Supported UC Endpoints and Client Devices: SIP endpoint, Wave
app for desktop (Windows 10+, macOS 10+), web browsers (Firefox,
Chrome, Safari, Edge, Opera), mobile (Android & iOS), Google
Chrome extensions
Call Features: Built-in SBC, Call with WebRTC Trunk,
Collaboration features

Customer Service Support: Integration with third-party
platforms, built-in live chat
CRM Integration: Supports various CRM systems and CloudUCM App
Store

Microsoft Integration: Supports integration with Microsoft
Teams, Outlook, AD Contact, Office 365
Computer Telephony Integration: CTI Mode for IP Phones, Headset
support

High Availability: Amazon Web Services, HA between UCM6300
Series IP PBX and CloudUCM
Firmware Upgrade and Provision: Supported by Grandstream Device
Management System (GDMS)

Product Usage Instructions

Supported UC Endpoints and Client Devices

The CloudUCM supports various UC endpoints including SIP
endpoints and the Wave app. It is compatible with desktops (Windows
10+, macOS 10+), web browsers (Firefox, Chrome, Safari, Edge,
Opera), and mobile devices (Android & iOS). Additionally, it
offers Google Chrome extensions for added functionality.

Call Features

The CloudUCM comes with built-in SBC services to enhance
security against external attacks. It also supports calls with
WebRTC trunks, allowing seamless communication via web browsers and
select mobile applications.

Collaboration

Enjoy a range of collaboration features including audio and
video meetings/conferences, instant messaging, group chats with
end-to-end encryption, file sharing, screen sharing, meeting
recordings, polls, surveys, and more. The system also supports
integration with third-party customer service platforms.

CRM Integration

The CloudUCM seamlessly integrates with various CRM systems such
as ACT!, Bitrix24, Freshdesk, Hubspot, Salesforce, Sugar, Vtiger,
Zendesk, Zoho, Dynamics 365, and more. It also supports custom
applications and cloud storage services for enhanced
productivity.

Microsoft Integration

Users can integrate the CloudUCM with Microsoft Teams, Outlook,
AD Contact, and Office 365 to streamline communication and
collaboration within Microsoft environments.

Computer Telephony Integration

The system supports CTI Mode to control IP phones from the GXP,
GRP, GXV, GHP series. Users can also connect wired or Bluetooth
headsets for hands-free communication.

High Availability

The CloudUCM offers high availability through Amazon Web
Services and ensures service continuity with HA setups between
UCM6300 Series IP PBX and CloudUCM systems.

Firmware Upgrade and Provision

Grandstream Device Management System (GDMS) facilitates firmware
upgrades and provisioning tasks through a cloud-based zero-touch
management system. Users can centrally manage and troubleshoot
Grandstream products with ease.

Frequently Asked Questions (FAQ)

Q: Can I use the CloudUCM with mobile devices?

A: Yes, the CloudUCM is compatible with mobile devices running
Android & iOS. You can access its features on the go.

Q: How secure is the CloudUCM system?

A: The CloudUCM comes with built-in SBC services for enhanced
security and protection against external attacks.

Q: Does the CloudUCM support integration with CRM systems?

A: Yes, the CloudUCM seamlessly integrates with various CRM
systems such as ACT!, Bitrix24, Freshdesk, Hubspot, Salesforce,
Sugar, Vtiger, Zendesk, Zoho, Dynamics 365, and more.

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CloudUCM – User Manual
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)
Call Center
Customizable Auto Attendant Property Management System (PMS) Cloud Storage
CloudUCM App Store

Supports integration with ACT!, Bitrix24, Freshdesk, Hubspot, Salesforce, Sugar, Vtiger, Zendesk, Zoho, Dynamics 365, and more.
Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ work-load, in-queue announcement.
Up to 5 layers of IVR (Interactive Voice Response) in multiple languages
Local PMS Supports Integration with Hmobile PMS Systems.
Included, varies by plan, additional add-ons available.
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 Android and iOS SDK Wave H5 Embedded for MAC/Windows application

Multi-Language Support Security

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

Network Protocols
Internet Protocol Standards
DTMF Methods Transmission Encryption Voice-over-Packet Capabilities Voice and Fax Codecs Video Codecs QoS
Administration

SIP, TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, HTTP/HTTPS, STUN, SRTP, TLS, LDAP, IPv4/6
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
In-band audio, RFC 2833, and SIP INFO
SRTP, DTLS-SRTP, TLS, HTTPS
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
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
H.264, H.263, H263+, VP8
Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
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

System Information

CloudUCM Dashboard

System Information
Remark
Active Calls
The active calls on the CloudUCM are displayed in the Web GUISystem StatusActive 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 GUISystem StatusActive 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, timezone settings, extension settings, trunks and routes configuration, etc.

EXTENSION/TRUNK

Setup Wizard

Extensions

Create a New SIP Extension
To manually create a new SIP user, go to Web GUIExtension/TrunkExtensions. 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 CallerID Number
Call Privileges SIP Password

The extension number associated with the user.
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.
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.
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.

Sync Contact

If enabled, this extension will be displayed in the CloudUCM and Wave contact list. If disabled, it will not be shown in the contact list, but Wave users will still be able to manually dial the extension number.
SIP Extension Configuration ParametersBasic Settings

DTMF Mode TEL URI

SIP Settings
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.
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 Enable T.38 UDPTL FECC Codec Preference
Jitter Buffer

When present in an INVITE request, the alert-Info header field specifies an alternative ring tone to the UAS.
Enable or disable T.38 UDPTL support.
Configure to enable Remote Camera Management.
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
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 Audio FEC Silence Suppression
SRTP
SRTP Crypto Suite

Check to enable Forward Error Correction (FEC) for Video.
Check to enable Forward Error Correction (FEC) for Audio.
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.
RTP Encryption
Enable SRTP for the extension. Disabled Enabled and Enforced Optional Note: By default, the SRTP is disabled.
The following encryption algorithms can be used to encrypt an RTP stream. AES_CM_128_HMAC_SHA1_80 (This is the default algorithm selected) AES_256_CM_HMAC_SHA1_80 AEAD_AES_128_GCM AEAD_AES_256_GCM

ZRTP Encryption Mode

ZRTP, also known as Media Path Key Agreement for Secure RTP, is an encryption protocol which allows negotiating the encryption key for RTP traffic. ZRTP uses Diffie-Hellman exchange to establish an encrypted and secure connection between the UCM and the SIP endpoint. If the SIP endpoint has both SRTP and ZRTP enabled, ZRTP will always be prioritized.
SIP Extension Configuration ParametersMedia

Presence Status

Call Transfer
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”,

Call Forward Unconditional CFU Time Condition Call Forward No Answer CFN Time Condition

“DND” and “Unavailable”. More details at [PRESENCE].
Internal Calls & External Calls
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”.
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 SettingsTime
SettingsOffice Time/Holiday page.
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”.
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 SettingsTime
SettingsOffice Time/Holiday page.

Call Forward Busy
CFB Time Condition Do Not Disturb DND Time Condition DND Whitelist FWD Whitelist Enable CC CC Mode CC Max Agents

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”.
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 SettingsTime
SettingsOffice Time/Holiday page.
If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.
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 SettingsTime SettingsOffice Time/Holiday page.
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.
Calls from users in the forward whitelist will not be forwarded. 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.
CC Settings
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.
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”.
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

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

Missed Call Type

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.

Enable SCA Emergency CID Language

If enabled, (1) Call Forward, Call Waiting and Do Not Disturb settings will not work, (2) Concurrent Registrations can be set only to 1, and (3) Private numbers can be added in Advanced Call Features->SCA page.
CallerID name and number that will be used when making emergency calls and receiving direct callbacks. If ELIN subnet mapping has been configured, and the extension is registered to a device in a mapped subnet, the configured ELIN will be used for CID number instead.
Select voice prompt language for this extension. If set to “Default”, the global setting for voice prompt language will be used.
SIP Extension Configuration ParametersFeatures

Specific Time

Time Condition

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

Normal Enable Wave

SIP Extension Configuration ParametersSpecific Time
Enable Wave for the specific extension.

Allow Concurrent Logins from the Same Client Type

Enables/disables the ability to login to Wave from different sessions on the same type of client. Note: This option is disabled by default.

Wave Welcome Email

Wave Welcome Email template.

Wave Permission Settings

Clicking the path will direct you to Wave Permission configuration.

Wave

Download Link

https://fw.gdms.cloud/wave/download/
SIP Extension Configuration ParametersWave

Enable
Skip Trunk Auth
Music On Hold Class
Confirm When Answering
Enable Destination
Default Destination
Use Callee DOD for Follow Me
Play Follow Me Prompt

Follow Me Configure to enable or disable Follow Me for this user. 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. Configure the Music On Hold class that the caller would hear while tracking the user. If enabled, call will need to be confirmed after answering. Configure to enable destination. The call will be routed to this destination if no one in the Follow Me answers the call.
Use the callee DOD number as CID if configured Follow Me numbers are external numbers. If enabled, the Follow Me prompt tone will be played.

New Follow Me Number
Dialing Order

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

SIP Extension Configuration ParametersFollow Me

Search and Edit Extension
All the CloudUCM extensions are listed under Web GUIExtension/TrunkExtensions, 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 to reset the extension parameters to default (except concurrent registration). Other settings will be restored to default in MaintenanceUser ManagementUser 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 Information includes:

Export Basic Settings

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 4. 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. 5. Click on “Choose file to upload” to select a CSV file from a local directory on the PC.

6. Click on “Apply Changes” to apply the imported file on the CloudUCM. Example of a file to import:

Field Extension

Supported Values Digits

Import File

Technology

SIP/SIP(WebRTC)

Enable Voicemail

yes/no/remote

CallerID Number

Digits

SIP Password

Alphanumeric characters

Voicemail Password

Digits

Skip Voicemail Password yes/no
Verification

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.72632,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

Field
Call Forward Unconditional

Supported Values 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 All time/Office time/out of office time/holiday/out of holiday/out of office time or

Simultaneously

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

Field Enable Wave

Supported Values Yes/no

Alert-Info

None/Ring 1/Ring2/Ring3/Ring 4/Ring 5/Ring 6/Ring 7/ Ring 8/Ring 9/Ring 10/bellcoredr1/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 GUISystem SettingsEmail 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 GUIExtension/TrunkEdit 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 GUIExtension/TrunkExtension Groups.

Click on Click on Click on

to create a new extension group. to edit the extension group. to delete the extension group.

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

Click on

Edit Extension Group 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 GUIExtension/TrunkOutbound Routes and select “Enable Source Caller ID Whitelist”. Both single extensions and extension groups will show up for users to select.

VoIP Trunks

Select Extension Group in Outbound Route

VoIP Trunk Configuration
VoIP trunks can be configured in CloudUCM under 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 Type

Check this box to disable this trunk.
Select the VoIP trunk type.
Peer SIP Trunk: A direct IP-to-IP connection between the PBX and another SIP server or device, without requiring registration.
Register SIP Trunk: A trunk that requires the PBX to register with the SIP server or provider using credentials (username and password).

Provider Name Host Name Dedicated VLAN Transport
Trunk Mode
DID Number Server Address Keep Original CID Keep Trunk CID NAT
TEL URI Need Registration

Account SIP Trunk: A trunk where the PBX acts as the registrar, allowing remote devices or endpoints to register with it.
Configure a unique label (up to 64 characters) to identify this trunk when listed in outbound rules, inbound rules, etc.
Configure the IP address or URL for the VoIP provider’s server of the trunk.
After selecting the corresponding VLAN, the traffic related to the relay will go through the VLAN interface.
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.
Set the trunk mode for incoming calls. In certain scenarios, service providers do not include a domain in “To” SIP header. In other scenarios, the service providers do not accept SIP INVITE messages from a different port than 5060. The trunk mode options allow to resolve such issues.
DID Access: When a domain is not included in “To” SIP header, the user can configure a DID which will be used to verify incoming calls.
Port Access: Choose this option to allow outbound SIP traffic to be sent from port 5060. Choosing this option will change the port used to receive SIP requests for this specific trunk to 6040, this should taken into consideration when interconnecting two PBXs
Note: This option is available only for Register Trunk and Peer Trunk when using UDP as transport protocol.
Enter the DID number which will be included in the “To” SIP header. This option is mandatory when Trunk Mode is set to DID Access.
Use the indicated address to register/peer trunks when configuring other PBXs with the CloudUCM.
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.
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”.
Enable this setting if the PBX is using public IP and communicating with devices behind NAT. Note 1: This setting will overwrite the Contact header of received messages, which may affect the ability to establish calls when behind NAT. Consider changing setttings in PBX Settings > SIP Settings > NAT instead. Note 2: If one is experiencing one-way audio issues, please check the NAT configuration and SIP/RTP ports in the firewall.
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.
Defines Whether to register the trunk on the external server. Enabled by default. Note: This option appears when the Type is set to Register Trunk.

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. Note: This option appears when the Type is set to Register Trunk.

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 PBX 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 PBX 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 PBX 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 PBX for outbound calls. Priority is based on order of configuration.

ZRTP Encryption Mode

If disabled, the PBX will not support ZRTP encryption. Otherwise, ZRTP will be supported, and if the registered endpoint supports both ZRTP and SRTP, ZRTP will be used first.

IPVT Mode

Similar to Enable Direct Media. The PBX will attempt to redirect the RTP media streams to bypass the PBX and to go directly between caller and callee. Primarily for use with trunks to IPVT.

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-AssertedIdentity (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:
1. On incoming calls, the Passthrough PAI Header value will be preferred for this PBX’s 18x and 200 SIP messages to the caller.
2. On outbound calls, the Send PAI Header value will be preferred for this PBX’s INVITE SIP message to the callee.

Passthrough PAI Header Send PANI Header Send Anonymous DID Mode
DTMF Mode

If enabled and “Send PAI Header” is disabled, PAI headers will be preserved as calls pass through the PBX.
If checked, the INVITE sent to the trunk will contain P-Access-Network-Info header.
If checked, the “From” header in outgoing INVITE message will be set to anonymous.
Configure to obtain the destination ID of an incoming SIP call from SIP Request-line or To header.
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.List Item 2
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 PBX. 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.

Disable This Trunk Type Provider Name Host Name Transport
Keep Original CID Keep Trunk CID
TEL URI
Need Registration Allow outgoing calls if registration fails

Trunk Group Configuration
Check this box to disable this trunk
Register Trunk
Configure a unique label to identify the trunk when listed in outbound rules and incoming rules.
Enter the IP address or hostname of the VoIP provider’s server.
Configure the SIP Transport method. Only TLS is supported, and TLS service must be enabled on the other end.
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.
Always use trunk CID if specified even if extension has DOD number or CID configured.
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.
Whether to register on the external server.
Uncheck to block outgoing calls 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 GUIExtension/TrunkVoIP 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.

7. Click “Save” at the bottom.

DOD extension selection

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

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

Edit DOD

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

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 GUIExtension/TrunkOutbound 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
Pattern
Disable This Route Password Local Country Code Call Duration Limit Maximum Call Duration Warning Time

Configure the name of the calling rule (e.g., local, long_distance, etc.). Letters, digits, _ and are allowed.
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.
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.
Configure the password for users to use this rule when making outbound calls.
If your local country code is affected by the outbound blacklist, please enter it here to bypass the blacklist.
Enable to configure the maximum duration for the call using this outbound route.
Configure the maximum duration of the call (in seconds). The default setting is 0, which means no limit.
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 Source Caller ID Allowlist”. The following two methods can be used at the same time to define the extensions as the source caller ID.
1. Select available extensions/extension groups from the list. This allows users to specify arbitrary single extensions available in the PBX.
2. 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: [123459] 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
Strip Prepend Time Condition Time Condition Mode

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. 10 failover trunks are supported.
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.
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 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
Don’t Call Me Blacklist Integration
Don’t Call Me database is a database on which people can register their numbers to prevent being called by marketers and salespersons. When CloudUCM is integrated with this database and one of the extensions dials a phone number, it will be verified in the database. In case the number exists in the database, the call will not be permitted. To access the integration page, please navigate to Extension/Trunk > Outbound Routes, then click on “Outbound Blacklist” button and click on Integrate Don’t Call Me Blacklist.

Parameter Integrate Don’t Call Me Blacklist Authorization Token Query Timeout Time (s)
Query Timeout Handling
Test Connection

Don’t Call Me Database Integration
Description Enable or disable Don’t Call Me integration Enter the authorization token generated by the Don’t Call Me database. Enter the duration after which the query is considered timed out. Select the action to perform after the query timeout. Allow Dialing: If the query times out, the call will be allowed. Prohibit Dialing: If the query times out, the call will be prohibited. Click on this button to test that the integration is working as intended. Note: If the database or Internet access is momentarily down, this test will fail.

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

Once the user clicks

Outbound Routes/PIN Group , 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/TrunkOutbound RoutesPIN Groups and click on the “Upload” button.

Importing PIN Groups from CSV files
2. Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in error messages such as the one below:

Incorrect CSV File 3. To ensure a successful import, please follow the format in the sample image below

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 GUIExtension/TrunkInbound 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

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.

Allowed DID Destination
Default Destination
Strip Prepend Time Condition Start Time End Time Date Week Destination

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.
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
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
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.
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.
Select the start time “hour:minute” for the trunk to use the inbound rule.
Select the end time “hour:minute” for the trunk to use the inbound rule.
Select “By Week” or “By Day” and specify the date for the trunk to use the inbound rule.
Select the day in the week to use the inbound rule.
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/TrunkInbound 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

4. 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 5. 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: Third-party Database Search
This feature allows the user to enter to integrate the UCM with a third-party database which contains the phone numbers and their matching names. When a call is received on a specific inbound route, the callerID will be checked against the database, if it’s found, then the corresponding name will be displayed.
Important Note This feature uses MySQL queries, therefore, it will function only with MySQL databases.
Inbound Routes Once the user clicks on “Third-party Database Search”, it will open the configuration page, as seen in the figure below.

Third-party Database Search

Third-party Enable or disable the feature.
Database Search

MySQL Host

Specifies the hostname or IP address of the MySQL server.

Database

The name of the MySQL database that stores caller information.

Username

Enter the username used to connect to the MySQL database.

MySQL Password Enter the password for the specified MySQL username.

Character Set

Specifies the character set for MySQL connections.

Query Key

Enter the 3 information of the target phonebook in the database, you can contact the database administrator to get the appropriate keywords for the query. For example, if the table name is “phonebook” , the caller name is “name” , the number is “number” , the SQL statement will be executed: SELECT name FROM phonebook WHERE number LIKE `% [NUMBER]%`;

Test Connection Test the connection to the database

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.

Blocklist Configurations

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

Select the checkbox for “Blocklist Enable” to turn on the Blocklist feature for all inbound routes. The blocklist is disabled by default.

Enter a number in the “Add Blocklist Number” field and then click “Add” to add to the list. Anonymous can also be added as a Blocklist Number by typing “Anonymous” in Add Blacklist Number field.

To remove a number from the Blocklist, select the number in the “Blocklist list” and click on button to remove all the numbers on the blocklist.

or click on the” Clear”

Users can also export the inbound route blocklist by pressing the

button.

Blocklist Configuration Parameters To add blocklisted numbers in batch, click on “Import” to upload the blocklist 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 Blocklist by dialing the feature code for “Blocklist Add’ (default: *40) and “Blocklist Remove” (default: *41) from an extension. The feature code can be configured under Web GUI > Basic Call Features > Feature Codes.
BASIC 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 > Basic 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 Configuration
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 Meeting Name Privilege Allow User Invite Allowed to Override Most Mute Auto Record

The number to dial to reach the meeting room.
Meeting Name
Please select the permission for outgoing calls.
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 Host Mute
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 Basic 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 Number of Meeting Rooms & Total Number of Participants

Plan

Startup

SOHO

Plus

Pro

Business

Number of Public Meeting Rooms

Total Number of Meeting Participants

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
Meeting Voice Indicator Sensitivity Meeting Audio Quality Meeting Record Prompt

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.
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.
Audio quality of multimedia meetings
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 GUIBasic 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

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

Mute/unmute yourself.

Lock/unlock the conference room.

Kick the last joined user from the conference.

Decrease the volume of the conference call.

Decrease your volume.

Increase the volume of the conference call.

7
8
Meeting User IVR Menu 1 4 5 6 7 8

Increase your volume. 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.
Mute/unmute yourself. Decrease the volume of the conference call. Decrease your volume. Increase the volume of the conference call. Increase your volume. 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 > Basic Call Features > Multimedia Meeting > Google Service Settings > Google Service Settings.

Google Service SettingsOAuth2.0 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 ServiceNew Project 2. Enable Calendar API from API Library. 3. Click “Credentials” on the left drop-down menu to create new OAuth2.0 login credentials.
Google ServiceCreate New Credential 4. Use the newly created login credential to fill in “OAuth2.0 Client ID” and “OAuth2.0 Client Secret”. 5. Click “Get Authentication Code” to obtain an authentication code from Google Service.

Google ServiceOAuth2.0 Login 6. 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.
Schedule Meeting
Log in to the UCM Web GUI, open the Basic Call Features Multimedia Meeting Meeting 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 Meeting Room Time Time Zone Password Host Password Host
Repeat

Configure the name of the scheduled meeting. Letters, digits, Other special characters are also supported. such as #%&@*=
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.
Configure the meeting date and time.
Select the meeting time zone.
Configure the meeting’s login password.
Configure the Host Password. Note: It is randomly generated when first creating a new meeting Schedule.
Configure Host.
Choose when to repeat a scheduled meeting. No Repeat Every Day Weekly Monthly Custom: it specifies how often the meeting is repeated per days/weeks. E.g., every 3
days/weeks.

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 51440. 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 PBX. 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.

Click on Click on

to edit the Meeting Schedule. to delete the Meeting Schedule.

Meeting details

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:

Click the button Click the button

Meeting Schedule Completed to download the Meeting Report of the meeting. 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 GUIBasic 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 GUIBasic 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 > Basic 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.

Basic Settings

Create New IVR

Name

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

Extensio Enter the extension number for users to access the IVR.
n

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.

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.

Permissi 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.

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

Dial Other Extensio ns

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

IVR Black/W hitelist

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

Internal Black/W hitelist

Contain numbers, either of Blacklist or Whitelist.

External Black/W hitelist

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 GUIPBX SettingsVoice PromptCustom 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.

Respons e 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.

Respons e 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.

Respons e 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.

Languag e

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 GUIPBX SettingsVoice PromptLanguage 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 GUIPBX SettingsVoice PromptLanguage Settings.

Key Pressing Events

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

Key Press Event: Press 0 Press 1 Press 2 Press 3 Press 4 Press 5 Press 6 Press 7 Press 8 Press 9 Press *

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.

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

Invalid Configure the destination when the Invalid Repeat Loop is done.

Time Conditio n

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 Basic Call FeaturesIVRCreate/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 GUIPBX SettingsVoice PromptCustom 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 > Basic Call Features > Voicemail.

Max Greeting Time (s)

Voicemail Settings
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 CallerID

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.

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.

Voicemail Remote Access

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 1 New messages
2 Change folders 3 Advanced options

Sub Menu 1
3 – Advanced options
5 – Repeat the current message 7 – Delete this message 8 – Forward the message to another user 9 Save * – Help # – Exit 0 – New messages 1 – Old messages 2 – Work messages 3 – Family messages 4 – Friend messages # – Cancel 1 – Send a reply

Sub Menu 2 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

2 – Call the person who sent this message 3 – Hear the message envelop 4 – Leave a message * – Return to the main menu 1 – Record your unavailable message
2 – Record your busy message
3 – Record your name
4 – Record temporary greeting 5 – Change your password * – Return to the main menu

1 – Accept this recording 2 – Listen to it 3 – Re-record your message 1 – Accept this recording 2 – Listen to it 3 – Re-record your message 1 – Accept this recording 2 – Listen to it 3 – Re-record your message 1 – Accept this recording 2 – Listen to it 3 – Re-record your message

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 Basic Call Features Voicemail Voicemail Group. Click on “Add” to configure the group.

Voicemail Group

Extension Name
Method
Voicemail Password Email Address Shared Voicemail Status

Enter the Voicemail Group Extension. The voicemail messages left to this extension will be forwarded to all the voicemail group members.
Configure the Name to identify the voicemail group. Letters, digits, _ and – are allowed.
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.
Configure the voicemail password for the users to check voicemail messages.
Configure the Email address for the voicemail group extension.
If enabled, voicemail group status can be monitored via BLF. Green indicates no unread voicemail, and red indicates existing unread voicemail.

Members Greet Prompt Temporary Prompt Unavailable Prompt

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.
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.
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.
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 GUIBasic Call FeaturesRing Group.

Ring Group

Click on Click on Click on

to add ring group. to edit the ring group. The following table shows the ring group configuration parameters. to delete the ring group.

Ring Group Name Extension Members LDAP Phonebook
Ring Strategy
Music On Hold Custom Prompt

Ring Group Configuration
Configure ring group name to identify the ring group. Letters, digits, _ and are allowed.
Configure the ring group extension.
Select available users from the left side to the ring group member list on the right side. Click on to arrange the order.
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.
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.
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.
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 recor

Documents / Resources

CloudUCM CloudUCM Cloud PBX Solution [pdf] User Manual
CloudUCM Cloud PBX Solution, Cloud PBX Solution, PBX Solution

References

User Manual

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