller before the call
reaches the drop through destination Value range is from 0 to 9. Default
value is 4.
Drop Through Destination Defines where the call lands if no action is performed by the caller even
after playing the menu for the defined number of repeats. You can
configure one of the following as the drop through destination:
• Extension—All extensions are listed in the drop-down.
• Direct to Voicemail—All extensions that have "Voicemail" enabled
are listed in the drop-down.
• Hunt Group—All Hunt Groups are listed in the drop-down.
• Group Mailbox—All Group Mailboxes are listed in the drop-down.
Note During the initial site deployment (in the Setup
Assistant), the drop-down shows an option, only if you
create a Group Mailbox on the Hunt Groups page, by
choosing "Route to Group Mailbox" from the When No
Member is Available drop-down list.
Audio Prompt (Welcome Message) Add an audio prompt for welcome Message. The BE4000 provides a
default audio file. This audio message is played first when a call is
answered by the auto attendant. You can also upload a new .wav file.
To select a new file, click Upload.
Note BE4000 supports only .wav audio file with G.711 u-law,
8kHz, 8 bit, Mono format. The file cannot be larger than 1
MB (about 2 minutes). The filename cannot have space and
special characters.
Audio Prompt (Open Message) Add an audio prompt for open message. The BE4000 provides a default
audio file. This audio message is played when a call is answered during
the open business hours. You can also upload a new .wav file. To select
a new file, click Upload.
Note BE4000 supports only .wav audio file with G.711 u-law,
8kHz, 8 bit, Mono format. The file cannot be larger than 1
MB (about 2 minutes). The filename cannot have space and
special characters.
Field Description
Add Menu Option Add customized menu options. You can add 0-9 menu options in addition
to a * menu. Each menu option can be labeled in a meaningful way to
help identify locations or users in your system using any one of the
following: Dial by Name, Pilot Number, Dial by Number, Call Hunt
Group, Repeat this Menu, Return to Main Menu, or Submenu.
Audio Prompt (Closed Message) Displays the default audio file that is played for all calls received during
closed hours. You can play the existing file or upload a new .wav file.
To select a new audio file, click Upload.
Note BE4000 supports only .wav audio file with G.711 u-law,
8kHz, 8 bit, Mono format. The file cannot be larger than 1
MB (about 2 minutes). The filename cannot have space and
special characters.
Add Menu Option Add customized menu options. You can add 0-9 menu options in addition
to a * menu. Each menu option can be labeled in a meaningful way to
help identify locations or users in your system using any one of the
following: Dial by Name, Pilot Number, Dial by Number, Call Hunt
Group, Repeat this Menu, Return to Main Menu, or Submenu.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Personal Mailbox.
Step 3 Click Modify from the Actions menu for the desired user.
Step 4 Navigate to Groups page and click Groups.
Step 5 Check PrompAdministrators. (Hover the mouse on PromptAdm to view the complete text)
Step 6 Click OK.
• The BE4000 portal displays AA_cPrompt3.wav as the default prompt for closed prompt and contains the
message "Prompt number 3, Auto Attendant, Prompt Number 3"
Procedure
Step 4 Press 2.
You hear the following:
• Press 1 record a new prompt
• Press 2 to edit previously recorded prompt
• Press * to go back previous menu
Step 5 Press 2.
You hear the total number of prompts available for your network and the message recorded for each prompt.
You hear the following options for each prompt:
• Press 2 to rerecord the prompt
• Press 3 to delete it
• Press # To go to next prompt
• Press * to skip playback of prompts
Step 6 Press 2 when you hear the prompt you want to rerecord.
You hear the following:
Record the new prompt at the tone. Press # to finish recording.
Step 10 Repeat steps 6 to 9 for the prompts that you want to edit.
Step 11 End the call.
Alternate Greeting
Alternate greeting is a voice message recorded from a phone by the customer administrator or a phone user
with the "PromptAdministrators" privileges. Alternate greeting is used if there is a disaster or a sudden business
need to replace the existing prompts. Alternate greeting is played first followed by the welcome prompt and
the open or closed prompts. Alternate greeting is played irrespective of open and closed business hours.
Note Alternate Greeting does not show up in the BE4000 portal and all administration must be done via the Prompt
Administrator Phone Menu 70397.
Procedure
Step 4 Press 1.
You hear the following prompts:
• Press 1 to record a new prompt
Step 5 Press 1.
You hear the following:
Record the new prompt at the tone. Press # to finish recording.
Procedure
Step 4 Press 1.
You hear the following prompts:
Your alternate greeting is currently active.
Step 5 Press 2.
You hear the following:
Record the new prompt at the tone. Press # to finish recording.
Procedure
Step 4 Press 2.
Step 5 Press 3.
You hear the following:
This operate is not reversable. Are you sure you wish to continue?
• Press 1 for yes
• Press 2 for no
Procedure
Step 1 Click Managed Site from Actions menu for the desired site on the dashboard.
Step 2 Click Business Hours.
Step 3 Click one of the following options:
• 24/7 (No Closed Hours)—Business is functional on 24 hours on all days.
• Dual Hours (Open and Closed)—Set open and closed hours for each day based on the organization
requirements.
Note • Auto Attendant menu is played based on the Open and Closed hours
• You must enter time in 24-hour format only (17:00 for example). Time must be either full (:00)
or half hours (:30).
Step 4 (Optional) Click Add New Holiday to add the holidays for the organization.
Note • You can add holidays only for the current year and a year ahead.
• Date should not be less than the present date.
Step 5 Click the arrow at the bottom-right corner of the screen and click Yes to apply the changes made.
Users
You can view all the existing users' information under Manage Site > User Management. You can also, add,
delete, modify email address, resend registration email, and reset password for the users.
Add a User
Before you begin
Ensure that you have:
• First name, last name, display name, extension, and email address of the user
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click User Management. The User Management page is displayed.
Step 3 Click Add User.
Step 4 Enter the information in the fields. For field descriptions, refer Extensions - Field Descriptions, on page 80.
Step 5 Click the arrows at the bottom of the screen to move to next screen or to go back to the previous screen.
Step 6 Click Yes.
What to do next
Provision the Phone for the user.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click User Management. The User Management page is displayed.
Step 3 Click Modify Email from the Actions menu for the desired user.
Step 4 Change the email address as required.
Step 5 Click Yes.
The registration email is sent to the newly entered email address.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click User Management. The User Management page is displayed.
Step 3 Click Resend Registration Email from the Actions menu for the desired user.
Step 4 Click OK.
The registration email is sent to the user. User must re-register to log in to the Cisco Business Edition Selfcare
Portal.
Resend Cisco Business Edition Selfcare Portal Registration Email to All Users
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click User Management. The User Management page is displayed.
Step 3 Choose Resend Registration Email from the Bulk Edit drop-down list.
Step 4 Click Resend All.
The registration email is sent to all the users. Users must re-register to log in to Cisco Business Edition Selfcare
Portal.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click User Management. The User Management page is displayed.
Step 3 Click Reset Password from the Actions menu for the desired user.
Step 4 Click OK.
The current password is erased and password reset instruction email is sent. User must change the password
to log in to Cisco Business Edition Selfcare Portal.
Reset Cisco Business Edition Selfcare Portal Password for All Users
Before you begin
The user must be registered to the Cisco Business Edition Selfcare Portal.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click User Management. The User Management page is displayed.
Step 3 Choose Reset Password from the Bulk Edit drop-down list.
Step 4 Click Reset All.
All the current passwords are erased and password reset instruction email is sent to all registered users. Users
must change their passwords to log in to Cisco Business Edition Selfcare Portal.
Delete a User
BE4000 supports deleting a user. Deleting a user releases the extension and removes the phone settings from
the system.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click User Management. The User Management page is displayed.
Step 3 Choose Delete User from the Actions menu for the desired user.
Step 4 Click Delete.
Extensions
An extension number is a unique number that is assigned to an employee in an organization. An employee
can have more than one extension number.
Dial Plan for your organization defines the number of digits for an extension and the maximum number of
extensions that can be configured at a site. Cisco partner configures the dial plan during the site creation.
Contact Cisco partner if you need more details on dial plan configured for a site.
Add an Extension
While adding an extension, you can also configure call forwarding, single number reach, voicemail capabilities,
and other calling features for an extension.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Extensions. The Manage Extensions page is displayed.
Step 3 Click Add Extension.
Step 4 Enter the information in the fields. For field descriptions, refer Extensions - Field Descriptions, on page 80
Step 5 Click the arrows at the bottom of the screen to move to next screen or to go back to the previous screen.
Step 6 Click Yes.
Field Description
Display Name Display name for the end user. The name entered here is displayed on the
called phone device when a call is received from this extension.
Field Description
User Type • User—An extension that is assigned to the end user. You must configure
an email address associated with the end user.
• Public—An extension that is assigned to a phone that is meant for
general use by many end users. You need not configure an email address.
For example, the extension assigned to a phone in the conference room.
Field Description
Email Address Email address of the end user. Cisco Cisco Business Edition Selfcare Portal
registration link is sent to this email address.
Enable Voicemail Voicemail to email feature is enabled for this extension. Any voicemail
coming to this extension is sent as an email attachment to the registered email
address.
Field Description
All All the incoming calls are forwarded to another extension, voicemail address,
or an E.164 number.
Night Service The incoming calls are forward calls to another extension during night service
hours.
Busy The incoming calls are forwarded to another extension, voicemail address,
or an E.164 number, only when the extension is busy.
No-Answer The incoming calls are forwarded to another extension, voicemail address,
or an E.164 number, when the calls are not answered.
No-Answer Timeout Time in seconds up to which a call rings on the extension when no one
(3-60,000 secs) answers. After this time out period, the call gets forwarded to configured
extension, voicemail address, or an E.164 number.
Unregistered When the extension is unregistered to BE4000, the calls are forward to another
extension, voicemail address, or an E.164 number. An extension can be in
unregistered state when the phone device is unplugged or the network between
the BE4000 and the extension is not functional.
Field Description
Check the Enable SNR check box to enable Single Number Reach Functionality.
Single Number Reach (SNR) provides end users the choice of answering an incoming call on the desk phone
using a mobile phone (cellular network). On enabling SNR, an incoming call rings both on the desk phone
and the mobile phone. An end user can answer the call either on the desk phone or from a mobile phone
based on the convenience. An active call can be swapped between the desk phone and the mobile phone
without disconnecting the call.
SNR Number The number for Single Number Reach (SNR) functionality.
Note When entering the SNR number, you must start with the prefix to
dial an outside line, followed by Country Code, National
Destination Code (Area Code), and Subscriber Number. For
example, if 9 is the digit to dial an outside line, 1 is the country
code, 555 is the area code, and 9999999 is the subscriber number,
you must enter 915559999999.
Call Forward No-Answer An incoming call is forwarded to the SNR number when the call is not
answered on the desk phone.
Calling Number Local Calling party number displayed on the configured mobile phone is replaced
with the SNR extension number.
SNR Delay (0-10 secs) Number of seconds up to which a phone rings before transferring the call to
configured SNR number. Range = 0 to 10 seconds. Default = 1.
SNR Timeout (5-60 secs) Number of seconds up to which phone rings after the configured SNR delay.
When the timeout value is reached, incoming call display is stopped on the
desk phone and the call gets forwarded to the configured SNR number. Range
= 5 to 60 seconds. Default = 10.
Ring Stop An incoming call stops ringing on the desk phone after the call is answered
from a mobile phone and conversely.
Answer Too Soon (1-5 secs) Number of seconds up to which Single Number Reach (SNR) calls are
prevented from being diverted to the voice mailbox of a mobile phone. Range
= 1 to 5 seconds.
Field Description
Not Register Number Extension is not associated with an external proxy server.
Pickup Call Answer an incoming call on the extensions belonging to any pick-up group.
Press GPickup followed by * on the phone to answer the pickup group call.
Field Description
Pickup Group Associate the extension with a pick-up group. 1 is the default pickup group
created by the BE4000.
Night Service Bell Incoming calls that ring during the night service period on the extension sends
an alert indication to all extensions that are marked to receive night service
bell notification. The alert notification is in the form of a splash ring (not
associated with any of the individual lines on the phone) and a visible display
of the extension. The phone users retrieve the call by pressing the Pickup
softkey.
Shared Line One extension can be shared across multiple phone devices. You can have a
minimum of 2 and a maximum of 16 calls.
Field Description
Note Every extension must be associated with a phone for it to be available for provisioning through
the Extension Assigner. Hence, we recommend you to add a phone while adding the extension.
You can also add phone after adding extension, under Manage Site > Phones.
Modify an Extension
If there is any change in an employee's phone system or calling features, you can modify the extension.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Extensions. The Manage Extensions page is displayed.
Step 3 Click Modify Extension from the Actions menu for the desired Extension.
Step 4 Modify the information as desired.
Note You can modify all field information except for the Extension.
Phones
You can add new phones and provision the phones using extension assigner. A maximum of 200 phones can
be configured at your site.
Adding a phone at your site involves the following steps:
1. Add a New Phone, on page 84
2. Provision the Phone Using Extension Assigner, on page 86
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Phones.
Step 3 Click Add Phone.
Step 4 Enter the information in the fields. For field description, refer Phones - Field Descriptions, on page 84
Step 5 Click the arrows at the bottom of the screen to move to next screen or to go back to the previous screen.
Step 6 Click Yes.
Field Description
COR Class of Restriction (COR). Enables the restrictions on the type of calls placed
from the phone.
Busy Trigger per Button The maximum number of calls allowed on an octo-line directory number
(1-50) before activating Call Forward Busy or a busy tone.
Field Description
MAC Address Note This field is applicable only for Cisco ATA 190 Analog Telephone
Adapter.
Field Descriptions
Block Conference Pattern Phone is prevented from initiating a conference to external numbers.
Overlap signal Called digits are sent one by one as they are received from the calling device.
Privacy-button Privacy capabilities are enabled for individual phones. Privacy prevents other
phone users from viewing the call information or barging into a call on a
shared-line number.
Note Privacy is supported for calls on shared-lines only.
Block Transfer-pattern Transfer restrictions are applied when a call transfer or conference is initiated
toward external parties such as a PSTN trunk, SIP trunk.
Note Transfer to local extensions are exempted from this restriction.
ATA-IVR-Password Password to access Interactive Voice Response (IVR) and change the default
phone settings on Analog Telephone Adaptors.
Local Directory Local directory service on the phones are available. By default, Local
Directory check box is checked.
My Phone Apps Local services on a phone’s My Phone Apps interface is made available. By
default, My Phone Apps check box is checked.
Field Description
You can assign an extension, speed dial, or BLF speed dial for the buttons on the phone. Click the Actions
column corresponding to the desired button to edit or add a task for button.
Type • Extension
• Speed Dial
• BLF Speed Dial
Procedure
Step 1 Connect the phone to the network. The phone gets assigned with a temporary extension.
Step 2 Dial the Extension Assigner directory number 70000; When prompted for password, enter 1234.
Step 3 Enter the pound (#) key.
Step 4 Enter the permanent extension that has been configured on the portal for this user followed by the pound (#)
key.
Step 5 Enter 1 followed by the pound (#) key to assign the extension.
Replace a Phone
Note Do not delete the phone from the Portal. Deleting the phone from the portal removes all user-specific
customizations such as speed dials, single number reach, call forward.
You can replace a faulty phone device or upgrade to a new phone model. Replacing the phone involves the
following steps:
1. Unplug the existing or faulty phone from the network.
2. Connect the new phone to the network.
3. Provision the new phone using Extension Assigner. For detailed steps, refer Provision the Phone Using
Extension Assigner, on page 86.
Procedure
Step 1 Delete the extension that was previously associated with the phone from the BE4000 Partner Portal. Go to
Manage Sites > Extensions > Manage Extensions and click Delete from the Actions menu corresponding to
the desired extension.
Step 2 Add an extension for the new user who uses the phone device in BE4000 Partner Portal. For details on adding
an extension, refer Add an Extension, on page 80.
Step 3 Use Extension Assigner to provision the phone with the extension associated with the new user. For details
on Extension Assigner, refer Provision the Phone Using Extension Assigner, on page 86.
Note KEM cannot be added during initial site deployment in the Setup Assistant. You can add KEMs only after
the site is successfully deployed.
To add a KEM to your Cisco IP Phone 8800 Series, perform the following tasks:
Step 1 Add a KEM on the Add a KEM for the Site is in the Online Add a KEM on the
BE4000 Portal Cisco IP Phone 8800 state BE4000 Portal, on
Series on the page 89
BE4000 portal.
Step 2 (Task performed on Connect the KEM KEM is adding on Connect a KEM to a
the site) Connect a module to the Cisco the BE4000 portal Cisco IP Phone, on
KEM to a Cisco IP IP Phone 8800 page 89
Phone Series phone. After
successfully
connected, KEM
pulls the
configurations from
the portal.
Procedure
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Phones.
Step 3 Click Modify Phone from the Actions menu for the desired phone.
Step 4 Choose the desired number of KEMs from the Add-on Module drop-down list on the Basic Info page.
Step 5 Click the arrows at the bottom of the screen to move to next screens.
Step 6 Choose an action for each button from the Type drop-down list on the Buttons page. Enter the details based
on your selection.
Step 7 Click Yes.
Personal Mailbox
Personal mailbox is assigned to a specific user and is accessible only by that user. When a caller leaves a
message in the mailbox, the message waiting indicator (MWI) light turns on.
You can specify the maximum quota for the voice mailbox for every user using the portal. The default mailbox
size is 10 minutes with maximum message size of 2 minutes.
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Personal Mailbox.
Step 3 Click Add User.
Step 4 Enter the information in the fields. For field descriptions, refer Personal Mailbox - Field Descriptions, on
page 90.
Step 5 Click the arrows at the bottom of the screen to move to next screen or to go back to the previous screen.
Step 6 Click Yes.
Field Description
Display Name Display name for the end user. The name entered here is displayed on called
phone device when a call is received from this extension.
Pin Personal Identification Number (PIN). To manage your voicemail box from
the phone, enter this PIN.
Confirm Pin
Field Description
Field Description
Message Expiry Time in days Maximum number of days up to which messages are stored in the mailbox.
Play Tutorial • Yes—Voicemail tutorial is played when the user enters the mailbox for
the first time.
• No—Voicemail tutorial is not played when the user enters the mailbox
for the first time.
Default—Yes.
Enabled Enable or disable mailbox. Enabled check box should be checked by default
for mailbox to function. Default—Yes.
Default—Standard.
Default—User-Recording.
Field Description
Add Groups Groups to which the user belongs. Search for a group and click Add.
Step 1 Created an extension for the user. For information on how to create an extension, refer Add an Extension, on
page 80.
Step 2 Create a Personal Mail for the user. Ensure that you specify PIN on the User Profile page. For information
on how to create a mailbox, refer Set Up Personal Mailbox for a User, on page 90.
Note If you do not enter PIN, the user cannot configure the mailbox.
What to do next
Users can access the configured the mailbox from any phone. For information on how to access the configured
mailbox, refer Access Personal Mailbox-Users Without an Assigned Phone, on page 92.
Procedure
Step 4 Enter the PIN that is configured in the portal while creating the personal mailbox.
Configuring system operator for the site, enables all the personal mailboxes to allow its callers to dial zero
for reaching the system operator.
Procedure
Step 1 Log in to the Cisco Business Edition 4000 Management Portal (https://be4000.cisco.com/).
Step 2 Click Manage Site for the desired site from the Actions menu.
Step 3 Click Setting > System Settings.
Step 4 Choose the one of the following as the system operator target type from the Target Type drop-down list:
• Extension—Any extension that is configured for the site.
• Direct to Voicemail—Voicemail of any extension.
• Hunt Group—Any hunt group that is configured for the site.
Step 5 Choose the number corresponding to the target type selected from the Target Number drop-down list.
All the extensions that have voicemail enabled (or have personal mailbox configured) allows the incoming
callers to dial zero to reach the system operator.
What to do next
Cisco Partner or Customer Administrator communicates to all the users who have voicemail enabled (or have
personal mailbox configured) to record their personal greeting to include instructions on dialing zero to reach
system operator. For example, the user, Edward records his personal greeting as “Edward is out of office.
Please dial zero to reach the system operator”.
Group Mailbox
Group mailbox is assigned to a group of users. All members in the group have access to the group mailbox.
When a caller leaves a message in a group mailbox, message waiting indicator (MWI) is not turned on.
However, when members log in their personal mailbox, the mailbox menu allows the members to access the
messages in each General delivery mailbox (GDM) to which the member belongs. Only one person can access
the GDM at a time.
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Group Mailbox.
Step 3 Click Add Group Mailbox.
Step 4 Enter the information in the fields. For field description, refer Group Mailbox - Field Descriptions, on page
94.
Step 5 Click the arrows at the bottom of the screen to move to next screen or to go back to the previous screen.
Step 6 Click Yes.
Field Description
Field Description
Full Name Long name of the group as it appears on other phone display.
E.164 Number Phone number (including country and area code) associated with the group.
Add Privilege Privilege associated with the group. Search for the privilege and click Add.
Field Description
Add Owners Users who own the group. Search for a user and click Add.
Note You must have at least one user as owner of the group.
Field Description
Message Expiry Time in days Maximum number of days up to which messages are stored in the mailbox.
Play Tutorial • Yes—Voicemail tutorial is played when the user enters the mailbox for
the first time.
• No—Voicemail tutorial is not played when the user enters the mailbox
for the first time.
Enabled Enable or diable mailbox. Enabled check box should be checked by default
for mailbox to function.
Field Description
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Group Mailbox.
Step 3 Click Modify Group from the Actions menu for the desired Group Mailbox.
Step 4 Navigate to Owners/Members page.
Step 5 Search for a user in the Add Members field and click Add.
Step 6 Click the arrows at the bottom of the screen to move to next screen or to go back to the previous screen.
Step 7 Click Yes.
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard
Step 2 Click Group Mailbox.
Step 3 Click Modify Group from the Actions menu for the desired Group Mailbox.
Step 4 Navigate to Owners/Members page.
Step 5 Search for a user in the Modify Owners field and click Add.
Step 6 Click the arrows at the bottom of the screen to move to next screen or to go back to the previous screen.
Step 7 Click Yes.
Hunt Groups
Hunt Groups allow incoming calls to a specific number (pilot number) to be directed to a defined group of
extension numbers. Incoming calls are redirected from the pilot number to the first extension number as
defined in the configuration. If the first number is busy or does not answer, the call is redirected to the next
phone in the list. A call remains redirected on busy or no answer from number to number in the list until it is
answered or until the call reaches the number that is defined as the final number.
A Hunt Group can have static and dynamic members.
• Static Members—Permanent members belonging to the Hunt Group.
• Dynamic Members—Not the permanent members, but they can join or unjoin a Hunt Group on a need
basis using the softkeys available on the phone.
Note • The total number of members in a Hunt Group, including static and dynamic members cannot exceed
32.
• If you check the "Allow dynamic members" check box on the Hunt Groups page, ensure that you check
the "Hunt Group Login" check box for each dynamic member extension under Manage Site > Extensions
> Basic Info page.
A Cisco Partner can configure the hunt groups using the Setup Assistant in Cisco Business Edition 4000
Management Portal during the initial site deployment. If the Cisco Partner has already created hunt groups
during the initial deployment, a customer administrator can see them on Hunt Groups screen. A customer
administrator can also create a new hunt group and also modify the existing hunt groups.
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Hunt Groups.
Step 3 Click Add Hunt Groups.
Step 4 Enter the information in the fields. For field descriptions, refer Hunt Groups - Field Descriptions, on page
98.
Step 5 Click the arrows at the bottom of the screen to move to next screen or to go back to the previous screen.
Group Name Enter a unique name for the hunt list. To easily identify the hunt list,
consider appending the pilot extension to the name; for example, hl5001.
Group name must contain a minimum of two characters.
Pilot Number Enter a number to access the Hunt Group that serves as the pilot for the
hunt list. This number serves as the trigger for hunting to begin. Pilot
number must contain a minimum of four digits and be within the range
of 1000 to 9999.
Note The Pilot Number of the Hunt Groups cannot be the same as
any existing extension and cannot start with the digit that is
used for sending calls to voicemail automatically and for
placing intercom calls.
Add Members Click Add to add members to this hunt group from the Stations page
(Show Member Directory). You can also search for the users by entering
member name or extension. All extensions that are assigned to users or
departments can be included as members of a Hunt Group. You must
add a minimum of two members for a hunt group.
Allow dynamic members Allows members that are not part of the Hunt Group to join and unjoin
the Hunt Group on a need basis using the softkeys displayed on the
screen.
Max dynamic members Note This field is visible only when "Allow dynamic members" is
checked.
Show Member Directory From the list of extensions that display, select which extensions must
be included in the hunt list.
Click Show Member Directory, to select the list of extensions for the
hunt list. Check the respective member's name and click OK.
Field Description
Hunt Method Select how BE4000 distributes the calls to members of the hunt list
based on one of the following hunt methods:
• Longest-idle—BE4000 only distributes a call to idle members,
starting from the longest idle member to the least idle member of
a hunt list.
• Parallel—Calls ring all numbers in that hunt group simultaneously.
The extension to first answer the call is connected.
• Sequential—Call hunting always starts with the first member in
the hunt group. Continues to reach number in the group in the order
in which they are listed, from top to bottom, in the hunt group.
• Peer—Call hunting starts with the extension immediately after the
one that just took the last call. Ringing proceeds in a circular
manner, that is from left to right. That is, BE4000 distributes a call
to idle or available members starting from the (n+1)th member of
a hunt list, where the nth member is the member to which BE4000
most recently extended a call. If the nth member is the last member
of a hunt list, BE4000 distributes a call starting from the top of the
hunt list.
Max Waiting Time Enter the maximum time to wait before disconnecting the call when the
queue is busy or full. The range is from 0 to 100 seconds.
When No Member is Available If no members of the hunt list are available to answer a call, you can
choose to perform one of the following:
• Disconnect—the call is disconnected.
• Route to Group Mailbox—the call is forwarded to a group mailbox.
Enter the email address and extension associated with the group
mailbox.
• Route to Number—the call is forwarded to an extension. Choose
the desired extension from the drop-down list.
Step 1 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 2 Click Hunt Groups.
Step 3 Click Modify from the Actions menu for the desired Hunt Group.
Step 4 Add the extension associated with the users in Extension List field. Use commas to separate extensions.
Night Service
Night service allows you to transfer the incoming calls to a designated set of extensions during closed hours.
During the night service hours (also known as closed hours), calls coming in to the designated extension,
known as night service extensions, sends a special "burst" ring to night-service phones (phones that receive
the calls coming from the night service extension) that have been specified to receive the special ring. Phone
users at the night-service phones can then answer the incoming calls for the night-service extensions.
Note You can configure only one night-service phone per night-service extension.
Example
Nancy is a receptionist with extension 1234 at ABC organization having business hours from 0900
to 1700. When night service is configured, the calls received on extension 1234 between 17:00 and
08:59 is transferred to extension 5678 that is designated to receive calls during closed hours.
Example
ABC organization has set the night service activation code as *1234. Business hours is from Monday
to Friday 09:00 to 17:00. If Nancy at the reception wants to log out for a couple of hours during the
business hours, she can dial *1234 to trigger the night service mode on the phone. The incoming
calls to her extension are transferred to the designated night service phone when she is away. When
she is back, she can dial *1234 to log out of the night service mode and enter the business hours
mode to receive the incoming calls.
Procedure
Step 6 Add the business holidays when you want the Night Service to be active.
Step 7 Click Yes to save the changes.
Procedure
Step 4 Enter the number to which calls must be forwarded during night service hours in the Night Service field on
Call Forward page.
Step 5 Check the Night Service Bell check box on the Additional Settings page.
Note You must configure Night Service Bell for all the Night Service phones designated to receive Night
Service calls. The Night Service bell ensures that all the phones designated for Night Service receive
the burst ring. For more information, refer to Designate an Extension to Receive Night Service
Calls, on page 102.
Procedure
Pickup Group
Pickup Group is a group of extensions, where the phone users can answer an incoming call on any of the
extensions belonging to the same pickup group. By default, BE4000 creates 1 as the pickup group for all the
extensions. You can change the pickup group based on your preference. You can assign only one pickup
group per extension.
Procedure
Step 5 Enter a number for the pickup group in the Pickup Group field.
Step 6 Click Yes to save the changes.
Step 7 Repeat steps 3 to 6 for all the extensions belonging to the group.
Procedure
Note Call Detail Records and Emergency Alerting—These features are currently in development and are not
intended for production use.
BE4000 allows you to generate the call detail records (CDR) reports after the site is successfully deployed.
The CDR report is generated based on the successful incoming and outgoing external calls that are made per
customer site for a specific date range. The report includes details such as the calling and called numbers,
location, time, and duration of each call.
You can download a local copy of the Call History CDR reports.
Currently, all CDR reports are retained indefinitely, unless CDR collection is disabled. However, the CDR
report retention period is subjected to change in the future.
Note If CDR collection is disabled, all previously collected call detail records get deleted and cannot be recovered.
Procedure
Step 2 Click Manage Site from Actions menu for the desired site on the dashboard.
Step 3 Click Reports > Call Detail Records. The Manage Express Reports page is launched in a new tab.
Step 4 Choose a location from the Location drop-down list.
Step 5 Choose Call History from Call History drop-down list.
Step 6 Select the required date range. The report is displayed on the screen.
Step 7 View the call history details. You can also perform a search based on the calling and called numbers.
What to do next
(Optional) Download Call History Report for a BE4000 Site .
Procedure
Procedure
Procedure
Step 8 (Optional) Click Grid View to view the number of concurrent calls on an hourly basis, peak time, and peak
count.
Emergency Alerting
Note Call Detail Records and Emergency Alerting—These features are currently in development and are not
intended for production use.
BE4000 allows you to monitor the calls that are made from within the organization to specific external numbers,
such as (911). You can enter the numbers to monitor and the email address to receive the notification when
the specified numbers are dialed.
Note • Cisco ATA 190 Analog Telephone Adapter can be added only post site deployment. You cannot add
during initial site deployment.
• Cisco ATA 190 Analog Telephone Adapter can be added only under Manage Site > Phones. You cannot
add Cisco ATA 190 Analog Telephone Adapter under Manage Site > Extensions and Manage Site >
User Management.
• A fax call does not fall back to voice only call after the fax is sent.
• The following features are not supported:
• Paging
• Single Number Reach (SNR)
• Speed dial
• TFTP Address
• IP Address and subnet mask for Cisco ATA-190 Analog Telephone Adapter
Procedure
Note Do not add FXO line numbers while adding DID numbers as FXO line numbers can be added when line cards
are added.
• Extension—You can select any one of the existing extensions. The incoming calls on the DID number
ring on the specified extension.
• Hunt Group—You can map the incoming calls on the DID number to an existing hunt group. The
incoming calls on the DID numbers ring on the extensions belonging to the specified hunt group.
You can also set a default target for all the DID numbers belonging to a service provider. The default target
can be auto attendant, extension, or hunt group. If there are any DID numbers that are not mapped to auto
attendant, extension, or hunt group, the call is directed to the default target set for the service provider.
Outbound Caller ID
BE4000 allows you to configure a specific DID number to be displayed on the called phone when an outbound
call is placed from an extension within the organization.
You can also set a default outbound DID number for a service provider. In such a case, an extension without
a specified DID number displays the default outbound DID number configured for the service provider.
Example
ABC organization has 12345 and 67890 as DID numbers from service provider XYZ. The default
outbound DID number set for XYZ service provider is 12345. ABC organization has two extensions
4501 and 4502. Extension 4501 is mapped to 67890 as Outbound DID. Extension 4502 is mapped
to default outbound DID.
When an outbound call is placed from extension 4501, the called phone displays the incoming call
number as 67890 without revealing the extension number within the organization ABC.
When a call is placed from extension 4502, the called phone displays the incoming call number as
12345 which is the default outbound DID number set for the service provider XYZ.
Procedure
Step 4 Choose a default target for each service provider from the Default Target drop-down list on the Inbound
Call Mapping page. Based on your preference, choose a target type (auto attendant, extension, or hunt group)
for each registered number from Target Type drop-down and choose a corresponding target number from
Target Number drop-down list.
Step 5 Choose a default outbound DID for each service provider from the Default Outbound DID drop-down list
on the Outbound Caller ID page. Based on your preference, choose a DID number as Caller ID for each
extension from the Caller ID drop-down list.
Step 6 Click Yes to save the changes.
Procedure
Note BE4000 supports only .au and .wav audio file with G.711; ITU-T a-law or u-law, 8kHz, 8 bit, Mono
format. The file cannot be larger than 1 MB (about 2 minutes). The filename cannot have space and
special characters.
Step 4 Click the arrow at the bottom of the screen and click Yes to save.
Procedure
Step 1 Click Managed Site from Actions menu for the desired site on the dashboard.
Step 2 Click Maintenance Schedule under Settings.
Step 3 Enter the Maintenance Schedule Beginning time for each day of the week.
Note The two hour duration of maintenance schedule is auto adjusted in the Ending field based on the
value you entered in the Beginning field.
Procedure
• In the “IP Address or Domain Name” field, enter one of the following:
Note You can configure Fully Qualified Domain Name (FQDN) of public SMTP servers only.
BE4000 handles all Domain Name System (DNS) resolutions through the internet and thus
only public FQDNs are accepted. For example, smtp.office365.com, smtp.gmail.com.
• If the customer has Office 365, enter the MX FQDN record copied from Office 365 portal
• If the customer has SMTP relay other than Office 365, enter the SMTP server IP Address or Domain
Name
• In the "Sender's Email Address", enter the email address that is used as "From" address while sending
the emails containing the voicemail attachments.
• (Optional) Check the Authenticate check box and enter the username and password. Use the credentials
for authenticating to the SMTP server.
What to do next
Deploy the BE4000 appliance. Refer Deploy a Customer Site for detailed steps.
Procedure
Procedure
Note • You cannot schedule more than one backup per day.
• You must configure at least one day in a week to back up your site configuration. If you have not scheduled
a backup for your site, by default, the site is backed up every Saturday during the maintenance schedule.
Procedure
What to do next
Restore the backed-up site configuration. Refer to, Restore the Backed-Up Site Configuration, on page 116
for more details.
Procedure
Step 4 Choose a backed-up configuration from the Please select a backup drop-down list.
The previous five consecutive back-ups are available to restore.
Step 5 Choose the configurations that you want to restore by clicking one from the following:
• All Data—Restores the complete site configuration that includes, phone, extension, dial plan, voicemail,
and auto attendant data.
• Phone, Extensions, Dial Plan—Restores the phone, extensions, and dial plan data.
• Voicemail & Auto Attendant—Restores the voicemail and auto attendant data.
Note We recommend you to click All Data and restore the complete site configuration.
Procedure
Note When the partner administrator performs a Delete Site on the BE4000 portal, the BE4000 is automatically
reset to factory defaults, and a manual reset is not required.
If a manual factory reset is desired, perform it only after deleting the site from the BE4000 portal. If the factory
reset was performed before deleting a site, you must reload (power cycle) the appliance after deleting the site
from BE4000 portal and before it is deployed again.
Choose one of the following two methods to connect to and reset the BE4000:
Step 1 Connect your computer back to back to MGMT port (use a normal Cat5e or Cat6 cable).
Step 2 Use SSH client to connect to the BE4000 using either the MGMT IP Address (169.254.100.1) or Host Name
(status@be4000).
Step 3 Log in with username status. No password is required.
Step 4 Type h to see the available options.
Reset OTP
You can now reset the One Time Password (OTP) account of another user within the same partner organization.
Procedure
Procedure
Procedure
Step 1 From the dashboard, click on your name displayed on the top-right corner.
Step 2 Click Settings. The Profile page is displayed.
Step 3 Enter your primary contact number in the Primary Phone field.
Note The primary phone number must contain a minimum of 10 digits.
Note You cannot edit the email address. The email address is auto populated based on the information
provided during partner registration.
Procedure
Step 1 From the dashboard, click on your name displayed on the top-right corner.
Step 2 Click Settings. The Settings page is displayed.
Step 3 Click Add Device. Two-Step Authentication page is displayed with the list of configured devices.
Note The nicknames for the devices are displayed.
Step 10 (Optional) From the drop-down for the secondary device, click Primary to make the secondary device primary.
Note You can set a secondary device as your primary device, from the sign-in screen as well.
Procedure
Procedure
Step 1 Click Manage Site from the Actions column for the desired site.
Step 2 Click edit (pen icon) under Contact Information.
Step 3 Click Reset OTP.
Note Resetting the OTP account mandates the customer administrator to create a new OTP application
account when logging in to the Cisco Business Edition Selfcare Portal.
Step 1 Click Manage Site from the Actions column for the desired site.
Step 2 Click edit (pen icon) under Contact Information.
Step 3 Enter the phone number. Provide the mobile number or a number that can be reached by outside the BE4000
system. Do not enter the extensions within the BE4000 system.
Step 4 Click Save.
Field Description
Customer Name Enter the name of the customer. You can enter a maximum of 15 characters.
Note One customer can have multiple sites. Each site needs one BE4000
appliance. Each site is configured separately.
Location Enter the location of the customer. You can enter a maximum of 15 characters.
Customer Admin Email Enter the email address of the customer administrator.
Template Name Select an existing customer site template. Template saves the site configurations
(connectivity details, dial plans, stations, call routing and feature details). Using
template, you can avoid rekeying the configuration details while creating a new
site similar to an existing site configuration.
Note If you are creating a site for the first time, you may not find any
existing templates listed in the drop-down.
Field Description
Network Details
BE4000 IP Address Enter IP address of the BE4000 appliance. Ensure that the IP address matches
with that of the customer subnet address and ports (UDP 500, UDP 4500, and
ESP 50) are connected, active, and reachable to the internet. Check the status of
the ports using Cisco Business Edition 4000 Port Check Tool.
Voicemail IP Address Enter the IP address of the voicemail server. Voicemail IP address cannot be the
same as BE4000 IP address.
Subnet Mask Enter the IP subnet mask. For example, 255.255.255.0 is an IP subnet mask.
ISP Enter the ISP label. In case, there are issues in deploying the BE4000, the ISP
label helps technicians to check if the fault is with the service provider. ISP label
is an arbitrary identifier.
Field Description
IP Address or Domain Enter the IP Address or Domain Name of the SMTP server.
Name
Note You can configure Fully Qualified Domain Name (FQDN) of public
SMTP servers only. BE4000 handles all Domain Name System (DNS)
resolutions through the internet and thus only public FQDNs are
accepted. For example, smtp.office365.com, smtp.gmail.com.
Sender's Email Address Enter an email address that is used as “From” address to send emails containing
voicemail as an attachment. Ensure that the SMTP server allows receiving emails
from the entered email address.
Authenticate Check the "Authenticate" check box to ensure that the voicemail to email
functionality is secure. Enter a username and password that needs to be filled by
the customers while accessing the voicemail from email.
Note • DID numbers must be entered in E.164 format. A minimum of 10 digits is required for the DID number.
For example: +14155552671
• Only up to 98 DID numbers are supported for SIP trunks and 98 DID numbers for ISDN trunks (PRI
and BRI).
Field Description
Choose file Click to browse and upload the file containing DID numbers.
Service Name Enter a service name for each DID. Provide any name that is easy to identify the
service to which each DID number belongs to.
Registered Numbers Add the DID numbers received from your service provider for each row.
Replace this list Click to remove all the DID numbers displayed on the page.
Field Description
Service Name Choose a service name from the drop-down list. The service names added in the
Direct Inward Dial (DID) Numbers page are listed in the drop-down.
Provider Template Choose a preconfigured provider template based on your SIP service provider.
If your SIP service provider is not in the drop-down list, choose "Custom".
Field Descriptions
Use Secondary Interface Choose the type of interface connectivity for SIP trunk. Primary interface refers
for Trunk? to GE 0/0/0 and Secondary interface refers to GE 0/0/1. Primary interface is
always connected to the internet service provider. If the internet service provider
and SIP trunk service provider are different, use secondary interface for SIP trunk
connectivity.
• Proxy Server Field Descriptions—SIP trunk and internet connectivity is
provided by the same service provider.
• Secondary Interface with Static Address—SIP trunk and internet are provided
by two separate service providers. Internet service provider is connected
using the primary interface and SIP trunk service provider is connected using
the secondary interface. The SIP trunk service provider provides static IP
address for the network connectivity.
• Secondary Interface with Dynamic Address—SIP trunk and internet are
provided by two separate service providers. Internet service provider is
connected using the primary interface and SIP trunk service provider is
connected using the secondary interface. The SIP trunk service provider
provides dynamic IP address for the network connectivity.
External Public Address Enter the public IP address assigned by your internet service provider so that SIP
services work across Network Address Translation (NAT).
No Secondary Interface
Field Description
Proxy Address Enter an IP address and Port, Fully Qualified Domain Name (FQDN) and Port,
or SRV record for your service proxy.
Proxy Port Optional. If you have provided an IP address for the proxy, you may also specify
a non-standard SIP port if necessary. Leave blank to use port 5060.
Outbound Proxy
Outbound Proxy Address Enter an IP Address, fully qualified domain name, or domain SRV for your
service outbound proxy if one is used.
Outbound Proxy Port Optional. If you have provided an IP address for the outbound proxy, you may
also specify a non-standard SIP port if necessary. Leave blank to use port 5060.
Call Authentication
Username and Password Enter the username and password if your service provider requires authentication
for every call.
Field Description
Authentication Realm Enter the authentication realm for call authentication. Typically, authentication
realm is the service domain name.
Include in Invite Check the "Include in Invite" check box, if your service provider requires
authentication details to be sent in the initial invite. If unchecked, authentication
is provided in the response to a 407 challenge.
Field Description
Min-SE Enter the minimum value for the session expiry parameter sent in the initial invite.
Range is from 90 to 86,400 seconds. Unless instructed by your SIP service
provider, the default value of 90 seconds must be used.
Session Expires Enter the maximum duration of a session in seconds. During a call, the session
expiry time is periodically refreshed based on the value entered here. Range is
from 90 to 86,400 seconds. Unless instructed by your provider, the default value
of 1800 seconds should be used.
RTP Port Range Limit the range of ports used for RTP. Enter even numbers between 8,000 to
48,198.
Transport Layer Choose the protocol used for transport layer by your service provider.
Fax Transmission Choose one of the ITU-T T.38 standard Fax Transmission Protocols to be used
Protocol (Optional) for a specific VoIP dial peer. Available options are:
• T.38
• T.38 fall back to G.711 u-law
• T.38 fall back to G.711 a-law
• Pass Through G711u
• Pass Through G711a
DTMF Signaling Protocol Choose one of the following as the DTMF signaling mechanism based on the
protocol offered by your SIP service provider.
• RFC2833
• sip-notify
Calling Party Header Choose one of the following for calling party header selection:
Selection
• From
• Remote Party ID (RPID)
• P-AID Pilot DID
• P-AID Assigned DIDs
Field Description
Calling Party Domain Leave the "Calling Party Domain" field blank to send the BE4000 IP address
with calling party headers. Enter a domain name or full qualified domain if you
want to replace the BE4000 IP address.
Pilot Number Note "Pilot Number" field is displayed only when "Calling Party Header
Selection" drop-down is chosen as "P-AID Pilot DID".
Choose the "Pilot Number" from the drop-down list if the service provider requires
a specific number to be used for P-Asserted Identity Headers.
CLI Restriction Prefix Enter the dialing prefix if the service provider allows calling line ID to be withheld
on a call by call basis.
RFC3555 Compliant Uncheck the "RFC3555 Compliant G.729 Annex B" check box if the call server
G.729 Annex B is not RFC3555 compliant for G.729 Annex B SDP formatting (Adds g729-annexb
override). Check if you are unsure.
Two way media override Check the "Two way media override" check box to override modification of
media stream from send/receive to sendonly or inactive. When checked, two way
media is always be requested.
Redirection (Optional) Check the Redirection option to reset the default processing of 3xx messages.
By default, SIP gateways process all incoming 3xx redirect messages according
to RFC 2543. However if the Redirection option is disabled, the gateway treats
the incoming 3xx responses as 4xx error class responses.
Redirection should be selected by default and only unselected if required by the
SIP trunk provider.
Options Ping
Enable to monitor the SIP service availability allowing traffic to be rerouted, if possible, in the event of
failure.
Service Up Interval Enter the period between Options packets being sent while the service is
considered to be up. Range is from 5 to 1,200 seconds. Default is 60 seconds.
Service Down Interval Enter the period between Options packets being sent while the service is
considered to be down. Range is from 5 to 1,200 seconds. Default is 30 seconds.
Retries Enter the number of missed responses allowed before a service is considered
unavailable. Range is from 1 to 10. Default is 5.
Registrar Server
Registrar server can be configured either through DHCP or by providing IP address and port. Click one of
the following options based on your network:
• Configure via DHCP
• Configure address and port
Field Description
Registrar Address Enter an IP Address, fully qualified domain name, or domain SRV for service
registrar.
Registrar Port If you have provided an IP address for the registrar, you may also specify a
non-standard SIP port if necessary. Leave blank to use port 5060.
Authentication Realm Enter the authentication realm used for registration by your service provider.
Mandatory if Registrar Address or DHCP is configured.
Registrar with realm Check the "Registrar with Realm" configuring the registrar with the realm
information provided for the proxy. Uncheck to remove the configuration.
Username and Password Enter the username and password, if the service provider requires per call
authentication.
Include DID Select appropriate DID for each username if the service provider requires a DID
to be included with registration authentication.
Add Row Click to add multiple rows for username, password, and include DID. You can
add a maximum of 12 rows.
Field Description
Registration Timeout Enter the Registration Timeout period. Determines how frequently the system
registers. Range is from
Transport Layer Choose TCP or UDP from the "Transport Layer" drop-down list as the transport
protocol used by the service provider.
Security
Add at least one trusted IP Address. The BE4000 accepts incoming VoIP (SIP) calls only if the remote IP
address of an incoming VoIP call matches an address in the trusted IP address list. Enter the IP addresses
provided for proxy, outbound proxy, and registrar from your service provider. IP addresses must be provided
if hostnames are used. Entries can be provided either as a host address (x.x.x.x) or subnet (x.x.x.x /nn)."
Field Description
Trusted IP Address Enter a trusted IP address to authenticate incoming SIP trunk calls for toll fraud
prevention.
Add Row Click "Add Row" and enter the trusted IP addresses.
Interface Settings
Field Description
Interface Options Ethernet ports usually use the auto-negotiate protocol settings. If your switch
does not support this option by itself, choose from the following interface options:
• Auto Negotiate
• Gigabit Ethernet
• Fast Ethernet Full Duplex
• Fast Ethernet Half Duplex
• Ethernet Full Duplex
• Ethernet Half Duplex
IP Address and Mask Enter IP address and subnet mask of the secondary interface. The fields are
mandatory.
Default Gateway Enter the IP address of the default gateway. This field is mandatory.
Name Servers Enter the IP address of the dedicated, private DNS used by your service provider.
Ensure that you enter the name server addresses even if they are provided via
DHCP. You can enter a maximum of 6 IP addresses separated by spaces.
External Public Address Enter To ensure that SIP services work across Network Address Translation,
provide the public IP address provided by your service provider.
Field Description
Proxy Address Enter an IP address and Port, Fully Qualified Domain Name (FQDN) and Port,
or SRV record for your service proxy.
Proxy Port Optional. If you have provided an IP address for the proxy, you may also specify
a non-standard SIP port if necessary. Leave blank to use port 5060.
Outbound Proxy
Outbound Proxy Address Enter an IP Address, fully qualified domain name, or domain SRV for your
service outbound proxy if one is used.
Outbound Proxy Port Optional. If you have provided an IP address for the outbound proxy, you may
also specify a non-standard SIP port if necessary. Leave blank to use port 5060.
Call Authentication
Field Description
Username and Password Enter the username and password if your service provider requires authentication
for every call.
Authentication Realm Enter the authentication realm for call authentication. Typically, authentication
realm is the service domain name.
Include in Invite Check the "Include in Invite" check box, if your service provider requires
authentication details to be sent in the initial invite. If unchecked, authentication
is provided in the response to a 407 challenge.
Field Description
Min-SE Enter the minimum value for the session expiry parameter sent in the initial invite.
Range is from 90 to 86,400 seconds. Unless instructed by your SIP service
provider, the default value of 90 seconds must be used.
Session Expires Enter the maximum duration of a session in seconds. During a call, the session
expiry time is periodically refreshed based on the value entered here. Range is
from 90 to 86,400 seconds. Unless instructed by your provider, the default value
of 1800 seconds should be used.
RTP Port Range Limit the range of ports used for RTP. Enter even numbers between 8,000 to
48,198.
Transport Layer Choose the protocol used for transport layer by your service provider.
Fax Transmission Choose one of the ITU-T T.38 standard Fax Transmission Protocols to be used
Protocol (Optional) for a specific VoIP dial peer. Available options are:
• T.38
• T.38 fall back to G.711 u-law
• T.38 fall back to G.711 a-law
• Pass Through G711u
• Pass Through G711a
DTMF Signaling Protocol Choose one of the following as the DTMF signaling mechanism based on the
protocol offered by your SIP service provider.
• RFC2833
• sip-notify
Field Description
Calling Party Header Choose one of the following for calling party header selection:
Selection
• From
• Remote Party ID (RPID)
• P-AID Pilot DID
• P-AID Assigned DIDs
Calling Party Domain Leave the "Calling Party Domain" field blank to send the BE4000 IP address
with calling party headers. Enter a domain name or full qualified domain if you
want to replace the BE4000 IP address.
Pilot Number Note "Pilot Number" field is displayed only when "Calling Party Header
Selection" drop-down is chosen as "P-AID Pilot DID".
Choose the "Pilot Number" from the drop-down list if the service provider requires
a specific number to be used for P-Asserted Identity Headers.
CLI Restriction Prefix Enter the dialing prefix if the service provider allows calling line ID to be withheld
on a call by call basis.
RFC3555 Compliant Uncheck the "RFC3555 Compliant G.729 Annex B" check box if the call server
G.729 Annex B is not RFC3555 compliant for G.729 Annex B SDP formatting (Adds g729-annexb
override). Check if you are unsure.
Two way media override Check the "Two way media override" check box to override modification of
media stream from send/receive to sendonly or inactive. When checked, two way
media is always be requested.
Redirection (Optional) Check the "Redirection (Optional)" to reset the default processing of 3xx
messages. By default, SIP gateways process all incoming 3xx redirect messages
according to RFC 2543. However if the Redirection option is disabled, the gateway
treats the incoming 3xx responses as 4xx error class responses.
Redirection should be selected by default and only unselected if required by the
SIP trunk provider.
Options Ping
Enable to monitor the SIP service availability allowing traffic to be rerouted, if possible, in the event of
failure.
Service Up Interval Enter the period between Options packets being sent while the service is
considered to be up. Range is from 5 to 1,200 seconds. Default is 60 seconds.
Service Down Interval Enter the period between Options packets being sent while the service is
considered to be down. Range is from 5 to 1,200 seconds. Default is 30 seconds.
Retries Enter the number of missed responses allowed before a service is considered
unavailable. Range is from 1 to 10. Default is 5.
Registrar Server
Registrar server can be configured either through DHCP or by providing IP address and port. Click one of
the following options based on your network:
• Configure via DHCP
• Configure address and port
Field Description
Registrar Address Enter an IP Address, fully qualified domain name, or domain SRV for service
registrar.
Registrar Port If you have provided an IP address for the registrar, you may also specify a
non-standard SIP port if necessary. Leave blank to use port 5060.
Authentication Realm Enter the authentication realm used for registration by your service provider.
Mandatory if Registrar Address or DHCP is configured.
Registrar with realm Check the "Registrar with Realm" configuring the registrar with the realm
information provided for the proxy. Uncheck to remove the configuration.
Username and Password Enter the username and password, if the service provider requires per call
authentication.
Include DID Choose appropriate DID for each username if the service provider requires a DID
to be included with registration authentication.
Add Row Click to add multiple rows for username, password, and include DID. You can
add a maximum of 12 rows.
Field Description
Registration Timeout Enter the Registration Timeout period. Determines how frequently the system
registers. Range is from
Transport Layer Choose TCP or UDP from the "Transport Layer" drop-down list as the transport
protocol used by the service provider.
Security
Add at least one trusted IP Address. The BE4000 accepts incoming VoIP (SIP) calls only if the remote IP
address of an incoming VoIP call matches an address in the trusted IP address list. Enter the IP addresses
provided for proxy, outbound proxy, and registrar from your service provider. IP addresses must be provided
if hostnames are used. Entries can be provided either as a host address (x.x.x.x) or subnet (x.x.x.x /nn)."
Field Description
Trusted IP Address Enter a trusted IP address to authenticate incoming SIP trunk calls for toll fraud
prevention.
Add Row Click "Add Row" and enter the trusted IP addresses.
Interface Settings
Field Description
Interface Options Ethernet ports usually use the auto-negotiate protocol settings. If your switch
does not support this option by itself, choose from the following interface options:
• Auto Negotiate
• Gigabit Ethernet
• Fast Ethernet Full Duplex
• Fast Ethernet Half Duplex
• Ethernet Full Duplex
• Ethernet Half Duplex
Name Servers Enter the IP address of the dedicated, private DNS used by your service provider.
Ensure that you enter the name server addresses even if they are provided via
DHCP. You can enter a maximum of 6 IP addresses separated by spaces.
External Public Address Enter To ensure that SIP services work across Network Address Translation,
provide the public IP address provided by your service provider.
Field Description
Proxy Address Enter an IP address and Port, Fully Qualified Domain Name (FQDN) and Port,
or SRV record for your service proxy.
Proxy Port Optional. If you have provided an IP address for the proxy, you may also specify
a non-standard SIP port if necessary. Leave blank to use port 5060.
Outbound Proxy
Outbound Proxy Address Enter an IP Address, fully qualified domain name, or domain SRV for your
service outbound proxy if one is used.
Field Description
Outbound Proxy Port Optional. If you have provided an IP address for the outbound proxy, you may
also specify a non-standard SIP port if necessary. Leave blank to use port 5060.
Call Authentication
Username and Password Enter the username and password if your service provider requires authentication
for every call.
Authentication Realm Enter the authentication realm for call authentication. Typically, authentication
realm is the service domain name.
Include in Invite Check the "Include in Invite" check box, if your service provider requires
authentication details to be sent in the initial invite. If unchecked, authentication
is provided in the response to a 407 challenge.
Field Description
Min-SE Enter the minimum value for the session expiry parameter sent in the initial invite.
Range is from 90 to 86,400 seconds. Unless instructed by your SIP service
provider, the default value of 90 seconds must be used.
Session Expires Enter the maximum duration of a session in seconds. During a call, the session
expiry time is periodically refreshed based on the value entered here. Range is
from 90 to 86,400 seconds. Unless instructed by your provider, the default value
of 1800 seconds should be used.
RTP Port Range Limit the range of ports used for RTP. Enter even numbers between 8,000 to
48,198.
Transport Layer Choose the protocol used for transport layer by your service provider.
Fax Transmission Choose one of the ITU-T T.38 standard Fax Transmission Protocols to be used
Protocol (Optional) for a specific VoIP dial peer. Available options are:
• T.38
• T.38 fall back to G.711 u-law
• T.38 fall back to G.711 a-law
• Pass Through G711u
• Pass Through G711a
DTMF Signaling Protocol Choose one of the following as the DTMF signaling mechanism based on the
protocol offered by your SIP service provider.
• RFC2833
• sip-notify
Field Description
Calling Party Header Choose one of the following for calling party header selection:
Selection
• From
• Remote Party ID (RPID)
• P-AID Pilot DID
• P-AID Assigned DIDs
Calling Party Domain Leave the "Calling Party Domain" field blank to send the BE4000 IP address
with calling party headers. Enter a domain name or full qualified domain if you
want to replace the BE4000 IP address.
Pilot Number Note "Pilot Number" field is displayed only when "Calling Party Header
Selection" drop-down is chosen as "P-AID Pilot DID".
Choose the "Pilot Number" from the drop-down list if the service provider requires
a specific number to be used for P-Asserted Identity Headers.
CLI Restriction Prefix Enter the dialing prefix if the service provider allows calling line ID to be withheld
on a call by call basis.
RFC3555 Compliant Uncheck the "RFC3555 Compliant G.729 Annex B" check box if the call server
G.729 Annex B is not RFC3555 compliant for G.729 Annex B SDP formatting (Adds g729-annexb
override). Check if you are unsure.
Two way media override Check the "Two way media override" check box to override modification of
media stream from send/receive to sendonly or inactive. When checked, two way
media is always be requested.
Redirection (Optional) Check the "Redirection (Optional)" to reset the default processing of 3xx
messages. By default, SIP gateways process all incoming 3xx redirect messages
according to RFC 2543. However if the Redirection option is disabled, the gateway
treats the incoming 3xx responses as 4xx error class responses.
Redirection should be selected by default and only unselected if required by the
SIP trunk provider.
Options Ping
Enable to monitor the SIP service availability allowing traffic to be rerouted, if possible, in the event of
failure.
Service Up Interval Enter the period between Options packets being sent while the service is
considered to be up. Range is from 5 to 1,200 seconds. Default is 60 seconds.
Service Down Interval Enter the period between Options packets being sent while the service is
considered to be down. Range is from 5 to 1,200 seconds. Default is 30 seconds.
Retries Enter the number of missed responses allowed before a service is considered
unavailable. Range is from 1 to 10. Default is 5.
Registrar Server
Registrar server can be configured either through DHCP or by providing IP address and port. Click one of
the following options based on your network:
• Configure via DHCP
• Configure address and port
Field Description
Registrar Address Enter an IP Address, fully qualified domain name, or domain SRV for service
registrar.
Registrar Port If you have provided an IP address for the registrar, you may also specify a
non-standard SIP port if necessary. Leave blank to use port 5060.
Authentication Realm Enter the authentication realm used for registration by your service provider.
Mandatory if Registrar Address or DHCP is configured.
Registrar with realm Check the "Registrar with Realm" configuring the registrar with the realm
information provided for the proxy. Uncheck to remove the configuration.
Username and Password Enter the username and password, if the service provider requires per call
authentication.
Include DID Choose appropriate DID for each username if the service provider requires a DID
to be included with registration authentication.
Add Row Click to add multiple rows for username, password, and include DID. You can
add a maximum of 12 rows.
Field Description
Registration Timeout Enter the Registration Timeout period. Determines how frequently the system
registers. Range is from
Transport Layer Choose TCP or UDP from the "Transport Layer" drop-down list as the transport
protocol used by the service provider.
Security
Add at least one trusted IP Address. The BE4000 accepts incoming VoIP (SIP) calls only if the remote IP
address of an incoming VoIP call matches an address in the trusted IP address list. Enter the IP addresses
provided for proxy, outbound proxy, and registrar from your service provider. IP addresses must be provided
if hostnames are used. Entries can be provided either as a host address (x.x.x.x) or subnet (x.x.x.x /nn)."
Field Description
Trusted IP Address Enter a trusted IP address to authenticate incoming SIP trunk calls for toll fraud
prevention.
Add Row Click "Add Row" and enter the trusted IP addresses.
Field Description
NIM-2FXS or NIM-4FXS
Field Description
Class of Restriction decides the type of calls that can be placed from the FXS
phone line.
Law Choose the type of algorithm used for modifying an input signal for digitization:
• u-law
• a-law
NIM-2FXO or NIM-4FXO
Field Description
Direction Mark the line as incoming only or bidirectional. The system builds the trunk
groups based on what you select.
• In + Out—Allows the phone line to receive and make calls.
• Inbound Only—Allows the phone line to receive the calls.
Law Choose the type of algorithm used for modifying an input signal for digitization:
• u-law
• a-law
NIM-2FXS/4FXO
Field Description
FXS
Class of Restriction decides the type of calls that can be placed from the FXS
phone line.
Law Choose the type of algorithm used for modifying an input signal for digitization:
• u-law
• a-law
Field Description
FXO
Direction Mark the line as incoming only or bidirectional. The system builds the trunk
groups based on what you select.
• In + Out—Allows the phone line to receive and make calls.
• Inbound Only—Allows the phone line to receive the calls.
Law Choose the type of algorithm used for modifying an input signal for digitization:
• u-law
• a-law
NIM-2BRI-NT/TE or NIM-4BRI-NT/TE
Field Description
Service Name Choose a service name from the drop-down list. The drop-down list contains the
list of service providers that you added in the DID page.
Note You cannot choose the same service provider for SIP and Line Cards.
Static TEI If your service provider requires that your line use a static Terminal Endpoint
Identifier, enter the value between 0 and 63. If the field is left blank, the line
attempt to negotiate a TEI.
Overlap Receiving Choose whether you want your call setup to work based on overlap receiving.
You can enable or disable this option. If your service provider does not use
“enbloc” signaling, this option allows BE4000 to wait for additional digits to be
received before the call is routed.
Send Redirecting IE Check the "Send Redirecting IE Number" check box to include the Redirecting
Number Number Information Element in the outbound Setup messages. Leave unchecked
if you are not sure about your service provider supporting this feature.
ISDN SPID Enter the ISDN SPID. Some service providers use service profile identifiers
(SPIDs) to define the services subscribed to by the ISDN device that is accessing
the ISDN service provider. A SPID is usually a seven-digit phone number with
some optional numbers.
Field Description
TEI Negotiation Method Choose a method for TEI negotiation based on your service provider requirements.
Setting a static TEI overrides TEI negotiation.
The default behavior is TEI to be negotiated on power-up. The following options
are provided to preserve or remove a negotiated TEI when the interface is reset:
• Power Up and Remove
• Power Up and Preserve
• First Call and Remove
• First Call and Preserve
Field Description
Service Name Choose a service name from the drop-down list. The drop-down list contains the
list of service providers that you added in the DID page.
Note You cannot choose the same service provider for SIP and Line Cards.
Card Type Choose the card type based on your customer network requirement. E1 PRI is
chosen by default. The available options are:
• T1 PR1
• E1 PRI
ISDN Switch Type Choose one of the following ISDN Service Provider PRI Switch Types:
• primary-4ess
• primary-5ess
• dms100
• primary-net5
• primary-ni
Controller Setup Defines the controller setup for configuring channelized T1 or E1 controllers.
Choose either Full PRI or partial PRI.
Line Code Choose a line code. By default, the line code for E1 PRI is high-density bipolar
3 (hdb3).
Framing Choose the framing from the drop-down list. This option defines the framing
characteristics.
Field Description
Send Redirecting IE Check the "Send Redirecting IE Number" check box to include the Redirecting
Number Number Information Element in the outbound Setup messages. Leave unchecked
if you are not sure about your service provider supporting this feature.
Field Description
Telephony Port Tones Choose your home country. This is used to display the date, time, currency, and
other dial plan tones and numbers.
Time Zone Choose your relevant time zone. Typically, your time zone is linked to the area
code of your main company number. For example, for area code 919 (RTP),
the time zone defaults to Pacific Time.
Phone Display Language Choose your phone display language as the default language used for all accounts
and notifications from the drop-down list.
Phone Tones Select the country to define dial tone for your phones.
Voicemail and System Select the language in which you want to receive your phone greetings.
Prompt Language
Selfcare Portal Select the language preference for your customers self care portal.
Time Format Select the time format as 12-or 24 hour. For example, the default format for the
United States is 12 hours.
Date Format Select the date format to suit your needs. For example, the default format for
the United States is MM/DD/YY.
DST Auto Adjust Enables or disables the automatic adjustment of daylight saving time on your
phones.
Note Choose unique digits for dialing an outside line, sending a call to voicemail automatically, and dialing an
intercom extension.
Field Description
Dial an Outside Line Choose a digit to make a call to an outside phone number. You can set any digit
between 0 to 6, 8, and 9. You cannot set * and 7. Default is 9. Users should dial
this digit before dialing an external phone number.
Field Description
Extension length Choose the total number of digits in an extension. You can set your extension
to contain 3, 4, or 5 digits. Default is 4.
Interdigit Timeout Choose the number of seconds to wait after each digit is entered, before assuming
the caller has finished entering digits. Range is from 0 to 9. Default is 5.
Send to Voicemail Choose a digit to dial for sending a call to voicemail automatically. Range is
Automatically from 1 to 6. Default is 2.
Intercom Choose a digit to dial for making an intercom call. Range is from 1 to 6. Default
is 4.
Advanced Options
Forwarding Local Choose to enable or disable forwarding local. This decides if internal (local)
calls can be forwarded.
Phone Redirect Limit Set the phone redirect limit. Limits the maximum number of 3XX responses
that can be accepted for a single call. Range is from 5 to 20. By default, 5 is
entered.
Demo Enables and configures the NIM switch module in NIM slot 1. This converts
the BE4000 into a demo box that does not require an external switch.
Field Description
Country Choose the country and locale that you want for your
system.
Local Dialling Options Select the option for local dialing as per customer
requirement. The local area length value depends on
the regulation set up by the service providers in your
region.
Local Area Code Enter a valid “Local Area Code” for your main
number. This field appears based on the local dialing
option selected.
Field Description
Field Description
Display Name Enter the display name of the user. The name entered here is displayed on the phone
along with the extension number.
Email Enter the email address of the user. The top-level domain in the email address can
contain up to six characters.
Note Email address must not be more than 32 characters in length. Only letters,
numbers, and the characters underscore (_), dot (.), and dash (-) are allowed
in the user ID portion of the email address. Do not use spaces in the email
address.
Field Description
Phone Type Choose the phone model associated with the extension. For the list of supported phone
models, refer to “Supported Phones” section in the Cisco Business Edition 4000 Release
Notes.
COR Choose the Class of Restriction (COR) for the extension. COR allows you to choose
one of the calling privileges:
• Internal
• Local
• local-plus
• national
• national-plus
• international
Voicemail Note You must select the Class of Restriction (COR) for every line while adding
FXS cards in the Setup Assistant.
SNR Enter the Single Number Reach (SNR) number for an extension.
SNR allows you to answer the incoming calls on the desk phone or from a mobile phone.
You can also swap active calls on a desk phone or at a remote destination without
disconnecting the call. You should include the area code and any additional digits that
are required to obtain an outside line prefix to your destination number. Example—If
9 is the digit to dial outside line, 1 is the country code, 555 is the area code, and 9999999
is the subscriber number, you must enter 915559999999.
Replace this list Replace an exiting list with an entirely new list.
Download Allows you to download a customized template. Template should be of .csv format.
Template
Field Description
When you select Dual Hours (Open and Closed), the following menu is displayed:
Field Description
Hours of Operation Customize your business hours for your various departments. You can specify
the open hours for each day of the week.
Note You must enter time in 24-hour format only (17:00 for example). Time
must be either full (:00) or half hours (:30).
Add Open Hours Custom hours let you add and specify hours for each day of the week.
Add New Holiday Add the list of holidays for the organization.
Note • You can add holidays only for the current year and a year ahead.
• Date should not be less than the present date.
Hunt Group
Hunt Groups allow incoming calls to a specific number (pilot number) to be directed to a defined group of
extension numbers. Incoming calls are redirected from the pilot number to the first extension number as
defined in the configuration. If the first number is busy or does not answer, the call is redirected to the next
phone in the list. A call remains redirected on busy or no answer from number to number in the list until it is
answered or until the call reaches the number that is defined as the final number.
A Hunt Group can have static and dynamic members.
• Static Members—Permanent members belonging to the Hunt Group.
• Dynamic Members—Not the permanent members, but they can join or unjoin a Hunt Group on a need
basis using the softkeys available on the phone.
Note • The total number of members in a Hunt Group, including static and dynamic members cannot exceed
32.
• If you check the "Allow dynamic members" check box on the Hunt Groups page, ensure that you check
the "Hunt Group Login" check box for each dynamic member extension under Manage Site > Extensions
> Basic Info page.
Field Description
Group Name Enter a unique name for the hunt list. To easily identify the hunt list,
consider appending the pilot extension to the name; for example, hl5001.
Group name must contain a minimum of two characters.
Pilot Number Enter a number to access the Hunt Group that serves as the pilot for the
hunt list. This number serves as the trigger for hunting to begin. Pilot
number must contain a minimum of four digits and be within the range
of 1000 to 9999.
Note The Pilot Number of the Hunt Groups cannot be the same as
any existing extension and cannot start with the digit that is
used for sending calls to voicemail automatically and for
placing intercom calls.
Add Members Click Add to add members to this hunt group from the Stations page
(Show Member Directory). You can also search for the users by entering
member name or extension. All extensions that are assigned to users or
departments can be included as members of a Hunt Group. You must
add a minimum of two members for a hunt group.
Allow dynamic members Allows members that are not part of the Hunt Group to join and unjoin
the Hunt Group on a need basis using the softkeys displayed on the
screen.
Max dynamic members Note This field is visible only when "Allow dynamic members" is
checked.
Field Description
Show Member Directory From the list of extensions that display, select which extensions must
be included in the hunt list.
Click Show Member Directory, to select the list of extensions for the
hunt list. Check the respective member's name and click OK.
Hunt Method Select how BE4000 distributes the calls to members of the hunt list
based on one of the following hunt methods:
• Longest-idle—BE4000 only distributes a call to idle members,
starting from the longest idle member to the least idle member of
a hunt list.
• Parallel—Calls ring all numbers in that hunt group simultaneously.
The extension to first answer the call is connected.
• Sequential—Call hunting always starts with the first member in
the hunt group. Continues to reach number in the group in the order
in which they are listed, from top to bottom, in the hunt group.
• Peer—Call hunting starts with the extension immediately after the
one that just took the last call. Ringing proceeds in a circular
manner, that is from left to right. That is, BE4000 distributes a call
to idle or available members starting from the (n+1)th member of
a hunt list, where the nth member is the member to which BE4000
most recently extended a call. If the nth member is the last member
of a hunt list, BE4000 distributes a call starting from the top of the
hunt list.
Max Waiting Time Enter the maximum time to wait before disconnecting the call when the
queue is busy or full. The range is from 0 to 100 seconds.
When No Member is Available If no members of the hunt list are available to answer a call, you can
choose to perform one of the following:
• Disconnect—the call is disconnected.
• Route to Group Mailbox—the call is forwarded to a group mailbox.
Enter the email address and extension associated with the group
mailbox.
• Route to Number—the call is forwarded to an extension. Choose
the desired extension from the drop-down list.
Auto Attendant
Auto Attendant service (also referred to as a virtual receptionist), is a phone system that enables your callers
to be automatically transferred to an extension, eliminating the need for a receptionist and avoiding extended
waiting period. BE4000 provides you an automated phone answering facility to communicate effectively with
customers and improve your business operations. An auto attendant answers all incoming calls with an audio
greeting and options menu (different for open and closed hours). A maximum of five submenus with a
maximum depth of 3 levels can be configured. The caller can select a menu option to reach to the desired
extension.
You can define the number of times the menu options is played to the caller before the call reaches the drop
through destination. You can also define where the call lands if no action is performed by the caller even after
the defined number menu repetitions.
Field Description
Pilot Number Enter the number that callers dial to reach auto attendant. A minimum
of four digits is required. Range is from 1000 to 9999.
Number of repeats through menu Number of times the audio file is played to the caller before the call
reaches the drop through destination Value range is from 0 to 9. Default
value is 4.
Drop Through Destination Defines where the call lands if no action is performed by the caller even
after playing the menu for the defined number of repeats. You can
configure one of the following as the drop through destination:
• Extension—All extensions are listed in the drop-down.
• Direct to Voicemail—All extensions that have "Voicemail" enabled
are listed in the drop-down.
• Hunt Group—All Hunt Groups are listed in the drop-down.
• Group Mailbox—All Group Mailboxes are listed in the drop-down.
Note During the initial site deployment (in the Setup
Assistant), the drop-down shows an option, only if you
create a Group Mailbox on the Hunt Groups page, by
choosing "Route to Group Mailbox" from the When No
Member is Available drop-down list.
Field Description
Audio Prompt (Welcome Message) Add an audio prompt for welcome Message. The BE4000 provides a
default audio file. This audio message is played first when a call is
answered by the auto attendant. You can also upload a new .wav file.
To select a new file, click Upload.
Note BE4000 supports only .wav audio file with G.711 u-law,
8kHz, 8 bit, Mono format. The file cannot be larger than 1
MB (about 2 minutes). The filename cannot have space and
special characters.
Audio Prompt (Open Message) Add an audio prompt for open message. The BE4000 provides a default
audio file. This audio message is played when a call is answered during
the open business hours. You can also upload a new .wav file. To select
a new file, click Upload.
Note BE4000 supports only .wav audio file with G.711 u-law,
8kHz, 8 bit, Mono format. The file cannot be larger than 1
MB (about 2 minutes). The filename cannot have space and
special characters.
Add Menu Option Add customized menu options. You can add 0-9 menu options in addition
to a * menu. Each menu option can be labeled in a meaningful way to
help identify locations or users in your system using any one of the
following: Dial by Name, Pilot Number, Dial by Number, Call Hunt
Group, Repeat this Menu, Return to Main Menu, or Submenu.
Audio Prompt (Closed Message) Displays the default audio file that is played for all calls received during
closed hours. You can play the existing file or upload a new .wav file.
To select a new audio file, click Upload.
Note BE4000 supports only .wav audio file with G.711 u-law,
8kHz, 8 bit, Mono format. The file cannot be larger than 1
MB (about 2 minutes). The filename cannot have space and
special characters.
Add Menu Option Add customized menu options. You can add 0-9 menu options in addition
to a * menu. Each menu option can be labeled in a meaningful way to
help identify locations or users in your system using any one of the
following: Dial by Name, Pilot Number, Dial by Number, Call Hunt
Group, Repeat this Menu, Return to Main Menu, or Submenu.
Night Service
Night service allows you to transfer the incoming calls to a designated set of extensions during closed hours.
During the night service hours (also known as closed hours), calls coming in to the designated extension,
known as night service extensions, sends a special "burst" ring to night-service phones (phones that receive
the calls coming from the night service extension) that have been specified to receive the special ring. Phone
users at the night-service phones can then answer the incoming calls for the night-service extensions.
Note You can configure only one night-service phone per night-service extension.
Field Description
Manual Activation Code Enter a code to trigger Night Service feature manually during open
business hours. The code must start with * followed by at least 4
numbers, and a maximum of 16 numbers. Default value is *1234.
Active Hours Enter the hours during which the Night Service must be active.
You cannot overlap the end time of Night Service and the start of
business open hours. For example, if your business closes at 17:00 and
opens next day at 09:00AM, enter the Night Service hours as 17:00 and
08:59. You cannot enter 09:00 as it overlaps with the open business
hours. Night service hours must be entered in 24-hour time format.
Note After enabling night service, you must the extensions for night service hours and to receive night service calls.
Refer Night Service section for more details.
Field Descriptions
Default Target Choose a default target for all incoming calls belonging to a service provider. If
there is a registered number that is not assigned with any target type, then the
incoming calls are place on the default target set for the service provider.
Choose one of the following: Hunt Group, Auto Attendant, Extension.
Provider Send Digits Specify the number of digits provided by the service provider.
Registered Number The DID numbers that are registered to SIP Trunk and Line Cards are listed by
default.
Field Descriptions
• Extension—You can select any one of the existing extensions. The incoming
calls on the DID number ring on the specified extension.
• Hunt Group—You can map the incoming calls on the DID number to an
existing hunt group. The incoming calls on the DID numbers ring on the
extensions belonging to the specified hunt group.
Outbound Caller ID
You can configure a specific DID number to be displayed on the called phone when an outbound call is placed
from an extension within the organization. You can also set a default outbound DID number for a service
provider.
Field Description
Default Outbound DID Choose a default target for all outgoing calls of the service provider. An extension
without an assigned DID number displays the default outbound DID number
configured for the service provider.
Mapped Extension Displays the list of extensions available for the site.
Caller ID Choose a DID number to be displayed on the called phone when an outbound
call is placed from the extension within the organization.
System Operator
You can configure a number to be reached when a caller dials zero after listening to the personal mailbox
greeting.
Field Description
Target Type Choose the one of the following as the system operator target type from the
Target Type drop-down list:
• Extension—Any extension that is configured for the site.
• Direct to Voicemail—Voicemail of any extension.
• Hunt Group—Any hunt group that is configured for the site.
• Group Mailbox—Any group mailbox that is configured for the site.
• Pilot Number—Any pilot number, such as Auto Attendant pilot number.
Target Number Choose the number corresponding to the target type selected from the Target
Number drop-down list.
Field Description
Music on Hold (MoH) MoH allows you to play audio for incoming and outgoing calls placed on hold.
You can play the default audio file or upload a new audio file. To select a new
audio file, click Upload.
Note BE4000 supports only .au and .wav audio file with G.711; ITU-T
a-law or u-law, 8kHz, 8 bit, Mono format. The file cannot be larger
than 1 MB (about 2 minutes). The filename cannot have space and
special characters.
Note The previous five consecutive backups are stored in the BE4000 portal.
Licensing
You can associate the site with the Cisco Smart Account by providing the smart license token. The Smart
License Token field is a placeholder for entering the smart license token ID. Currently, the BE4000 does not
register to the Cisco Smart Account. If you enter a smart license token ID, we recommend setting a validity
date of at least 180 days when the token is created. Enter the smart license token ID received from Cisco in
the Smart License Token field.