CloudUCM - User Manual

  • Updated on June 20, 2024

INTRODUCTION

CloudUCM is a cloud PBX solution that provides a scalable and secure business communication and collaboration platform with powerful features and integrations that enable teams to be more productive than ever before. This Cloud PBX unifies all business communication into one centralized solution that provides voice and video calling, meetings, chat, data, analytics, mobility, surveillance, facility access, intercoms and more. CloudUCM supports all SIP endpoints and the Wave app for desktop, mobile, and web, allowing teams to communicate and collaborate from anywhere on nearly any device. This scalable solution can be easily expanded at any time without the need for extra equipment, provides enterprise-level security and reliability, and supports powerful third-party integrations and expansions. By providing a state-of-the-art suite of communication and collaboration features, bank-grade security, advanced customization, and a variety of plan options, CloudUCM is the ideal PBX solution for small-to-medium sized businesses, retail, hospitality, and residential deployments.

Note

To see the plans offered in CloudUCM in detail, please refer to the following link: https://ucmrc.gdms.cloud/clouducm/plans

TECHNICAL SPECIFICATIONS

The following table resumes all the technical specifications including the protocols/standards supported, voice codecs, telephony features, and languages for the CloudUCM.

Supported UC Endpoints and Client Devices

Support all SIP endpoint

Wave app for desktop (Windows 10+, macOS 10+), web (Firefox, Chrome, Safari, Edge, Opera) and mobile (Android & iOS)

Google Chrome extensions.

Call Features

Call park, call forward, call transfer, call waiting, caller ID, call record, call history, ringtone, IVR, music on hold, call routes, DID, DOD, DND, DISA, ring group, ring simultaneously, time schedule, PIN groups, call queue, pickup group, paging/intercom, voicemail, call wakeup, SCA, BLF, voicemail to email, fax to email, speed dial, call back, dial by name, emergency call, call follow me, blacklist/whitelist, voice conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control, post-meeting reports, virtual fax sending/receiving, email to fax

Built-in SBC

Free

All plans default to supporting built-in SBC services to protect CloudUCM systems from external attacks

Call with WebRTC Trunk

Supports mobile and desktop web browsers: Chrome, Edge , Safari, Firefox, Opera

Supports mobile application which built-in WebRTC WebView, such as Whatsapp, Facebook, Weixin and more

Collaboration

Audio and Video Meetings/Conferences, Instant Messaging and Group Chats with End-to-End Encryption, File Sharing, Screen Sharing, In-Meeting Chat, Voice Detection, Meeting Recording, Polls, Surveys, Message status, Advance Whiteboard with Multiplayer Annotation, Meeting Assistant, Onsite Meeting Room Scheduling, and more

Customer Service Support

  • Supports integration with third-party customer service platforms, including Whatsapp and Telegram. And built-in live chat

  • Includes a built-in live online web chat platform to provide customer service

  • Provides a web link that can be added to any web page or any browser that supports WebRTC

  • Compatible with computers, mobile browsers, and mobile apps

Customer Relationship Management (CRM)

Supports integration with ACT!, Bitrix24, Freshdesk, Hubspot, Salesforce, Sugar, Vtiger, Zendesk, Zoho, Dynamics 365, and more.

Call Center

Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ work-load, in-queue announcement.

Customizable Auto Attendant

Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

Property Management System (PMS)

Local PMS

Supports Integration with Hmobile PMS Systems.

Cloud Storage

Included, varies by plan, additional add-ons available.

CloudUCM App Store

Supports more than 20+ customized applications, with new apps being regularly added.

  • CRM add-ons

  • Google Drive and Office 365

  • Whatsapp and Telegram

  • Hotdesking (coming soon)

Microsoft Integration

Supports integration with Microsoft Teams (via TeamMate), Outlook, AD Contact, and Office 365

Computer Telephony Integration

CTI Mode to Control GXP, GRP, GXV, GHP Series' IP Phones

Wired and Bluetooth Headset

Supports docking with different types of headphones

Supports Microsoft Teams certified Headsets, supports phone call control

High Availability (HA)

Amazon Web Services (AWS) provides 99.99% service guarantee

HA between UCM6300 Series IP PBX and CloudUCM (coming soon)

HA between multiple CloudUCM systems (coming soon)

Firmware Upgrade and Provision

Supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system, GDMS provides a centralized interface to provision, manage, monitor and troubleshoot Grandstream products.

IP Cameras, Intercom and Door Access Integration

Supports Grandstream GSC Series IP Cameras and Intercom/Public Address devices, supports GDS Series Door Access Solutions

Supports third party devices, including Hikvision, Dahua, and more

API and SDK

Full CGI API available for third-party platform and application integration

Wave add-in SDK

Wave Andriod and iOS SDK

Wave H5 Embedded for MAC/Windows application

Multi-Language Support

  • Web User Interface: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish

  • Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic

  • Customizable language pack to support any other languages

Security

Frequency Restriction, Fail2ban, Ping Defense, Ping of Death, SYN-Flood, Remote Login Interception, Multi-factor Authentication, SMS Login Authentication

Network Protocols

SIP, TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, NTP, HTTP/HTTPS, STUN, SRTP, TLS, LDAP, IPv4/6

Internet Protocol Standards

RFC 3261, RFC 3262, RFC 3263, RFC 3264, RFC 3515, RFC 3311, RFC 4028. RFC 2976, RFC 3842, RFC 3892, RFC 3428, RFC 4733, RFC 4566, RFC 2617, RFC 3856, RFC 3711, RFC 4582, RFC 4583, RFC 5245, RFC 5389, RFC 5766, RFC 6347, RFC 6455, RFC 8860, RFC 4734, RFC 3665, RFC 3323, RFC 3550

DTMF Methods

In-band audio, RFC 2833, and SIP INFO

Transmission Encryption

SRTP, DTLS-SRTP, TLS, HTTPS

Voice-over-Packet Capabilities

LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711, NetEQ, FEC 2.0, jitter resilience up to 50% audio packet loss

Voice and Fax Codecs

Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38

Video Codecs

H.264, H.263, H263+, VP8

QoS

Layer 3 (ToS, DiffServ, MPLS) QoS

Administration

Call Detail Record, event alert and SMS notifications, event logs, export import extensions, feature codes, LDAP, feedback system, PBX monitor, resource monitor, system prompt, user permission, web-based control panel, user portal, trunk cluster, voice prompt customization, firewall, Fail2ban, IP blacklist, Syslog, gateway and endpoint provisioning, Wave permissions (deploy & configure Wave Desktop, installations en masse, pre-install Wave Add-ons for extensions, manage Wave feature, access permissions), local backup

SYSTEM STATUS

Dashboard

  • Storage Usage
  • CloudUCM
  • PBX Status
  • Trunks
CloudUCM Dashboard

System Information

System Information
Remark

Active Calls

The active calls on the CloudUCM are displayed in the Web GUI🡪System Status🡪Active Calls page. Users can monitor the status, hang up a call, and barge in the active calls in a real-time manner.

Active Calls Status

To view the status of active calls, navigate to Web GUI🡪System Status🡪Active Calls. The following figure shows extension 1004 is calling 1000. 1000 is ringing.

Active Calls – Ringing Status

The following figure shows the call between 1000 and 5555 is established.

Active Call – Established Status

The gray color of the active call means the connection of call time is less than half an hour. It means this call is normal.

The orange color of the active call means the connection of call time is greater than half an hour but less than one hour. It means this call is a bit long.

The red color of the active call means the connection of call time is more than one hour. It means this call could be abnormal.

Setup Wizard

When you log in to the CloudUCM Web GUI interface for the first time, the system will automatically start the setup wizard and expand the description.

The setup wizard guides users to complete basic configuration, such as administrator password modification, Email Delivery settings, time zone settings, extension settings, trunk&routes configuration, etc.

Setup Wizard

EXTENSION/TRUNK

Extensions

Create a New SIP Extension

To manually create a new SIP user, go to Web GUI🡪Extension/Trunk🡪Extensions. Click on “Add” and a new window will show for users to fill in the extension information.

Create New Extension

Extension options are divided into five categories:

  • Basic Settings
  • Media
  • Features
  • Voicemail
  • Specific Time
  • Wave Client
  • Follow me
  • Advanced Settings

The configuration parameters are as follows.

General

Extension

The extension number associated with the user.

CallerID Number

Configure the CallerID Number that would be applied for outbound calls from this user.

Note:

The ability to manipulate your outbound Caller ID may be limited by your VoIP provider.

Call Privileges

Assign permission level to the user. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal".

Note: Users need to have the same level as or higher level than an outbound rule's privilege to make outbound calls using this rule.

SIP Password

Configure the password for the user. A random secure password will be automatically generated. It is recommended to use this password for security purposes.

Concurrent Registrations

The maximum endpoints which can be registered into this extension. For security concerns, the default value is 3.

Note: When this option is set to "1(seize)"

Auth ID

Configure the authentication ID for the user. If not configured, the extension number will be used for authentication.

Disable This Extension

If selected, this extension will be disabled on the CloudUCM.

Note: The disabled extension still exists on the PBX but cannot be used on the end device.

User Settings

First Name

Configure the first name of the user. The first name can contain characters, letters, digits, and _.

Last Name

Configure the last name of the user. The last name can contain characters, letters, digits, and _.

Email Address

Fill in the Email address for the user. Voicemail will be sent to this Email address.

User/Wave Password

Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purposes.

Mobile Number

Configure the phone number for the extension, user can type the related star code for the phone number followed by the extension number to directly call this number.

For example, the user can type *881000 to call the mobile number associated with extension 1000.

Department

Configure the user's department. The department can be configured in User Management->Address Book Management->Department Management.
Job Title: The user's department position.

Job Title

Enter the job title of the user of the extension.

Contact Privileges

Same as Department Contact Privileges

When enabled, The extension will inherit the same privilege attributed to the department it belongs to.

Contact View Privileges

Select the privileges regarding the contact view in SIP endpoints and Wave.

SIP Extension Configuration Parameters🡪Basic Settings

SIP Settings

DTMF Mode

Select DTMF mode for the user to send DTMF. The default setting is "RFC4733". If "Info" is selected, the SIP INFO message will be used. If "Inband" is selected, a-law or u-law are required. When "Auto" is selected, RFC4733 will be used if offered, otherwise "Inband" will be used.

TEL URI

If the phone has an assigned PSTN telephone number, this field should be set to “User=Phone”. The “User=Phone” parameter will be attached to the Request-Line and “TO” header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel” will be used instead of “SIP” in the SIP request.

Alert-Info

When present in an INVITE request, the alert-Info header field specifies an alternative ring tone to the UAS.

Enable T.38 UDPTL

Enable or disable T.38 UDPTL support.

FECC

Configure to enable Remote Camera Management.

Codec Preference

Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p, RTX and VP8.

QoS

Jitter Buffer

Select the jitter buffer method.

  • Disable: Jitter buffer will not be used.

  • Fixed: Jitter buffer with a fixed size (equal to the value of "jitter buffer size")

  • Adaptive: Jitter buffer with an adaptive size (no more than the value of "max jitter buffer").

  • NetEQ: Dynamic jitter buffer via NetEQ.

Packet Loss Retransmission

Configure to enable Packet Loss Retransmission.

  • NACK

  • NACK+RTX(SSRC-GROUP)

  • OFF

Video FEC

Check to enable Forward Error Correction (FEC) for Video.

Audio FEC

Check to enable Forward Error Correction (FEC) for Audio.

Silence Suppression

If enabled, the UCM will send CN packets for silence suppression after a successful CN negotiation in the SIP SDP. If the client endpoint's OPUS codec supports the reception of DTX packets, the UCM will send DTX packets instead.

SRTP

SRTP

Enable SRTP for the extension.

  • Disabled

  • Enabled and Enforced

  • Optional

The default setting is disabled.

SRTP Crypto Suite

The following encryption protocols can be used to encrypt an RTP stream.

  • AES_CM_128_HMAC_SHA1_80 (This is the default used protocol)

  • AES_256_CM_HMAC_SHA1_80

  • AEAD_AES_128_GCM

  • AEAD_AES_256_GCM

SIP Extension Configuration Parameters🡪Media

Call Transfer

Presence Status

Select which presence status to set for the extension and configure call forward conditions for each status. Six possible options are possible: “Available”, “Away”, “Chat”, “Custom”, “DND” and “Unavailable”. More details at [PRESENCE].

Internal Calls & External Calls

Call Forward Unconditional

Enable and configure the Call Forward Unconditional target number. Available options for target number are:

  • None”: Call forward deactivated.

  • Extension”: Select an extension from the dropdown list as CFU target.

  • Custom Number”: Enter a customer number as a target. For example: *97.

  • Voicemail”: Select an extension from the dropdown list. Incoming calls will be forwarded to the voicemail of the selected extension.

  • Ring Group”: Select a ring group from the dropdown list as CFU target.

  • Queues”: Select a queue from the dropdown list as CFU target.

  • Voicemail Group”: Select a voicemail group from the dropdown list as CFU target.

  • Custom Prompt: The call will be forwarded to a custom prompt.

The default setting is “None”.

CFU Time Condition

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are ‘All’, ‘Office Time’, ‘Out of Office Time’, ‘Holiday’, ‘Out of Holiday’, ‘Out of Office Time or Holiday’, ‘Office Time and Out of Holiday’, ‘Specific Time’, ‘Out of Specific Time’, ‘Out of Specific Time or Holiday’, ‘Specific Time and Out of Holiday’.

Notes:

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward No Answer

Configure the Call Forward No Answer target number. Available options for target number are:

  • “None”: Call forward deactivated.

  • “Extension”: Select an extension from the dropdown list as CFN target.

  • “Custom Number”: Enter a customer number as a target. For example: *97.

  • “Voicemail”: Select an extension from the dropdown list. Incoming calls will be forwarded to the voicemail of the selected extension.

  • “Ring Group”: Select a ring group from the dropdown list as CFN target.

  • “Queues”: Select a queue from the dropdown list as CFN target.

  • “Voicemail Group”: Select a voicemail group from the dropdown list as CFN target.

  • Custom Prompt: The call will be forwarded to a custom prompt.

The default setting is “None”.

CFN Time Condition

Select time condition for Call Forward No Answer. The available time conditions are ‘All’, ‘Office Time’, ‘Out of Office Time’, ‘Holiday’, ‘Out of Holiday’, ‘Out of Office Time or Holiday’, ‘Office Time and Out of Holiday’, ‘Specific Time’, ‘Out of Specific Time’, ‘Out of Specific Time or Holiday’, ‘Specific Time and Out of Holiday’.

Notes:

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward Busy

Configure the Call Forward Busy target number. Available options for target number are:

  • None”: Call forward deactivated.

  • “Extension”: Select an extension from the dropdown list as CFB target.

  • Custom Number”: Enter a customer number as a target. For example: *97

  • “Voicemail”: Select an extension from the dropdown list. Incoming calls will be forwarded to the voicemail of the selected extension.

  • “Ring Group”: Select a ring group from the dropdown list as CFB target.

  • “Queues”: Select a queue from the dropdown list as CFB target.

  • “Voicemail Group”: Select a voicemail group from dropdown list as CFB target.

  • Custom Prompt:

The default setting is “None”.

CFB Time Condition

Select time condition for Call Forward Busy. The available time conditions ‘All’, ‘Office Time’, ‘Out of Office Time’, ‘Holiday’, ‘Out of Holiday’, ‘Out of Office Time or Holiday’, ‘Office Time and Out of Holiday’, ‘Specific Time’, ‘Out of Specific Time’, ‘Out of Specific Time or Holiday’, ‘Specific Time and Out of Holiday’.

Notes: 

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Do Not Disturb

If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.

DND Time Condition

Select time condition for Do Not Disturb. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”, and “Specific”.

Notes:

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

DND Whitelist

If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.

  • Z match any digit from 1-9.

  • N match any digit from 2-9.

  • X match any digit from 0-9.

FWD Whitelist

Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported.

  • match any digit from 1-9.

  • N match any digit from 2-9.

  • X match any digit from 0-9.

CC Settings

Enable CC

If enabled, CloudUCM will automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason. By default, it is disabled.

CC Mode

Two modes for Call Completion are supported:

  • Normal: This extension is used as an ordinary extension.

  • For Trunk: This extension is registered from a PBX.

The default setting is “Normal”.

CC Max Agents

Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel can make.

The minimum value is 1.

CC Max Monitors

Configure the maximum number of monitor structures that may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time.

The minimum value is 1.

Ring Simultaneously

Ring Simultaneously

Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as the caller ID number.

External Number

Set the external number to ring simultaneously. ‘-’ is the connection character that will be ignored.

This field accepts only letters, numbers, and special characters + = * #.

Time Condition for Ring Simultaneously

Ring the external number simultaneously along with the extension based on this time condition.

Use callee DOD on FWD or RS

Use the DOD number when calls are being diverted/forwarded to external destinations or when ring simultaneous is configured.

Monitor Privilege Control

Call Montoring Whitelist

Add members from “Available Extensions” to “Selected Extensions” so that the selected extensions can spy on the used extension using feature code.

Allow Operator Panel Monitoring

Configure whether this extension can be monitored by the Operator Panel administrator.

Seamless Transfer Privilege Control

Allowed Seamless Transfer

Any extensions on the CloudUCM can perform a seamless transfer. When using the Pickup Incall feature, only extensions available on the “Selected Extensions” list can perform a seamless transfer to the edited extension.

PMS Remote Wakeup Whitelist

Select the extensions that can set wakeup service for other extensions

Selected extensions can set a PMS wakeup service for this extension via feature code.

Other Settings

Ring Timeout

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the CloudUCM. The valid range is between 5 seconds and 600 seconds.

Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Auto Record

Enable automatic recording for the calls using this extension. The default setting is disabled. The recordings can be accessed under Web GUI🡪CDR🡪Recording Files.

Skip Trunk Auth

  • If set to “yes”, users can skip entering the password when making outbound calls.

  • If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition.

  • If set to “No”, users will be asked to enter the password when making outbound calls.

Time Condition for Skip Trunk Auth

If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering the password when making outbound calls.

Dial Trunk Password

Configure personal password when making outbound calls via the trunk.

Support Hot-Desking Mode

Check to enable Hot-Desking Mode on the extension. Hot-Desking allows using the same endpoint device and logs in using extension/password combination. This feature is used in scenarios where different users need to use the same endpoint device during a different time of the day for instance. If enabled, SIP Password will accept only alphabet characters and digits. Auth ID will be changed to the same as Extension.

Enable LDAP

If enabled, the extension will be added to the LDAP Phonebook PBX list.
Default is enabled.

Use MOH as IVR ringback tone

If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as a ringback tone instead of the regular ringback tone.

Music On Hold

Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.

Call Settings

Call Duration Limit

Check to enable and set the call limit the duration.

Maximum Call Duration (s)

The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds

The Maximum Number of Call Lines

The maximum number of simultaneous calls that the extension can have.
0 indicates no limit.

Outgoing Call Frequency Limit

If enabled, if the number of outbound calls exceed the configured threshold within the specified period, further outbound calls will be not be allowed.

Send PCPID Header

If enabled, this extension's SIP INVITE messages will contain the P-Called-Party-ID (PCPID) header if the callee is a SIP device.

Period (m)

The period of outgoing call frequency limit. The valid range is from 1 to 120. The default value is 1.

Max Number of Calls

Set the maximum number of outgoing calls in a period. The valide tange is from 1 to 20. The default value is 5.

Enable Auto-Answer Support

If enabled, the extension will support auto-answer when indicated by Call-info/Alert-info headers.

Call Waiting

Allows calls to the extension even when it is already in a call. This only works if the caller is directly dialing the extension. If disabled, the CC service will take effect only for unanswered and timeout calls.

Stop Ringing

If enabled, when the extension has concurrent registrations on multiple devices, upon incoming call or meeting invite ringing, if one end device rejects the call, the rest of the devices will also stop ringing. By default, it’s disabled.

Email Missed Call Log

If enabled, the log of missed calls will be sent to the extension’s configured email address.

Missed Call Type

If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:

  • Default: All missed calls will be sent in email notifications.

  • Missed Internal Call: Only missed local extension-to-extension calls will be sent in email notifications.

  • Missed External Call: Only missed calls from trunks will be sent in email notifications.

SIP Extension Configuration Parameters🡪Features

Specific Time

Time Condition

Click to add Time Condition to configure a specific time for this extension.

SIP Extension Configuration Parameters🡪Specific Time

Normal

Enable Wave

Enable Wave for the specific extension.

Wave Welcome Email

Wave Welcome Email template.

Wave

Download Link

SIP Extension Configuration Parameters🡪Wave

Follow Me

Enable

Configure to enable or disable Follow Me for this user.

Skip Trunk Auth

If the outbound calls need to check the password, we should enable this option or enable the option “Skip Trunk Auth” of the Extension. Otherwise, this Follow Me cannot call out.

Music On Hold Class

Configure the Music On Hold class that the caller would hear while tracking the user.

Confirm When Answering

If enabled, call will need to be confirmed after answering.

Enable Destination

Configure to enable destination.

Default Destination

The call will be routed to this destination if no one in the Follow Me answers the call.

Use Callee DOD for Follow Me

Use the callee DOD number as CID if configured Follow Me numbers are external numbers.

Play Follow Me Prompt

If enabled, the Follow Me prompt tone will be played.

New Follow Me Number

Add a new Follow Me number which could be a “Local Extension” or an “External Number”. The selected dial plan should have permissions to dial the defined external number.

Dialing Order

This is the order in which the Follow Me destinations will be dialed to reach the user.

SIP Extension Configuration Parameters🡪Follow Me

Search and Edit Extension

All the CloudUCM extensions are listed under Web GUI🡪Extension/Trunk🡪Extensions, with status, Extension, CallerID Name, IP, and Port. Each extension has a checkbox for users to “Edit” or “Delete”. Also, options “Edit” , “Reboot” and “Delete” are available per extension. Users can search for an extension by specifying the extension number to find an extension quickly.

Manage Extensions

  • Status

Users can see the following icon for each extension to indicate the SIP status.

Green: Idle

Blue: Ringing

Yellow: In Use

Grey: Unavailable (the extension is not registered or disabled on the PBX)

  • Edit single extension

Click on to start editing the extension parameters.

  • Reset single extension

Click on A close up of a logo

Description generated with high confidence to reset the extension parameters to default (except concurrent registration).

Other settings will be restored to default in Maintenance🡪User Management🡪User Information except for username and permissions and delete the user voicemail prompt and voice messages.

Note

This is the expected behavior when you reset an extension: 

  • All the data and configuration on the user side will be deleted. That includes user information, call history, call recordings, faxes, voice mails, meeting schedules, and recordings, as well as chat history. However, the data related to the user will be kept on the UCM side.
  • The extension will be removed from group chats and the messages sent previously by the extension will be kept. However, only other users can search through those messages while the new user of the extension cannot. 
  • If the extension was in a meeting schedule, the meeting will still be present. The extension will be removed from the meeting and will not be notified about the meeting.

  • Reboot the user

Click on to send NOTIFY reboot event to the device that has a CloudUCM extension already registered. To successfully reboot the user.

  • Delete single extension

Click on to delete the extension. Or select the checkbox of the extension and then click on “Delete Selected Extensions”.

Notes

This is the expected behavior when you delete an extension:

  • The system will delete all the data of the extension except the CDR and meetings records. All the data on the user side will be erased.
  • The extension will be removed from group chats and the messages sent previously by the extension will be kept. However, only other users can search through those messages while the new user of the extension cannot. 
  • If the extension was in a meeting schedule, the meeting will still be present. The extension will be removed from the meeting and will not be notified about the meeting. 

  • Modify selected extensions

Select the checkbox for the extension(s). Then click on “Edit” to edit the extensions in a batch.

  • Delete selected extensions

Select the checkbox for the extension(s). Then click on “Delete ” to delete the extension(s).

Export Extensions

The extensions configured on the CloudUCM can be exported to a CSV format file. Click on the “Export Extensions” button and select technology in the prompt below.

Export Extensions
Export Basic Settings

Export Basic Information includes:

  • Extension
  • CallerID Number
  • Privilege
  • SIP Password
  • AuthID
  • Voicemail
  • Voicemail Password
  • Sync Contact
  • First Name
  • Last Name
  • Email Address
  • User/Wave Password

If importing extensions with no values for settings, the following will occur:

  • If importing new extensions, or if Replace is selected as the duplicate import option, the default values for those settings will be used.
  • If Update is selected as the duplicate import option, no changes will be made to the existing settings.

The exported CSV file can serve as a template for users to fill in desired extension information to be imported to the CloudUCM.

Import Extensions

The capability to import extensions to the CloudUCM provides users the flexibility to batch-add extensions with similar or different configurations quickly into the PBX system.

  1. Export the extension CSV file from the CloudUCM by clicking on the “Export Extensions” button.
  2. Fill up the extension information you would like in the exported CSV template.
  3. Click on the “Import Extensions” button. The following dialog will be prompted.
Import Extensions

  1. Select the option in “On Duplicate Extension” to define how the duplicate extension(s) in the imported CSV file should be treated by the PBX.
  • Skip: Duplicate extensions in the CSV file will be skipped. The PBX will keep the current extension information as previously configured without change.
  • Delete and Recreate: The current extension previously configured will be deleted and the duplicate extension in the CSV file will be loaded to the PBX.
  • Update Information: The current extension previously configured in the PBX will be kept. However, if the duplicate extension in the CSV file has a different configuration for any options, it will override the configuration for those options in the extension.
  1. Click on “Choose file to upload” to select a CSV file from a local directory on the PC.
  2. Click on “Apply Changes” to apply the imported file on the CloudUCM.

Example of a file to import:

Import File

FieldSupported Values

Extension

Digits

Technology

SIP/SIP(WebRTC)

Enable Voicemail

yes/no/remote

CallerID Number

Digits

SIP Password

Alphanumeric characters

Voicemail Password

Digits

Skip Voicemail Password Verification

yes/no

Ring Timeout

Empty/ 3 to 600 (in second)

SRTP

yes/no

Skip Trunk Auth

yes/no/bytime

Codec Preference

PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus

Permission

Internal/Local/National/International

DTMF Mode

RFC4733/info/inband/auto

Insecure

Port

Enable Keep-alive

Yes/no

Keep-alive Frequency

Value from 1-3600

AuthID

Alphanumeric value without special characters

TEL URI

Disabled/user=phone/enabled

Call Forward Busy

Digits

Call Forward No Answer

Digits

Call Forward Unconditional

Digits

Support Hot-Desking Mode

Yes/no

Dial Trunk Password

Digits

Disable This Extension

Yes/no

CFU Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFN Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFB Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Music On Hold

Default/ringbacktone_default

CC Agent Policy

If CC is disabled use: never

If CC is set to normal use: generic

If CC is set to trunk use: native

CC Monitor Policy

Generic/never

CCBS Available Timer

3600/4800

CCNR Available Timer

3600/7200

CC Offer Timer

60/120

CC Max Agents

Value from 1-999

CC Max Monitors

Value from 1-999

Ring simultaneously

Yes/no

External Number

Digits

Time Condition for Ring Simultaneously

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Time Condition for Skip Trunk Auth

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Enable LDAP

Yes/no

Enable T.38 UDPTL

Yes/no

Max Contacts

Values from 1-10

Enable Wave

Yes/no

Alert-Info

None/Ring 1/Ring2/Ring3/Ring 4/Ring 5/Ring 6/Ring 7/ Ring 8/Ring 9/Ring 10/bellcore-dr1/bellcore-dr2/ bellcore-dr3/ bellcore-dr4/ bellcore-dr5/custom

Do Not Disturb

Yes/no

DND Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Custom Auto answer

Yes/no

Do Not Disturb Whitelist

Empty/digits

User Password

Alphanumeric characters.

First Name

Alphanumeric without special characters.

Last Name

Alphanumeric without special characters.

Email Address

Email address

Language

Default/en/zh

Phone Number

Digits

Call-Barging Monitor

Extensions allowed to call barging

Seamless Transfer Members

Extensions allowed to seamless transfer

SIP extensions Imported File Example

The CSV file should contain all the above fields, if one of them is missing or empty, the CloudUCM will display the following error message for missing fields.

Import Error

Extension Details

Users can click on an extension number in the Extensions list page and quickly view information about it such as:

  • Extension: This shows the Extension number.
  • Status: This shows the status of the extension.
  • Presence status: Indicates the Presence Status of this extension.
  • Terminal Type: This shows the type of the terminal using this extension
  • Caller ID Name: Reveals the Caller ID Name configured on the extension.
  • Messages: Shows the messages’ stats.
  • IP and Port: The IP address and the ports of the device using the extension.
  • Email status: Show the Email status (sent, to be sent…etc.).
  • Ring Group: Indicates the ring groups that this extension belongs to.
  • Call Queue: Indicates the Cal Queues that this extension belongs to.
  • Call Queue (Dynamic): Indicates the Call Queues that this extension belongs to as a dynamic agent.
Extension Details

E-mail Notification

Once the extensions are created with Email addresses, the PBX administrator can click on the button “E-mail Notification” to send the account registration and configuration information to the user. Please make sure the Email setting under Web GUI🡪System Settings🡪Email Settings is properly configured and tested on the CloudUCM before using “E-mail Notification”.

When clicking on ”More” > “E-mail Notification” button, the following message will be prompted on the web page. Click on OK to confirm sending the account information to all users’ Email addresses.

E-mail Notification – Prompt Information

The user will receive an Email including account registration information as well as the Wave Settings with the QR code:

Wave Settings and QR Code
Important Note

For security and confidentiality reasons, it is highly advisable for the user to change the Wave login extension after the first time log in.

The CloudUCM admin can also send “Extension Information” mail and “Wave Welcome” mail as the figure below shows

Send Email Notification

Multiple Registrations per Extension

CloudUCM supports multiple registrations per extension so that users can use the same extension on devices in different locations.

This feature can be enabled by configuring the option “Concurrent Registrations” under Web GUI🡪Extension/Trunk🡪Edit Extension. The default value is set to 3 registrations. The maximum is 10. When the option “1(allowed to seize) is selected, the UCM will allow newer registration attempts to seize the extension from a previously registered endpoint. To prevent this behavior, please select the option 1.

Extension – Concurrent Registration

SMS Message Support

The CloudUCM provides built-in SIP SMS message support. For SIP end devices such as Grandstream GXP or GXV phones that support SIP messages, after a CloudUCM account is registered on the end device, the user can send and receive SMS messages. Please refer to the end device documentation on how to send and receive SMS messages.

Extension Groups

The CloudUCM extension group feature allows users to assign and categorize extensions in different groups to better manage the configurations on the CloudUCM. For example, when configuring the “Enable Source Caller ID Whitelist”, users could select a group instead of each person’s extension to assign. This feature simplifies the configuration process and helps manage and categorize the extensions for a business environment.

Configure Extension Groups

Extension groups can be configured via Web GUI🡪Extension/Trunk🡪Extension Groups.

  • Click on to create a new extension group.
  • Click on to edit the extension group.
  • Click on to delete the extension group.

Select extensions from the list on the left side to the right side.

Edit Extension Group

Click on to change the ringing priority of the members selected on the group.

Using Extension Groups

Here is an example where the extension group can be used. Go to Web GUI🡪Extension/Trunk🡪Outbound Routes and select “Enable Source Caller ID Whitelist”. Both single extensions and extension groups will show up for users to select.

Select Extension Group in Outbound Route

VoIP Trunks

VoIP Trunk Configuration

VoIP trunks can be configured in CloudUCM under Web GUI🡪Extension/Trunk🡪VoIP Trunks. Once created, the VoIP trunks will be listed with the Provider Name, Type, Hostname/IP, Username, and Options to edit/detect the trunk.

  • Click on “Add SIP Trunk” to add a new VoIP trunk.
  • Click on to configure detailed parameters for the VoIP trunk.
  • Click on to configure Direct Outward Dialing (DOD) for the SIP Trunk.
  • Click on to start LDAP Sync.
  • Click on to delete the VoIP trunk.

The VoIP trunk options are listed in the table below.

Disable This Trunk

Check this box to disable this trunk.

Type

Select the VoIP trunk type.

  • Peer SIP Trunk: A direct IP-to-IP connection between the CloudUCM and another SIP server or device, without requiring registration.

  • Register SIP Trunk: A trunk that requires the CloudUCM to register with the SIP server or provider using credentials (username and password).

  • Account SIP Trunk: A trunk where the CloudUCM acts as the registrar, allowing remote devices or endpoints to register with it.

Provider Name

Configure a unique label (up to 64 characters) to identify this trunk when listed in outbound rules, inbound rules, etc.

Host Name

Configure the IP address or URL for the VoIP provider’s server of the trunk.

Transport

Select the transport protocol to use.

  • UDP: if selected, then the option Enabe UDP should be checked, under PBX Settings => SIP Settings => Transport Protocol

  • TCP: If selected, then the option TCP Enable should be checked under PBX Settings => SIP Settings => Transport Protocol

  • TLS: The default Transport protocol

Server Address

Defines the CloudUCM server address
Note: Please provide the server address to the ITSP supplier for relay service binding. This information is not displayed on the Register SIP Trunk Type.

Keep Original CID

Keep the CID from the inbound call when dialing out. This setting will override the “Keep Trunk CID” option. Please make sure that the peer PBX at the other side supports to match user entry using the “username” field from the authentication line.

Keep Trunk CID

If enabled, the trunk CID will not be overridden by the extension’s CID when the extension has CID configured. The default setting is “No”.

TEL URI

If "Enabled" option is selected, TEL URI and Remove OBP from Route cannot be enabled at the same time. If the phone has an assigned PSTN telephone number, this field should be set to "User=Phone". A "User=Phone" parameter will then be attached to the Request-Line and "TO" header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request.

Need Registration

Defines Whether to register the trunk on the external server. Enabled by default

Allow outgoing calls if registration fails

Disable to block outgoing calls if registration fails. If "Need Registration" option is disabled, this setting will be ignored.
This option is enabled by default.

Caller ID Number

Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored.

Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call:
From the user (Register Trunk Only) 🡪 CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID (Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled) 🡪 Global Outbound CID.

CallerID Name

Configure the new name of the caller when the extension has no CallerID Name configured.

Username

The number or username used for registration and authentication with the service provider.
Note: You can configure this option for "Account SIP Trunk" and "Register SIP Trunk only"

Password

The password used for registration and authentication with the service provider.
Note: You can configure this option for "Account SIP Trunk" and "Register SIP Trunk only"

Auth ID

Enter the Authentication ID for the "Register SIP Trunk" type.

AuthTrunk

If enabled, the UCM will send a 401 response to the incoming call to authenticate the trunk.

Auto Record

If enabled, calls handled with this extension/trunk will automatically be recorded.
Note: the recording functionality is not available on the startup plan.

Direct Callback

Allows external numbers the option to get directed to the extension that last called them.

For Example, User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination.

Domain Connection Mode

If enabled, the following options will be automatically configured: TLS transport, From Domain, Enable Heartbeat Detection and ICE Support. Please ensure that the trunk host name is a GDMS-assigned address and supports TLS.

Monitor Concurrent Calls

If enabled and when the number of concurrent calls exceeds any trunk's configured concurrent call thresholds, an alarm notification will be generated. Note: Please make sure the system alert event "Trunk Concurrent Calls" is enabled.

Concurrent Call Threshold

Threshold of all incoming and outgoing concurrent calls through this trunk.

Outgoing Concurrent Calls Threshold

Threshold of all outgoing concurrent calls passing through this trunk.

Incoming Concurrent Calls Threshold

Threshold of all incoming concurrent calls passing through this trunk.

Total Time Limit For Outbound Calls

Enable Total Time Limit For Outgoing Calls

When this setting is activated, the user can set a time limit before calls cannot be initiated through this trunk

Period

This setting defines how long until the time allowed for outgoing calls is reset.


  • Monthly: The time allowed will reset every month.

  • Quarterly: The time allowed will reset every 3 months.

Example: If the time limit has been set to 4320 minutes, the allowed time will always revert back to 4320 after a month or 3 month based on the period configured.

Total Time

Total time allowed in minutes

Advanced Settings

Codec Preference

Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p and VP8.

Packet Loss Retransmission

Configure to enable Packet Loss Retransmission.

Audio FEC

Configure to enable Forward Error Correction (FEC) for audio.

Video FEC

Configure to enable Forward Error Correction (FEC) for video.

ICE Support

Toggles ICE support. For peer trunks, ICE support will need to be enabled on the other end.

FECC

Configure to enable Far-end Camera Control

Silence Suppression

If enabled, the UCM will send CN packets for silence suppression after a successful CN negotiation in the SIP SDP. If the client endpoint's OPUS codec supports the reception of DTX packets, the UCM will send DTX packets instead.

SRTP

Enable SRTP for the VoIP trunk. The default setting is "No".

SRTP Crypto Suite

SRTP encryption suite used by UCM for outbound calls. Priority is based on order of configuration.

Enable T.38 UDPTL

Enable or disable T.38 UDPTL support.

Include Special Attributes

If enabled, this trunk's SIP SDP will contain ssrc/msid/mid/as/tias/record attributes. These attributes may cause incompatibility when connecting to other devices and services.

Send PPI Header

If checked, the invite message sent to trunks will contain PPI (P-Preferred-Identity) Header.

Send PAI Header

If checked, the INVITE, 18x and 200 SIP messages sent to trunks will contain P-Asserted-Identity (PAI) header. It is not possible to send both PPI and PAI headers. If both Send PAI Header and Passthrough PAI Header are enabled, the following will occur:

  • On incoming calls, the Passthrough PAI Header value will be preferred for this UCM's 18x and 200 SIP messages to the caller.

  • On outbound calls, the Send PAI Header value will be preferred for this UCM's INVITE SIP message to the callee.

Passthrough PAI Header

If enabled and "Send PAI Header" is disabled, PAI headers will be preserved as calls pass through the UCM.

Send PANI Header

If checked, the INVITE sent to the trunk will contain P-Access-Network-Info header.

Send Anonymous

If checked, the "From" header in outgoing INVITE message will be set to anonymous.

DID Mode

Configure to obtain the destination ID of an incoming SIP call from SIP Request-line or To header.

DTMF Mode

Configures the mode for sending DTMF.
RFC4733 (default): DTMF is transmitted as audio in the RTP stream but is encoded separately from the audio stream. Backward-compatible with RFC2833.
Inband: DTMF is transmitted as audio and is included in the audio stream. Requires alaw/ulaw codecs.
Info: DTMF is transmitted separely from the media streams.
RFC4733_info: DTMF is transmitted through both RFC4733 and SIP INFO.
Auto: DTMF mode will be negotiated with the remote peer, only supports RFC4733 and inband. RFC4733 will be used by default unless the remote peer does not indicate support.

Enable Heartbeat Detection

If enabled, the PBX will regularly send SIP OPTIONS to check if the device is online.

Max Outgoing Calls

The number of current outgoing calls over the trunk at the same time. The default value 0 means no limit.

Max Incoming Calls

The max allowed number of concurrent incoming calls through the trunk. Default is 0 (no limit).

Sync LDAP Enable

Automatically sync local LDAP phonebooks to a remote peer (SIP peer trunk only). To ensure successful syncing, the remote peer must also enable this service and set the same password as the local UCM. Port 873 is used by default.

STIR/SHAKEN

Block Spam Calls.

Disabled: Disable STIR/SHAKEN.
Outgoing Attest: Enable STIR/SHAKEN attestation for outgoing calls.
Incoming Verify: Enable STIR/SHAKEN verification for incoming calls.
Both: Enable STIR/SHAKEN for both outgoing and incoming calls.

Enable CC

Check this box to allow the system to automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason.

Trunk Group

Users can create VoIP Trunk Groups to register and easily apply the same settings on multiple accounts within the same SIP server. This can drastically reduce the amount of time needed to manage accounts for the same server and improve the overall cleanliness of the web UI.

Trunk Group

Once creating the new trunk group and configuring the SIP settings, users can add multiple accounts within the configured SIP server by pressing the button and configuring the username, password, and authentication ID fields.

Trunk Group Configuration

Disable This Trunk

Check this box to disable this trunk

Type

Register Trunk

Provider Name

Configure a unique label to identify the trunk when listed in outbound rules and incoming rules.

Host Name 

Enter the IP address or hostname of the VoIP provider's server.

Transport

Configure the SIP Transport method. Only TLS is supported, and TLS service must be enabled on the other end.

Keep Original CID

Keep CID from the inbound call when dialing out even if option "Keep Trunk CID" is enabled. Please make sure the peer PBX at the other end supports matching user entry using the "username" field from the authentication line.

Keep Trunk CID

Always use trunk CID if specified even if extension has DOD number or CID configured.

TEL URI

if "Enabled" option is selected, TEL URI and remove OBP from Route cannot be enabled at the same time. If the phone has an assigned PSTN telephone number, this field should be set to "User=Phone". A "User=Phone" parameter will the be attached to the Request-Line and "TO" header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request.

Need Registration

Whether to register on the external server.

Allow outgoing calls if registration fails

Uncheck to block outgoing cakks if registration fails. If "Need Registration" option is unchecked, this settting will be ignored.

CallerID Name

To display the caller ID name of the trunk, you must configure the caller ID number of the trunk.

Trunk Registration Number

The number used to register with the provider server, and the VoIP provider will authenticate the number based on the trunk registration number.

Line Selection Strategy

  • Linear: Use lines in the list order for outbound calls.

  • Round Robin: Use lines based on rotary line selection for outbound calls. Previously used lines will be remembered.

AuthTrunk

If enabled, the UCM will send a 401 response to the incming call to authenticate the trunk.

Auto Record

If enabled, calls handled with this extension/trunk will automatically be recorded. 

Direct Callback

Allows external numbers the option to get directed to the extension that last called them.

Monitor Concurrent Calls

If enabled, the number of concurrent calls on this trunk will be monitored. If the "Trunk Concurrent Calls" system alert is enabled, alert notifications will be generated if the number of concurrent calls exceeds this trunk's configured concurrent call thresholds.

Concurrent Call Threshold

Threshold of all incoming and outgoing concurrent calls in this trunk.

Outgoing Concurrent Call Threshold

Threshold of all outgoing concurrent calls passing through this trunk.

Incoming Concurrent Call Threshold

Threshold of all incoming concurrent calls passing through this trunk.

Enable Total Time Limit For Outbound Calls

If enabled,  a limit will be placed on the cumulative duration of outbound calls within a specific period. Once this limit has been reached, further outbound calls from this trunk will not be allowed.

Direct Outward Dialing (DOD)

The CloudUCM provides Direct Outward Dialing (DOD), which is a service of a local phone company (or local exchange carrier) that allows subscribers within a company’s PBX system to connect to outside lines directly.

Example of how DOD is used:

Company ABC has a SIP trunk. This SIP trunk has 4 DIDs associated with it. The main number of the office is routed to an auto attendant. The other three numbers are direct lines to specific users of the company. Now when a user makes an outbound call their caller ID shows up as the main office number. This poses a problem, as the CEO would like their calls to come from their direct line. This can be accomplished by configuring DOD for the CEO’s extension.

Steps to configure DOD on the CloudUCM:

  1. To setup DOD go to CloudUCM Web GUI🡪Extension/Trunk🡪VoIP Trunks page.
  2. Click to access the DOD options for the selected SIP Trunk.
  3. Click “Add DOD” to begin your DOD setup
  4. Enter a SIP trunk DID number in the “DOD number” field. In this example, ABC company has a total of 4 DID numbers. Enter the phone number used by the CEO here.
  5. When adding extensions, you can choose whether to “Enable Strip” according to your needs. If it is enabled, you can configure the number (0-64) that will be stripped from the extension number before being added to the DOD number. For example, if the entered digit is 2, and the DOD number for extension 4002 is 1122, then dialing out from 4002, 112202 will be used as the caller ID (DOD).
  6. Select an extension from the “Available Extensions” list. Users have the option of selecting more than one extension. In this case, Company ABC would select the CEO’s extension. After making the selection, click on the button to move the extension(s) to the “Selected Extensions” list.
DOD extension selection
  1. Click “Save” at the bottom.

Once completed, the user will return to the EDIT DOD page which shows all the extensions that are associated with a particular DOD.

Edit DOD

: Add a DOD.
: Import DODs using a csv file.
: Export the DODs using a csv file.
: Filter DODs by number or name.

For DOD importing, please refer to the screenshot below for the template used.

DOD CSV file Template

WebRTC Trunks

WebRTC, Web Real-Time Communication, is a real-time audio/video chatting framework that allows real-time audio/video chatting through the web browser. WebRTC usually does not refer to the web application itself but to the set of protocols and practices bundled with a graphical interface. Our CloudUCM supports creating WebRTC trunks to use exclusively with web applications, this allows the users to join calls and meetings just by clicking a link to a web page.

Below is a figure that shows the options to configure when setting up this feature:

Create WebRTC Trunk

Trunk Name

Create a unique label to easily identify the trunk for inbound route configuration.

Disable This Trunk

Check this box to disable this trunk.

Auto Record

If enabled, calls handled with this extension/trunk will automatically be recorded.

Jitter Buffer

Select jitter buffer method for temporary accounts such as meeting participants who joined via link.

Disable: Jitter buffer will not be used.

Fixed: Jitter buffer with a fixed size (equal to the value of "Jitter Buffer Size")

Adaptive: Jitter buffer with a adaptive size that will not exceed the value of "Max Jitter Buffer").

NetEQ: Dynamic jitter buffer via NetEQ.

Monitor Concurrent Calls

If enabled, the number of concurrent calls on this trunk will be monitored. If the "Trunk Concurrent Calls" system alert is enabled, alert notifications will be generated if the number of concurrent calls exceeds this trunk's configured concurrent call thresholds.

Incoming Concurrent Call Threshold

Threshold of all incoming concurrent calls passing through this trunk.

WebRTC Inbound Link Address

This link can be embedded onto a web page. Clicking the link will connect to a pre-configured WebRTC trunk destination. You can also enter this link in the browser address bar to directly access and test WebRTC calls.

Outbound Routes

In the following sections, we will discuss the steps and parameters used to configure and manage outbound rules in CloudUCM, these rules are the regulating points for all external outgoing calls initiated by the UCM through the SIP trunks.

Configuring Outbound Routes

In the CloudUCM, an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern. This allows different patterns to be dialed through different trunks. Users can also set up a fail-over trunk to be used when the primary trunk fails.

Go to Web GUI🡪Extension/Trunk🡪Outbound Routes to add and edit outbound rules.

  • Click on to add a new outbound route.
  • Click the “Import” button to upload the outgoing route in .CSV format.
  • Click the “Export” button to generate outgoing routes in .CSV format.
  • Click to edit the outbound route.
  • Click to delete the outbound route.

On the CloudUCM, the outbound route priority is based on the “Best matching pattern”. For example, the CloudUCM has outbound route A with pattern 1xxx and outbound route B with pattern 10xx configured. When dialing 1000 for an outbound call, outbound route B will always be used first. This is because pattern 10xx is a better match than pattern 1xxx. Only when there are multiple outbound routes with the same pattern configured.

Outbound Rule Name

Configure the name of the calling rule (e.g., local, long_distance, etc.). Letters, digits, _ and – are allowed.

Pattern

 All patterns are prefixed by the “_” character, but please do not enter more than one “_” at the beginning. All patterns can add comments, such as “_pattern /* comment */”. In patterns, some characters have special meanings:

  • [12345-9] … Any digit in the brackets. In this example, 1,2,3,4,5,6,7,8,9 is allowed.

  • N … Any digit from 2-9.

  • . … Wildcard, matching one or more characters.

  • ! … Wildcard, matching zero or more characters immediately.

  • X … Any digit from 0-9.

  • Z … Any digit from 1-9.

  • - … Hyphen is to connect characters and it will be ignored

  • [] Contain special characters ([x], [n], [z]) represent letters x, n, z.

Disable This Route

After creating the outbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore. However, the route settings will remain in UCM. Users can enable it again when it is needed.

Password

Configure the password for users to use this rule when making outbound calls.

Local Country Code

If your local country code is affected by the outbound blacklist, please enter it here to bypass the blacklist.

Call Duration Limit

Enable to configure the maximum duration for the call using this outbound route.

Maximum Call Duration

Configure the maximum duration of the call (in seconds). The default setting is 0, which means no limit.

Warning Time

Configure the warning time for the call using this outbound route. If set to x seconds, the warning tone will be played to the caller when x seconds are left to end the call.

Auto Record

If enabled, calls using this route will automatically be recorded.

Warning Repeat Interval

Configure the warning repeat interval for the call using this outbound route. If set to X seconds, the warning tone will be played every x seconds after the first warning.

PIN Groups

Select a PIN Group

PIN Groups with Privilege Level

If enabled and PIN Groups are used, Privilege Levels and Filter on Source Caller ID will also be applied.

Privilege Level

Select the privilege level for the outbound rule.

  • Internal: The lowest level required. All users can use this rule.

  • Local: Users with Local, National, or International levels can use this rule.

  • National: Users with National or International levels can use this rule.

  • International: The highest level required. Only users with the international level can use this rule.

  • Disable: The default setting is “Disable”. If selected, only the matched source caller ID will be allowed to use this outbound route.

Please be aware of the potential security risks when using the “Internal” level, which means all users can use this outbound rule to dial out from the trunk.

Enable Filter on Source Caller ID

When enabled, users could specify extensions allowed to use this outbound route. “Privilege Level” is automatically disabled if using “Enable Filter on Source Caller ID”.

The following two methods can be used at the same time to define the extensions as the source caller ID.

  • Select available extensions/extension groups from the list. This allows users to specify arbitrary single extensions available in the PBX.

  • Custom Dynamic Route: define the pattern for the source caller ID. This allows users to define extension range instead of selecting them one by one.

  • All patterns are prefixed with the “_”.

  • Special characters

X: Any Digit from 0-9.
Z: Any Digit from 1-9.
N: Any Digit from 2-9.
“.“: Wildcard. Match one or more characters.
“!“: Wildcard. Match zero or more characters immediately.

Example: [12345–9] – Any digit from 1 to 9.

Note: Multiple patterns can be used. Patterns should be separated by a comma “,”. Example: _X. , _NNXXNXXXXX, _818X.

Outbound Route CID

Attempt to use the configured outbound route CID. This CID will not be used if DOD is configured.

Send This Call Through Trunk

Trunk

Select the trunk for this outbound rule.

Strip

Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk.

Example: The users will dial 9 as the first digit of long-distance calls. In this case, 1 digit should be stripped before the call is placed.

Prepend

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.

Use Failover Trunk

Failover Trunk

Failover trunks can be used to make sure that a call goes through an alternate route when the primary trunk is busy or down. If “Use Failover Trunk” is enabled and “Failover trunk” is defined, the calls that cannot be placed via the regular trunk may have a secondary trunk to go through.

CloudUCM supports up to 10 failover trunks.

Strip

Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk.

Example: The users will dial 9 as the first digit of long-distance calls. In this case, 1 digit should be stripped before the call is placed.

Prepend

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.

Time Condition

Time Condition Mode

Use Main Trunk or Failover Trunk: Use the Main Trunk and its settings during the configured time conditions. If the main trunk is unavailable, the Failover Trunk and its settings will be used instead.

Use Specific Trunks: Use specific trunks during the configured time conditions. The Strip and Prepend settings of the Main Trunk will be used. If a trunk is unavailable during its time condition, no failover trunks will be used.

Failover Trunk Toggles

Inbound Routes

This option controls whether failover trunks will be used if receiving specific responses to outgoing calls.

Failover Trunk Toggles

If a call receives the selected response codes, the UCM will redirect it to the call route’s failover trunk.

Note

Due to the addition of this option, the Enable 486 to Failover Trunks option under PBX Settings > General Settings page has been removed.

Outbound Routes DOD

It is possible to specify the DOD number based on the Outbound Route, as displayed in the screenshot below. For each outbound route.

Outbound Routes Page
DOD Configuration by Outbound Route

Outbound Blacklist

The CloudUCM allows users to configure a blacklist for outbound routes. If the dialing number matches the blacklist numbers or patterns, the outbound call will not be allowed. The outbound blacklist can be configured under UCM Web GUI > Extension/Trunk > Outbound Routes: Outbound Blacklist.

Users can configure numbers, patterns or select country code to add to the blacklist. Please note that the blacklist settings apply to all outbound routes.

Country Codes

Users can export outbound route blacklists and delete all blacklist entries. Additionally, users can also import blacklists for outbound routes.

Blacklist Import/Export

PIN Groups

The CloudUCM supports the pin group. Once this feature is configured, users can apply pin groups to specific outbound routes. When placing a call on pin-protected outbound routes, the caller will be asked to input the group PIN, this feature can be found on the Web GUI > Extension/Trunk > Outbound Routes > PIN Groups.

Name

Specify the name of the group

Record In CDR

Specify whether to enable/disable the record in CDR

PIN Number

Specify the code that will be asked once dialing via a trunk

PIN Name

Specify the name of the PIN

Outbound Routes/PIN Group

Once the user clicks , the following figure shows to configure the new PIN.

Create a New PIN Group

The following screenshot shows an example of created PIN Groups and members:

PIN Members

If the PIN group is enabled on the outbound route level, the password, privilege level and enable the filter on source caller ID will be disabled, unless you check the option “PIN Groups with Privilege Level” where you can use the PIN Groups and Privilege Level or PIN Groups and Enable Filter on Source Caller ID.

Outbound PIN

If PIN group CDR is enabled, the call with PIN group information will be displayed as part of CDR under the Account Code field.

CDR Record

  • Importing PIN Groups from CSV files:

Users can also import PIN Groups by uploading CSV files for each group. To do this:

  1. Navigate to Extension/Trunk🡪Outbound Routes🡪PIN Groups and click on the “Upload” button.
Importing PIN Groups from CSV files
  1. Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in error messages such as the one below:
Screenshot (5)
Incorrect CSV File
  1. To ensure a successful import, please follow the format in the sample image below
pingroup_format
CSV File Format
  • The top-left value (A1) is the PIN Group name. In this case, it is “ALPHA”.
  • Row 2 contains the labels for the modifiable fields: pin and pin_name. These values should not be changed and will cause an upload error otherwise.
  • Rows 3+ contain the user-defined values with Column A holding the PINs and Column B holding the PIN names. PIN values must consist of at least four digits.
  • Once the file is successfully uploaded, the entry will be added to the list of PIN Groups.
CSV File Successful Upload

Inbound Routes

Inbound routes can be configured via Web GUI🡪Extension/Trunk🡪Inbound Routes.

  • Click on to add a new inbound route.
  • Click on “Blacklist” to configure the blacklist for all inbound routes.
  • Click on to edit the inbound route.
  • Click on to delete the inbound route.

Inbound Route Configuration

Trunks

Select the trunk to configure the inbound rule.

Inbound Route Name

Configure the name of the Inbound Route. For example, “Local”, “LongDistance” etc.

Pattern

All patterns are prefixed with the “_”.

Special characters:

X: Any Digit from 0-9. Z: Any Digit from 1-9. N: Any Digit from 2-9. “.“: Wildcard. Match one or more characters. “!“: Wildcard. Match zero or more characters immediately. Example: [12345-9] – Any digit from 1 to 9.

Notes:
Multiple patterns can be used. Each pattern should be entered in a new line.

Example:
_X.
_ NNXXNXXXXX /* 10-digit long distance */

Disable This Route

After creating the inbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore. However, the route settings will remain in UCM. Users can enable it again when it is needed.

CID Source

Configures the source of the CID to match with the configured CallerID Pattern.
None: CID is not obtained from any source. Only applicable if no CallerID Pattern is configured.
DiversionID: CID is obtained from the Diversion header. Only applicable to SIP trunks.
CallerID: If the call is from a SIP trunk, the CID is obtained from the From header. Otherwise, the CID will be obtained from other related signaling.

Seamless Transfer Whitelist

Allows the selected extension to use this function. If an extension is busy, and a mobile phone is bound to that extension, the mobile phone can pick up calls to that extension.

Ringback tone

Choose the custom ringback tone to play when the caller reaches the route.

Auto Record

If enabled, calls using this route will automatically be recorded.

Block Collect Call

If enabled, collect calls will be blocked.

Note: Collect calls are indicated by the header “P-Asserted-Service-Info: service-code=Backward Collect Call, P-Asserted-Service-Info: service-code=Collect Call”.

Alert-Info

Configure the Alert-Info, when UCM receives an INVITE request, the Alert-Info header field specifies an alternative ring tone to the UAS.

Fax Detection

If enabled, fax signals from the trunk during a call will be detected.

Fax Destination

Configures the destination of faxes.

  • Extension: send the fax to the designated FAX extension.

  • Fax to Email: send the fax as an email attachment to the designated extension’s email address. If the selected extension does not have an associated email address, it will be sent to the default email address configured in the Call Features->Fax/T.38->Fax Settings page.

Note: please make sure the sending email address is correctly configured in System Settings->Email Settings.

Auto Answer

If enabled, the UCM will automatically answer calls and receive faxes through the inbound route. If disabled, the UCM will not receive a fax until after the call has been answered. Enabling this option will slow down the answering of non-fax calls on the inbound route. The alert tone heard during the detection period can be customized.

Block Collect Calls

If enabled, collect calls will be blocked.
Note: Collect calls are indicated by the header "P-Asserted-Service-Info: service-code=Backward Collect Call, P-Asserted-Service-Info: service-code=Collect Call".
Note: This is affected by Block Set Calls on the SIP Settings -> General Settings page.

Prepend Trunk Name

If enabled, the trunk name will be added to the caller id name as the displayed caller id name.

Set Caller ID Info

Manipulates Caller ID (CID) name and/or number within the call flow to help identify who is calling. When enabled two fields will show allowing to manipulate the CalleID Number and the Caller ID Name.

CallerID Number

Configure the pattern-matching format to manipulate the numbers of incoming callers or to set a fixed CallerID number for calls that go through this inbound route.

  • ${CALLERID(num)}: Default value which indicates the number of an incoming caller (CID). The CID will not be modified.

  • ${CALLERID(num):n}: Skips the first n characters of a CID number, where n is a number.

  • ${CALLERID(num):-n}: Takes the last n characters of a CID number, where n is a number.

  • ${CALLERID(num):s:n}: Takes n characters of a CID number starting from s+1, where n is a number and s is a character position (e.g. ${CALLERID(num):2:7} takes 7 characters after the second character of a CID number).

  • n${CALLERID(num)}: Prepends n to a CID number, where n is a number.

CallerID Name

The default string is ${CALLERID(name)},which means the name of an incoming caller, it is a pattern-matching syntax format.

A${CALLERID(name)}B means Prepend a character ‘A’ and suffix a character ‘B’ to ${CALLERID(name)}.

Not using pattern-matching syntax means setting a fixed name to the incoming caller.

Enable Route-Level Inbound Mode

Gives uses the ability to configure inbound mode per individual route. When enabled two fields will show allowing to set the Inbound mode and the Inbound mode Suffix.

Note: Global inbound mode must be enabled before users can configure route-level inbound mode.

Inbound Mode

Choose the inbound mode for this route.

Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If all routes have the option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be affected by the global inbound mode change.

Inbound Mode Suffix

Dial “Global Inbound Mode feature code + Inbound Mode Suffix” or a route’s assigned suffix to toggle the route’s inbound mode.

The BLF subscribed to the inbound mode suffix can monitor the current inbound mode.

Inbound Multiple Mode

Multiple mode allows users to switch between destinations of the inbound rule by feature codes. Configure related feature codes as described in [Inbound Route: Multiple Mode]. If this option is enabled, the user can use feature code to switch between different modes/destinations.

CallerID Name Lookup

If enabled, the callerID will be resolved to a name through local LDAP. Note, if a matched name is found, the original callerID name will be replaced. The name lookup is performed before other callerID or callerID name modifiers (e.g., Inbound Route's Set CallerID Info or Prepend Trunk Name). Note: Name lookup may impact system performance.

Dial Trunk

This option shows up only when “By DID” is selected. If enabled, the external users dialing into the trunk via this inbound route can dial outbound calls using the UCM’s trunk.

Privilege Level

This option shows up only when “By DID” is selected.

  • Disable: Only the selected Extensions or Extension Groups are allowed to use this rule when enabled Filter on Source Caller ID.

  • Internal: The lowest level required. All users are allowed to use this rule, checking this level might be risky for security purposes.

  • Local: Users with Local level, National or International level are allowed to use this rule.

  • National: Users with National or International Level are allowed to use this rule.

  • International: The highest level required. Only users with an international level are allowed to use this rule.

Allowed DID Destination

This option shows up only when “By DID” is selected. This controls the destination that can be reached by the external caller via the inbound route. The DID destination is:

  • Extension

  • Conference

  • Call Queue

  • Ring Group

  • Paging/Intercom Groups

  • IVR

  • Voicemail Groups

  • Dial By Name

  • All

Default Destination

Select the default destination for the inbound call. 

  • Extension

  • Voicemail

  • Conference Room

  • Call Queue

  • Ring Group

  • Paging/Intercom

  • Voicemail Group

  • DISA

  • IVR

  • External Number

  • By DID

When “By DID” is used, the UCM will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR, and voicemail groups as configured in “DID destination”. If the dialed number matches the DID pattern, the call will be allowed to go through.

  • Dial By Name

  • Callback

Strip

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.

Prepend

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.

Time Condition

Start Time

Select the start time “hour:minute” for the trunk to use the inbound rule.

End Time

Select the end time “hour:minute” for the trunk to use the inbound rule.

Date

Select “By Week” or “By Day” and specify the date for the trunk to use the inbound rule.

Week

Select the day in the week to use the inbound rule.

Destination

Select the destination for the inbound call under the defined time condition.

  • Extension

  • Voicemail

  • Conference Room

  • Call Queue

  • Ring Group

  • Paging/Intercom

  • Voicemail Group

  • DISA

  • IVR

  • By DID

When “By DID” is used, the UCM will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR, and voicemail groups as configured in “DID destination”. If the dialed number matches the DID pattern, the call will be allowed to go through.

Configure the number of digits to be stripped in the “Strip” option.

  • Dial By Name

  • External Number

  • Callback

Inbound Route: Prepend Example

CloudUCM allows users to prepend digits to an inbound DID pattern, with strip taking precedence over prepend. With the ability to prepend digits in the inbound route DID pattern, the user no longer needs to create multiple routes for the same trunk to route calls to different extensions. The following example demonstrates the process:

  1. If Trunk provides a DID pattern of 18005251163.
  2. If Strip is set to 8, UCM will strip the first 8 digits.
  3. If Prepend is set to 2, UCM will then prepend a 2 to the stripped number, now the number becomes 2163.
  4. The UCM will forward the incoming call to extension 2163.
Inbound Route feature: Prepend

Inbound Route: Multiple Mode

In the UCM, the user can configure an inbound route to enable multiple mode to switch between different destinations. The inbound multiple mode can be enabled under Inbound Route settings.

Inbound Route – Multiple Mode

When Multiple Mode is enabled for the inbound route, the user can configure a “Default Destination” and a “Mode 1” destination for all routes. By default, the call coming into the inbound routes will be routed to the default destination.

SIP end devices that have registered on the UCM can dial feature code *62 to switch to the inbound route “Mode 1” and dial feature code *61 to switch back to “Default Destination”. Switching between different modes can be easily done without a Web GUI login.

For example, the customer service hotline destination has to be set to a different IVR after 7 PM. The user can dial *62 to switch to “Mode 1” with that IVR set as the destination before off work.

To customize feature codes for “Default Mode” and “Mode 1”, click on under the “Inbound Routes” page, check the “Enable Inbound Multiple Mode” option, and change “Inbound Default Mode” and “Inbound Mode 1” values (By default, *61 and *62 respectively).

Inbound Route – Multiple Mode Feature Codes

Inbound Route: Route-Level Mode

In the UCM, users can enable Route-Level Inbound Mode to switch between different destinations for each inbound route. The inbound Route-Level mode can be enabled under Inbound Route settings.

Inbound Route – Route-Level Mode

The global inbound mode must be enabled before configuring Route-Level Inbound Mode. Additionally, Mode 1 must be configured as well.

When Route-Level Inbound Mode is enabled, the user can configure a “Default Destination” and a “Mode 1” destination for each specific route. By default, the call coming into this specific inbound route will be routed to the default destination.

Users can toggle the route’s inbound mode by dialing “Global Inbound Mode feature code + Inbound Mode Suffix” and the current inbound route can be monitored by subscribing a BLF to the Inbound Mode Suffix.

For example, the Inbound Default Mode feature code is set to *61 and the Inbound Mode suffix for route 1 is set to 1010. To switch the mode of route 1 to Default Mode, users can dial *611010.

Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If all routes have the option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be affected by the global inbound mode change.

Inbound Route: Inbound Mode BLF Monitoring

Users can assign MPKs and VPKs to monitor and toggle the current global inbound mode of the UCM.

To do this, please refer to the following steps:

  1. Access the UCM web GUI and navigate to Extension/Trunk🡪Inbound Routes.
  2. Click on the button and enable Inbound Multiple Mode.
  3. Edit the subscribe number field to the desired BLF value.
Global Inbound Mode

  1. Configure the BLF value on a phone’s MPK/VPK. As an example, a GXP2140 with the BLF configured will show the Inbound Mode status on its screen once configured. The 777 BLF is lit green, indicating that the current inbound mode is “Default Mode”.
Inbound Mode – Default Mode
  1. Pressing the key will toggle the inbound mode to “Mode 1”, and the button’s color will change to red.
Inbound Mode – Mode 1

Inbound Route: Import/Export Inbound Route

Users can now import and export inbound routes to quickly set up inbound routing on a UCM or to back up an existing configuration. An exported inbound route configuration can be directly imported without needing any manual modifications.

Import/Export Inbound Route

The imported file should be in CSV format and using UTF-8 encoding, the imported file should contain the below columns, and each column should be separated by a comma (It is recommended to use Notepad++ for the imported file creation):

  • Disable This Route: Yes/No.
  • Pattern: Always prefixed with _
  • CallerID Pattern: Always prefixed with _
  • Prepend Trunk Name: Yes/No.
  • Prepend User Defined Name Enable: Yes/No.
  • Prepend User Defined Name: A string.
  • Alert-info: None, Ring 1, Ring 2… The user should enter an Alert-info string following the values we have in the Inbound route Alert-Info list.
  • Allowed to seamless transfer: [Extension_number]
  • Inbound Multiple Mode: Yes/No.
  • Default Destination: By DID, Extension, Voicemail… Users should enter a Default Destination string following the values we have in the Inbound route Default Destination list.
  • Destination: An Extension number, Ring Group Extension…
  • Default Time Condition.
  • Mode 1: By DID, Extension, Voicemail… Users should enter a Default Destination string following the values we have in the mode 1 Default Destination list.
  • Mode 1 Destination: An Extension number, Ring Group Extension…
  • Mode 1 Time Condition.

Blacklist Configurations

In the UCM, Blacklist is supported for all inbound routes. Users could enable the Blacklist feature and manage the Blacklist by clicking on “Blacklist”.

  • Select the checkbox for “Blacklist Enable” to turn on the Blacklist feature for all inbound routes. The blacklist is disabled by default.
  • Enter a number in the “Add Blacklist Number” field and then click ”Add” to add to the list. Anonymous can also be added as a Blacklist Number by typing “Anonymous” in Add Blacklist Number field.
  • To remove a number from the Blacklist, select the number in the “Blacklist list” and click on or click on the” Clear” button to remove all the numbers on the blacklist.
  • Users can also export the inbound route blacklist by pressing the button.
Blacklist Configuration Parameters
  • To add blacklisted numbers in batch, click on “Import” to upload the blacklist file in CSV format. The supported CSV format is as below.
Blacklist CSV File

Users could also add a number to the Blacklist or remove a number from the Blacklist by dialing the feature code for “Blacklist Add’ (default: *40) and “Blacklist Remove” (default: *41) from an extension. The feature code can be configured under Web GUI > Call Features > Feature Codes.

CALL FEATURES

Multimedia Meeting

The UCM supports multimedia meeting room allowing multiple rooms used at the same time.

The multimedia meeting room configurations can be accessed under Web GUI🡪Call Features🡪 Multimedia Meeting. On this page, users can create, edit, view, invite, manage the participants, and delete multimedia meeting rooms. The multimedia meeting room status and meeting call recordings (if recording is enabled) will be displayed on this web page as well.

For video meeting, which is based on WebRTC, participants can join the meeting from a PC without installing extra plug-ins or software.

The UCM admin can create multiple multimedia meeting rooms for users to dial in.

Meeting room specifications affect user participation to a certain extent. UCM supports the forecasting of meeting resources. There will be corresponding judgments and adjustments in the following scenarios:

  1. When meeting resources are used up, scheduled meeting members cannot join the meeting in advance.
  2. When a point-to-point call is transferred to a conference, the conference resources are used up.
  3. When meeting resources are used up, do not join a group IM chat when you initiate a meeting.
  4. When meeting resources are used up, do not join an instant meeting.
  5. Close other instant meetings or scheduled meetings that have timed out to ensure that invited members can join the scheduled meeting.
  6. In an ongoing meeting, if the number of invited members exceeds the upper limit, members cannot be invited to join the meeting.
  7. Enable flow control for videos and presentations in the conference room.
Notes

The multimedia meeting room supports up to 4 video calls and one video presentation.

  • The administrator can set the number of videos to 9 parties. The increase in the number of videos will take up more system resources and affect the overall performance of the UCM system. Please set it according to your needs.
  • During a meeting, when the system detects that another scheduled meeting is about to be held, it will remind the meeting members that the subsequent meeting room has been reserved, please end the meeting in advance.
  • The use of video in the meeting will take up system resources and may cause performance problems when used.
  • The maximum meeting duration is 12 hours. If it exceeds 12 hours, the system will remind the current meeting and the host can continue to extend the meeting.

Multimedia Room Configurations

  • Click on “Add” to add a new meeting room.
  • Click on to edit the meeting room.
  • Click on to delete the meeting room.

Meeting Settings contains the following options:

Extension

The number to dial to reach the meeting room.

Meeting Name

Meeting Name

Privilege

Please select the permission for outgoing calls.

Allow User Invite

If enabled, participants will be able to invite other to the meeting by pressing 1 on their keypad or by clicking the Participants -> Invite option on the Wave bottom bar.

Allowed to Override Most Mute

Allowed to Override Host Mute

Auto Record

Meeting audio and video can be automatically recorded. These reconrdings can be found under the Meeting Recording or Meeting Video Recordings Page.

  • None: Auto record is disabled.

  • Record Audio: Record only the meeting Audio.

  • Record video (Focus Mode): Record the focus screen and all audio of the meeting. When a shared source is present in the meeting, only the shared screen is recorded.

Room Password

If meeting room password is configured, meeting participants will need to enter a password to enter the room. Scheduling meetings will not be supported for this room.

Log in to the UCM Web GUI and open the Call Features > Multimedia Meeting page to manage the conference room. Users can create, edit, view, invite, manage meeting members, and delete meeting rooms. The conference room status and conference call recording (if the recording function is enabled) will be displayed on the page. The meeting rooms in the list include public meeting rooms and random meeting rooms. For temporary meeting room administrators, only the “batch kicking people” function is supported. The temporary meeting room has no meeting password or host code. The member who initiates the group meeting is the host, and ordinary members have the right to invite.

Multimedia Meeting

Meetings Settings

To edit the general settings of the meeting rooms created in the UCM, the user can click on “Meetings Settings” button under the Room tab.

Meeting Max Concurrent Audio

Maximum number of partipants that can be heard simultaenously in multimedia meetings. If the number of participants talking at any given point exceeds this value, the audio of the excess participants will not be heard.

Meeting Voice Indicator Sensitivity

Configures the sensitivity of the talking indicator in multimedia meetings. Setting this higher will make the talking indicator appear more easily for lower volumes of audio. Note: This does not adjust audio input sensitivity itself. Lower volumes of sounds may still be heard even if the talking indicator does not show the source.

Meeting Audio Quality

Audio quality of multimedia meetings

Meeting Record Prompt

If enabled, system will prompt the user before the start of meeting recording that your meeting will be recorded.

Allow New Participants To View Chat History

Configure whether new attendees joining in the middle of a Wave meeting can view the chat content already in the meeting.

Meeting AGC (beta)

Enabling this option will toggle on Automatic Gain Control for meeting audio. AGC is a system that dynamically reduces the variability of sound levels by adjusting high and low volumes based on the average or peak sound level. High volume sounds will be lowered, and low volume sounds will be boosted.

Silence Suppression

Silence suppression for temporary accounts (e.g., meeting participants that joined the meeting via link). If enabled, the UCM will send CN packets for silence suppression after a successful CN negotiation in the SIP SDP. If the client endpoint's OPUS codec supports the reception of DTX packets, the UCM will send DTX packets instead.

Enable Talk Detection

If enabled, the AMI will send the corresponding event when a user starts or stops talking.

DSP Talking Threshold (ms)

The amount of time(ms) that sound exceeds what the DSP has established as the baseline for silence before a user is considered to be talking. This value affects several operations and should not be changed unless the impact on call quality is fully understood.

DSP Silence Threshold (ms)

The amount of time(ms) that sound falls within what the DSP has established as the baseline for silence before a user is considered be silent. This value affects several operations and should not be changed unless the impact on call quality is fully understood.

Max Number of Video Feeds

Set the maximum number of video feeds supported per meeting room.

Audio Codec Preference

Configures the preferred codecs for temporary accounts such as meeting participants who joined via link.

Packet Loss Retransmission

Packet Loss Retransmission configuration for temporary accounts (meeting participants without registered extensions who entered the meeting via link).

Jitter Buffer

Select the jitter buffer method for temporary accounts such as meeting participants who joined via link.

  • Disabled: Jitter buffer will not be used.

  • Fixed: Jitter buffer with a fixed size (equal to the value of "Jitter Buffer Size")

  • Adaptive: Jitter buffer with an adaptive size that will not exceed the value of "Max Jitter Buffer").

  • NetEQ: Dynamic jitter buffer via NetEQ.

Multimedia Meeting Call Operations

Join a Meeting Call

Users could dial the meeting room extension to join the meeting. If the password is required, enter the password to join the meeting as a normal user, or enter the admin password to join the meeting as an administrator.

Invite Other Parties to Join a Meeting

When using the UCM meeting room., there are two ways to invite other parties to join the meeting.

  • Invite from Web GUI.

For each meeting room in CloudUCM Web GUI🡪Call Features🡪 Multimedia Meeting, there is an icon for option “Invite a participant”. Click on it and enter the number of the party you would like to invite. Then click on “Add”. A call will be sent to this number to join the conference. 

Meeting Invitation from Web GUI
  • Invite by dialing 0 or 1 during a conference call.

A meeting participant can invite other parties to the meeting by dialing from the phone during the meeting call. Please make sure the option “Enable User Invite” is turned on for the meeting room first. Enter 0 or 1 during the meeting call. Follow the voice prompt to input the number of the party you would like to invite. A call will be sent to this number to join the meeting.

0: If 0 is entered to invite another party, once the invited party picks up the invitation call, permission will be asked to “accept” or “reject” the invitation before joining the conference.

1: If 1 is entered to invite another party, no permission will be required from the invited party.

Conference administrators can always invite other parties from the phone during the call by entering 0 or 1. To join a conference room as an administrator, enter the admin password when joining the conference. A conference room can have multiple administrators.

During The Meeting

During the meeting call, users can manage the conference from Web GUI or IVR.

  • Manage the meeting call from Web GUI.

Log in UCM Web GUI during the meeting call, and the participants in each meeting room will be listed.

  1. Click on to kick a participant from the meeting.
  2. Click on to mute the participant.
  3. Click on to lock this meeting room so that other users cannot join it anymore.
  4. Click on to invite other users into the meeting room.
  5. Click on to Invite meeting rooms or Invite contact groups.
  • Manage the meeting call from IVR.

Please see the options listed in the table below.

Meeting Administrator IVR Menu

1

Mute/unmute yourself.

2

Lock/unlock the conference room.

3

Kick the last joined user from the conference.

4

Decrease the volume of the conference call.

5

Decrease your volume.

6

Increase the volume of the conference call.

7

Increase your volume.

8

More options.

  • 1: List all users currently in the conference call.
  • 2: Kick all non-administrator participants from the conference call.
  • 3: Mute/Unmute all non-administrator participants from the conference call.
  • 4: Record the conference call.
  • 8: Exit the caller menu and return to the conference.

Meeting User IVR Menu

1

Mute/unmute yourself.

4

Decrease the volume of the conference call.

5

Decrease your volume.

6

Increase the volume of the conference call.

7

Increase your volume.

8

Exit the caller menu and return to the conference.

Meeting Caller IVR Menu

When there is a participant in the meeting, the meeting room configuration cannot be modified.

Google Service Settings Support

CloudUCM supports Google OAuth 2.0 authentication. This feature is used for supporting the CloudUCM meeting scheduling system. Once OAuth 2.0 is enabled, the CloudUCM conference system can access Google Calendar to schedule or update conference.

Google Service Settings can be found under Web GUI🡪Call Features🡪 Multimedia Meeting 🡪Google Service Settings🡪Google Service Settings.

Google Service Settings🡪OAuth2.0 Authentication

If you already have an OAuth2.0 project set up on the Google Developers web page, please use your existing login credentials for “OAuth2.0 Client ID” and “OAuth2.0 Client Secret” in the above figure for the CloudUCM to access Google Service.

If you do not have the OAuth2.0 project set up yet, please follow the steps below to create a new project and obtain credentials:

  1. Go to the Google Developers page https://console.developers.google.com/start Create a New Project on the Google Developers page.
Google Service🡪New Project
  1. Enable Calendar API from API Library.
  2. Click “Credentials” on the left drop-down menu to create new OAuth2.0 login credentials.
Google Service🡪Create New Credential

  1. Use the newly created login credential to fill in “OAuth2.0 Client ID” and “OAuth2.0 Client Secret”.
  2. Click “Get Authentication Code” to obtain an authentication code from Google Service.
Google Service🡪OAuth2.0 Login

  1. Once this has been done, the CloudUCM will connect to Google services.

You can also configure the Status update, which automatically refreshes your Google Calendar with the configured time (m). Note: Zero means disable.

Meeting Schedule

Log in to the UCM Web GUI, open the Call Features 🡪 Multimedia Meeting 🡪 Meeting Schedule page, and you can manage the reservation management of the meeting room. Users can create, edit, view, and delete conference room reservation records. The following is a set meeting room reservation, which shows the ongoing and pending reservations. Once the conference room is reserved, all users will be removed from the conference room at the start time, and extensions will no longer be allowed to enter the conference room. At the scheduled meeting time, UCM will send invitations to the extensions that have been selected to participate in the meeting. At the same time, it supports users to enter the meeting 10 minutes in advance. If the current meeting is occupied, enter the waiting room and wait (members joining the meeting in advance occupy global member resources, but it will be released after the scheduled meeting starts); otherwise, you can join the meeting directly and the meeting will be held in advance. After the meeting ends, the reservation record is transferred to the historical meeting list. History meeting displays the information of the ended and expired meetings.

  • Click the button “Schedule Meeting” to edit the meeting room reservation.
Schedule meeting Interface

Schedule Options

Meeting Subject

Configure the name of the scheduled meeting. Letters, digits, Other special characters are also supported. such as #%&@*=

Meeting Room

Choose which room to have this scheduled meeting.
If this option has been enabled, please select an existing room for this meeting. If this option has not been enabled, a new meeting room will be created.

Time

Configure the meeting date and time.

Time Zone

Select the meeting time zone.

Password

Configure the meeting's login password.

Host Password

Configure the Host Password.

Note: It is randomly generated when first creating a new meeting Schedule.

Host

Configure Host.

Repeat

Choose when to repeat a scheduled meeting.

Allow User Invite

If this option is enabled, the user can:

  • Press ‘0’ to invite others to join the meeting with invited party’s permission

  • Press ‘1’ to invite without invited party’s permission

  • Press ‘2’ to create a multi-meeting room to another meeting room

  • Press ‘3’ to drop all current multi-meeting rooms.

Note:

Meeting host is always allowed to access this menu.

Call Participant

If enabled, the invited participants will be called upon meeting start time.

Allowed to Override Host Mute

If enabled, participants will be able to unmute themselves if they have been muted by the host.

Email Reminder (m)

Email reminders will be sent out x minutes prior to the start of the meeting. Valid range is 5-1440. 60 is the default value. 0 indicates not to send out email reminders for the meeting.

Note: After editing the time of a single recurrence of a scheduled meeting, a cancelation email will now be sent out followed by a meeting update email.

Auto Record

If selected, the meeting will be recorded and saved as either a .WAV or .MKV file. The default filename is meeting-${Meeting Number}-${UNIQUEID}. Recordings can be downloaded from either the Meeting Recordings or the Meeting Video Recordings page. Video recordings require external storage to be available. When recording a screen share, only the screen share and meeting audio will be recorded.

Enable Google Calendar

Select this option to sync scheduled meeting with Google Calendar.

Note: Google Service Setting OAuth2.0 must be configured on the CloudUCM. Please refer to Google Services configuration section.

Meeting Agenda

Enter information about the meeting, e.g., the purpose of the meeting or the subjects that will be discussed in the meeting.

Invitees

Local extensions, remote extensions, and special extensions are supported.

Once the Meeting Schedule is configured, the scheduled meeting will be displayed as the below figure.

Meetings Schedule
  • Click the button to view the meeting details in the Meeting room. The meeting details of Meeting History include actual participant information.
Meeting details
  • Click on to edit the Meeting Schedule.
  • Click on to delete the Meeting Schedule.

At the scheduled meeting time, CloudUCM will send INVITE to the extensions that have been selected for the conference.

Once the meeting starts, it will be displayed under Pending Meeting with an “Ongoing” status, as displayed below:

Meeting Scheduled – Ongoing

Once the conference is finished, the conference will be displayed under Historical Meeting as below:

Meeting Schedule – Completed
  • Click the button to download the Meeting Report of the meeting.
  • Click the button to reschedule the Meeting.

In addition, once the meeting ends, the system will send a meeting report email to the host including a PDF file where he/she can view the meeting, participant information, device type, and trend graph of participant levels.

You can also choose to display the meetings that took place in a specific time frame. Please see the screenshot below:

Please make sure that the outbound route is properly configured for remote extensions to join the meeting.

Meeting Recordings

The CloudUCM allows users to record the audio of the meeting call and retrieve the recording from Web GUI🡪Call Features🡪 Multimedia Meeting🡪 Meeting Recordings.

To record the Meeting call, when the meeting room is idle, enable “Auto Record” from the meeting room configuration dialog. Save the setting and apply the change. When the meeting call starts, the call will be automatically recorded in .wav format.

The recording files will be listed below once available. Users could click on to download the recording or click on to delete the recording. Users could also delete all recording files by clicking on “Delete All Recording Files” or delete multiple recording files at once by clicking on “Delete” after selecting the recording files.

Meeting Recordings

Meeting Video Recordings

The CloudUCM allows users to record the audio and video of the meeting call and retrieve the recording from Web GUI🡪Call Features🡪 Multimedia Meeting🡪 Meeting Recordings.

To record the Meeting call, when the meeting room is idle, enable “Auto Record” from the meeting room configuration dialog. Save the setting and apply the change. When the meeting call starts, the call will be automatically recorded in .mkv format.

The recording files will be listed below once available. Users could click on to download the recording or click on to delete the recording. Users could also delete all recording files by clicking on “Delete All Recording Files” or delete multiple recording files at once by clicking on “Delete” after selecting the recording files.

Call Statistics

Meeting reports will now be generated after every conference. These reports can be exported to a .CSV file for offline viewing. The conference report page can be accessed by clicking on the Call Statistics button on the main Conference page.

Meeting Call Statistics
Meeting Report on Web
Meeting Report on CSV

IVR

Configure IVR

IVR configurations can be accessed under the CloudUCM Web GUI🡪Call Features🡪IVR. Users could create, edit, view, and delete an IVR.

  • Click on “Add” to add a new IVR.
  • Click on to edit the IVR configuration.
  • Click on to delete the IVR.
Create New IVR

Basic Settings

Name

Configure the name of the IVR. Letters, digits, _ and – are allowed.

Extension

Enter the extension number for users to access the IVR.

Dial Trunk

If enabled, all callers to the IVR can use the trunk. The permission must be configured for the users to use the trunk first. The default setting is “No”.

Auto Record

If enabled, calls to this IVR will automatically be recorded.

Permission

Assign permission level for outbound calls if “Dial Trunk” is enabled. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level.

The default setting is “Internal”. If the user tries to dial outbound calls after dialing into the IVR, the CloudUCM will compare the IVR’s permission level with the outbound route’s privilege level.

If the IVR’s permission level is higher than (or equal to) the outbound route’s privilege level, the call will be allowed to go through.

Dial Other Extensions

This controls the destination that can be reached by the external caller via the inbound route. The available destinations are:

  • Extension
  • Conference
  • Call Queue
  • Ring Group
  • Paging/Intercom Groups
  • Voicemail Groups
  • Dial by Name
  • All

IVR Black/Whitelist

If enabled only numbers inside of the Whitelist or outside of the Blacklist can be called from IVR.

Internal Black/Whitelist

Contain numbers, either of Blacklist or Whitelist.

External Black/Whitelist

This feature can be used only when Dial Trunk is enabled, it contains external numbers allowed or denied calling from the IVR, the allowed format is the following: Number1, number2, number3…

Replace Display Name

If enabled, the CloudUCM will replace the caller display name with the IVR name.

Return to IVR Menu

If enabled and if a call to an extension fails, the caller will be redirected to the IVR menu.

Alert Info

When present in an INVITE request, the alert-Info header field specifies an alternative ring tone to the UAS.

Prompt

Select an audio file to play as the welcome prompt for the IVR. Click on “Prompt” to add audio file under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt.

Digit Timeout

Configure the timeout between digit entries. After the user enters a digit, the user needs to enter the next digit within the timeout. If no digit is detected within the timeout, the CloudUCM will consider the entries complete. The default timeout is 3s.

Response Timeout

After playing the prompts in the IVR, the CloudUCM will wait for the DTMF entry within the timeout (in seconds). If no DTMF entry is detected within the timeout, a timeout prompt will be played. The default setting is 10 seconds.

Response Timeout Prompt

Select the prompt message to be played when the timeout occurs.

Invalid Input Prompt

Select the prompt message to be played when an invalid extension is pressed.

Response Timeout Prompt Repeats

Configure the number of times to repeat the prompt if no DTMF input is detected. When the loop ends, it will go to the timeout destination if configured, or hang up. The default setting is 3.

Invalid Input Prompt Repeats

Configure the number of times to repeat the prompt if the DTMF input is invalid. When the loop ends, it will go to the invalid destination if configured, or hang up. The default setting is 3.

Language

Select the voice prompt language to be used for this IVR. The default setting is “Default” which is the selected voice prompt language under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. The dropdown list shows all the currently available voice prompt languages on the CloudUCM. To add more languages in the list, please download the voice prompt package by selecting “Check Prompt List” under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Key Pressing Events

Key Press Event:

Press 0

Press 1

Press 2

Press 3

Press 4

Press 5

Press 6

Press 7

Press 8

Press 9

Press *

Select the event for each key pressing for 0-9, *, Timeout, and Invalid. The event options are:

  • Extension
  • Voicemail
  • Multimedia Meeting
  • Voicemail Group
  • IVR
  • Ring Group
  • Queues
  • Page Group
  • Custom Prompt
  • Hangup
  • DISA
  • Dial by Name
  • External Number
  • Callback

For each key event, time conditions can be configured. At the configured time condition, this IVR key event can be triggered. Office time, holiday time, or specific time can be configured for time conditions. Up to 5 time conditions can be added for each key.

The available time conditions are ‘All’, ‘Office Time’, ‘Out of Office Time’, ‘Holiday’, ‘Out of Holiday’, ‘Out of Office Time or Holiday’, ‘Office Time and Out of Holiday’, and ‘Specific Time’. If ‘Specific Time’ is selected, a new window will prompt for admin to configure start time, end time, and frequency.

Timeout

When exceeding the number of defined answer timeout, IVR will enter the configured event when timeout. If not configured, then it will hang up.

Invalid

Configure the destination when the Invalid Repeat Loop is done.

Time Condition

Configure the time condition for each key press event, so that it goes to the corresponding destination within a specified time.

IVR Configuration Parameters
Key Pressing Events

Black/Whitelist in IVR

In some scenarios, the IPPBX administrator needs to restrict the extensions that can be reached from IVR.
For example, the company CEO and directors prefer only receiving calls transferred by the secretary, and some special extensions are used on IP surveillance endpoints which should not be reached from external calls via IVR for privacy reasons. CloudUCM has now added blacklist and whitelist in IVR settings for users to manage this.

Up to 500 extensions are allowed on the back/whitelist.

To use this feature, log in to CloudUCM Web GUI and navigate to Call Features🡪IVR🡪Create/Edit IVR: IVR Black/Whitelist.

  • If the user selects “Blacklist Enable” and adds an extension to the list, the extensions in the list will not be allowed to be reached via IVR.
  • If the user selects “Whitelist Enable” and adds an extension to the list, only the extensions in the list can be allowed to be reached via IVR.
Black/Whitelist

Create Custom Prompt

To record a new IVR prompt or upload IVR prompt to be used in IVR, click on “Upload Audio File” next to the “Welcome Prompt” option and the users will be redirected to the Custom Prompt page. Or users could go to Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page directly.

Click on Prompt to Create IVR Prompt

Once the IVR prompt file is successfully added to the CloudUCM, it will be added to the prompt list options for users to select in different IVR scenarios.

Key Pressing Events

Standard Key Event

CloudUCM supports adding time conditions for different key events so that each key event of the IVR goes to the corresponding destination within a specified time.

Each key event supports up to five time conditions, the options available are: All time, Office Time, Out of Office Time, Holiday, Out of Holiday, Out of Office Time or Holiday, Office Time and Out Of Holiday, Specific time.

Key Pressing Events
Note

If you select “Specific time”, you need to select the start time and the end time.

The frequency supports two options: By week and By Month, by default, the specific time does not include the holidays.

Specific Time
Custom Key Event

Users can create custom IVR key press events, vastly increasing the options a business can provide to its customers and improving customer relations and accessibility.

This new feature supports the following:

  • Up to 100 custom key press events
  • Each key combination can contain up to 8 characters (numbers and star (*) only)
  • Supports Time Conditions
  • Different custom keys can have the same Destination and Time Condition
Note

Note: IVR option Dial Other Extensions will be disabled if using custom IVR keys.

Voicemail

Configure Voicemail

If the voicemail is enabled for CloudUCM extensions, the configurations of the voicemail can be globally set up and managed under Web GUI🡪Call Features🡪Voicemail.

Voicemail Settings

Max Greeting Time (s)

Configure the maximum number of seconds for the voicemail greeting. The default setting is 60 seconds.

Dial ‘0’ For Operator

If enabled, the caller can press 0 to exit the voicemail application and connect to the configured operator’s extension.

Operator Type

Configure the operator type; either an extension or a ring group.

Operator Extension

Select the operator extension, which will be dialed when users press 0 to exit the voicemail application. The operator extension can also be used in IVR.

Max Messages Per Folder

Configure the maximum number of messages per folder in users’ voicemail. The valid range is 10 to 1000. The default setting is 50.

Max Message Time

Select the maximum duration of the voicemail message. The message will not be recorded if the duration exceeds the maximum message time. The default setting is 15 minutes. The available options are:

  • 1 minute
  • 2 minutes
  • 5 minutes
  • 15 minutes
  • 30 minutes
  • Unlimited

Min Effective Message Time

Configure the minimum duration (in seconds) of a voicemail message. Messages will be automatically deleted if the duration is shorter than the Min Message Time. The default setting is 3 seconds. The available options are:

  • No minimum
  • 1 second
  • 2 seconds
  • 3 seconds
  • 4 seconds
  • 5 seconds

Note: Silence and noise duration are not counted in message time.

Announce Message Caller-ID

If enabled, the caller ID of the user who has left the message will be announced at the beginning of the voicemail message. The default setting is “No”.

Announce Message Duration

If enabled, the message duration will be announced at the beginning of the voicemail message. The default setting is “No”.

Play Envelope

If enabled, a brief introduction (received time, received from, etc.) of each message will be played when accessed from the voicemail application. The default setting is “Yes”.

Play Most Recent First

If enabled, it will play the most recent message first.

Allow User Review

If enabled, users can review the message following the IVR before sending.

Voicemail Remote Access

If enabled, external callers routed by DID and reaching VM will be prompted by the CloudUCM with 2 options:

  • Press 1 to leave a message.

To leave a message for the extension reached by DID.

  • Press 2 to access the voicemail management system.

This will allow the caller to access any extension VM after entering the extension number and its VM password.

Note: This option applies to inbound calls routed by DID only.

The default setting is “Disabled”.

Forward Voicemail to Peered UCMs

Enables the forwarding of voicemail to remote extensions on peered SIP trunks.

The default setting is “Disabled”.

Voicemail Password

Configures the default voicemail password that will be used when an extension is reset.

Format

Warning: WAV files take up significantly more storage space than GSM files.

Voicemail Settings

Resetting an extension will reset Voicemail Password, Send Voicemail to Email, and Keep Voicemail after Emailing values to default. Previous custom voicemail prompts and messages will be deleted.

Access Voicemail

If the voicemail is enabled for CloudUCM extensions, the users can dial the voicemail access number (by default *97) to access their extension’s voicemail. The users will be prompted to enter the voicemail password and then can enter digits from the phone keypad to navigate in the IVR menu for different options.

Otherwise, the user can dial the voicemail access code (by default *98) followed by the extension number and password to access that specific extension’s voicemail.

Main Menu

Sub Menu 1

Sub Menu 2

1 – New messages

3 - Advanced options

1 - Send a reply

2 - Call the person who sent this message

3 - Hear the message envelop

4 - Leave a message

* - Return to the main menu

5 - Repeat the current message

7 - Delete this message

8 - Forward the message to another user

9 – Save

* - Help

# - Exit

2 – Change folders

0 - New messages

1 - Old messages

2 - Work messages

3 - Family messages

4 - Friend messages

# - Cancel

3 – Advanced options

1 - Send a reply

2 - Call the person who sent this message

3 - Hear the message envelop

4 - Leave a message

* - Return to the main menu

0 – Mailbox options

1 - Record your unavailable message

1 - Accept this recording

2 - Listen to it

3 - Re-record your message

2 - Record your busy message

1 - Accept this recording

2 - Listen to it

3 - Re-record your message

3 - Record your name

1 - Accept this recording

2 - Listen to it

3 - Re-record your message

4 - Record temporary greeting

1 - Accept this recording

2 - Listen to it

3 - Re-record your message

5 - Change your password

* - Return to the main menu

Tips

  • While listening to the voicemail, press * or # to rewind and forward the voice message, respectively. Each press will forward or rewind 3 seconds.
  • Rewind can go back to the beginning of the message while forward will not work when there are 3 seconds or less left in the voice message.
  • Voice guidance will be automatically played when the voicemail is done playing.

Leaving Voicemail

If an extension has voicemail enabled under basic settings “Extension/Trunk 🡪 Extensions 🡪 Basic Settings” and after a ring timeout or the user is not available, the caller will be automatically redirected to the voicemail to leave a message on which case they can press # to submit the message.

In case the caller is calling from an internal extension, they will be directly forwarded to the extension’s voicemail box. But if the caller is calling from outside the system and the incoming call is routed by DID to the destination extension, then the caller will be prompted with the choice to either press 1 to access voicemail management or press 2 to leave a message for the called extension. This feature could be useful for remote voicemail administration.

Voicemail Email Settings

The CloudUCM can be configured to send the voicemail as an attachment to the Email. Click on the “Voicemail Email Settings” button to configure the Email attributes and content.

Send Voicemail to Email

If enabled, voicemail will be sent to the user’s email address.


Note: SMTP server must be configured to use this option.

Keep Voicemail after Emailing

Enable this option if you want to keep recording files after the Email is sent. The default setting is Enable.

Email Template

Fill in the “Subject:” and “Message:” content, to be used in the Email when sending to the user. The template variables are:

  • \t: TAB
  • ${VM_NAME}: Recipient’s first name and last name
  • ${VM_DUR}: The duration of the voicemail message
  • ${VM_MAILBOX}: The recipient’s extension
  • ${VM_CALLERID}: The caller ID of the person who has left the message
  • ${VM_MSGNUM}: The number of messages in the mailbox
  • ${VM_DATE}: The date and time when the message is left. (Format: MM/dd/yyyy hh:mm:ss)
Voicemail Email Settings
Voicemail Email Settings

Click on the “Email Template” button to view the default template as an example.

Configure Voicemail Group

The CloudUCM supports voicemail group and all the extensions added in the group will receive the voicemail to the group extension. The voicemail group can be configured under Web GUI 🡪 Call Features 🡪 Voicemail 🡪 Voicemail Group. Click on “Add” to configure the group.

Voicemail Group

Extension

Enter the Voicemail Group Extension. The voicemail messages left to this extension will be forwarded to all the voicemail group members.

Name

Configure the Name to identify the voicemail group. Letters, digits, _ and - are allowed.

Method

Select the preference for receiving and managing group voicemail.

  • Forwarded: Voicemail will be stored in the group voicemail box, and each voicemail group member will be forwarded a copy of it.

  • Shared: Voicemail will be stored in the group voicemail box, and voicemail status will be shared among all voicemail group members. If a member deletes a voicemail, it will also be deleted for all members. Likewise, if one member reads a voicemail, it will be considered read for the entire group.

Voicemail Password

Configure the voicemail password for the users to check voicemail messages.

Email Address

Configure the Email address for the voicemail group extension.

Shared Voicemail Status

If enabled, voicemail group status can be monitored via BLF. Green indicates no unread voicemail, and red indicates existing unread voicemail.

Members

Select available mailboxes from the left list and add them to the right list. The extensions need to have voicemail enabled to be listed in available mailboxes list.

Greet Prompt

This voicemail prompt will be played when the callee does not answer within their ring timeout period. Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with .mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

Temporary Prompt

This voicemail prompt will be played in all scenarios when it is configured (unregistered, unanswered/ring timeout, busy, DND). Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with .mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

Unavailable Prompt

This voicemail prompt will be played when user enters voicemail. Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with .mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

Ring Groups

The CloudUCM supports ring group feature with different ring strategies applied to the ring group members. This section describes the ring group configuration on the CloudUCM.

Configure Ring Group

Ring group settings can be accessed via Web GUI🡪Call Features🡪Ring Group.

Ring Group

  • Click on to add ring group.
  • Click on to edit the ring group. The following table shows the ring group configuration parameters.
  • Click on to delete the ring group.
Ring Group Configuration

Ring Group Name

Configure ring group name to identify the ring group. Letters, digits, _ and – are allowed.

Extension

Configure the ring group extension.

Members

Select available users from the left side to the ring group member list on the right side. Click on ⮝ ⮟ to arrange the order.

LDAP Phonebook

Select available remote users from the left side to the ring group member list on the right side. Click on ⮝ ⮟ to arrange the order. Note: LDAP Sync must be enabled first.

Ring Strategy

Select the ring strategy. The default setting is “Ring in order”.

  • Ring Simultaneously: Ring all the members at the same time when there is incoming call to the ring group extension. If any of the member answers the call, it will stop ringing.

  • Ring in Order: Ring the members with the order configured in ring group list. If the first member does not answer the call, it will stop ringing the first member and start ringing the second member.

Music On Hold

Select the “Music On Hold” Class of this Ring Group, “Music On Hold” can be managed from the “Music On Hold” panel on the left.

Custom Prompt

This option is to set a custom prompt for a ring group to announce to caller. Click on ‘Prompt’, it will direct the users to upload the customized voice prompts.

Note: Users can also refer to the page PBX Settings🡲 Voice Prompt🡲 Custom Prompt, where they could record new prompt or upload prompt files.

Ring Timeout on Each Member

Configure the number of seconds to ring each member. If set to 0, it will keep ringing. The default setting is 60 seconds.

Note: The actual ring timeout might be overridden by users if the phone has ring timeout settings as well.

Auto Record

If enabled, calls on this ring group will be automatically recorded. The default setting is No. The recording files can be accessed from WebGUI🡲 CDR🡲 Recording Files.

Endpoint Call Forwarding Support

This allows the UCM to work with endpoint-configured call forwarding settings to redirect calls to ring group. For example, if a member wants to receive calls to the ring group on his mobile phone, he will have to set his endpoint’s call forwarding settings to his mobile number. By default, it is disabled.

However, this feature has the following limitations:

  • This feature will work only when call forwarding is configured on endpoints, not on the UCM.

  • If the forwarded call goes through an analog trunk, and polarity reversal is disabled, the other ring group members will no longer receive the call after it is forwarded.

  • If the forwarded call goes through a VoIP trunk, and the outbound route for it is PIN-protected and requires authentication, the other ring group members will no longer receive the call after it is forwarded.

  • If the forwarded call hits voicemail, the other ring group members will no longer receive the call.

Replace Display Name

If enabled, the UCM will replace the caller display name with the Ring Group name the caller know whether the call is incoming from a direct extension or a Ring Group.

Skip Busy Agent

If enabled, skip busy agents regardless of call waiting settings.

Enable Destination

If enabled, users could select extension, voicemail, ring group, IVR, call queue, voicemail group as the destination if the call to the ring group has no answer. Secret and Email address are required if voicemail is selected as the destination.

Default Destination

The call would be routed to this destination if no one in this ring group answers the call.

Note: Users can now set the voicemail of ring groups as routing destinations and IVR key press event destinations and to do so ring group must have their Default Destination set to Voicemail with Ring Group Extensions.

Voicemail

Whether to enable the voicemail for the ring group or not.

Voicemail Password

Configure the voicemail password (only numbers).

Email Address

Fill in the user's Email address (s), the voice message will be sent to this address (s).

Busy Prompt

This voicemail prompt will be played when the callee is in another call or is in DND mode. Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with .mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

Greet Prompt

This voicemail prompt will be played when the callee does not answer within their ring timeout period. Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with .mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

Temporary Prompt

This voicemail prompt well be played in all scenarios when it is configured (unregistered, unanswered/ring timeout, busy, DND). Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with .mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

Unavailable Prompt

This voicemail prompt will only be played when the callee’s extension is unregistered. Priority: Temporary Prompt > Busy Prompt/Unavailable Prompt > Greet Prompt

Sound file must be PCM encoded, 16 bits at 8000Hz mono with mp3/wav format, or raw ulaw/alaw/gsm file with .mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

Remote Extension in Ring Group

Remote extensions from the peer trunk of a remote UCM can be included in the ring group with local extensions. An example of Ring Group with peer extensions is presented in the following:

  1. Creating SIP Peer Trunk between both UCM_A and UCM_B. SIP Trunk can be found under Web GUI🡪Extension/Trunk🡪VoIP Trunks. Also, please configure their Inbound/Outbound routes accordingly.
  2. Click edit button in the menu , and check if Sync LDAP Enable is selected, this option will allow UCM_A update remote LDAP server automatically from peer UCM_B. In addition, Sync LDAP Password must match for UCM_A and UCM_B to sync LDAP contact automatically. Port number can be anything between 0~65535, and use the outbound rule created in step 1 for the LDAP Outbound Rule option.
Sync LDAP Server Options
  1. In case if LDAP server does not sync automatically, user can manually sync LDAP server. Under VoIP Trunks page, click sync button shown in the following figure to manually sync LDAP contacts from peer UCM.
Manually Sync LDAP Server
  1. Under Ring Groups setting page, click “Add”. Ring Groups can be found under Web GUI🡪Call Features🡪Ring Groups.
  2. If LDAP server is synced correctly, Available LDAP Numbers box will display available remote extensions that can be included in the current ring group. Please also make sure the extensions in the peer UCM can be included into that UCM’s LDAP contact.
Ring Group Remote Extension

Restrict Calls

Restrict calls is a feature that can be used to restrict calls between internal extensions besides those in the Allowed List.

This section describes the configuration of this feature in the Call Features->Restrict Calls page.

Restrict Calls

Configure Restrict Calls

  • Click on “Add” to add a rule for restrict calls.
  • Click on to edit the rule of restrict calls.
  • Click on to delete the rule of restrict calls.

Name

Configure Restrict call’s name

Restrict Calls between extensions

When enabled, members of the group cannot dial other extension, only the numbers in the Allowed List. By default it’s enabled.

Members

Configure the members that will not be able to call any extensions besides those in the Allowed List.

Allowed list

Select the extensions that the Members list can be able to call.

Paging/Intercom

Paging and Intercom Group can be used to make an announcement over the speaker on a group of phones. Targeted phones will answer immediately using speaker. The CloudUCM paging and intercom can be used via feature code to a single extension or a paging/intercom group. This section describes the configuration of paging/intercom group under Web GUI🡪Call Features🡪Paging/Intercom.

Paging/Intercom Groups

2-way Intercom
2-way Intercom
NameConfigure paging/intercom group name.
TypeSelect “2-way Intercom”.
ExtensionConfigure the paging/intercom group extension.
Auto RecordEnable this option to record in WAV format.
Replace Display NameIf enabled, the UCM will replace the caller display name with the Paging/Intercom name.
Maximum Call DurationSpecify the maximum call duration in seconds. The default value 0 means no limit.
Custom PromptThis option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct the users to upload the customized voice prompts. Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom Prompt, where they could record new prompt or upload prompt files.
MembersSelect available users from the left side to the paging/intercom group member list on the right.
Paging/Intercom WhitelistSelect which extensions are allowed to use the paging/intercom feature for this paging group.
2-way Intercom Configuration Parameters
1-way Paging
1-way Paging
NameConfigure paging/intercom group name.
TypeSelect “1-way Paging”.
ExtensionConfigure the paging/intercom group extension.
Video BroadcastIf checked, video paging will be supported. If the caller sends a video page, the paging group members will be able to receive and view the video.
Auto RecordEnable this option to record in WAV format.
Delayed PagingAllows the announcement to be played after the configured delay paging. If there are many messages, they will be played in sequence.
LaggedTime(seconds)Set the delay paging duration, the default is 5 seconds.
Replace Display NameIf enabled, the UCM will replace the caller display name with Paging/Intercom name.
Maximum Call DurationSpecify the maximum call duration in seconds. The default value 0 means no limit.
Custom PromptThis option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct the users to upload the customized voice prompts.Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom Prompt, where they could record new prompt or upload prompt files.
MembersSelect available users from the left side to the paging/intercom group member list on the right.
Paging/Intercom WhitelistSelect which extensions are allowed to use the paging/intercom feature for this paging group.

In case the user wants to broadcast a video, these requirements should be respected.

  • H.264 video encoding
  • .mkv or .tar/.tgz/tar.gz format  
  • MKV files must be 30 MB file or less
  • Compressed files (.tar/.tgz/tar.gz) must be 50 MB or less.
  • File name can only contain alphanumeric characters, hyphens (-), and period (.)

If Auto Record is enabled, recorded video pages will be saved in MKV file format. Saved recordings can be found on the CDR🡪Recordings🡪Video Recordings page.

Scheduled Paging/Intercom

Pending Paging/Intercom

In this page, the user can create scheduled intercom/paging to be played automatically when the time scheduled arrives.

Paging/IntercomSelect existing paging/intercom groups.
NameEnter the name of the scheduled Intercom/Paging.
CallerOnce a caller is selected, and the specified start time is reached, the system will contact the caller. If this call is rejected, the page/intercom will be cancelled. If caller is set to None, the system will call all group members and play the configured prompt.
Start DateSelect the date of the start of the paging/intercom
Start TimeSelect the start time of the paging/intercom.
RepeatSelect the repeat interval of the paging/intercom.
No Repeat: The intercom/paging will play once on the scheduled date and time
Everyday: The intercom/paging will play daily starting from the scheduled day and on the time scheduled every day.
Weekly: The intercom/paging will play weekly on the selected day(s) of the week.
Monthly: The intercom/paging will play monthly on the selected date of the month.
Sync to Google CalendarThis feature cannot be used if Google Services have not been authorized. Please resolve this in the Integrations > Google Services page.

Once the paging and intercom has been created, it can be viewed on the same page.

Pending Paging/Intercom

Paging/Intercom Schedule

This section displays the schedule of the paging/intercom which have been scheduled. The user can choose to display per day, week, or per month.

Scheduled Paging/Intercom

Operator Panel

Configure Operator Panel

Operator Panel settings can be accessed via Web GUI🡪Call Features🡪Operator Panel.

The CloudUCM supports the operator panel so that UCM extension can be used as admin to manage calls and activities such as extension status, call queue status, transfer, barge-in, hangup, etc. On Grandstream Wave client, it can display the extensions, ring group, voicemail, call queue, call park status under the management of the extension. This section describes how to configure the operator panel.

Graphical user interface, application

Description automatically generated
Operator Panel Configuration Page
  • Click on “Add” to create the operator panel.
  • Click on to edit the operator panel.
  • Click on to delete the operator panel.

Name

The Operator panel name.

Administrator

The operator of the call console can select extensions, extension groups, and departments. For the selected extension groups and departments, subsequent extensions will automatically become administrators.

Management Module

Extension

The selected extensions will be supervised by the administrator, and you can choose extensions, extension groups, and departments. For the selected extension groups and departments, subsequent extensions will be automatically supervised by the administrator.

Ring Groups

The checked Ring Groups will be supervised by the administrator. Select “All”, all Ring Groups and subsequent updates will be automatically supervised by the administrator.

Voicemail Groups

The checked Call Queue will be supervised by the administrator. Select “All”, all Call Queue and subsequent updates will be automatically supervised by the administrator.

Call Queue

The checked Call Queue will be supervised by the administrator. Select “All”, all Call Queue and subsequent updates will be automatically supervised by the administrator.

Parking Lot

The checked Parking Lot will be supervised by the administrator. Select “All”, all Parking Lot and subsequent updates will be automatically supervised by the administrator.

Call Queue

CloudUCM supports call queue by using static agents or dynamic agents. Call Queue system can accept more calls than the available agents. Incoming calls will be held until next representative is available in the system. This section describes the configuration of call queue under Web GUI🡪Call Features🡪Call Queue.

Configure Call Queue

Call queue settings can be accessed via Web GUI🡪Call Features🡪Call Queue.

Call Queue

CloudUCM supports custom prompt feature in call queue. This custom prompt will active after the caller waits for a period of time in the Queue. Then caller could choose to leave a message/ transfer to default extension or keep waiting in the queue.

To configure this feature, please go to UCM Web GUI🡪Call Features🡪Call Queue🡪Create New Queue/Edit Queue🡪Queue Options🡪set Enable Destination to Enter Destination with Voice Prompt. Users could configure the wait time with Voice Prompt Cycle.

  • Click on “Add” to add call queue.
  • Click on to edit the call queue. The call queue configuration parameters are listed in the table below.
  • Click on to delete the call queue.

Basic Settings

General

Extension

Configure the call queue extension number.

Name

Configure the call queue name to identify the call queue.

Strategy

Select the strategy for the call queue.

  • Ring All: Ring all available Agents simultaneously until one answers.

  • Linear: Ring agents in the specified order.

  • Least Recent: Ring the agent who has been called the least recently.

  • Fewest Calls: Ring the agent with the fewest completed calls.

  • Random: Ring a random agent.

  • Round Robin: Ring the agents in Round Robin scheduling with memory.

The default setting is "Ring All".

Music On Hold

Select the Music On Hold class for the call queue.
Note: Music On Hold classes can be managed from Web GUI🡪PBX Settings🡪Music On Hold.

Max Queue Length

Configure the maximum number of calls to be queued at once. This number does not include calls that have been connected with agents, only calls that are still in queue. When this maximum value is exceeded, the caller will hear a busy tone and be forwarded to the configured failover destination. Default value is 0 (unlimited).

Agent Rest Time (s)

Configure the amount of time in seconds after ending a call where the agent will not receive additional calls. Once this time has passed, the agent will be able to receive calls again. If set to 0, agents can receive additional calls immediately after ending a call.

Retry Time (s)

Configure the number of seconds to wait before ringing the next agent. The minimum is 1.

Agent Ring Time

Configure the number of seconds to ring an agent. The minimum is 5.

Auto Record

If enabled, the calls on the call queue will be automatically recorded. The recording files can be accessed in Queue Recordings under Web GUI🡪Call Features🡪Call Queue.

Welcome Prompt

Enable

Enable the welcome prompt.

Custom Prompt

Initial tone that plays when the user dials the queue number.

Note: The user can upload a custom prompt directly from this parameter.

Play Full Welcome Prompt

If enabled, queue agents will not be rung until after the welcome prompt is done playing. Otherwise, queue agents will be rung while the playing the welcome prompt.

Satisfaction Survey Prompt

Custom Prompt

After a queue agent hangs up a call, a prompt will play asking the caller to rate their satisfaction on a scale of 1 to 5, with 5 being the highest.

Note: The user can upload a custom prompt directly from this parameter.

Max Wait Time

Max Wait Time

Configures the amount of time a caller will be kept in queue before the the call is automatically routed to the configured Max Wait Time Destination. If set to 0, callers will be kept in queue indefinitely.

Destination

The call will be routed to this destination if no one in this queue answers the call.

Destination Prompt Cycle

Enable

Enable Destination Prompt Cycle

Destination Prompt Cycle

Configure the voice prompt cycle (in seconds) of this call queue. When playing the voice prompt, you can press 1 to transfer to failover destination.

Custom Prompt

When playing a custom prompt, press 1 to enter the failover destination or continue waiting in queue.

Note: The user can upload a custom prompt directly from this parameter.

Destination

After the specified amount of time, the caller will be prompted to press 1 to immediately get redirected to the configured failover destination.

Advanced Settings

Virtual Queue

Enable Virtual Queue

If enabled, system will enable a virtual queue for users waiting in queue.

Virtual Queue Mode

When in DTMF mode, pressing 2 will manually trigger virtual queue. When in Timeout mode, virtual queue will automatically be triggered when the configured Virtual Queue Period has passed. DTMF mode and Timeout mode require the caller to manually set a callback number. When in Auto mode, virtual queue will automatically be triggered when the configured Virtual Queue Period has passed. The callback number will automatically be set to the caller's detected CID number.

Virtual Queue Period (s)

The amount of time in seconds that must pass before virtual queue is offered to callers when using Timeout mode or Auto mode.

Virtual Queue Outbound Prefix

System will add this prefix to dialed numbers when calling back users.

Enable Virtual Queue Position Announcement

If enabled, the system will inform callers waiting in the queue of their positions in line.

Enable Virtual Queue Wait Time Announcement

If enabled, the estimated wait time for the call to get answered will periodically be announced to the caller.

Enable Virtual Queue Callback Timeout

If enabled, agents will have a set amount of time to answer a virtual queue callback.

Virtual Queue Welcome Prompt

Upload the file of your welcome prompt of the virtual queue.

Announcement Settings

Enable Position Announcement

If enabled, the system will inform callers waiting in the queue of their positions in line.

Enable Wait Time Announcement

If enabled, the estimated wait time for the call to get answered will periodically be announced to the caller. Note: Wait time will not be announced if less than one minute.

Announcement Interval

The interval at which caller positions and estimated wait times will be announced.

Agent ID Announcement

If enabled, a system prompt coontaining the agent ID will be played to the caller when answered by an agent.

Empty Queue

Leave When Empty

Configure whether the callers will be disconnected from the queue or not if the queue has no agent anymore. The default setting is "Strict".

  • Yes: Callers will be disconnected from the queue if all agents are paused or invalid.

  • No: Never disconnect the callers from the queue when the queue is empty.

  • Strict: Callers will be disconnected from the queue if all agents are paused, invalid or unavailable.

Dial in Empty Queue

Configure whether the callers can dial into a call queue if the queue has no agent. The default setting is "No".

  • Yes: Callers can always dial into a call queue.

  • No: Callers cannot dial into a queue if all agents are paused or invalid.

  • Strict: Callers cannot dial into a queue if the agents are paused, invalid or unavailable.

Failover Destination

Choose the destination where the call will be directed when the queue is empty or when all the agents are not logged in, here are the destinations that can be configured:

  • Play Sound.

  • Extension.

  • Voicemail.

  • Queues.

  • Ring Group.

  • Voicemail Group.

  • IVR

  • External Number.

CTI

Enable Agent Login

Enabling agent login will cause the dynamic agents to be unavailable.

Queue Chairman

The queue chairman can log into his web portal to operate the queue.

Service Level Agreement (SLA)

Enable SLA

Toggles Service Level Agreement (SLA), which is percentage measurement of the queue group's ability to answer incoming calls within a defined amount of time. If a queue group's calculated SLA percentage is below the configured threshold value, alerts will be generated and sent out via email to the specified recipients. Example: The SLA goal is 80% of calls (Threshold) within 20 seconds (SLA Time). If less than 80% of queue calls are answered within 20 seconds, the specified users will be notified of it.

SLA Time (s)

Configures the amount of time in seconds that agents must answer incoming queue calls within to satisfy service quality requirements. Answering calls past this time will negatively affect the SLA measurement, and an alert will be generated once it hits below the specified SLA alert threshold. Supported values are 1 to 180. Default value is 20.

SLA Alert Email Notification

Enable SLA alert email notification.

Alert Threshold (%)

Configures the SLA alert threshold. If the percentage of queue calls answered within the configured SLA Time go below this value, an alert email will be generated and sent to the configured recipients. Supported values are 1 to 100. Default value is 80.

SLA Alert Interval (m)

Configures the minimum amount of time (in minutes) between alert sending. If a new alert is generated within this period, it will not be sent to recipients until the next alert interval. The valid range is from 1 to 120. The default value is 120.

SLA Alert Email Template

The template of the SLA alert email notifications. 

Alert Email Recipients

Send SLA alert notifications to the configured alert email recipients. If a recipient does not have an email address configured, they will not receive the alert notifications.

Other Settings

Report Hold Time

If enabled, the CloudUCM will report (to the agent) the duration of time of the call before the caller is connected to the agent. The default setting is "No".

Replace Display Name

If enabled, the UCM will replace the caller display name with the Call Queue name so that the caller knows the call is incoming from a Call Queue.

Enable Feature Codes

Enable feature codes option for call queue. For example, *83 is used for “Agent Pause”

Autofill

Configure to enable autofill.

Dynamic Login Password

If enabled, the configured PIN number is required for dynamic agent to log in. The default setting is disabled.

Alert-Info

When present in an INVITE request, the Alert-info header field specifies an alternative ring tone to the UAS.

Agents

Static Agents

Go to “Agents” Tab and Select the available users to be the static agents in the call queue. Choose from the available users on the left to the static agents list on the right. Click on ⮜ or ⮞ to choose. And use UP and Down arrow to select the order of the agent within the call queue.

Click on “Global Queue Settings” to configure Agent Login Extension Postfix and Agent Logout Extension Postfix. Once configured, users could log in the call queue as dynamic agent.

Agent Login Settings

For example, if the call queue extension is 6500, Agent Login Extension Postfix is * and Agent Logout Extension Postfix is **, users could dial 6500* to login to the call queue as dynamic agent and dial 6500** to logout from the call queue. Dynamic agent does not need to be listed as static agent and can log in/log out at any time.

  • Call queue feature code “Agent Pause” and “Agent Unpause” can be configured under Web GUI🡪Call Features🡪Feature Codes. The default feature code is *83 for “Agent Pause” and *84 for “Agent Unpause”.
    Note: When dialing the “Agent Pause” feature code, users can specify the reason for it. The following reasons are available: (1) Lunch, (2) Hourly Break, (3) Backoffice, (4) Email, and (5) Wrap.
    The agent can also dial the feature with the number of the reason of the pause. E.g., if the agent want to perform a pause for lunch, he/she can dial *831 directly instead of waiting for the IVR response.
  • Queue recordings are shown on the Call Queue page under “Queue Recordings” Tab. Click on to download the recording file in .wav format; click on to delete the recording file. To delete multiple recording files by one click, select several recording files to be deleted and click on “Delete Selected Recording Files” or click on “Delete All Recording Files” to delete all recording files.

Call Center Settings and Enhancements

UCM supports light weight call center features including virtual queue and position announcement, allowing the callers to know their position on the call queue and giving them the option to either stay on the line waiting for their turn or activate a callback which will be initiated by the UCM one an agent is free.

To configure call center features, press on an existing call queue and go under the advanced settings tab.

The following parameters are available:

Enable Virtual Queue

Enable virtual queue to activate call center features.

Virtual Queue Period

Configure the time in (s) after which the virtual queue will take effect and the menu will be presented to the caller to choose an option. Default is 20s.

Virtual Queue Mode

Offered to caller after timeout: After the virtual queue period passes, the caller will enter the virtual call queue and be presented with a menu to choose an option, the choices are summarized below:

  • Press * to set current number as callback number.
  • Press 0 to set a callback number different than current caller number.
  • Press # to keep waiting on the call queue.

Triggered on user request: In this mode, the callers can activate the virtual queue by pressing 2, then they will be presented with the menu to choose an option as below:

  • Press * to set the current number as a callback number.
  • Press 0 to set a callback number different than current caller number.
  • Press # to keep waiting on the call queue.

Virtual Queue Outbound Prefix

The system will add this prefix to dialed numbers when calling back users.

Enable Virtual Queue Timeout

When this option is enabled and after a caller registers a call back request on the virtual queue. While all the agents are busy, the UCM will call an agent once he/she is idle again, this timeout is used for how long the UCM continues calling the agent and if the agent doesn’t answer the call then the callback request will timeout and expire.

Write Timeout

Configure the virtual queue callback timeout period in seconds.

Enable Virtual Queue Position Announcement

Enable the announcement of the caller’s position periodically.

Note: The queue position will now be announced to the caller upon entering the queue.

Position Announcement Interval

Configure the period of time in (s) during which the UCM will announce the caller’s position in the call queue.

Enable Virtual Queue Wait Time Announcement

When enabled the UCM will announce the estimated queue wait time to callers if the estimated wait time is longer than 1 minute.

Queue Chairman

Select the extension to act as chairman of the queue (monitoring).

Virtual Queue Welcome Prompt

Click on “Upload Audio File” to upload the VQ welcome prompt.

Enable Agent Login

When enabled, statics agents can conveniently log in and out of a queue by configuring a programmable key on their phones as a shortcut.

Notes:

  • This feature is currently available only for GXP21xx phones on firmware 1.0.9.18 or greater.
  • After enabling the feature, users need to set the option on GXP21XX phone under “Account🡪SIP Settings🡪Advanced Features🡪Special Feature” to “UCM Call Center”. A softkey labeled “UCM-CC” will appear on the bottom of the phone’s screen.
  • When this option is enabled, dynamic agent login will be no longer supported.
  • In case of concurrent registrations, changing agent status on one phone (login/logout) will be reflected on all phones.
Call Center Parameters

Queue Auto fill enhancement:

The waiting callers are connecting with available members in a parallel fashion until there are no more available members or no more waiting callers.

For example, in a call queue with linear method, if there are two available agents, when two callers call in the queue at the same time, UCM will assign the two callers to each of the two available agents at the same time, rather than assigning the second caller to second available agent after the first agent answers the call from the first caller.

Call Queue Statistics

Along with call center features, users can also gather detailed call queue statistics allowing them to make better changes/decisions to manage the call distribution and handling based on time, agent, and queue.

To access call queue statistics, go to Web GUI🡪Call Features🡪Call Queue and click on “Call Queue Statistics”, the following page will be displayed:

Call Queue Statistics
  • Agent statistics: shows the number of calls and call-related information of agents;
  • Queue Statistics: counts the number of calls in the queue and information such as calls, waiting, and callback;
  • Agent satisfaction statistics used for user’s rating of agents;
  • Queue satisfaction statistics count the score survey statistics.

The overview page performs seat statistics, queue statistics, seat satisfaction statistics, and queue satisfaction statistics according to the business. Agent statistics record the number of calls and call-related information of agents; queue counts the number of calls in the queue and information such as calls, waiting, and callback; agent satisfaction statistics are survey statistics based on user ratings of agents; queue satisfaction statistics are user-queue The score survey statistics.

By selecting a time interval, administrators can get detailed statistics for agent(s) such as total calls, answered calls, etc, as well as for the queue(s) such as ABANDONED CALLS also a detailed information for the queue’s call log by clicking on Options🡪Information button and the below window will pop up:

Queue’s call log details

User can download statistics on CSV format by clicking on the “Download”, also the statistics can be cleared using “Reset Statistics” button.

The statistics can be automatically sent to a specific email address on a pre-configured Period, this can be done by clicking on “Automatic Download”, and the user will be directed to the below page where he can configure the download period (Day/Week/Month) and the Email where the statistics will be sent (Email settings should be configured correctly):

Automatic Download Settings – Queue Statistics

Significantly more information is now available UCM’s queue statistics page. In addition to the information presented in previous firmware, users can now view a call log that displays calls to all agents and queues, a dynamic agent login/logout record, and a pause log. Statistics reports for these new pages can be obtained by pressing the Download button in the top left corner of the Call Queue Statistics page. The reports are in .CSV format and will be packaged into a single tar.gz file upon download.

Agent Details is a call log that shows every call to each agent from all queues. The following information is available:

  • Time – the date and time the call was received.
  • Agent – the agent that was rung for the call.
  • Queue – the queue that the call went to.
  • Caller ID Number – the CID of the caller
  • Abandoned – indicates whether the call was picked up or not by that specific agent. If the call rang several agents simultaneously, and this specific agent did not pick up the call, the call will be considered abandoned even if a different agent in the same queue picked it up.
  • Wait Time – the amount of time that the call was waiting in the queue after dialing in.
  • Talk Time – the duration of the call after it was picked up by agent.
Agent details

Login Record is a report that shows the timestamps of dynamic agent logins and logouts and calculates the amount of time the dynamic agents were logged in. Dynamic agents are extensions that log in and out either via agent login/logout codes (configured in the Global Queue Settings page) or by using the GXP21xx call queue softkey. A new record will be created only when an agent logs out. The following information is available:

  • Agent – the extension that logged in and out.
  • Queue – the queue that the extension logged in and out of.
  • Login Time – the time that the extension logged into the queue.
  • Logout Time – the time that the extension logged out of the queue.
  • Login Duration – the total length of time that the extension was logged in.
Login Record

Pause Log is a report that shows the times of agent pauses and unpauses and calculates the amount of time that agents are paused. If an agent is part of several queues, an entry will be created for each queue. An entry will only be created after an agent unpauses. The following information is available:

  • Agent: The extension that paused/unpaused.
  • Name: The name of the agents that paused/unpaused.
  • Queue: The queue that the agent is in.
  • Pause Time: The time when the agent paused.
  • Resume Time: The time when the agent unpaused.
  • Pause Duration: The total length of time the agent was paused for.
  • Pause Reason: The reason of the pause (e.g., lunch, coffee break, etc…)
Pause Log

Switchboard

Switchboard is a Web GUI tool for call queue monitoring and management, admin can access to it from the menu Call Features🡪Call Queue then press “Switchboard”.

Following page will be displayed:

Switchboard Summary

The page above summarizes the available queue statistics and if one of the queues is clicked the user will be directed to the page below:

Call Queue Switchboard

The table below gives a brief description of the main menus:

Waiting

This menu shows the current waiting calls along with the caller id and the option to hang-up call by pressing on the button.

Proceeding

Shows the currently established calls along with the caller ID and the callee (agent) as well as the option to hang up, transfer, add conference, or barge-in the call.

Agents

Displays the list of agents in the queue and the extension status (idle, ringing, in use or unavailable) along with some basic call statistics and agent’s mode (static or dynamic).

Note: the dashboard will show the number of calls (answered and abandoned) of each agent. For dynamic agents, it will count the number of calls starting from the last login time.

Switchboard Parameters

There are three different privilege levels for Call Queue management from the switchboard: Super Admin, Queue Chairman, and Queue Agent.

  • Super Admin – Default admin of the UCM. Call queue privileges include being able to view and edit all queue agents, monitor, and execute actions for incoming and ongoing calls for each extension in Switchboard, and generate Call Queue reports to track performance.
  • Queue Chairman – User appointed by Super Admin to monitor and manage an assigned queue extension via Switchboard. The Queue Chairman can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on “Other Features” in the side menu and click on “Call Queue”. In the image below, User 1001 is the Queue Chairman appointed to manage Queue Extension 6500 and can see all the agents of the queue in the Switchboard and their related information (Extension Status, Agent, Name, Answered, Abandoned, Login/Logout Time, Pause/Resume Time, Talk Time, Agent Status, Pause Reason, and Options). The Chairman is also able to log out dynamic agents from call queues.
  • Queue Agent – User appointed by Super Admin to be a member of a queue extension. A queue agent can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on “Other Features” in the side menu and click on “Call Queue”. However, a queue agent can view and manage only his own calls and statistics, but not other agents’ in the queue extension. In the image below, User 1000 is a queue agent and can see only his own information in the Switchboard.

Global Queue Settings

As explained before, under this section users can configure the feature codes for Dynamic agent login and logout, and also can now customize the keys for virtual queue options like shown below.

Global Queue Settings

Dynamic Agent Login Settings

Agent Login Code Suffix

Configure the code to dial after the queue extension to log into the queue (i.e. queue extension + suffix).
If no suffix is configured, dynamic agents will not be able to log in

Agent Logout Code Suffix

Configure the code to dial after the queue extension to log out of the queue (i.e. queue extension + suffix).
If no suffix is configured, dynamic agents will not be able to log out.

Virtual Queue Callback Key Settings

Enable

Select whether to enable or disable virtual queue callback feature. By default it’s disabled.

Call Back Current Number

Press the feature key configured to set your current number as callback number.

Custom Callback Number

Press these feature keys configured to set a custom callback number.

Continue Waiting

Press the feature key configured to continue waiting.

Global Queue Settings

Pickup Groups

The CloudUCM supports the pickup group feature which allows users to pick up incoming calls for other extensions if they are in the same pickup group, by dialing the “Pickup Extension” feature code (by default *8).

Configure Pickup Groups

Pickup Groups interface

Pickup groups can be configured via Web GUI🡪Call Features🡪Pickup Groups.

  • Click on to create a new pickup group.
  • Click the button to upload the pickup group information in CSV format.
  • Click the button to generate pickup group information in .CSV format.
  • Click on to edit the pickup group.
  • Click on to delete the pickup group.

Select extensions from the list on the left side to the right side.

Edit Pickup Group

Configure Pickup Feature Code

When picking up the call for the pickup group member, the user only needs to dial the pickup feature code. It is not necessary to add the extension number after the pickup feature code. The pickup feature code is configurable under Web GUI🡪Call Features🡪Feature Codes.

The default feature code for call pickup extension is *8, otherwise if the person intending to pick up the call knows the ringing extension they can use ** followed by the extension number to perform the call pickup operation. The following figure shows where you can customize these feature codes.

Edit Pickup Feature Code

Dial By Name

Dial by Name is a feature on the PBX that allows the caller to search a person by first or last name via his/her phone’s keypad. The administrator can define the Dial by Name directory including the desired extensions in the directory and the searching type by “first name” or “last name”. After dialing in, the PBX IVR/Auto Attendant will guide the caller to spell the digits to find the person in the Dial by Name directory. This feature allows customers/clients to use the guided automatic system to contact the enterprise employees without having to know the extension number, which brings convenience and improves the business image for the enterprise.

Dial by Name Configuration

The administrators can create the dial by name group under Web GUI🡪Call Features🡪Dial By Name.

Create Dial by Name Group

Configure Extension First Name and Last Name

  1. Name

Enter a Name to identify the Dial by Name group.

  1. Extension

Configure the direct dial extension for the Dial By Name group.

  1. Custom Prompt

This option sets a custom prompt for directory to announce to a caller. The file can be uploaded from the page “Custom Prompt”. Click “Upload Audio File” to add additional record.

  1. Available Extensions/Selected Extensions

Select available extensions from the left side to the right side as the directory for the Dial By Name group. Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group. The extensions here must have a valid first name and last name configured under Web GUI🡪Extension/Trunk🡪Extensions in order to be searchable in Dial By Name directory through IVR. By specifying the extensions here, the administrators can make sure unscreened calls will not reach the company employee if he/she does not want to receive them directly.

  1. Prompt Wait Time

Configure “Prompt Wait Time” for Dial By Name feature. During Dial By Name call, the caller will need to input the first letters of First/Last name before this wait time is reached. Otherwise, timeout will occur, and the call might hang up. The timeout range is between 3 and 60 seconds.

  1. Query Type

Specify the query type. This defines how the caller will need to enter to search the directory.

By First Name: enter the first 3 digits of the first name to search the directory.

By Last Name: enter the first 3 digits of the last name to search the directory.

  1. Select Type

Specify the select type on the searching result. The IVR will confirm the name/number for the party the caller would like to reach before dialing out.

By Order: After the caller enters the digits, the IVR will announce the first matching party’s name and number. The caller can confirm and dial out if it is the destination party, or press * to listen to the next matching result if it is not the desired party to call.

By Menu: After the caller enters the digits, the IVR will announce 8 matching results. The caller can press number 1 to 8 to select and call or press 9 for results in next page.

The Dial by Name group can be used as the destination for inbound route and key pressing event for IVR. The group name defined here will show up in the destination list when configuring IVR and inbound route. If Dial by Name is set as a key pressing event for IVR, user could use ‘*’ to exit from Dial by Name, then re-enter IVR and start a new event. The following example shows how to use this option.

Dial By Name Group In IVR Key Pressing Events
Dial by Name Group In Inbound Rule

Please refer to [Username Prompt Customization] for Username Prompt Customization.

Speed Dial

Add Speed Dial

The CloudUCM supports Speed Dial feature that allows users to call a certain destination by pressing one or four digits on the keypad. This creates a system-wide speed dial access for all the extensions on the CloudUCM.

To enable Speed Dial, on the CloudUCM Web GUI, go to page Web GUI🡪Call Features🡪Speed Dial.

User should first click on . Then decide from one digit up to four digits combination used for Speed Dial and select a dial destination from “Default Destination”. The supported destinations include extension, voicemail, conference room, voicemail group, IVR, ring group, call queue, page group, DISA, Dial by Name and external number.

Note

The maximum number of speed dial entries that can be configured is 1000 speed dial entries.

Speed Dial Destinations
List of Speed Dial

Import Speed Dial

The user can import speed dial entries from a csv file, this reduces the amount of configuring the same speed dial entries on different UCMs. To do this, please click on “Import” as the figure below shows.

Import Dial Speed

Then select the csv file of the speed dial entries and click

Important

Please use UTF-8 encoding when importing a CSV file. CSV files can be opened using programs such as Notepad and saved as a UTF-8 encoded file.

Alert

Importing speed dial entries will overwrite the existing speed dials, if you wish to import new speed dial entries to the already existing ones, you will have to export them then combine them together in one file before you import it.

Export Speed Dial

To export speed dial entries, please click on export as the screenshot below shows, then choose the location where to save the csv file.

Export Dial Speed

DISA

In many situations, the user will find the need to access his own IP PBX resources, but he is not physically near one of his extensions. However, he does have access to his own cell phone. In this case, we can use what is commonly known as DISA (Direct Inward System Access). Under this scenario, the user will be able to call from the outside, whether it is using his cell phone, pay phone, etc. After calling into the CloudUCM, the user can then dial out via the SIP trunk connected to the CloudUCM as it is an internal extension.

The CloudUCM supports DISA to be used in IVR or inbound route. Before using it, create new DISA under Web GUI🡪Call Features🡪DISA.

  • Click on to add a new DISA.
  • Click on to edit the DISA configuration.
  • Click on to delete the DISA.
Create New DISA

The following table details the parameters to set and configure DISA feature on CloudUCM.

Name

Configure DISA name to identify the DISA.

Password

Configure the password (digit only) required for the user to enter before using DISA to dial out.

Note: The password must be at least 4 digits.

Permission

Configure the permission level for DISA. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level.
The default setting is "Internal". If the user tries to dial outbound calls after dialing into the DISA, the CloudUCM will compared the DISA's permission level with the outbound route's privilege level.
If the DISA's permission level is higher than (or equal to) the outbound route's privilege level, the call will be allowed to go through.

Response Timeout

Configure the maximum amount of time the CloudUCM will wait before hanging up if the user dials an incomplete or invalid number. The default setting is 10 seconds.

Digit Timeout

Configure the maximum amount of time permitted between digits when the user is typing the extension. The default setting is 5 seconds.

Allow Hangup

If enabled, during an active call, users can enter the CloudUCM hangup feature code (by default it is *0) to disconnect the call or hang up directly. A new dial tone will be heard shortly for the user to make a new call. The default setting is "No".

Replace Display Name

If enabled, the UCM will replace the caller display name with the DISA name.

Once successfully created, users can configure the inbound route destination as “DISA” or IVR key event as “DISA”. When dialing into DISA, users will be prompted with password first. After entering the correct password, a second dial tone will be heard for the users to dial out.

Callback

Callback is designed for users who often use their mobile phones to make long distance or international calls which may have high service charges. The callback feature provides an economic solution for reduce the cost from this.

The callback feature works as follows:

  1. Configure a new callback on the CloudUCM.
  2. On the CloudUCM, configure destination of the inbound route for callback.
  3. Save and apply the settings.
  4. The user calls number of the CloudUCM using the mobile phone, which goes to callback destination as specified in the inbound route.
  5. Once the user hears the ringback tone from the mobile phone, hang up the call on the mobile phone.
  6. The CloudUCM will call back the user.
  7. The user answers the call.
  8. The call will be sent to DISA or IVR which directs the user to dial the destination number.
  9. The user will be connected to the destination number.

In this way, the calls are placed and connected through trunks on the CloudUCM instead of to the mobile phone directly. Therefore, the user will not be charged on mobile phone services for long distance or international calls.

To configure callback on the CloudUCM, go to Web GUI🡪Call Features🡪Callback page and click on . Configuration parameters are listed in the following table.

Name

Configure a name to identify the Callback. (Enter at least two characters)

CallerID Pattern

Configure the pattern of the callers allowed to use this callback. The caller who places the inbound call needs to have the CallerID match this pattern so that the caller can get callback after hanging up the call.

Note: If leaving as blank, all numbers are allowed to use this callback.

Outbound Prepend

Configure the prepend digits to be added at before dialing the outside number. The number with prepended digits will be used to match the outbound route. ‘-’ is the connection character which will be ignored.

Delay Before Callback

Configure the number of seconds to be delayed before calling back the user.

Destination

Configure the destination which the callback will direct the caller to. Two destinations are available:

  • IVR

  • DISA

The caller can then enter the desired number to dial out via CloudUCM trunk.

Event List

Besides BLF, users can also configure the phones to monitor event list. In this way, both local extensions on the same CloudUCM and remote extensions on the VOIP trunk can be monitored. The event list setting is under Web GUI🡪Call Features🡪Event List.

  • Click on “Add” to add a new event list.
  • Sort selected extensions manually in the Eventlist
  • Click on to edit the event list configuration.
  • Click on to delete the event list.

URI

Configure the name of this event list (for example, office_event_list). Please note the URI name cannot be the same as the extension name on the CloudUCM. The valid characters are letters, digits, _ and -.

Local Extensions

Select the available extensions/Extension Groups listed on the local CloudUCM to be monitored in the event list.

Remote Extensions

If LDAP sync is enabled between the CloudUCM and the peer CloudUCM, the remote extensions will be listed under "Available Extensions". If not, manually enter the remote extensions under "Special Extensions" field.

Special Extensions

Manually enter the remote extensions in the peer/register trunk to be monitored in the event list. Valid format: 5000,5001,9000

Create New Event List

Remote extension monitoring works on the UCM via event list BLF, among Peer SIP trunks or Register SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the UCM first before using remote BLF feature. Please note the SIP end points need support event list BLF in order to monitor remote extensions.

When an event list is created on the UCM and remote extensions are added to the list, the UCM will send out SIP SUBSCRIBE to the remote UCM to obtain the remote extension status. When the SIP end points register and subscribe to the local UCM event list, it can obtain the remote extension status from this event list. Once successfully configured, the event list page will show the status of total extension and subscribers for each event list. Users can also select the event URI to check the monitored extension’s status and the subscribers’ details.

  • To configure LDAP sync, please go to CloudUCM Web GUI🡪Extension/Trunk🡪VoIP Trunk. You will see “Sync LDAP Enable” option. Once enabled, please configure password information for the remote peer UCM to connect to the local UCM. Additional information such as port number, LDAP outbound rule, LDAP Dialed Prefix will also be required. Both PBXs need enable LDAP sync option with the same password for successful connection and synchronization.
  • Currently LDAP sync feature only works between two UCM.
  • (Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is non-UCM PBX. However, it might not work the other way around depending on whether the non-UCM PBX supports event list BLF or remote monitoring feature.

Feature Codes

Feature Maps

Blind Transfer

- Default code: #1

- Enter the code during active call. After hearing "Transfer", you will hear dial tone. Enter the number to transfer to. Then the user will be disconnected, and transfer is completed.
- Options:

  • Disable

  • Allow Caller: Enable the feature code on caller side only.

  • Allow Callee: Enable the feature code on callee side only.

  • Allow Both: Enable the feature code on both caller and callee.

Attended Transfer

- Default code: *2
- Enter the code during active call. After hearing "Transfer", you will hear the dial tone. Enter the number to transfer to and the user will be connected to this number. Hang up the call to complete the attended transfer. In case of the called party does not answer, users could press *0 to cancel the call and retrieve the first call leg.
- Options:

  • Disable

  • Allow Caller: Enable the feature code on caller side only.

  • Allow Callee: Enable the feature code on callee side only.

  • Allow Both: Enable the feature code on both caller and callee.

Transfer Dialing Timeout Period (s)

Configures the dial timeout period of blind and attended transfers.

Seamless Transfer

  • Default code: *44 (Disabled by default).

  • Seamless Transfer allows user to perform blind transfer using UCM feature code without having music on hold presented during the transfer process, it minimizes the interruption during transfer, making the process smooth and simple.

  • During an active call use the feature code (*44 by default) followed by the number you want to transfer to in order to perform the seamless transfer.

Disconnect

- Default code: *0
- Enter the code during active call. It will disconnect the call.
- Options:

  • Disable

  • Allow Caller: Enable the feature code on caller side only.

  • Allow Callee: Enable the feature code on callee side only.

  • Allow Both: Enable the feature code on both caller and callee.

Call Park

- Default code: #72

- Enter the code during active call to park the call.
- Options:

  • Disable

  • Allow Caller: Enable the feature code on caller side only.

  • Allow Callee: Enable the feature code on callee side only.

  • Allow Both: Enable the feature code on both caller and callee.

Feature Code Input Timeout (ms)

Configure the maximum interval (ms) between digits for feature code activation.

Start/Stop Call Recording

-Default code: *3
- Enter the code followed by # or SEND to start recording the audio call and the PBX will mix the streams natively on the fly as the call is in progress.

- Options:

  • Disable

  • Allow Caller: Enable the feature code on caller side only.

  • Allow Callee: Enable the feature code on callee side only.

  • Allow Both: Enable the feature code on both caller and callee.

Enable Recording Whitelist

Enable the Recording Whitelist feature

Recording Operation Whitelist

Select extension in the whitelist that can use the *3 recording function.

Feature Code Digits Timeout

Set the maximum interval (ms) between digits for feature code activation

DND/Call Forward

Do Not Disturb (DND) Activate

Default code: *77

Do Not Disturb (DND) Deactivate

Default code: *78

Call Forward Busy Activate

- Default Code: *90
- Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.

Call Forward Busy Deactivate

Default Code: *91

Call Forward No Answer Activate

- Default Code: *92
-
Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.

Call Forward No Answer Deactivate

Default Code: *93

Call Forward Unconditional Activate

- Default Code: *72

Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.

Call Forward Unconditional Deactivate

Default Code: *73

Remote Call Forward Enable

Enable this option and configure the Remote Call Forward Whitelist below to allow specific extensions to dial the remote call forwarding feature codes to set call forwarding for any extension.

Remote DND / Call Forward Settings

Enable

Enable this option and configure the Whitelist below to allow specific extensions to dial feature codes to set DND or call forwarding for any extension.

Remote Call Forward Busy Enable

Configures and enables CFB for any extension.

Remote Call Forward No Answer Enable

Configures and enables CFNA for any extension.

Remote Call Forward Always Enable

Configures and enables CFU for any extension.

Remote DND Enable

Enables Do Not Disturb for any extension.

Remote Call Forward Busy Disable

Disables CFB for any extension.

Remote Call Forward No Answer Disable

Disables CFNA for any extension.

Remote Call Forward Always Disable

Disables CFU for any extension.

Remote DND Disable

Disables Do Not Disturb for any extension.

Whitelist

Extensions in this whitelist can configure DND or call forwarding for any extension via feature codes.

Feature Codes

Voicemail

Voicemail Access Code

- Default Code: *98
- Enter *98 and follow the voice prompt. Or dial *98 followed by the extension and # to access the entered extension's voicemail box.

My Voicemail

- Default Code: *97
- Press *97 to access the voicemail box.

Voicemail Group Access Code

Dial this code to access group voicemail. If password is required, enter password followed by the pound (#) key.

Direct Dial Voicemail Prefix

Prefix used to dial directly to voicemail.

Call Queue

Agent Pause

- Default Code: *83

- Pause the agent in all call queues.

Agent Unpause

- Default Code: *84

- Unpause the agent in all call queues.

Dynamic Agent Logout

Log the dynamic agent out of all queues.

Call Pickup

Pickup on Ringing Prefix

Picks up a ringing call for another extension.

Example: If the prefix is **, and there is a call ringing ext 1008, dial **1008 from a different extension to pick up the call to 1008.

Pickup In-call Prefix

Picks up an ongoing call for another extension.
Example: If the feature code is *45, and ext 1008 is in a call, dialing *45 and then 1008 following the prompt will take that call.
Note: The feature code user must be in the extension's Allowed to seamless transfer list to pick up calls for it.

Pickup Extension

This is the feature code to pick up incoming calls for other extensions in the same pickup group. The default setting is *8.

Call Barging

Enable Spy

Check this box to enable spy feature codes.

Listen Spy

This is the feature code to listen in on a call to monitor performance. Your line will be muted, and neither party will hear you. The default setting is *54.

Barge Spy

This is the feature code to join in on the call to assist both parties. The default setting is *56.

Whisper Spy

This is the feature code to speak to only one party in the call. For example, you could whisper to employees to help them handle a call. Only an employee on your account will be able to hear you. The default setting is *55.

PMS

PMS Wakeup Service

Dial this feature code to access PMS Wakeup Service. You can add, update, activate or deactivate PMS Wakeup Service.

PMS Remote Wakeup Service

Dial this code to add, update, activate, and deactivate PMS wakeup service for other extensions.

Update PMS Room Status

2 methods are available:

1. Dial the room status feature code + housekeeper code, listen to the prompt and then the dial the appropriate key for the desired room status. Example: The housekeeper with the housekeeper code 0001 dials *230001, listens to the room status options prompt, and then dials 1 to change room status to Available.

2. Dial room status feature code*housekeeper code*desired room status option key to quickly change the room status without needing to go through the system voice prompts. Example: The housekeeper with housekeeper code 0001 dials *23*0001*1 to change room status Available.

Misc

Paging Prefix

Configure the paging prefix for paging. For example, if the Paging Prefix is set to *81, dial *816000 to initiate a paging call to extension 6000.

Intercom Prefix

Configure the intercom prefix for intercom calls. For example, if the Intercom Prefix is set to *80, dial *806000 to initiate an intercom call to extension 6000.

Blacklist Add

Follow the voice prompt to add a caller ID to blacklist.

Blacklist Last Caller

Add the last inbound caller ID number to blacklist.

Blacklist Remove