UCM630x Audio Series – User Manual

Thank you for purchasing Grandstream UCM630xA series IP PBX appliance. The UCM6300A series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on one centralized network, including voice, video calling, voice meeting, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300A series supports up to 1500 users and includes a built-in web meetings and meeting solution that allows employees to connect from the desktop, mobile, GVC series devices and IP phones. It can be paired with the UCM6300A ecosystem to offer a hybrid platform that combines the control of an on-premises IP PBX with the remote access of a cloud solution. The UCM630xA ecosystem consists of the Wave app for desktop and mobile, which provides a hub for collaborating remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections. The UCM6300A series also offers cloud setup and management through GDMS and an TableAPI for integration with third-party platforms. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, meeting and collaboration tools, the UCM6300A series provides a powerful platform for any organization.

PRODUCT OVERVIEW

Technical Specifications

The following table resumes all the technical specifications including the protocols / standards supported, voice codecs, telephony features, languages, and upgrade/provisioning settings for UCM630xA series.

Technical Specifications

INSTALLATION

Before deploying and configuring the UCM630xA series, the device needs to be properly powered up and connected to a network. This section describes detailed information on installation, connection, and warranty policy of the UCM630xA series.

Equipment Packaging

Main Case	1
Power Adaptor	1
Ethernet Cable	1
Quick Installation Guide	1

UCM6300A front and back view

UCM6302A front and back view

UCM6304A front and back view

UCM6308A front and back view

GETTING STARTED

To get started with the UCM630xA setup process, use the following available interfaces: LCD display, and web portal.

• The LCD display shows hardware, software, interface status and network information and can be navigated via the Slide control and Touch keys. From here, users can configure basic network settings, run diagnostic tests, and factory reset.
• The web portal (may also be referred to as web UI in this guide) is the primary method of configuring the UCM.

This section will provide step-by-step instructions on how to use these interfaces to quickly set up the UCM and start making and receiving calls with it.

Use the LCD Menu

• Idle Screen

Once the device has booted up completely, the LCD will show the UCM model, hardware version and IP address. Upon menu key press timeout (30 seconds), the screen will default back to this information.

• Menu

Pressing the Home button will show the main menu. All available menu options are found in the table [LCD Menu Options].

• Menu Navigation

Scrolling sown using slide control through the menu options. Press the OK button to select an option.

• Exit

Selecting the Back option will return to the previous menu. For the Device Info, Network Info, and Web Info screens that have no Back option, pressing the OK button will return to the previous menu.

• LCD Backlight

The LCD backlight will turn on upon button press and will go off when idle for 30 seconds.

The following table summarizes the layout of the LCD menu of UCM630x.

View Events	Critical Events Other Events
Device Info	Hardware: Hardware version number Software: Software version number P/N: Part number WAN MAC: WAN side MAC address LAN MAC: LAN side MAC address Uptime: System uptime
Network Info	WAN Mode: DHCP, Static IP, or PPPoE WAN IP: IP address WAN Subnet Mask LAN IP: IP address LAN Subnet Mask
Network Menu	WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE Static Route Reset: Select this to reset static route settings.
Factory Menu	Reboot Factory Reset LCD Test Patterns Press DOWN and OK buttons to scroll through and select different LCD patterns to test. Once a test is done, press the OK button to return to the previous menu. Fan Mode Select Auto or On. LED Test Patterns All On, All Off, and Blinking are the available options. Selecting Back in the menu will revert the LED indicators back to their actual status. RTC Test Patterns Select either 2022-02-22 22:22 or 2011-01-11 11:11 to start the RTC (Real-Time Clock) test pattern. Check the system time from either the LCD idle screen or in the web portal System Status🡪System Information🡪General page. To revert back to the correct time, manually reboot the device. Hardware Testing Select Test SVIP to verify hardware connections within the device. The result will display on the LCD when the test is complete.
Web Info	Protocol: Web access protocol (HTTP/ HTTPS). HTTPS is used by default. Port: Web access port number, which is 8089 by default.
SSH Switch	Enable SSH Disable SSH SSH access is disabled by default

Use the LED Indicators

The UCM6300A/6302A has LED indicators on the network port to display connection status and the following picture shows the other ports status.

The UCM6304A/6308A has LED indicators in the front to display connection status. The following table shows the status definitions.

LED Indicator	LED Status
Power 1/Power 2   PoE LAN WAN USB SD FXS ports FXO ports	Solid: Connected Fast Blinking: Data Transferring Slow Blinking: Trying to connect OFF: Not Connected

Using the Web UI

Accessing the Web UI

The UCM’s web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow users to configure the device through a web browser such Microsoft IE (version 8+), Mozilla Firefox, Google Chrome, etc. To access the UCM’s web portal, follow the steps below:

1. Make sure your computer is on the same network as the UCM.
2. Make sure that the UCM’s IP address is displayed on its LCD.
3. Enter the UCM’s IP address into a web browsers’ address bar. The login page should appear (please see the above image).
4. Enter default administrator username “admin” and password can be found on the sticker at the back of the UCM.

Setup Wizard

After logging into the UCM web portal for the first time, the setup wizard will guide the user through basic configurations such as time zone, network settings, trunks, and routing rules.

The setup wizard can be closed and reopened at any time. At the end of the wizard, a summary of the pending configuration changes can be reviewed before applying them.

New Feature Guide

The UCM630xA includes an interactive New Feature Guide to inform administrators of major platform updates upon upgrading system firmware. When logging in for the first time following a firmware upgrade, an illustrated modal window automatically appears to highlight the key enhancements and value-added capabilities introduced in the new release.

The interactive guide presents update highlights using visual card layouts with concise descriptions. Administrators can navigate through the card pages using the directional arrows or dismiss the dialog by clicking OK.

Once closed, the guide remains permanently accessible from the Web GUI header bar. To revisit the guide at any time, click on your username in the upper-right corner and select New Feature Guide from the drop-down menu.

Web GUI Languages

Currently the UCM630xA series Web GUI supports English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, Russian, Italian, Polish, German etc.

Users can select the UCM’s web UI display language in the top-right corner of the page.

Web GUI Search Bar

Users can search for options in the web portal with the search bar on the top right of the page.

Saving and Applying Changes

After making changes to a page, click on the “Save” button to save them and then the “Apply Changes” button that finalizes the changes. If a modification requires a reboot, a prompt will appear asking to reboot the device.

Setting Up an Extension

Power on the UCM630xA and your SIP endpoint. Connect both devices to the same network and follow the steps below to set up an extension.

1. Log into the UCM web portal and navigate to Extension/Trunk🡪Extensions
2. Click on the “Add” button to start creating a new extension. The Extension and SIP/IAX Password information will be used to register to this extension. To set up voicemail, the Voicemail Password will be required.
3. To register an endpoint to this extension, go into your endpoint’s web UI and edit the desired account. Enter the newly created extension’s number, SIP user ID, and password into their corresponding fields on the endpoint. Enter the UCM’s IP address into the SIP server field. If setting up voicemail, enter *97 into the Voice Mail Access Number field. This field may be named differently on other devices.
4. To access the extension’s voicemail, use the newly registered extension to dial *97 and access the personal voicemail system. Once prompted, enter the voicemail password. If successful, you will now be prompted with various voicemail options.
5. You have now set up an extension on an endpoint.

SYSTEM SETTINGS

This section will explain the available system-wide parameters and configuration options on the UCM630xA series. This includes settings for the following items: General Settings, HTTP server, network Settings, OpenVPN, DDNS Settings, Security Settings, LDAP server, Time settings, Email settings and TR-069.

General Settings

System administrators can prevent the UCM from making calls and/or writing to the data partition (e.g., CDR, recordings, etc.) once the system reaches a specified threshold of storage usage and CPU usage respectively. These options are located in the System Settings 🡪 General Settings page.

General Settings
Device Name	Configure the name of the UCM.
Enable CPU Flow Control	Enables the CPU flow control.
CPU Flow Control Threshold	Used to set the threshold generated by the CPU Flow Control. When the system CPU reaches the threshold, it will prohibit the new calls.   Default value is 90%.
Data Partition Write Threshold	Used to set a threshold to stop writing data partition. When the disk data partition reaches the threshold configured, the data partition writing will be stopped. Default value is 90%.

IM Settings

IM Settings

The IM Settings page allows administrators to globally configure instant messaging parameters and chat behavior for Wave client users across the system.

The table below describes the configuration parameters available under the IM Settings page:

Cloud IM Service

After enabling Cloud IM, it means that all IM data in Grandstream Wave is stored in the external server Cloud IM, and is no longer stored locally in UCM.GDMS can configure Cloud IM service for UCM devices. At this time, the UCM device synchronizes the configuration item information.

IM Server

If Enable IM Server Mode is toggled on, UCM will function only as an IM server. The UCM management portal will remove PBX related services and supports the binding of multiple cross-region UCM devices. The UCM device that wants to bind the IM server address is also bound by turning on the Cloud IM mode, and the IM data in his Grandstream Wave is stored in this IM server.

Company name	The entered company name
Server Address	The domain name or IP address of the Cloud IM server.
Service ID	The service ID of the Cloud IM server.
Key	The Key of the Cloud IM server.
Trusted User	The trusted user of the cloud IM. Only letter, number, and special characters are allowed.
Bound device information
Department	The department represented by the bound UCM.
MAC Address	MAC address of the bound UCM device.
Dial prefix	Extension prefix

HTTP Server

The UCM630X’s embedded web server responds to HTTPS GET/POST requests and allows users to configure the UCM via web browsers such as Microsoft IE, Mozilla Firefox, and Google Chrome. By default, users can access the UCM by just typing its IP address into a browser address bar. The browser will automatically be redirected to HTTPS using port 8089. For example, typing in “192.168.40.50” into the address bar will redirect the browser to “https://192.168.40.50:8089”. This behavior can be changed in the System Settings🡪HTTP Server page.

If the protocol or port has been changed, the user will be logged out and redirected to the new URL.

Network Settings

After successfully connecting the UCM630xA to the network for the first time, users could login the Web GUI and go to System Settings🡪Network Settings to configure the network parameters for the device.

• UCM630xA supports Route/Switch/Dual mode functions.

In this section, all the available network setting options are listed for all models. Select each tab in Web GUI🡪System Settings🡪Network Settings page to configure LAN settings, WAN settings, 802.1X and Port Forwarding.

Basic Settings

Please refer to the following tables for basic network configuration parameters on UCM6300A, UCM6302A, UCM6304A and UCM6308A, respectively.

• Method: Route

When the UCM630xA has, method set to Route in network settings, WAN port interface is used for uplink connection and LAN port interface is used as a router. Please see a sample diagram below.

• Method: Switch

WAN port interface is used for uplink connection; LAN port interface is used as room for PC connection.

• Method: Dual

Both WAN port and LAN port are used for uplink connection. Users will need assign LAN 1 or LAN 2 as the default interface in option “Default Interface” and configure “Gateway IP” if static IP is used for this interface.

802.1X

IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication mechanism to device before the device can access Internet or other LAN resources. The UCM630xA supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure show UCM630xA use 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet.

The following table shows the configuration parameters for 802.1X on UCM630xA. Identity and MD5 password are required for authentication, which should be provided by the network administrator obtained from the RADIUS server. If “EAP-TLS” or “EAP-PEAPv0/MSCHAPv2” is used, users will also need to upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated from the RADIUS server.

802.1X Mode	Select 802.1X mode. The default setting is “Disable”. The supported 802.1X mode are:   EAP-MD5 EAP-TLS EAP-PEAPv0/MSCHAPv2
Identity	Enter 802.1X mode Identity information.
MD5 Password	Enter 802.1X mode MD5 password information.
802.1X CA Certificate	Select 802.1X certificate from local PC and then upload.
802.1X Client Certificate	Select 802.1X client certificate from local PC and then upload.

Static Routes

The UCM630xA provides users static routing capability that allows the device to use manually configured routes, rather than information only from dynamic routing or gateway configured in the UCM630xA Web GUI🡪System Settings🡪Network Settings🡪Basic Settings to forward traffic. It can be used to define a route when no other routes are available or necessary, or used in complementary with existing routing on the UCM630xA as a failover backup, etc.

• 
  Click on “Add IPv4 Static Route” to create a new IPv4 static route or click on ”Add IPv6 Static Route” to create a new IPv6 static route. The configuration parameters are listed in the table below.
• Once added, users can select
  

  

  
  to edit the static route.
• Select
  

  

  
  to delete the static route.

Destination	Configure the destination IPv4 address or the destination IPv6 subnet for the UCM630xA to reach using the static route.   Example: IPv4 address – 192.168.66.4 IPv6 subnet – 2001:740:D::1/64
Subnet Mask	Configure the subnet mask for the above destination address. If left blank, the default value is 255.255.255.255.   Example: 255.255.255.0
Gateway	Configure the IPv4 or IPv6 gateway address so that the UCM630xA can reach the destination via this gateway. Gateway address is optional.   Example: 192.168.40.5 or 2001:740:D::1
Interface	Specify the network interface on the UCM630xA to reach the destination using the static route.   LAN interface is eth0; WAN interface is eth1.

Static routes configuration can be reset from LCD menu🡪Network Menu.

The following diagram shows a sample application of static route usage on UCM6304A.

The network topology of the above diagram is as below:

• Network 192.168.69.0 has IP phones registered to UCM6304A LAN 1 address
• Network 192.168.40.0 has IP phones registered to UCM6304A LAN 2 address
• Network 192.168.66.0 has IP phones registered to UCM6304A via VPN
• Network 192.168.40.0 has VPN connection established with network 192.168.66.0

In this network, by default the IP phones in network 192.168.69.0 are unable to call IP phones in network 192.168.66.0 when registered on different interfaces on the UCM6304A. Therefore, we need configure a static route on the UCM6304A so that the phones in isolated networks can make calls between each other.

Port Forwarding

The UCM network interface supports router function which provides users the ability to do port forwarding. If LAN mode is set to “Route” under Web GUI🡪System Settings🡪Network Settings🡪Basic Settings page, port forwarding is available for configuration.

The port forwarding configuration is under Web GUI🡪System Settings🡪Network Settings🡪Port Forwarding page. Please see related settings in the table below.

WAN Port	Specify the WAN port number or a range of WAN ports. Unlimited number of ports can be configured.   Note: When it is set to a range, WAN port and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access from WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000 will be port forwarding to LAN port 1000.
LAN IP	Specify the LAN IP address.
LAN Port	Specify the LAN port number or a range of LAN ports.   Note: When it is set to a range, WAN port and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access from WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000 will be port forwarding to LAN port 1000.
Protocol Type	Select protocol type “UDP Only”, “TCP Only” or “TCP/UDP” for the forwarding in the selected port. The default setting is “UDP Only”.

The following figures demonstrate a port forwarding example to provide phone’s Web GUI access to public side.

• UCM630xA network mode is set to “Route”.
• UCM630xA WAN port is connected to uplink switch, with a public IP address configured, e.g. 1.1.1.1.
• UCM630xA LAN port provides DHCP pool that connects to multiple phone devices in the LAN network 192.168.2.x. The UCM60X is used as a router, with gateway address 192.168.2.1.
• There is a GXP2160 connected under the LAN interface network of the UCM630xA. It obtains IP address 192.168.2.100 from UCM630xA DHCP pool.
• On the UCM630xA Web GUI🡪System Settings🡪Network Settings🡪Port Forwarding, configure a port forwarding entry as the figure shows below.

• 

  
  Click on

WAN Port: This is the port opened on the WAN side for access purpose.

LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM630xA.

LAN Port: This is the port opened on the GXP2160 side for access purpose.

Protocol Type: We select TCP here for Web GUI access using HTTP.

This will allow users to access the GXP2160 Web GUI from public side, by typing in public IP address (example: 1.1.1.1:8088).

ARP Settings

The ARP settings can be configured under Web GUI🡪System Settings🡪Network Settings🡪ARP Settings

ARP GC Threshold 1	Minimum number of entries to keep. Garbage collector will not purge entries if there are fewer than this number. The default value is 128.
ARP GC Threshold 2	Threshold when garbage collector becomes more aggressive about purging entries. Entries older than 5 seconds will be cleared when over this number. The default value is 512.
ARP GC Threshold 3	Maximum number of non-PERMANENT neighbor entries allowed. Increase this when using large numbers of interfaces and when communicating with large numbers of directly connected peers. The default value is 1024.

OpenVPN®

OpenVPN® settings allow the users to configure UCM630xA to use VPN features, the following table gives details about the various options in order to configure the UCM as OpenVPN Client.

DDNS Settings

DDNS setting allows user to access UCM630xA via domain name instead of IP address.

The UCM supports DDNS service from the following DDNS provider:

• dydns.org
• noip.com
• freedns.afraid.org
• zoneedit.com
• oray.net

Here is an example of using noip.com for DDNS.

1. Register domain in DDNS service provider. Please note the UCM630xA needs to have public IP access.

2. On Web GUI🡪System Settings🡪Network Settings🡪DDNS Settings, enable DDNS service and configure username, password, and host name.

3. Now you can use domain name instead of IP address to connect to the UCM630xA Web GUI.

Security Settings

The UCM630xA provides users firewall security configurations to prevent certain malicious attack to the UCM630xA system. Users could configure to allow, restrict, or reject specific traffic through the device for security and bandwidth purpose. The UCM630xA also provides Fail2ban feature for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE. To configure firewall settings in the UCM630xA, go to Web GUI🡪System Settings🡪Security Settings page.

Static Defense

Under Web GUI🡪System Settings🡪Security Settings🡪Static Defense page, users will see the following information:

• Current service information with port, process, and type.
• Typical firewall settings.
• Custom firewall settings.

The following table shows a sample current service status running on the UCM630xA.

Port	Process	Type	Protocol or Service
7777	Asterisk	TCP/IPv4	SIP
389	Slapd	TCP/IPv4	LDAP
6060	zero_config	UDP/IPv4	UCM630xA zero_config service
5060	Asterisk	UDP/IPv4	SIP
4569	Asterisk	UDP/IPv4	IAX
38563	Asterisk	udp/ipv4	SIP
10000	gs_avs	udp/ipv4	gs_avs
10001	gs_avs	udp/ipv4	gs_avs
10002	gs_avs	udp/ipv4	gs_avs
10003	gs_avs	udp/ipv4	gs_avs
10004	gs_avs	udp/ipv4	gs_avs
10005	gs_avs	udp/ipv4	gs_avs
10006	gs_avs	udp/ipv4	gs_avs
10007	gs_avs	udp/ipv4	gs_avs
10010	gs_avs	udp/ipv4	gs_avs
10012	gs_avs	udp/ipv4	gs_avs
10013	gs_avs	udp/ipv4	gs_avs
10014	gs_avs	udp/ipv4	gs_avs
10015	gs_avs	udp/ipv4	gs_avs
10018	gs_avs	udp/ipv4	gs_avs
10019	gs_avs	udp/ipv4	gs_avs
10020	gs_avs	udp/ipv4	gs_avs
6066	Python	udp/ipv4	python
3306	Mysqld	tcp/ipv4	mysqld
45678	Python	udp/ipv4	python
8439	Lighttpd	tcp/ipv4	HTTP
8088	asterisk	tcp/ipv4	SIP
8888	Pbxmid	tcp/ipv4	pbxmid
25	Master	tcp/ipv4	master
636	Slapd	tcp/ipv4	SLDAP
4569	asterisk	udp/ipv6	IAX
42050	asterisk	udp/ipv6	SIP
7681	Pbxmid	tcp/ipv4	pbxmid

For typical firewall settings, users could configure the following options on the UCM630xA.

Ping Defense Enable	If enabled, ICMP response will not be allowed for Ping request. The default setting is disabled. To enable or disable it, click on the check box for the LAN or WAN (UCM630xA) interface.
SYN-Flood Defense Enable	Allows the UCM630xA to handle excessive amounts of SYN packets from one source and keep the web portal accessible. There are two options available and only one of these options may be enabled at one time.   eth(0)LAN defends against attacks directed to the LAN IP address of the UCM630xA. eth(1)WAN defends against attacks directed to the WAN IP address of the UCM630xA. SYN Flood Defense will limit the amount of SYN packets accepted by the UCM from one source to 10 packets per second. Any excess packets from that source will be discarded.
Ping-of-Death Defense Enable	Enable to prevent Ping-of-Death attack to the device. The default setting is disabled. To enable or disable it, click on the check box for the LAN or WAN (UCM630xA) interface.

Under “Custom Firewall Settings”, users could create new rules to accept, reject or drop certain traffic going through the UCM630xA. To create new rule, click on “Create New Rule” button and a new window will pop up for users to specify rule options.

Right next to “Create New Rule” button, there is a checkbox for option “Reject Rules”. If it is checked, all the rules will be rejected except the firewall rules listed below. In the firewall rules, only when there is a rule that meets all the following requirements, the option “Reject Rules” will be allowed to check:

• Action: “Accept”
• Type “In”
• Destination port is set to the system login port (e.g., by default 8089)
• Protocol is not UDP

Rule Name	Specify the Firewall rule name to identify the firewall rule.
Action	Select the action for the Firewall to perform.   ACCEPT REJECT DROP
Type	Select the traffic type.   IN If selected, users will need specify the network interface “LAN” or “WAN” (for UCM630xA) for the incoming traffic. OUT
Interface	Select the interface to use the Firewall rule.
Service	Select the service type.   FTP SSH Telnet HTTP LDAP Custom If “Custom” is selected, users will need specify Source (IP and port), Destination (IP and port) and Protocol (TCP, UDP or Both) for the service. Please note if the source or the destination field is left blank, it will be used as “Anywhere”.
Source IP Address and Port	Configure a source subnet and port. If set to “Anywhere” or left empty, traffic from all addresses and ports will be accepted. A single port or a range of ports can be specified (e.g., 10000, 10000-20000).
Destination IP Address and Port	Configure a destination subnet and port. If set to “Anywhere” or left empty, traffic can be sent to all addresses and ports. A single port or a range of ports can be specified (e.g., 10000, 10000-20000).
Protocol	Select the protocol for the rule to be used.

Save the change and click on “Apply” button. Then submit the configuration by clicking on “Apply Changes” on the upper right of the web page. The new rule will be listed at the bottom of the page with sequence number, rule name, action, protocol, type, source, destination, and operation. More operations below:

• Click on
  

  

  
  to edit the rule.
• Click on
  

  

  
  to delete the rule.

Dynamic Defense

Dynamic defense is supported on the UCM630xA series. It can blacklist hosts dynamically when the LAN mode is set to “Route” under Web GUI🡪System Settings🡪Network Settings🡪Basic Settings page. If enabled, the traffic coming into the UCM630xA can be monitored, which helps prevent massive connection attempts or brute force attacks to the device. The blacklist can be created and updated by the UCM630xA firewall, which will then be displayed in the web page. Please refer to the following table for dynamic defense options on the UCM630xA.

Dynamic Defense Enable	Enable dynamic defense. The default setting is disabled.
Blacklist Update Interval	Configure the blacklist update time interval (in seconds). The default setting is 120.
Connection Threshold	Configure the connection threshold. Once the number of connections from the same host reaches the threshold, it will be added into the blacklist. The default setting is 100.
Dynamic Defense Whitelist	Allowed IPs and ports range, multiple IP addresses and port range.   For example: 192.168.2.100-192.168.2.105, 1000:9999

The following figure shows a configuration example like this:

• If a host at IP address 192.168.5.7 initiates more than 20 TCP connections to the UCM630xA it will be added into UCM630xA blacklist.
• This host 192.168.5.7 will be blocked by the UCM630xA for 500 seconds.
• Since IP range 192.168.5.100-192.168.5.200 is in whitelist, if a host initiates more than 20 TCP connections to the UCM630xA it will not be added into UCM630xA blacklist. It can still establish TCP connection with the UCM630xA.

Geo-IP Access Control

This feature allows administrators to control access to the UCM based on the source country or region of incoming IP addresses. When enabled, access from selected countries can be restricted for all external services, including Web GUI access, API access, SIP signaling, and extension registration.

The country/region IP database can be updated manually or automatically to ensure accurate IP-to-location mapping.

Country/Region IP Defense supports two enforcement methods, which determine how and when traffic from blocked countries is handled.

1. Non-Strict Defend Mode: In this mode, Country/Region IP Defense is enforced with business priority. Traffic from blocked countries is rejected at the application/service layer. The UCM returns explicit responses (for example, SIP error responses or HTTP error pages). Services remain reachable, but access is denied.
   
2. Strict Defend Mode: In this mode, Country/Region IP Defense is enforced with security priority. Traffic from blocked countries is dropped at the network level, and no SIP, HTTP, or HTTPS response is returned. Requests do not reach the Web server, SIP stack, API services, or Wave services.

Fail2ban

Fail2Ban feature on the UCM630xA provides intrusion prevention for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE. Once the entry is detected within “Max Retry Duration”, the UCM630xA will act to forbid the host for a certain period as defined in “Banned Duration”. This feature helps prevent SIP brute force attacks on the PBX system.

The table below descibes the main parameters of the Fail2Ban feature:

SSH Access

SSH switch now is available via Web GUI and LCD. User can enable or disable SSH access directly from Web GUI or LCD screen. For web SSH access, please log in UCM630xA web interface and go to Web GUI🡪System Settings🡪Security Settings🡪SSH Access.

The “Enable SSH access” option is for system debugging. If you enable this option, the system will allow SSH access. The SSH connection also requires the username and password of the super administrator. This option is turned off by default. It is recommended to turn off this option when debugging is not required.

Tick “Enable remote SSH” option, the system will allow remote SSH access via the GDMS platform. This option is turned off by default, and it is strongly recommended to turn off this option when remote troubleshooting is not required.

Data/File Encryption

The UCM6300 Series offers encryption for your data stored on the internal storage space of the UCM or on the attached storage units (e.g., SD Card, USB HDD/SSD, USB Flash Drive). This feature will harden the security and confidentiality of your data, which renders it impossible to read data in case they end up acquired by unauthorized parties.

Enabling Data/File Encryption

To choose which files are going to encrypted, use the following settings to enable the encryption for specific types of data.

Once the user chooses which data to encrypt, the user can save the setting. This means that any data created after this configuration will be encrypted, however, all the data which has been created prior to this configuration is not encrypted. To encrypt it please see the sections below.

Encrypting/Decrypting Files

When the user click on “Export” button, a CSV file which contains the secret key will be downloaded. This key can be stored safely and used to decrypt the data later on.

You can update the secret key using the “Update” button, this would require entering the administrator’s password to allow the user to change the secret key.

In the following section, the user can view the status of the files, whether they are encrypted or not. The user can also select the different data types to encrypt. This will encrypt the existing data.

The user can select the types of stored files to encrypt, then use the button to start the encryption. Or you can encrypt the files using the following encrypting button .

To open and play encrypted audio files outside of the UCM system, users need a decryption tool (found on the Grandstream Tools page), a key file and the password for it (set when exporting the key file).

For more information on how to use the decryption tool, please refer to the following guide :

UCM6300/A Series IP PBX – CDR Tool Guide

LDAP Server

The UCM630xA has an embedded LDAP/LDAPS server for users to manage corporate phonebook in a centralized manner.

• By default, the LDAP server has generated the first phonebook with PBX DN “ou=pbx,dc=pbx,dc=com” based on the UCM630xA user extensions already.
• Users could add new phonebook with a different Phonebook DN for other external contacts. For example, “ou=people,dc=pbx,dc=com”.
• All the phonebooks in the UCM630xA LDAP server have the same Base DN “dc=pbx,dc=com”.

Term Explanation:

cn= Common Name

ou= Organization Unit

dc= Domain Component

These are all parts of the LDAP data Interchange Format, according to RFC 2849, which is how the LDAP tree is filtered.

If users have the Grandstream phone provisioned by the UCM630xA, the LDAP directory will be set up on the phone and can be used right away for users to access all phonebooks.

Additionally, users could manually configure the LDAP client settings to manipulate the built-in LDAP server on the UCM630xA. If the UCM630xA has multiple LDAP phonebooks created, in the LDAP client configuration, users could use “dc=pbx,dc=com” as Base DN to have access to all phonebooks on the UCM630xA LDAP server, or use a specific phonebook DN, for example “ou=people,dc=pbx,dc=com”, to access to phonebook with Phonebook DN “ou=people,dc=pbx,dc=com ” only.

UCM can also act as a LDAP client to download phonebook entries from another LDAP server.

To access LDAP server and client settings, go to Web GUI🡪Settings🡪LDAP Server.

LDAP Server Configurations

The following figure shows the default LDAP server configurations on the UCM630xA.

The UCM630xA LDAP server supports anonymous access (read-only) by default. Therefore, the LDAP client does not have to configure username and password to access the phonebook directory. The “Root DN” and “Root Password” here are for LDAP management and configuration where users will need provide for authentication purpose before modifying the LDAP information.

The default phonebook list in this LDAP server can be viewed and edited by clicking on

for the first phonebook under LDAP Phonebook.

The UCM630xA support secure LDAP (LDAPS) where the communication is encrypted and secure.

LDAP Phonebook

Users could use the default phonebook, edit the default phonebook, add new phonebook, import phonebook on the LDAP server as well as export phonebook from the LDAP server.

The user can use the search bar on the top right corner of page, please see the screenshot below.

You can enter the extension number or the name of the contact; if you are unsure about the name or the extension, the search feature supports fuzzy matching.

• Add new phonebook

A new sibling phonebook of the default PBX phonebook can be added by clicking on “Add” under “LDAP Phonebook” section.

Configure the “Phonebook Prefix” first. The “Phonebook DN” will be automatically filled in. For example, if configuring “Phonebook Prefix” as “people”, the “Phonebook DN” will be filled with “ou=people,dc=pbx,dc=com”.

Once added, users can select

to edit the phonebook attributes and contact list (see figure below) or select

to delete the phonebook.

• Import phonebook from your computer to LDAP server

Click on “Import Phonebook” and a dialog will prompt as shown in the figure below.

The file to be imported must be a CSV, VCF or XML file with UTF-8 encoding. Users can open the file with Notepad and save it with UTF-8 encoding.

Here is how a sample file looks like. Please note “Account Number” and “Phonebook DN” fields are required. Users could export a phonebook file from the UCM630xA LDAP phonebook section first and use it as a sample to start with.

The Phonebook DN field is the same “Phonebook Prefix” entry as when the user clicks on “Add” to create a new phonebook. Therefore, if the user enters “phonebook” in “Phonebook DN” field in the CSV file, the actual phonebook DN “ou=phonebook,dc=pbx,dc=com” will be automatically created by the UCM630xA once the CSV file is imported.

In the CSV file, users can specify different phonebook DN fields for different contacts. If the phonebook DN already exists on the UCM630xA LDAP Phonebook, the contacts in the CSV file will be added into the existing phonebook. If the phonebook DN does not exist on the UCM630xA LDAP Phonebook, a new phonebook with this phonebook DN will be created.

The sample phonebook CSV file in above picture will result in the following LDAP phonebook in the UCM630xA.

As the default LDAP phonebook with DN “ou=pbx,dc=pbx,dc=com” cannot be edited or deleted in LDAP phonebook section, users cannot import contacts with Phonebook DN field “pbx” if existed in the CSV file.

• Export phonebook to your computer from UCM630xA LDAP server

Select the checkbox for the LDAP phonebook and then click on “Export Selected Phonebook” to export the selected phonebook. The exported phonebook can be used as a record or a sample CSV, VFC or XML file for the users to add more contacts in it and import to the UCM630xA again.

LDAP Settings

Prerequisites to support contacts sync-up to IP Phones, UCM needs to support the following:

1. If Cloud IM is enabled, UCM can send remote UCM’s contacts to each end device.
2. Contacts from remote UCM can be synced by Cloud IM or LDAP sync via trunk. The contacts data must be complete and consistent.
3. If Cloud IM is enabled, the contacts sent from UCM to end device should integrate Cloud IM contacts.
4. If Cloud IM is disabled, the contacts sent from UCM to end device should only contain contacts on the UCM.

To support contacts sync-up to Wave, it allows Wave to obtain enterprise contacts from Cloud IM or LDAP. On UCM SIP peer trunk, if LDAP sync is enabled, end point can obtain remote UCM extensions’ info via LDAP. Also, it will allow configuring whether to sync up LDAP contacts on Wave so that Wave doesn’t receive duplicate contacts info.

Under UCM webUI🡪 System Settings🡪 LDAP Server, click on “LDAP Settings”, option “Wave enable LDAP phonebook” is available for configuration. If enabled, all Wave users on this UCM will display LDAP contacts. Otherwise, it will not display.

Please note the LDAP contacts displayed on Wave will exclude the duplicate contacts from Cloud IM.

LDAP Client Configurations

The configuration on LDAP client is useful when you use other LDAP servers. Here we provide an example on how to configure the LDAP client on the UCM.

Assuming the remote server base dn is “dc=pbx,dc=com”, configure the LDAP client as follows:

The UCM can automatically update the phonebook, by configuring the ‘LDAP Automatic Update Cycle’. Available options are: 1 day/2days/7 days. It is set to ‘None’ by default.

The following figure gives a sample configuration for UCM acting as a LDAP client.

To configure Grandstream IP phones as the LDAP clients for UCM, please refer to the following example:

• Server Address: The IP address or domain name of the UCM
• Base DN: dc=pbx,dc=com
• Username: Please leave this field empty
• Password: Please leave this field empty
• LDAP Name Attribute: CallerIDName Email Department FirstName LastName
• LDAP Number Attribute: AccountNumber MobileNumber HomeNumber Fax
• LDAP Number Filter: (AccountNumber=%)
• LDAP Name Filter: (CallerIDName=%)
• LDAP Display Name: AccountNumber CallerIDName
• LDAP Version: If existed, please select LDAP Version 3
• Port: 389

The following figure shows the configuration information on a Grandstream GXP2170 to successfully use the LDAP server as configured in [Figure: LDAP Server Configurations].

AD Client Type

Time Settings

Automatic Date and Time

The current system time on the UCM630xA can be found under Web GUI🡪System Status🡪Dashboard🡪PBX Status.

To configure the UCM630xA to update time automatically, go to Web GUI🡪System Settings🡪Time Settings🡪Automatic date and Time.

Set Date and Time

To manually set the time on the UCM630xA, go to Web GUI🡪System Settings🡪Time Settings🡪Set Date and Time. The format is YYYY-MM-DD HH:MM:SS.

NTP Server

The UCM630xA can be used as an NTP server for the NTP clients to synchronize their time with. To configure the UCM630xA as the NTP server, set “Enable NTP server” to “Yes” under Web GUI🡪System Settings🡪Time Settings🡪NTP Server. On the client side, point the NTP server address to the UCM630xA IP address or host name to use the UCM630xA as the NTP server.

Office Time

On the UCM630xA, the system administrator can define “office time”, which can be used to configure time condition for extension call forwarding schedule and inbound rule schedule. To configure office time, go to Web GUI🡪System Settings🡪Time Settings🡪Office Time. Click on “Add” to create an office time.

Start Time	Configure the start time for office hour.
End Time	Configure the end time for office hour
Week	Select the workdays in one week.
Show Advanced Options	Check this option to show advanced options. Once selected, please specify “Month” and “Day” below.
Month	Select the months for office time.
Day	Select the workdays in one month.

Select “Start Time”, “End Time” and the day for the “Week” for the office time. The system administrator can also define month and day of the month as advanced options. Once done, click on “Save” and then “Apply Change” for the office time to take effect. The office time will be listed in the web page as the figure shows below.

• Click on
  

  

  
  to edit the office time.
• Click on
  

  

  
  to delete the office time.
• Click on “Delete” to delete multiple selected office times at once.

Holiday

On the UCM, the system administrator can define Holidays, which can be used by Inbound Routes, Call Forwarding, and other time based features such as Paging/Intercom. To configure holidays, log in to the Web GUI and navigate to System Settings→Time Settings→Holiday.

From this page, the administrator can:

• Configure new new holidays using the “Add” button.
• Edit existing holidays by clicking the “Edit” icon for a specific holiday.
• Delete a single holiday by clicking the “Delete” icon.
• Delete multiple holidays at once by selecting them and clicking the “Delete” button.
• Import holiday configurations using the “Import” button.
• Export holiday configurations using the “Export” button.

When creating/editing a holiday, users can configure the parameters described in the table below:

Parameter	Description
Name	Specifies the name of the holiday. Used to identify the holiday in related features.
Description	Optional description for the holiday.
Custom Prompt	Selects a custom voice prompt or generate a prompt using AI to play during the holiday period for inbound calls or time-based routing rules. Custom prompts can be managed under PBX Settings → Voice Prompt → Custom Prompt.
Year	Specifies the year in which the holiday applies. Select All to configure a recurring annual holiday.
Month	Select the month in which the holiday occurs.
Configuration Method	Defines how the holiday date is determined. The options are: By Date: Used for holidays with a fixed calendar date. By Week: Used for holidays defined by a weekday pattern within a month (for example, the fourth Thursday of a month).
Day	Displayed when Configuration Method is “By Date”. Choose one or more calendar dates within the selected month to be treated as holidays.
Sequence	Displayed when Configuration Method is “By Week”. Select the occurrence of the weekday within the month, such as First, Second, Third, Fourth, or Fifth.
Week	Displayed when Configuration Method is “By Week”. Select the day of the week (Sunday through Saturday).
Time	Specifies the effective time range of the holiday. The holiday is active from the start time to the end time on the configured date(s).

Custom Time Groups

Users can create custom time frames which can be used as a routing condition in the inbound routes. Multiple time ranges can be added and the frequency can be customized to be every specific weekday or every specific day/week of the selected months.

To access Custom Time Groups, please navigate to System Settings > Time Settings > Custom Time Groups

Email Settings

Email Settings

The Email application on the UCM630xA can be used to send out alert event Emails, Voicemail (Voicemail-To-Email) etc. The configuration parameters can be accessed via System Settings > Email Settings > Email Settings.

The following figure shows a sample Email setting on the UCM630XA, assuming the Email is using 192.168.6.202 as the SMTP server.

Once the configuration is finished, click on “Test”. In the prompt, fill in a valid Email address to send a test Email to verify the Email settings on the UCM630xA.

Email Templates

Users can customize email templates for password reset, voicemail, meeting scheduling, extensions, fax,

meeting report, PMS, CDR, emergency call, missed calls, alert events, call queue statistics and etc.

• Click on

icon to edit the template.

Email Footer Hyperlink

Under UCM Web GUI🡪 System Settings🡪 Email Settings🡪 Email Footer Hyperlink, users could edit the text and URL to modify the email footer hyperlink.

Email Send Log

Under UCM Web GUI🡪System Settings🡪Email Settings🡪Email Send Log, users could search, filter and check whether the Email is sent out successfully or not. This page will also display the corresponding error message if the Email is not sent out successfully.

Field	Description
Start Time	Enter the start time for filter
End Time	Enter the end time for filter
Receivers	Enter the email recipient, while searching for multiple recipients, please separate then with comma and no spaces.
Send Result	Enter the status of the send result to filter with
Return Code	Enter the email code to filter with
Email Send Module	Select the email module to filter with from the drop-down list, which contains:   All Modules Extension Voicemail Meeting Schedule User Password Alert Events CDR Test

Email logs will be shown on bottom of the “Email Send Log” page, as shown on the following figure.

Below are the codes returned when sending emails and their description:

Code	Description
250	Mail sent successfully
501	Address format parsing error, 501 will be returned when there are unacceptable characters in the recipient’s email address in MTA mode. Please check if the recipient’s email address format is correct. The “sender” configured on the client is your mail account.
535	The user name and password verification in the client mode is incorrect. Please check whether the user name and password are configured correctly.
550	Possible reasons:   1. The recipient’s mailbox user name does not exist or is in a banned state, please check whether the email recipient is the correct email address. 2. The number of destination addresses sent by the sender exceeds the maximum limit per day and is temporarily blacklisted. Please reduce the sending frequency or try again the next day. 3. The sender’s IP does not pass the SPF permission test of the sending domain. Emails sent in MTA mode may return this error code even if they are sent.
552	The sent email is too large or the email attachment type is prohibited
553	The sender and the email account are inconsistent, please configure the sender as your email account correctly.
554	The email was identified as spam. Please reduce the sending frequency or try again the next day
none	Indicates that there is no return code.   If the sending result is “deferred”, the general reason is that the mail service area is configured incorrectly. Please check whether the server configuration is correct. If the sending result is “bounced”, the general reason is that the receiving email address domain name is wrong, please check whether the email recipient is the correct email address. If it is in MTA mode, please check whether the “domain” is configured to be in the same domain name as the “recipient”.

SMS Settings

SMS Configuration

Configuring the SMS feature on the UCM6300 series allows the administrators to enable two-factor authentication and to send alerts and meeting notices.

The UCM supports SMS integration through multiple third-party service providers, including Twilio, Amazon SNS, VoiceMeUp, and Voxtelesys.

When SMS is enabled, users can configure the following parameters based on the selected carrier:

SMS Carrier	Parameter	Description
Twilio	Account ID	The Account SID provided in your Twilio Console dashboard used to authenticate API requests.
	Auth Token	The secret authentication token associated with your Twilio Account SID.
	Messages Server ID	The Messaging Service SID created within Twilio for handling outgoing message pools.
	From	The authorized Twilio phone number (including country code) assigned to send outbound SMS messages.
Amazon	Region	The AWS region where your instance is hosted (e.g., us-east-1).
	Account ID	The AWS Access Key ID generated from the AWS IAM console for API access.
	Secret	The AWS Secret Access Key corresponding to the Access Key ID used for request signing.
VoiceMeUp	Account ID	The primary account or username credential assigned by VoiceMeUp for portal and API access.
	Auth Token	The API authentication token generated within the VoiceMeUp portal under API settings.
	From	The SMS-enabled VoiceMeUp DID or phone number configured as the originating sender ID.
Voxtelesys	API Key	The bearer/API authorization key generated in the Voxtelesys Portal under API Credentials.
	From	The SMS-enabled Voxtelesys phone number assigned to your account used as the sender ID.

SMS Template

SMS templates can be managed under the SMS Template tab. SMS templates are subject to carrier specifications. Please refer to operator requirements before sending outbound messages. Additional details can be found in the documentation for Amazon, Twilio, VoiceMeUp, and Voxtelesys.

SMS Templates are categorized into two sections:

• System SMS Templates: Includes default templates for system-level notifications, such as Verification Code and Alarm Notification.
• Custom SMS Templates: Allows administrators to create custom SMS templates containing URLs or informational text. These templates can be assigned as destinations under IVR Key Events. When a caller presses a designated key during an IVR call, the system automatically sends the custom SMS message containing the configured link or text directly to the caller’s mobile device.

To configure IVR SMS destination, please navigate to Basic Call Features→IVR→Add/Edit IVR→Key Pressing Events.

SMS Delivery Log

All the SMS messages sent will be logged in the following tab.

HA

Dual-system hot standby provides a highly reliable and fault-tolerant solution for enterprises using the UCM6300 series/UCM6300A series. Based on two UCM devices of the same product model and software version, one of them is in the “Active” working state in real-time, and the other is in the “Standby” working state. The daily data on the host server will be synchronized to the standby machine in real-time, and the standby machine always monitors the running status of the host. When the host fails, including hardware failures and severe software failures, the standby machine will immediately take over the business and enter the “Active” working state, and Upgrade to a host to ensure that the business is not interrupted, and the call will automatically resume.

The HA function provides two modes of operating. The first mode of function is Local Hot Standby, which offers a deployment that switches dynamically when the primary UCM encounters an issue. The second mode is Remote Disaster Recover, it offers deploying a back up UCM remotely, this offers an architecture that would not be affected by any disaster that might occur on the geolocation of the primary UCM.

HA settings

The users can configure the HA under System Settings→HA settings page.

This table describes parameters used when configuring local HA between two UCM systems within the same site.

Local Hot Standby

The table below describes parameters used for Remote Disaster Recovery HA, where two UCM systems are deployed at different sites.

Remote Disaster Recovery

HA Status

Once the HA is configured, the user can view its status under System Settings 🡪 HA 🡪 HA Status as shown below

HA Log

The user can view the HA log through the system settings 🡪 HA 🡪 HA log page. The HA log effectively records the execution results of past full backup actions, as well as the historical records that triggered the active/standby switchover.

SNMP

UCM63xx supports SNMP in case the system administrator chooses to use third party monitoring tools. These are the options available when setting up SNMP.

SNMP Settings

SNMP Community

You can also create SNMP communities and affect a certain level of access. An SNMP community is a group created to aggregate many management stations. The community name is used to authenticate and identify these machines in the NMS (Network Management System).

SNMP Trap Destinations

SNMP Traps is a very useful feature when there are many network components to manage. Instead of sending requests to all the machines in the network in order to view their SNMP logs risking slowing down or bringing the network to a complete halt, SNMP Traps can be configured so these machines can send unrequested messages to the manager to notify it about critical events and general failures.

SNMP Version 3

UCM 63xx also supports SNMP v3 in case the system administrator decides to add more security to the monitoring process. SNMP v3 is a very good solution to monitor devices that interface directly with Internet. SNMP v3 offers more security than its predecessors by hashing the authentication information, encrypting the SNMP messages exchanged between the managed devices and the network management system which prevent eavesdropping. Also, it prevents any data tampering which protects the integrity of the data exchanged.

SNMP Trap Proxy

RADIUS

The UCM6300 offers Radius-based authentication for the super administrator and other administrators. This requires configuring a Radius server then enabling Radius client on the UCM6300 which can be found under System Settings → RADIUS

TR-069

To configure TR-069 on Grandstream devices, set following parameters:

ZERO CONFIG

Overview

Grandstream SIP Devices can be configured via Web interface as well as via configuration file through TFTP/HTTP/HTTPS download. All Grandstream SIP devices support a proprietary binary format configuration file and XML format configuration file.

The UCM630xA provides a Plug and Play mechanism to auto-provision the SIP devices in a zero-configuration manner by generating an XML config file and having the device download it within the LAN area. This allows users to finish the installation with ease and start using the SIP devices in a managed way.

Zero Config supports centralized provisioning for Grandstream SIP devices and selected third-party enterprise SIP phones. Supported third-party devices can be discovered, configured, and managed using the same Zero Config workflow, except where explicitly limited.

To provision a phone, three steps are involved, i.e., discovery, configuration, and provisioning. This section explains how Zero Config works on the UCM630xA. The settings for this feature can be accessed via Device Management→Zero Config.

Zero Config Supported Devices

The following table lists all Grandstream and third-party SIP device series that support Zero Config.

Vendor	Supported Device Series & Models
Grandstream	GXP Series
	GRP Series
	GXV Series
	GHP Series
	GVC Series
	GAC Series
	GSC Series
	GDS Series
	GXW Series
	HT Series
	DP Series
	WP Series
Avaya	J100 Series
Cisco	CP-3905
	CP-6921
	CP-78xx Series
	CP-88xx Series
	DP-98xx Series
Poly	Edge-E Series
	Edge-B Series
	VVX Series (non-OBi)
Yealink	AX Series
	T4 Series
	T5 Series
	T7 Series
	T8 Series

Third-party enterprise device provisioning (Avaya, Cisco, Poly, and Yealink) can be managed and enabled under Device Management → Zero Config → Zero Config Settings → Manage 3rd-Party Enterprise Phones.

Configuration Architecture for Endpoint Device

The end point device configuration in ZeroConfig is divided into the following three layers with priority from the lowest to the highest:

• Global
  

This is the lowest layer. Users can configure the most basic options that could apply to all SIP devices during provisioning via Zero config.

• Model
  

In this layer, users can define model-specific options for the configuration template.

• Device
  

This is the highest layer. Users can configure device-specific options for the configuration for the individual device here.

Each layer also has its own structure at different levels. Please see the figure below. The details for each layer are explained in sections [Global Configuration], [Model configuration] and [Device Configuration].

The configuration options available in the Model and Device layers fully include those defined in the Global layer. In other words, the Global layer options are a subset of the options available at both the Model and Device layers.

When the same option is configured in multiple layers with different values, the value from the higher-priority layer overrides those from lower-priority layers. Layer priority is as follows (from lowest to highest): Global→Model→Device.

Example: if English is selected as the language in the Global Policy, and Spanish is selected in the Default Model Template, the device being provisioned will use Spanish, because the Model layer has higher priority than the Global layer.

Auto-Provisioning Settings

By default, the Zero Config feature is enabled on the UCM630xA for auto-provisioning. Three methods of auto-provisioning are used.

• SIP SUBSCRIBE
  

When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The UCM630xA discovers it and then sends a NOTIFY with the XML config file URL in the message body. The phone will then use the path to download the config file generated in the UCM630xA and take the new configuration.

• DHCP OPTION 66
  

Route mode needs to be set to use this feature. When the phone restarts (by default DHCP Option 66 is turned on), it will send out a DHCP DISCOVER request. The UCM630xA receives it and returns DHCP OFFER with the config server path URL in Option 66, for example, https://192.168.2.1:8089/zccgi/. The phone will then use the path to download the config file generated in the UCM630xA.

To start the auto-provisioning process, under Device Management→Zero Config→Zero Config Settings, fill in the auto provision information.

The table below explains all the configurable parameters that can be found on the Zero Config Settings page:

Discovery

SIP endpoints are automatically discovered after bootup. Users could also manually discover devices by specifying the IP address or scanning the entire LAN network. Three methods are supported to scan the devices.

• PING
• ARP
• SIP Message (NOTIFY)

To start the discover process:

1. Click on “Auto Discover” under Device Management→Zero Config→Zero Config.
2. Choose the “Scan Method” and enter the “Scan IP”. The IP address segment will be automatically filled in based on the network mask detected on the UCM630xA. If users need to scan the entire network segment, enter 255 (for example, 192.168.40.255) instead of a specific IP address.
3. Click on “Save” to start discovering the devices within the same network. To successfully discover the devices, “Zero Config” needs to be enabled on the Device Management→Zero Config→Auto Provisioning Settings.

The following figure shows a list of discovered phones, displaying the MAC address, IP address, extension (if assigned), version, vendor, model, connection status, Create Config option, and available actions (Edit, Delete, Update, Reboot, and Access Device Web GUI).

Firmware

In the Firmware tab, users can upload and manage endpoint firmware. Upgrades are performed per device, not per MAC address.

After you click “Upload”, a window appears displaying the configuration parameters below.

Option	Description
Enable	Toggles whether the UCM will provision this firmware to endpoints using the UCM as their firmware server. If disabled, the UCM rejects firmware requests for this file.
Choose File to Upload	Select the firmware file to upload from the user’s PC. The file name must match the firmware file name requested by the endpoint.
Model	The device model for which this firmware is intended. For self-reference only; has no effect on provisioning.
Firmware	The firmware version of the file being uploaded. For self-reference only; has no effect on provisioning.
Remark	Add a comment about the uploaded firmware. For self-reference only; has no effect on provisioning.

Administrators can perform local firmware updates across one or multiple managed IP phones and gateways simultaneously via Zero Config.

To execute a local firmware upgrade:

1. Navigate to Extension/Trunk → Zero Config → Zero Config.
2. Select the target devices from the device list using the checkboxes.
3. Click the Upgrade button dropdown and select Local.
4. Choose the desired locally uploaded firmware file from the prompt to trigger the upgrade across all selected targets.

Uploading Devices List

Besides the built-in discovery method on the UCM, users could prepare a list of devices on a xlsx or csv file and upload it by clicking on the button ”Import”, after which a success message prompt should be displayed.

Users need to make sure that the CSV file respects the format as shown on the following figure and that the entered information is correct (valid IP address, valid MAC address, device model, and an existing account), otherwise the UCM will reject the file and the operation will fail:

Managing Discovered Devices

Users can manage the discovered devices in the Zero Config tab as follows:

• Sorting: Press or to sort per MAC Address, IP Address, Version, Vendor, Model or Create Config columns from lower to higher or higher to lower, respectively.
• Filter: Select a filter to display corresponding results. Users can filter by registration staus, models, or vendors. The “All” filter displays all discovered devices.
• Search: Search in the list of discovered devices by model, extension number, MAC address, or IP address.
• Add: Use the “Add” button to add a new device to the Zero Config database using its MAC address.
• Delete: Use the “Delete” button delete selected devices from the Zero Config database.
• Edit: Modify selected devices by clicking the “Edit” button.
• Upgrade: Upgrade the software of the discovered devices to the latest version. Select the devices which you want to upgrade. If no devices are selected, all the discovered devices will be upgraded.
• Update Config: Batch update a list of devices, the UCM on this case will send SIP NOTIFY message to all selected devices to update them at once.
• Reboot: Use the “Reboot” option for selected devices (the selected devices should have provisioned extensions since the phone will authenticate the server sending the reboot command).
• More: Perform additional action like:
  
  • Import: Import a list of devices. Devices with duplicate MAC addresses can either be skipped, replaced or updated. The imported file should be in CSV or XLSX format.
  • Export: Export the list of the discovered devices in CSV or XLS format. Select the devices before exporting. If no devices are selected, all the discovered devices will be exported.
  • Reset: Reset the discovered devices. Select the devices to reset before clicking this button. If no devices are selected, all the discovered devices will be reset to their initial configuration.
  • Copy: Copy the configuration across devices.

Global Configuration

The global configuration will apply to all the connected SIP endpoint devices in the same LAN with the UCM630xA regardless of the device model. It is divided into two levels:

• Global Policy
  
• Global Templates
  

Global Policy

Global Policy can be accessed in Device Management→Zero Config→Global Policy page. On the top of the configuration table, users can select a category in the “Options” dropdown list to quickly navigate to the category or they can also complete the configuration by importing/exporting. The categories are:

• Localization: configure display language, data, and time.
• Phone Settings: configure the dial plan, call features, NAT, call progress tones, etc.
• Contact List: configure LDAP and XML phonebook download.
• Maintenance: configure upgrading, web access, Telnet/SSH access, and Syslog.
• Network Settings: configure the IP address, QoS, and STUN settings.
• Customization: customize LCD screen wallpaper for the supported models.
• Communication Settings: configure Email and FTP settings
  

Select the checkbox on the left of the parameter you would like to configure to activate the dropdown list for this parameter.

The following tables list the Global Policy configuration parameters for the SIP end device sorted by category.

Global Policy Parameters – Localization

Global Policy Parameters – Phone Settings

LDAP Phonebook
Source	Select “Manual” or “PBX” as the LDAP configuration source. If “Manual” is selected, the LDAP configuration below will be applied to the SIP end device. If “PBX” is selected, the LDAP configuration built-in from UCM630X Web GUI🡪System Settings🡪LDAP Server will be applied.
Address	Configure the IP address or DNS name of the LDAP server.
Port	Configure the LDAP server port. The default value is 389.
Base DN	This is the location in the directory where the search is requested to begin. Example: dc=grandstream, dc=com ou=Boston, dc=grandstream, dc=com
Username	Configure the bind “Username” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.
Password	Configure the bind “Password” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.
Number Filter	Configure the filter used for number lookups. Please refer to the user manual for more details.
Name Filter	Configure the filter used for name lookups. Please refer to the user manual for more details.
Version	Select the protocol version for the phone to send the bind requests. The default value is 3.
Name Attribute	Specify the “name” attributes of each record that are returned in the LDAP search result. Example: gn cn sn description
Number Attribute	Specify the “number” attributes of each record that are returned in the LDAP search result. Example: telephoneNumber telephoneNumber Mobile
Display Name	Configure the entry information to be shown on the phone’s LCD. Up to 3 fields can be displayed.  Example: %cn %sn %telephoneNumber
Max Hits	Specify the maximum number of results to be returned by the LDAP server. The valid range is 1 to 3000. The default value is 50.
Search Timeout	Specify the interval (in seconds) for the server to process the request and the client waits for the server to return. The valid range is 0 to 180. The default value is 30.
Sort Results	Specify whether the searching result is sorted or not. The default setting is No.
Incoming Calls	Configure to enable LDAP number searching when receiving calls. The default setting is No.
Outgoing Calls	Configure to enable LDAP number searching when making calls. The default setting is No.
Lookup Display Name	Configures the display name when LDAP looks up the name for an incoming call or outgoing call. It must be a subset of the LDAP Name Attributes.
XML Phonebook
Phonebook XML Server	Select the source of the phonebook XML server. Disable Disable phonebook XML downloading. Manual Once selected, users need to specify downloading protocol HTTP, HTTPS, or TFTP and the server path to download the phonebook XML file. The server path could be an IP address or URL, with up to 256 characters. Local UCM Server Once selected, click on the Server Path field to upload the phonebook XML file. Please note after uploading the phonebook XML file to the server, the original file name will be used as the directory name and the file will be renamed as phonebook.xml under that directory.
Phonebook Download Interval	Configure the phonebook download interval (in minutes). If set to 0, the automatic download will be disabled. The valid range is 5 to 720.
Remove manually edited entries on download	If set to “Yes”, when XML phonebook is downloaded, the entries added manually will be automatically removed.

Upgrade and Provision
Firmware Source	Firmware source via ZeroConfig provisioning could be a URL for external server address, local UCM directory, or USB media if plugged into the UCM630X. Select a source to get the firmware file: URL If selected to use URL to upgrade, complete the configuration for the following four parameters: “Upgrade Via”, “Server Path”, “File Prefix” and “File Postfix”. Local UCM Server Firmware can be uploaded to the UCM630X internal storage for firmware upgrade. If selected, click on the “Manage Storage” icon next to the “Directory” option, upload the firmware file, and select a directory for the end device to retrieve the firmware file. Local USB Media If selected, the USB storage device needs to be plugged into the UCM630X and the firmware file must be put under a folder named “ZC_firmware” in the USB storage root directory. Local SD Card Media If selected, an SD card needs to be plugged into the UCM630X and the firmware file must be put under a folder named “ZC_firmware” in the USB storage root directory.
Upgrade via	When URL is selected as firmware source, configure upgrade via TFTP, HTTP, or HTTPS.
Server Path	When URL is selected as firmware source, configure the firmware upgrading server path.
File Prefix	Configure the Config Server Path.
Config Server Path	When URL is selected as firmware source, configure the firmware file postfix. If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone.
Allow DHCP Option 43/66	If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can be redirected.
Automatic Upgrade	If enabled, the end point device will automatically upgrade if a new firmware is detected. Users can select automatic upgrading by day, by week, or by minute. By week Once selected, specify the day of the week to check the HTTP/TFTP server for firmware upgrades or configuration files changes. By day Once selected, specify the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes. By minute Once selected, specify the interval X that the SIP end device will request for new firmware every X minutes.
Firmware Upgrade Rule	Specify how firmware upgrading and provisioning requests are to be sent.
Zero Config	Select either to enable or disable Zero Config.
Web Access
Admin Password	Configure the administrator password for admin-level login.
End-User Password	Configure the end-user password for the end-user level login.
Web Access Mode	Select HTTP or HTTPS as the web access protocol.
Web Server Port	Configure the port for web access. The valid range is 1 to 65535.
RTSP Port	Configure the RTSP Port.
Enable UPnP Discovery	Select either to enable or disable Enable UPnP Discovery
Login Settings	Configure the login settings.
User Login Timeout	Configure User Login Timeout.
Maximum Consecutive Failed Login Attempts	Configure Maximum Consecutive Failed Login Attempts.
Login Error Lock Time	Configure Login Error Lock Time.
Security
Disable Telnet/SSH	Enable Telnet/SSH access for the SIP end device. If the SIP end device supports Telnet access, this option controls the Telnet access of the device; if the SIP end device supports SSH access, this option controls the SSH access of the device.
Syslog
Syslog Server	Configure the URL/IP address for the Syslog server.
Syslog Level	Select the level of logging for Syslog.
Send SIP Log	Configure whether the SIP log will be included in the Syslog message.

Basic Settings
IP Address	Configure how the SIP end device shall obtain the IP address. DHCP or PPPoE can be selected. DHCP Once selected, users can specify the Host Name (option 12) of the SIP end device as DHCP client, and Vendor Class ID (option 60) used by the client and server to exchange vendor class ID information. PPPoE Once selected, users need to specify the Account ID, Password, and Service Name for PPPoE.
Host Name	Specify the name of the client. This field is optional but may be required by Internet Service Providers.
Vendor Class ID	Used by clients and servers to exchange vendor class ID.
Account ID	Enter the PPPoE account ID.
Password	Enter the PPPoE Password.
Service Name	Enter the PPPoE Service Name.
Advanced Setting
Layer 3 QoS	Define the Layer 3 QoS parameter. This value is used for IP Precedence, Diff-Serv, or MPLS. The valid range is 0-63.
Layer 3 QoS For RTP	Assign the priority value of the Layer 3 QoS for RTP packets. The valid range is 0 -63.
Layer 3 QoS For SIP	Assign the priority value of the Layer 3 QoS for SIP packets. The valid range is 0 -63.
Layer 2 QoS Tag	Assign the VLAN Tag of the Layer 2 QoS packets. The valid range is 0 -4095.
Layer 2 QoS Priority Value	Assign the priority value of the Layer 2 QoS packets. The valid range is 0-7.
STUN Server	Configure the IP address or Domain name of the STUN server. Only non-symmetric NAT routers work with STUN.
Keep-Alive	Select either to enable or disable Keep Alive.
Keep Alive Interval	Specify how often the phone will send a blank UDP packet to the SIP server to keep the “ping hole” on the NAT router open. The valid range is 10-160.
Register Expiration	Specify the Register Expiration.
Local SIP Port	Configure Local SIP Port.
Local RTP Port	Configure Local RTP Port.
Auto On-Hook Timer(s)	Configure Auto On-Hook Timer(s).
Ring Timeout	Configure Ring Timeout.
SIP Transport	Select either UDP, TCP, or TLS/TCP as SIP transport protocol.
Direct IP Call	Select either to disable or enable Direct IP Call support.
SIP Proxy Compatibility Mode	Select either to disable or enable SIP Proxy Compatibility Mode.
Unregister On Reboot	Select either to disable or enable Unregister On Reboot.
Whitelist
Whitelist	Select either to enable or disable Whitelist
SIP Phone Number Whitelist	Configure the SIP Phone Number Whitelist.

Email Settings
SMTP Settings	Check this option to configure the email settings that will be sent to the provisioned phones: Server The IP address of the SMTP server Port SMTP server port From E-Mail address Email address Sender Username Username of the sender Password Recovery Email Email where the recovered password will be sent Alarm receive Email 1 Email address where alarms notifications will be sent Alarm receive Email 1 Email address where alarms notifications will be sent Enable SSL Enable SSL protocol for SMTP
FTP
FTP	Check this option to configure the FTP settings that will be sent to the provisioned phones: Storage Server Type Either FTP or Central Storage Server FTP server address Port FTP port to be used Username FTP username Path FTP Directory path

Wallpaper
Screen Resolution 1024 x 600	Check this option if the SIP end device shall use 1024 x 600 resolution for the LCD screen wallpaper. Source Configure the location where wallpapers are stored. File If “URL” is selected as the source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as the source, click to upload wallpaper file to the UCM630X.
Screen Resolution 800 x 400	Check this option if the SIP end device shall use 800 x 400 resolution for the LCD screen wallpaper. Source Configure the location where wallpapers are stored. File If “URL” is selected as the source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as the source, click to upload wallpaper file to the UCM630X.
Screen Resolution 480 x 272	Check this option if the SIP end device shall use 480 x 272 resolution for the LCD screen wallpaper. Source Configure the location where wallpapers are stored. File If “URL” is selected as the source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as the source, click to upload wallpaper file to the UCM630X.
Screen Resolution 320 x 240	Check this option if the SIP end device supports 320 x 240 resolution for the LCD screen wallpaper. Source Configure the location where wallpapers are stored. File If “URL” is selected as the source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as the source, click to upload wallpaper file to the UCM630X.

Global Templates

Global Templates can be accessed in Device Management→Zero Config→Global Templates. Users can create multiple global templates with different sets of configurations and save the templates, or click on the “Import/Export” button to add multiple global templates.

When the user configures the device in the Edit Device dialog→Advanced Settings, they can select to use one of the global templates for the device. Please refer to section [Manage Devices] for more details on using the global templates.

When creating a Global Template, users can select the categories and specific parameters within each category to include in the template. Both the Global Policy and the selected Global Template are applied when generating the configuration file. However, for overlapping configuration options, the Global Template takes precedence over the Global Policy.

1. Click on “Add” to add a global template. Users will see the following configurations.

Template Name	Create a name to identify this global template.
Description	Describe the global template. This is optional.
Active	Check this option to enable the global template.

2. Click on the “Edit” icon to edit the global template.
3. In the “Options” field, after entering the option name keyword, the options containing the keyword will be listed. Users could then select the options to be modified under the global template.

1. Enter or select the value for each option to be used in the global template.
2. On the left side of each added option, click the “Trash” icon to delete this option from the template, or click on the “Reset” icon to reset the option value to the default value.
3. Click on “Save” to save this global template.

The created global templates will show in the Device Management→Zero Config→Global Templates page. Users can delete one or multiple templates or click on “Toggle Selected Template Status” to toggle between enabled/disable status for the selected templates.

Model Configuration

Model templates

The Model layer configuration allows users to apply model-specific settings to devices. Users can create, edit, or delete Model Templates through the Web GUI at Device Management→Zero Config→Model Templates.

If multiple Model Templates are created and enabled, users can select a specific template for a device in the Edit Device dialog under Advanced Settings. The selected Model Template is then applied to that device.

For each Model Template, users can designate it as the default template. When a Model Template is set as the default, its configuration values are applied to all devices of the corresponding model.

Both the default Model Template and the selected Model Template are applied during configuration generation. For overlapping options, the selected Model Template takes precedence and overrides the default Model Template.

To configure a model template:

1. Click on “Add” to add a model template, and configure the following parameters:

Model	Select a model to apply this template. The supported Grandstream models are listed in the dropdown list for selection.
Template Name	Create a name for the model template.
Description	Enter a description for the model template. This is optional.
Default Model Template	Select to assign this model template as the default model template. The value of the option in the default model template will be overridden if another selected model template has a different value for the same option.
Active	Check this option to enable the model template.

2. Click on the “Edit” icon to edit the model template. The editing window for a model template is shown in the following figure.

3. On the left side of each added option, click the “Trash” icon to delete this option from the template, or click on the “Reset” icon to reset the option value to the default value.
4. In the “Options” field, enter the option name key word, the option that contains the keyword will be listed. The user could then select the option to be modified under the model template.

5. Use “Add New Field” to add a P-value number and the value to the configuration. The following figure shows setting P-value “P1362” to “en”, which means the display language on the LCD is set to English.

6. Click on Save when done. The model template will be displayed on Device Management→Zero Config→Model Templates page.
7. Click on the “Trash” icon to delete the model template or use the “Delete” button delete multiple selected templates at once.
8. Click on “Toggle Selected Template Status” to toggle between enabled/disable status for the selected templates.
9. Use the “Import/Export” buttons to upload/export the model template list in CSV or XLS format.

The filter option allows to search through the model templates. Please see the screenshot below:

Model Update

UCM630xA Zero Config feature supports provisioning all models of Grandstream SIP end devices including OEM device models, and supported third-party models.

OEM Models

Users can associate OEM device models with their original Grandstream-branded models, allowing these OEM devices to be provisioned appropriately.

1. Click on “Add OEM Models” button.
2. In the Source Model field, select the Grandstream device that the OEM model is based on from the dropdown list.
3. For the Destination Model and Destination Vendor field, enter the custom OEM model name and vendor name.
4. The newly added OEM model should now be selectable as an option in Model fields.

Model Template Package List

Templates for most of the Grandstream models are built-in with the UCM630xA already. Templates for Wave, Grandstream surveillance products, and third-party devices require users to download and install under Device Management→Zero Config→Model Update first before they are available in the UCM630xA for selection.

After downloading and installing the model template to the UCM630xA, it will show in the dropdown list for “Model” selection when editing the model template.

1. Click on the “Download” icon to download the template.
2. Click on “Upgrade” to upgrade an existing model template.

Upload Model Template Package

By default, third-party model templates are not available on the UCM630xA and must be uploaded manually through the Web GUI.

In environments with restricted Internet connectivity, Grandstream model packages must also be uploaded manually.

Device Configuration

On Web GUI, page Device Management→Zero Config→Zero Config, users could create a new device, delete existing device(s), make special configuration for a single device, or send NOTIFY to an existing device(s).

Create New Device

Users can directly create a new device and configure basic settings before the device is discovered by the UCM630xA. Once the device is plugged in, it can then be discovered and provisioned. This gives the system administrator adequate time to set up each device beforehand.

Click on “Add” and the following dialog will show. Follow the steps below to create the configurations for the new device.

1. Firstly, select a model for the device to be created and enter its MAC address, IP address, and firmware version (optional) in the corresponding field.
2. Basic settings will show a list of settings based on the model selected in step 1. Users could assign extensions to accounts, assign functions to Line Keys and Multiple-Purposed Keys if supported on the selected model.
3. Click on “Save” to save the configuration for this device.

Manage Device

The device manually created or discovered from Auto Discover will be listed in the Device Management→Zero Config→Zero Config page. Users can see the devices with their MAC address, IP address, vendor, model, etc.

• Click on “Access Device Web UI” to access the Web GUI of the device.
• Click on the “Edit” icon to edit the device configuration.

A new dialog will be displayed for the users to configure “Basic” settings and “Advanced” settings. “Basic” settings have the same configurations as displayed when manually creating a new device, i.e., account, line key, and MPK settings; “Advanced” settings allow users to configure more details in a five-level structure.

A preview of the Advanced Settings is shown in the figure above. There are five configuration levels, as described in items (1) through (5) below, ordered from lowest to highest priority. All configuration levels are applied to the device.

If the same option is defined at multiple levels with different values, the value from the higher-priority level overrides the value from the lower-priority level.

1. Global Policy

This is the lowest level configuration. The global policy configured in Device Management→Zero Config→Global Policy will be applied here. Clicking on “Modify Global Policy” to redirect to page Device Management→Zero Config→Global Policy.

2. Global Templates

Select a Global Template and click the right arrow to add it to the device. Multiple Global Templates can be selected, and their priority can be adjusted using the up and down arrows. All selected templates take effect.

If the same option is defined in multiple selected Global Templates, the value from the higher-priority template overrides that of the lower-priority template. Click the left arrow to remove a Global Template from the selected list.

3. Default Model Template

Default Model Template will be applied to the devices of this model. Default model template can be configured in the model template under Device Management→Zero Config→Model Templates page.

4. Model Templates

Select a Model Template to apply to the device and click the right arrow to add it. Multiple Model Templates can be selected, and their priority can be adjusted using the up and down arrows. All selected templates take effect.

If the same option is defined in multiple selected Model Templates, the value from the higher-priority template overrides that of the lower-priority template. Click the left arrow to remove a Model Template from the selected list.

5. Customize Device Settings

This is the highest-level configuration for the device. Click on “Modify Customize Device Settings” and the following dialog will show.

Scroll down in the dialog to view and edit device-specific options. To add options that are not included in the predefined list, click Add New Field and specify the P-value and its corresponding value.

In the example shown above, P-value P1362 is set to en, which configures the LCD display language to English. The warning message on the right indicates that a predefined option matching this P-value already exists; clicking it navigates to the corresponding option.

Select multiple devices that need to be modified and then click on ”Update Config” to batch modify devices.

If selected devices are of the same model, the configuration dialog is like the following figure. Configurations in five levels are all available for users to modify.

If selected devices are of different models, the configuration dialog is like the following figure. Click on the arrows to view more devices of other models. Users are only allowed to make modifications to the Global Templates and Global Policy level.

After the above configurations, save the changes and go back to Device Management→Zero Config→Zero Config page. Users could then click on the “Update” icon to send NOTIFY to the SIP end point device and trigger the provisioning process. The device will start downloading the generated configuration file from the URL contained in the NOTIFY message.

On this web page, users can also click on “Reset All Extensions” to reset the extensions of all the devices.

Sample Application

Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1 GXV3275 phone used by customer support supervisor. 3 of the 8 customer support members speak Spanish and the rest speak English. We could deploy the following configurations to provision the office phones for the customer support team.

1. Go to Device Management→Zero Config→Zero Config Settings, select “Enable Zero Config”.
2. Go to Device Management→Zero Config→Global Policy, configure Date Format, Time Format, and Firmware Source as follows.

3. Go to Device Management→Zero Config→Model Templates, create a new model template “English Support Template” for GXP2170. Add option “Language” and set it to “English”. Then select the option “Default Model Template” to make it the default model template.
4. Go to Device Management→Zero Config →Model Templates, create another model template “Spanish Support Template” for GXP2170. Add option “Language” and set it to “Español”.
5. After 9 devices are powered up and connected to the LAN network, use the “Auto Discover” function or “Create New Device” function to add the devices to the device list on Device Management→Zero Config→Zero Config.
6. On Web GUI→Device Management→Zero Config→Zero Config page, users could identify the devices by their MAC addresses or IP addresses displayed on the list. Click on the “Edit” icon to edit the device settings.
7. For each of the 5 phones used by English-speaking customer support, in “Basic settings” select an available extension for account 1 and click on “Save”. Then click on the “Advanced settings” tab to bring up the following dialog. Users will see the English support template is applied since this is the default model template. A preview of the device settings will be listed on the right side.

8. For the 3 phones used by Spanish support, in “Basic settings” select an available extension for account 1 and click on “Save”. Then click on the “Advanced settings” tab to bring up the following dialog.
   

9. Select “Spanish Support Template” in “Model Template”. The preview of the device settings is displayed on the right side and we can see the language is set to “Español” since Model Template has the higher priority for the option “Language”, which overrides the value configured in the default model template.
10. For the GXV3275 used by the customer support supervisor, select an available extension for account 1 on “Basic settings” and click on “Save”. Users can see the preview of the device configuration in “Advanced settings”. There is no model template configured for GXV3275.

11. Click on “Apply Changes” to apply saved changes.
12. On the Device Management→Zero Config→Zero Config page, click on the“Update” icon to send NOTIFY to trigger the device to download the config file from UCM630xA.

Now all the 9 phones in the network will be provisioned with a unique extension registered on the UCM630xA. 3 of the phones will be provisioned to display Spanish on LCD and the other 5 will be provisioned to display English on LCD. The GXV3275 used by the supervisor will be provisioned to use the default language on the LCD display since it is not specified in the global policy.

EXTENSIONS

Create New Extension

Create New SIP Extension

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

Extension options are divided into the following categories:

• Basic Settings
• Media
• Call Policy
• Features
• Voicemail
• Custom Time
• AI
• Wave Client
• Follow me
• Advanced Settings

Select first which type of extension: SIP Extension, IAX Extension or FXS Extension. The configuration parameters are as follows.

SIP Extension Configuration Parameters > Basic Settings

SIP Extension Configuration Parameters > Media

SIP Extension Configuration Parameters > Call Policy

SIP Extension Configuration Parameters > Features

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

SIP Extension Configuration Parameters > AI

SIP Extension Configuration Parameters > Wave Client

Follow Me
Enable	Configure to enable or disable Follow Me for this user.
Skip Trunk Auth	If the outbound calls need to check the password, we should enable this option or enable the option “Skip Trunk Auth” of the Extension. Otherwise this Follow Me cannot call out.
Music On Hold Class	Configure the Music On Hold class that the caller would hear while tracking the user.
Confirm When Answering	If enabled, call will need to be confirmed after answering.
Enable Destination	Configure to enable destination
Default Destination	The call will be routed to this destination if no one in the Follow Me answers the call.
Use Callee DOD for Follow Me	Use the callee DOD number as CID if configured Follow Me numbers are external numbers.
Play Follow Me Prompt	If enabled, the Follow Me prompt tone will be played
New Follow Me Number	Add a new Follow Me number which could be a “Local Extension” or an “External Number”. The selected dial plan should have permissions to dial the defined external number.
Dialing Order	This is the order in which the Follow Me destinations will be dialed to reach the user.

Create New IAX Extension

The UCM630xA supports Inter-Asterisk eXchange (IAX) protocol. IAX is used for transporting VoIP telephony sessions between servers and terminal devices. IAX is like SIP but also has its own characteristic. For more information, please refer to RFC 5465.

To manually create new IAX user, go to Web GUI🡪Extension/Trunk🡪Extensions. Click on “Add” and a new dialog window will show for users which need to make sure first to select the extension type to be IAX Extension before proceeding to fill in the extension information. The configuration parameters are as follows.

Create New FXS Extension

The UCM630xA supports Foreign eXchange Subscriber (FXS) interface. FXS is used when user needs to connect analog phone lines or FAX machines to the UCM630xA.

To manually create new FXS user, go to Web GUI🡪Extension/Trunk🡪Extensions. Click on “Add” and a new dialog window will show for users which need to make sure first to select the extension type to be FXS Extension before proceeding to fill in the extension information. The configuration parameters are as follows.

Batch Add Extensions

Batch Add SIP Extensions

To add multiple SIP extensions, BATCH add can be used to create standardized SIP extension accounts. However, unique extension username cannot be set using BATCH add.

Under Web GUI🡪Extension/Trunk🡪Extensions, click on “Add” and select extension type as SIP extension, and “Select Add Method” as Batch.

General
Create Number	Specify the number of extensions to be added. The default setting is 5.
Extension Incrementation	Select how much to increment successive extensions. For example, if the value is 2, the extensions will be 1000,1002,1004,…… Note: Up to 3 characters.
Extension	Configure the starting extension number of the batch of extensions to be added.
Permission	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 from this rule.
Voicemail	Configure Voicemail.   There are three valid options and the default option is “Enable Local Voicemail”. Disable Voicemail: Disable Voicemail. Enable Local Voicemail: Enable voicemail for the user. Enable Remote Voicemail: Forward the notify message from remote voicemail system for the user, and the local voicemail will be disabled. Note: Remote voicemail feature is used only for Infomatec (Brazil).
SIP/IAX Password	Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions.   User Random Password. A random secure password will be automatically generated. It is recommended to use this password for security purpose. Use Extension as Password. Enter a password to be used on all the extensions in the batch.
Voicemail Password	Configure Voicemail password (digits only) for the users.   User Random Password. A random password in digits will be automatically generated. It is recommended to use this password for security purpose. Use Extension as Password. Enter a password to be used on all the extensions in the batch.
Send Voicemail to Email	Send voicemail messages to the configured email address. If set to “Default”, the global setting will be used. Global settings can be found in Voicemail->Voicemail Email Settings.
Keep Voicemail after Emailing	Only applies if extension-level or global Send Voicemail to Email is enabled.
CallerID Number	Configure CallerID Number when adding Batch Extensions.   Use Extension as Number Users can choose to use the extension number as CallerID Use as Number Users can choose to set a specific number instead of using the extension number.
Skip Voicemail Password Verification	When user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.
Enable Keep-alive	If enabled, the PBX will regularly send SIP OPTIONS to check if host device is online.
Keep-alive Frequency	Configure the keep-alive interval (in seconds) to check if the host is up.
Disable This Extension	Check this box to disable this extension.
Enable SCA	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.
Emergency Calls CID	CallerID number that will be used when calling out and receiving direct callbacks.
Enable Wave	If enabled, this extension number can register, log in and use Wave normally, otherwise it will not be able to use Wave, but the phone function will still be retained.
Sync Contact	If enabled, this extension number will be displayed in the Wave contact, otherwise it will not be displayed, and it cannot be found in the chat, but the user can still dial this number.
Language	Select voice prompt language for this extension. If set to “Default”, the global setting for voice prompt language will be used.
Media
NAT	Use NAT when the PBX is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it is related to NAT configuration or Firewall’s support of SIP and RTP ports.   The default setting is enabled.
Enable Direct Media	By default, the PBX will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to negotiate with the endpoints to route the media stream directly. It is not always possible for the PBX to negotiate endpoint-to-endpoint media routing. The default setting is “No”.
DTMF Mode	Select DTMF mode for the user to send DTMF. The default setting is “RFC4733”. If “Info” is selected, 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.
Alert-info	When present in an INVITE request, the Alert-info header field specifies an alternative ring tone to the UAS.
SRTP	Enable/disable SRTP for RTP stream encryption.
Packet Loss Retransmission	Configure to enable Packet Loss Retransmission.   NACK NACK+RTX(SSRC-GROUP) OFF
Video FEC	Check to enable Forward Error Correction (FEC) for Video.
FECC	Configure to enable FECC
Audio FEC	Check to enable Forward Error Correction (FEC) for Audio.
ACL Policy	Access Control List manages the IP addresses that can register to this extension.   Allow All: Any IP address can register to this extension. Local Network: Only IP addresses in the configured network segments can register to this extension. Press “Add Local Network Address” to add more IP segments.
Jitter Buffer	Select 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.
Codec Preference	Configure the codecs to be used.
Call Transfer
Presence Status	Select which presence status to set for the extension and configure call forward conditions for each status. Six possible options are possible: “Available”, “Away”, “Chat”, “Custom”, “DND” and “Unavailable”. More details at [PRESENCE].
Call Forward Unconditional	Enable and configure the Call Forward Unconditional target number. Available options for target number are:   “None”: Call forward deactivated. “Extension”: Select an extension from dropdown list as CFU target. “Custom Number”: Enter a customer number as target. For example: *97. “Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension. “Ring Group”: Select a ring group from dropdown list as CFU target. “Queues”: Select a queue from dropdown list as CFU target. “Voicemail Group”: Select a voicemail group from dropdown list as CFU target. The default setting is “None”.
CFU Time Condition	Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.   Note: “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period. Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
Call Forward No Answer	Configure the Call Forward No Answer target number. Available options for target number are:   “None”: Call forward deactivated. “Extension”: Select an extension from dropdown list as CFN target. “Custom Number”: Enter a customer number as target. For example: *97. “Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension. “Ring Group”: Select a ring group from dropdown list as CFN target. “Queues”: Select a queue from dropdown list as CFN target. “Voicemail Group”: Select a voicemail group from dropdown list as CFN target. The default setting is “None”.
CFN Time Condition	Select time condition for Call Forward No Answer. 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 on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
Call Forward Busy	Configure the Call Forward Busy target number. Available options for target number are:   “None”: Call forward deactivated. “Extension”: Select an extension from dropdown list as CFB target. “Custom Number”: Enter a customer number as target. For example: *97. “Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension. “Ring Group”: Select a ring group from dropdown list as CFB target. “Queues”: Select a queue from dropdown list as CFB target. “Voicemail Group”: Select a voicemail group from dropdown list as CFB target. The default setting is “None”.
CFB Time Condition	Select time condition for Call Forward Busy. 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 on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
Do Not Disturb	If Do Not Disturb is enabled, all incoming calls will be dropped.   All call forward settings will be ignored.
DND Time Condition	Select time condition for Do Not Disturb. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.   Notes: “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period. Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
DND Whitelist	If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.   Z match any digit from 1-9 N match any digit from 2-9 X match any digit from 0-9.
FWD Whitelist	Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported.   Z match any digit from 1-9 N match any digit from 2-9 X match any digit from 0-9.
CC Settings
Enable CC	If enabled, UCM630xA will automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason. By default, it is disabled.
CC Mode	Two modes for Call Completion are supported:   Normal: This extension is used as ordinary extension. For Trunk: This extension is registered from a PBX. The default setting is “Normal”.
CC Max Agents	Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel can make. The minimum value is 1.
CC Max Monitors	Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time. The minimum value is 1.
Ring Simultaneously
Ring Simultaneously	Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.
External Number	Set the external number to be rang simultaneously. ‘-’ is the connection character which 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 on the basis of 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
Allowed to   call-barging	Add members from “Available Extensions” to “Selected Extensions” so that the selected extensions can spy on the used extension using feature code.
Seamless transfer privilege control
Allowed to seamless transfer	Any extensions on the UCM can perform seamless transfer. When using Pickup Incall feature, only extensions available on the “Selected Extensions” list can perform seamless transfer to the edited extension.
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 UCM630xA, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt: General Preference. The valid range is between 3 seconds and 600 seconds.   Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.
Auto Record	Enable automatic recording for the calls using this extension. The default setting is disabled. The recordings can be accessed under Web GUI🡪CDR🡪Recording Files.
Skip Trunk Auth	If set to “yes”, users can skip entering the password when making outbound calls. If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition. If set to “No”, users will be asked to enter the password when making outbound calls.
Time Condition for Skip Trunk Auth	If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering password when making outbound calls.
Dial Trunk Password	Configure personal password when making outbound calls via trunk.
Enable LDAP	If enabled, the extension will be added to LDAP Phonebook PBX list.
Bind PMS Room	If enabled, the system will create a room whose room number, by default, will equal the extension number in PMS module. Note: If this room already exists, the configuration of the existing room will be overwritten.
Music On Hold	Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.
Call Duration Limit	The maximum duration of call-blocking.
Maximum Call Duration	The maximum call duration (in seconds). The default value 0 means no limit.
Call Waiting	If disabled, UCM will not invite the extension when it is already in a call and will do the same work as the user is busy.   Note: the option only works when the caller dials the extension directly.

Batch Add IAX Extensions

Under Web GUI🡪Extension/Trunk🡪Extensions, click on “Add”, then select extension type as IAX Extension and the add method to be Batch.

General
Create Number	Specify the number of extensions to be added. The default setting is 5.
Extension Incrementation	Select how much to increment successive extensions. For example, if the value is 2, the extensions will be 1000,1002,1004,……
Extension	The extension number associated with this particular user/phone.
Permission	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 in order to make outbound calls from this rule.
CallerID Number	Configure the Caller ID number displayed when dialing calls from this user. Note: The Caller ID usage might be limited by your VoIP provider. In Batch Add Method, “e” means to use the extension as the number.
Voicemail	Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”.   Disable Voicemail: Disable Voicemail. Enable Local Voicemail: Enable voicemail for the user.
SIP/IAX Password	Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions.   User Random Password. A random secure password will be automatically generated. It is recommended to use this password for security purpose. Use Extension as Password. Enter a password to be used on all the extensions in the batch.
Voicemail Password	Configure Voicemail password (digits only) for the users.   User Random Password. A random password in digits will be automatically generated. It is recommended to use this password for security purpose. Use Extension as Password. Enter a password to be used on all the extensions in the batch.
Send Voicemail to Email	Send voicemail messages to the configured email address. If set to “Default”, the global setting will be used. Global settings can be found in Voicemail->Voicemail Email Settings.
Keep Voicemail after Emailing	Only applies if extension-level or global Send Voicemail to Email is enabled.
Auto Record	Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.
Skip Voicemail Password Verification	When user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.
Disable This Extension	Check this box to disable this extension.
Language	Select voice prompt language for this extension. If set to “Default”, the global setting for voice prompt language will be used.
IAX Settings
Max Number of Calls	Configure the maximum number of calls allowed for each remote IP address.
Require Call Token	Configure to enable/disable requiring call token. If set to “Auto”, it might lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints.   The default setting is “Yes”.
SRTP	Enable/disable SRTP for RTP stream encryption.
ACL Policy	Access Control List manages the IP addresses that can register to this extension.   Allow All: Any IP address can register to this extension. Local Network: Only IP addresses in the configured network segments can register to this extension.
Codec Preference	Configure the codecs to be used.
Call Transfer
Call Forward Unconditional	Enable and configure the Call Forward Unconditional target number.
CFU Time Condition	Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.   Note: “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period. Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
Call Forward No Answer	Configure the Call Forward No Answer target number.
CFN Time Condition	Select time condition for Call Forward No Answer. 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 on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
Call Forward Busy	Configure the Call Forward Busy target number.
CFB Time Condition	Select time condition for Call Forward Busy. 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 on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
Do Not Disturb	If Do Not Disturb is enabled, all incoming calls will be dropped.   All call forward settings will be ignored.
DND Time Condition	Select time condition for Do Not Disturb. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.   Notes: “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period. Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.
DND Whitelist	If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.   Z match any digit from 1-9， N match any digit from 2-9， X match any digit from 0-9.
FWD Whitelist	Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported.   Z match any digit from 1-9， N match any digit from 2-9， X match any digit from 0-9.
Ring Simultaneously
Ring Simultaneously	Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.
External Number	Set the external number to be rang simultaneously. ‘-’ is the connection character which 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 on the basis of 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
Allowed to   call-barging	Add members from “Available Extensions” to “Selected Extensions” so that the selected extensions can spy on the used extension using feature code.
Seamless transfer privilege control
Allowed to seamless transfer	Any extensions on the UCM can perform seamless transfer. When using Pickup Incall feature, only extensions available on the “Selected Extensions” list can perform seamless transfer to the edited extension.
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 UCM630xA, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt: General Preference. The valid range is between 5 seconds and 600 seconds.   Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.
Auto Record	Enable automatic recording for the calls using this extension. The default setting is disabled. The recordings can be accessed under Web GUI🡪CDR🡪Recording Files.
Skip Trunk Auth	If set to “yes”, users can skip entering the password when making outbound calls. If set to “By Time”, users can skip entering the password when making outbound calls during the selected time condition. If set to “No”, users will be asked to enter the password when making outbound calls.
Time Condition for Skip Trunk Auth	If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering password when making outbound calls.
Dial Trunk Password	Configure personal password when making outbound calls via trunk.
Enable LDAP	If enabled, the extension will be added to LDAP Phonebook PBX list.
Music On Hold	Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.
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

Batch Extension Resetting Functionality

Users can select multiple extensions and reset their settings to default by pressing the reset button

and confirm the reset functionality. Once done, all settings in Basic Settings page will be restored to default values with the exception of Concurrent Registrations. User voicemail password will be reset to Random Password. User voicemail prompts and recordings will be deleted. User Management settings will also be restored to default with the exception of usernames and custom privileges

Search and Edit Extension

All the UCM630xA extensions are listed under Web GUI🡪Extension/Trunk🡪Extensions, with status, Extension, CallerID Name, Technology (SIP, IAX and FXS), IP and Port. Each extension has a checkbox for users to “Edit” or “Delete”. Also, options “Edit”

, “Reboot”

and “Delete”

are available per extension. User can search an extension by specifying the extension number to find an extension quickly.

• 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 Maintenance🡪User Management🡪User Information except username and permissions and delete the user voicemail prompt and voice messages.

• Reboot the user

Click on

to send NOTIFY reboot event to the device which has an UCM630xA extension already registered. To successfully reboot the user, “Zero Config” needs to be enabled on the UCM630xA Web GUI🡪 Device Management🡪Zero Config🡪Zero Config Settings.

• Delete single extension

Click on

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

• 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 UCM630xA can be exported to csv format file with selected technology “SIP”, “IAX” or “FXS”. Click on “Export Extensions” button and select technology in the prompt below.

The exported csv file can serve as a template for users to fill in desired extension information to be imported to the UCM630xA.

Import Extensions

The capability to import extensions to the UCM630xA provides users flexibility to batch add extensions with similar or different configuration quickly into the PBX system.

1. Export extension csv file from the UCM630xA by clicking on “Export Extensions” button.
2. Fill up the extension information you would like in the exported csv template.
3. Click on “Import Extensions” button. The following dialog will be prompted.

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 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 csv file from local directory in the PC.
6. Click on “Apply Changes” to apply the imported file on the UCM630xA.

Example of file to import:

Field	Supported values
Extension	Digits
Technology	SIP/SIP(WebRTC)
Enable Voicemail	yes/no/remote
CallerID Number	Digits
SIP/IAX Password	Alphanumeric characters
Voicemail Password	Digits
Skip Voicemail Password Verification	yes/no
Ring Timeout	Empty/ 3 to 600 (in second)
SRTP	yes/no
Strategy	Allow All/Local Subnet Only/A Specific IP Address
Local Subnet 1	IP address/Mask
Local Subnet 2	IP address/Mask
Local Subnet 3	IP address/Mask
Local Subnet 4	IP address/Mask
Local Subnet 5	IP address/Mask
Local Subnet 6	IP address/Mask
Local Subnet 7	IP address/Mask
Local Subnet 8	IP address/Mask
Local Subnet 9	IP address/Mask
Local Subnet 10	IP address/Mask
Specific IP Address	IP address
Skip Trunk Auth	yes/no/bytime
Codec Preference	PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus
Permission	Internal/Local/National/International
NAT	yes/no
DTMF Mode	RFC4733/info/inband/auto
Insecure	Port
Enable Keep-alive	Yes/no
Keep-alive Frequency	Value from 1-3600
AuthID	Alphanumeric value without special characters
TEL URI	Disabled/user=phone/enabled
Call Forward Busy	Digits
Call Forward No Answer	Digits
Call Forward Unconditional	Digits
Support Hot-Desking Mode	Yes/no
Dial Trunk Password	Digits
Disable This Extension	Yes/no
CFU Time Condition	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
CFN Time Condition	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
CFB Time Condition	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Music On Hold	Default/ringbacktone_default
CC Agent Policy	If CC is disabled use: never   If CC is set to normal use: generic If CC is set to trunk use: native
CC Monitor Policy	Generic/never
CCBS Available Timer	3600/4800
CCNR Available Timer	3600/7200
CC Offer Timer	60/120
CC Max Agents	Value from 1-999
CC Max Monitors	Value from 1-999
Ring simultaneously	Yes/no
External Number	Digits
Time Condition for Ring Simultaneously	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Time Condition for Skip Trunk Auth	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Enable LDAP	Yes/no
Enable T.38 UDPTL	Yes/no
Max Contacts	Values from 1-10
Enable Wave	Yes/no
Alert-Info	None/Ring 1/Ring2/Ring3/Ring 4/Ring 5/Ring 6/Ring 7/ Ring 8/Ring 9/Ring 10/bellcore-dr1/bellcore-dr2/ bellcore-dr3/ bellcore-dr4/ bellcore-dr5/custom
Do Not Disturb	Yes/no
DND Time Condition	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Custom Auto answer	Yes/no
Do Not Disturb Whitelist	Empty/digits
User Password	Alphanumeric characters.
First Name	Alphanumeric without special characters.
Last Name	Alphanumeric without special characters.
Email Address	Email address
Language	Default/en/zh
Phone Number	Digits
Call-Barging Monitor	Extensions allowed to call barging
Seamless Transfer Members	Extensions allowed to seamless transfer

Field	Supported values
Extension	Digits
Technology	IAX
Enable Voicemail	yes/no
CallerID Number	Digits
SIP/IAX Password	Alphanumeric characters
Voicemail Password	Digits
Skip Voicemail Password Verification	yes/no
Ring Timeout	Empty/ 3 to 600 (in second)
SRTP	yes/no
Strategy	Allow All/Local Subnet Only/A Specific IP Address
Local Subnet 1	IP address/Mask
Local Subnet 2	IP address/Mask
Local Subnet 3	IP address/Mask
Local Subnet 4	IP address/Mask
Local Subnet 5	IP address/Mask
Local Subnet 6	IP address/Mask
Local Subnet 7	IP address/Mask
Local Subnet 8	IP address/Mask
Local Subnet 9	IP address/Mask
Local Subnet 10	IP address/Mask
Specific IP Address	IP address
Skip Trunk Auth	yes/no/bytime
Codec Preference	PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus
Permission	Internal/Local/National/International
NAT	yes/no
Call Forward Busy	Digits
Call Forward No Answer	Digits
Call Forward Unconditional	Digits
Require Call Token	Yes/no/auto
Max Number of Calls	Values from 1-512
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
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
Ring simultaneously	Yes/no
External Number	Digits
Time Condition for Ring Simultaneously	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Time Condition for Skip Trunk Auth	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Enable LDAP	Yes/no
Limit Max time (s)	empty
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
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

Field	Supported values
Extension	Digits
Technology	FXS
Analog Station	FXS1/FXS2
Enable Voicemail	yes/no
CallerID Number	Digits
Voicemail Password	Digits
Skip Voicemail Password Verification	yes/no
Ring Timeout	Empty/ 3 to 600 (in second)
Auto Record	yes/no
Fax Mode	None/Fax Gateway/Fax Detection
Skip Trunk Auth	Yes/no/bytime
Permission	Internal/Local/National/International
Call Forward Busy	Digits
Call Forward No Answer	Digits
Call Forward Unconditional	Digits
Call Waiting	Yes/no
Use # as SEND	Yes/no
RX Gain	Values from -30🡪6
TX Gain	Values from -30🡪6
MIN RX Flash	Values from: 30 – 1000
MAX RX Flash	Values from: 40 – 2000
Enable Polarity Reversal	Yes/no
Echo Cancellation	On/Off/32/64/128/256/512/1024
3-Way Calling	Yes/no
Send CallerID After	1/2
Dial Trunk Password	digits
Disable This Extension	Yes/no
CFU Time Condition	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
CFN Time Condition	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
CFB Time Condition	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Music On Hold	Default/ringbacktone_default
CC Agent Policy	If CC is disabled use: never   If CC is set to normal use: generic If CC is set to trunk use: native
CC Monitor Policy	Generic/never
CCBS Available Timer	3600/4800
CCNR Available Timer	3600/7200
CC Offer Timer	60/120
CC Max Agents	Value from 1-999
CC Max Monitors	Value from 1-999
Ring simultaneously	Yes/no
External Number	Digits
Time Condition for Ring Simultaneously	All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Time Condition for Skip Trunk Auth
Enable LDAP	Yes/no
Enable Hotline	Yes/no
Hotline Type	Immediate hotline/delay hotline
Hotline Number	digits
Limit Max time (s)	empty
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
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

The CSV file should contain all the above fields, if one of them is missing or empty, the UCM630xA will display the following error message for missing fields.

Extension Details

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

• Extension: Shows the Extension number.
• Status: Shows the status of the extension.
• Presence status: Indicates the Presence Status of this extension.
• Terminal Type: Shows the Type of the terminal using this extension (SIP, FXS…etc.).
• 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.

E-mail Notification

Once the extensions are created with Email addresses, the PBX administrator can click on button “E-mail Notification” to send the account registration and configuration information to the user. Please make sure Email setting under Web GUI🡪System Settings🡪Email Settings is properly configured and tested on the UCM630xA before using “E-mail Notification”.

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

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

Multiple Registrations per Extension

UCM630xA 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 option “Concurrent Registrations” under Web GUI🡪Extension/Trunk🡪Edit Extension. The default value is set to 1 for security purpose. Maximum is 10.

SMS Message Support

The UCM630xA provides built-in SIP SMS message support. For SIP end devices such as Grandstream GXP or GXV phones that supports SIP message, after an UCM630xA account is registered on the end device, the user can send and receive SMS message. Please refer to the end device documentation on how to send and receive SMS message.

EXTENSION GROUPS

The UCM630xA extension group feature allows users to assign and categorize extensions in different groups to better manage the configurations on the UCM630xA. For example, when configuring “Enable Filter on Source Caller ID”, 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 business environment.

Configure Extension Groups

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

• Click on
  

  

  
  to create a new extension group.
• Click on
  

  

  
  to edit the extension group.
• Click on
  

  

  
  to delete the extension group.

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

Click on

in order to change the ringing priority of the members selected on the group.

Using Extension Groups

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

ANALOG TRUNKS

Go to Web GUI🡪Extension/Trunk🡪Analog Trunks to add and edit analog trunks.

• Click on “Create New Analog Trunk” to add a new analog trunk.
• Click on
  

  

  
  to edit the analog trunk.
• Click on
  

  

  
  to delete the analog trunk.

Analog Trunk Configuration

The analog trunk options are listed in the table below.

PSTN Detection

The UCM630xA provides PSTN detection function to help users detect the busy tone, Polarity Reversal and Current Disconnect by making a call from the PSTN line to another destination. The detecting call will be answered and up for about 1 minute. Once done, the detecting result will show and can be used for the UCM630xA settings.

1. Go to UCM630xA Web GUI🡪Extension/Trunk🡪Analog Trunks page.
2. Click to edit the analog trunk created for the FXO port.
3. In the window to edit the analog trunk, go to “Tone Settings” section and there are two methods to set the busy tone.

• Tone Country. The default setting is “United States of America (USA)”.
• PSTN Detection.

4. Click on “Detect” to start PSTN detection.

• If there are two FXO ports connected to PSTN lines, use the following settings for auto-detection.

Detect Model: Auto Detect.

Source Channel: The source channel to be detected.

Destination Channel: The channel to help detecting. For example, the second FXO port.

Destination Number: The number to be dialed for detecting. This number must be the actual PSTN number for the FXO port used as the destination channel.

• If there is only one FXO port connected to PSTN line, use the following settings for auto-detection.

Detect Model: Semi-auto Detect.

Source Channel: The source channel to be detected.

Destination Number: The number to be dialed for detecting. This number could be a cell phone number or other PSTN number that can be reached from the source channel PSTN number.

5. Click “Detect” to start detecting. The source channel will initiate a call to the destination number. For “Auto Detect”, the call will be automatically answered. For “Semi-auto Detect”, the UCM630xA Web GUI will display prompt to notify the user to answer or hang up the call to finish the detecting process.
6. Once done, the detected result will show. Users could save the detecting result as the current UCM630xA settings.

Detect Model	Select “Auto Detect” or “Semi-auto Detect” for PSTN detection.   Auto Detect Please make sure two or more channels are connected to the UCM630xA and in idle status before starting the detection. During the detection, one channel will be used as caller (Source Channel) and another channel will be used as callee (Destination Channel). The UCM630xA will control the call to be established and hang up between caller and callee to finish the detection. Semi-auto Detect Semi-auto detection requires answering or hanging up the call manually. Please make sure one channel is connected to the UCM630xA and in idle status before starting the detection. During the detection, source channel will be used as caller and send the call to the configured Destination Number. Users will then need follow the prompts in Web GUI to help finish the detection. The default setting is “Auto Detect”.
Source Channel	Select the channel to be detected.
Destination Channel	Select the channel to help detect when “Auto Detect” is used.
Destination Number	Configure the number to be called to help the detection.
Dump Call Progress Tone File	Choose whether to save the calling tone file, it is not checked by default.

VOIP TRUNKS

VoIP Trunk Configuration

VoIP trunks can be configured on the UCM under Web GUI→Extension/Trunk→VoIP Trunks.

Once created, all VoIP trunks will appear in a list showing the following key information:

• Provider Name: A user-defined label to identify the trunk.
• Status: The current state of the trunk (Registered, Reachable, Unreachable, or Unmonitored).
• Active Calls: The number of calls currently using the trunk.
• Endpoint Type: The protocol used for the trunk (SIP or IAX).
• Type: The trunk type (register, peer, or account).
• Hostname/IP Address: The server address of the trunk.
• Username: The authentication username for register/account trunks.
• Total Time for Outbound Calls (m): The cumulative outbound call duration limit.
• Options: Quick access to management functions using the icons described below:
  
  • Edit: Opens the trunk configuration page (Basic + Advanced settings).
  • DOD: Opens Direct Outward Dialing settings for that trunk (maps extensions to outbound CallerID).
  • Sync LDAP: Starts LDAP phonebook synchronization (only for SIP peer trunks with Sync LDAP enabled).
  • Delete: Removes the trunk from the UCM system.

The following sections provide a detailed explanation of the available configuration parameters for each trunk type.

Account SIP Trunk

This type of SIP trunk allows devices such as ATAs and gateways to trunk directly with the UCM6300 to handle and route calls.

This is especially useful when the devices do not have a static IP address for peering or if they cannot register to a UCM extension due to call handling and routing complications.

Register SIP Trunk

This type of SIP trunk allows registering the trunk to an account SIP trunk, which allows exchanging the calls between two PBXs. This deployment is often found when connecting the UCM to an ITSP service line.

Peer SIP Trunk

Peer trunk allows to create a peer-to-peer connection between two PBXs over the internet or in an internal network. This allows the calls to be exchanged between the PBXs.

Create New IAX Trunk

Trunk Groups

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.

Once creating the new trunk group and configuring the SIP settings, users can add multiple accounts within the configured SIP server by pressing

button and configuring the username, password, and authentication ID fields.

WebRTC Trunks

WebRTC (Web Real-Time Communication) is a real-time framework that enables audio and video communication directly through web browsers. The UCM63xx series supports creating WebRTC trunks to connect web applications with the PBX, allowing visitors or customers to initiate calls or join meetings directly from a website.

Administrators can configure parameter settings for each WebRTC trunk and easily integrate call options into their web pages using the generated link addresses or script code snippets.

Below is a table that describes the options to configure when setting up this feature:

Parameter	Description
Basic Settings
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 trunk will automatically be recorded.
Allow Video Calling	Enables video call support for WebRTC connections established over this trunk.
One-Click Dialing	If enabled, calls are initiated immediately upon clicking the web link without requiring secondary confirmation or user prompts.
Jitter Buffer	Configures the jitter buffer mechanism (e.g., NetEQ) to optimize audio packet handling and stability over web networks.
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 threshold.
Incoming Concurrent Call Threshold	Threshold of all incoming concurrent calls passing through this trunk.
WebRTC Inbound Link Address	Displays the unique URL generated for this trunk. This link can be embedded into a web page or entered directly in a browser address bar to initiate calls.
Advanced Settings
Packet Loss Retransmission	Configures packet loss recovery methods (e.g., NACK+RTX) to maintain call quality during network congestion.
Codec Preference	Specifies and prioritizes the supported audio and video codecs (e.g., PCMU, PCMA, Opus, G.722, H.264) used for WebRTC trunk calls.

Direct Outward Dialing (DOD)

The UCM630xA 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 to 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 UCM630X:

1. To setup DOD go to UCM630xA Web GUI→Extension/Trunk→VoIP Trunks page.
2. Click on the “DOD” icon 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.

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

From this page users can:

• Add Direct Outward Dialing (DOD) entries manually.
• Import DOD entries from CSV or XLSX files.
• Export DOD entries to CSV or XLSX files.
• Filter DOD entries by number or name.
• Select multiple DOD entries for bulk operations.
• Delete one or multiple selected DOD entries in batch.

SLA STATION

The UCM630xA supports SLA that allows mapping the key with LED on a multi-line phone to different external lines. When there is an incoming call and the phone starts to ring, the LED on the key will flash in red and the call can be picked up by pressing this key. This allows users to know if the line is occupied or not. The SLA function on the UCM630xA is like BLF but SLA is used to monitor external line i.e., analog trunk on the UCM630xA. Users could configure the phone with BLF mode on the MPK to monitor the analog trunk status or press the line key pick up call from the analog trunk on the UCM630xA.

Create/Edit SLA Station

SLA Station can be configured on Web GUI🡪Extension/Trunk🡪SLA Station.

• Click on
  

  

  
  to add an SLA Station.
• Click on
  

  

  
  to edit the SLA Station. The following table shows the SLA Station configuration parameters.
• Click on
  

  

  
  to delete the SLA Station.

SLA Station Configuration Parameters

Station Name	Configure a name to identify the SLA Station.
Station	Specify a SIP extension as a station that will be using SLA.
Available SLA Trunks	Existing Analog Trunks with SLA Mode enabled will be listed here.
Associated SLA Trunks	Select a trunk for this SLA from the Available SLA Trunks list. Click on   to arrange the order. If there are multiple trunks selected, when there are calls on those trunks at the same time, pressing the LINE key on the phone will pick up the call on the first trunk here.
SLA Station Options
Ring Timeout	Configure the time (in seconds) to ring the station before the call is considered unanswered. No timeout is set by default. If set to 0, there will be no timeout.
Ring Delay	Configure the time (in seconds) for delay before ringing the station when a call first coming in on the shared line. No delay is set by default. If set to 0, there will be no delay.
Hold Access	This option defines the competence of the hold action for one particular trunk. If set to “open”, any station could hold a call on that trunk or resume one held session; if set to “private”, only the station that places the trunk call on hold could resume the session. The default setting is “open”.

Sample Configuration

1. On the UCM630xA, go to Web GUI🡪Extension/Trunk🡪Analog Trunks page. Create analog trunk or edit the existing analog trunk. Make sure “SLA Mode” is enabled for the analog trunk. Once enabled, this analog trunk will be only available for the SLA stations created under Web GUI🡪Extension/Trunk🡪SLA Station page.

1. Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.

1. On the UCM630xA, go to Web GUI🡪Extension/Trunk🡪SLA Station page, click on “Add”. Please refer to section [Create/Edit SLA Station] for the configuration parameters. Users can create one or more SLA stations to monitor the analog trunk. The following figure shows two stations, 1002 and 1005, are configured to be associated with SLA trunk “fxo1”.

1. On the SIP phone 1, configure to register UCM630xA extension 1002. Configure the MPK as BLF mode and the value must be set to “extension_trunkname”, which is 1002_fxo1 in this case.
2. On the SIP phone 2, configure to register UCM630xA extension 1005. Configure the MPK as BLF mode and value must be set to “extension_trunkname”, which is 1005_fxo1 in this case.

Now the SLA station is ready to use. The following functions can be achieved by this configuration.

• Making an outbound call from the station/extension, using LINE key

When the extension is in idle state, pressing the line key for this extension on the phone to off hook. Then dial the station’s extension number, for example, dial 1002 on phone 1 (or dial 1005 on phone 2), to hear the dial tone. Then the users could dial external number for the outbound call.

• Making an outbound call from the station/extension, using BLF key

When the extension is in idle state, pressing the MPK and users could dial external numbers directly.

• Answering call using LINE key

When the station is ringing, pressing the LINE key to answer the incoming call.

• Barging-in active call using BLF key

When there is an active call between an SLA station and an external number using the SLA trunk, other SLA stations monitoring the same trunk could join the call by pressing the BLF key if “Barge Allowed” is enabled for the analog trunk.

• Hold/UnHold using BLF key

If the external line is previously put on hold by an SLA station, another station that monitors the same SLA trunk could UnHold the call by pressing the BLF key if “Hold Access” is set to “open” on the analog trunk and the SLA station.

CALL ROUTES

Outbound Routes

This section explains how to configure and manage outbound routes on the UCM630xA. Outbound routes control all external outgoing calls initiated by the UCM, regardless of trunk type (SIP, Analog, or Digital).

Configuring Outbound Routes

An outbound route in the UCM630xA links a dial pattern to a trunk, allowing different call types to use different trunks. For example:

• Local calls (7 digits) → FXO trunk
• Long-distance calls (10 digits) → low-cost SIP trunk

Users can also configure a failover trunk to handle calls if the primary trunk is unavailable.

To manage outbound routes:

1. Go to Web GUI→Extension/Trunk→Outbound Routes.

2. Click the Add button to create a new outbound route.
3. Click Import to upload routes in CSV or XLSX format.
4. Click Export to download routes in CSV or XLSX format.
5. Click the Edit icon to modify a route.
6. Click the Delete icon to remove a single route, or use the selection feature to delete multiple outbound routes at once.

On the UCM630xA, route selection is determined by the Outgoing Routing Priority feature, which can prioritize routes by either list order or pattern accuracy. For detailed configuration, see the Outgoing Routing Priority section in this guide.

When creating an outbound route, users can configure the parameters described in the table below:

Failover Trunk Toggles

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

If a call receives the selected response codes, the UCM will not be 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 DOD number based on Outbound Route, as displayed on the screenshot below. For each outbound route.

Outbound Blacklist

The UCM630xA allows users to configure 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 in the blacklist. Please note that the blacklist settings apply to all outbound routes.

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

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 the UCM6300 Series 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.

Scheduled Sync

The UCM630xA allows users to synchronize the outbound routes, this feature can be found on the Web GUI🡪Extension/Trunk🡪Outbound Routes🡪 Scheduled Sync.

Outbound Routes/Scheduled Sync

Scheduled Sync	Enable the Scheduled Sync feature
Server Address	Enter the TFTP server address. For example, “192.168.1.2:69”.
File Name	Specify the file name
Sync Time	Enter the sync time (24hr format). Valid range is 0-23.
Sync Frequency	Create new sync every x day(s). The valid range is 1 to 30.

PIN Groups

The UCM630xA supports pin group. Once this feature is configured, users can apply pin group to specific outbound routes. When placing a call on pin protected outbound routes, caller will be asked to input the group pin number, this feature can be found on the WebGUI🡪Extension/Trunk🡪Outbound Routes🡪PIN Groups.

Outbound Routes/PIN Group

Name	Specify the name of the group
Record In CDR	Specify whether to enable/disable record in CDR
PIN Number	Specify the code that will asked once dialing via a trunk
PIN Name	Specify the name of the PIN

Once user click on

the following figure shows to configure the new PIN.

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

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

• 
  Importing PIN Groups from CSV files:

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

1. Navigate to Extension/Trunk🡪Outbound Routes🡪PIN Groups and click on the “Upload” button.

1. Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in error messages such as the one below:

1. To ensure a successful import, please follow the format in the sample image below

• 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.

Carrier

LCR, Least Cost Routing, is a mechanism in which the user can set the priority for the trunks to use when emitting a call. Typically for Least Cost Routing, the user can set higher priority for the trunks which have a lower call rate. This is an effective approach to use the trunks which are have cheaper lower cost, and in the case when they cannot be used, like when they are down or many concurrent calls, the next trunk in the list will be used.

To configure this option, navigate to Extension/Trunk > Outbound Routes and click on Carrier.

On the list of carriers, please click on “Add”.

Select the trunks to apply the rates on, the add the rate in Carrier Rates section.

Click on “Add” button under Carrier Rates section.

Outgoing Routing Priority

Outgoing Routing Priority Policy feature allows defining the priority approach for the outbound routes. Using this feature, the administrator can select either the accuracy of the dial pattern, or the position of the outbound route in the list to determine the route of the call.

To configure this feature, please click on the following button on the Outbound Routes page.

Selecting one of the policies will result in the following behaviour.

• List Sorting Priority: Selecting this option will result in taking the sorting order of the list of outbound routes as a matching criteria for the dial pattern. The route with the first matching dial plan will be used to route the call. Any routes further down the list with a more resembling dial pattern will not be taken into consideration.
• Accuracy First: Selecting this option will result in choosing the route with the most resembling dial pattern. In contrast with List Sorting Priority, the position of the route in the list is irrelevant and will not be taken into consideration.

Inbound Routes

Inbound routes allow the UCM630X to control and forward incoming external calls to specific destinations based on defined criteria. These criteria can include source caller ID, destination number, and other route-level settings. A default destination can also be set to handle calls that do not match any specific criteria.

Configuring Inbound Routes

To manage inbound routes:

1. Go to Web GUI→Extension/Trunk→Inbound Routes.

2. Click the Add button to create a new inbound route.
3. Click Blocklist to configure numbers or patterns that should be blocked on all inbound routes.
4. Click Import to upload routes in CSV or XLSX format.
5. Click Export to download routes in CSV or XLSX format.
6. Click the Edit icon to modify an existing inbound route.
7. Click the Filter button to search in the list inbound routes using specific filters.
8. Click the Delete icon to remove a single route, or use the selection feature to delete multiple routes at once.

When creating an inbound route, users can configure the parameters described in the table below:

Inbound Route: Prepend Example

UCM630xA now allows user to prepend digits to an inbound DID pattern, with strip taking precedence over prepend. With the ability to prepend digits in inbound route DID pattern, 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, UCM630xA will strip the first 8 digits.
3. If Prepend is set to 2, UCM630xA will then prepend a 2 to the stripped number, now the number become 2163.
4. UCM630xA will now forward the incoming call to extension 2163.

Inbound Route: Multiple Mode

The UCM630X allows configuring an inbound route to enable multiple mode to switch between different destinations. The inbound multiple mode can be enabled under Inbound Route settings.

When Multiple Mode is enabled for the inbound route, the user can configure a “Default Destination” and up to 9 destination modes for all routes (Mode 1, Mode 2, etc). By default, calls coming into the inbound routes will be routed to the default destination.

SIP end devices registered on the UCM630X can dial feature codes (e.g., *62) to switch to “Mode 1” and dial feature code *61 to switch back to “Default Destination”. These feature codes can be customized under Extension/Trunk → Inbound Routes → Set Global Inbound Mode (ensure they do not conflict with other configured feature codes). Switching modes can be easily performed via phone keypads without logging into the Web GUI.

To configure Global Inbound Multi-Modes, navigate to Extension/Trunk → Inbound Routes, click on Set Global Inbound Mode, and check the Global Inbound Multi-Mode option.

The following options are available under the Inbound Mode configuration list:

Inbound Route: Custom Inbound Mode

In the UCM630X, users can enable Custom Inbound Mode to switch between different destinations for each specific inbound route. The Custom Inbound Mode can be enabled under Inbound Route settings.

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

When Custom Inbound Mode is enabled, the user can configure a “Default Destination” and or a “Mode X” destination for each specific route. This Inbound route will use the inbound mode specified in this page’s Inbound Mode setting instead of the configured Global Inbound Mode.

Users can toggle the route’s inbound mode by dialing “Global Inbound Mode feature code + Inbound Mode Feature Code” 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 Feature Code 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 Custom 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: Set Global Inbound Mode (Inbound Mode BLF Monitoring)

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

To do this, please refer to the following steps:

1. Access the UCM web GUI and navigate to Extension/Trunk🡪Inbound Routes.
2. Click on the

   

   button and enable Global Inbound Multi-Mode.
3. Edit the subscribe number field to the desired BLF value.

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”.

5. Pressing the key will toggle the inbound mode to “Mode 1”, and the button’s color will change to red.

Inbound Routes: 3rd Party Database Search

This feature allows the UCM to perform a real-time lookup using an external database when an inbound call is received. If the incoming caller number matches a record in the external database, the corresponding name is retrieved and displayed as the caller ID.

To access this feature, navigate to Extension/Trunk→Inbound Routes and click on the “3rd Party Database Search” button.

On this page, users can enable or disable third-party database lookup for inbound routes by enabling or disabling the 3rd Party Database Search option.

This page also allows users to specify the database objects used by the UCM to construct the SQL query for inbound caller name lookup. The parameters configured here must exactly match the corresponding table and column names in the external database.

The following table describes the parameters used by the UCM to generate the SQL lookup query for inbound caller name resolution.

Based on the configured query keys, the UCM automatically generates and executes a SQL query in the following format:

SELECT <Caller Name Column>
FROM <Table>
WHERE <Number Column> LIKE '%[INBOUND_NUMBER]%';

The inbound caller number is dynamically substituted at call time.

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.

The imported file should be on CSV format and using UTF-8 encoding, the imported file should contain 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… 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… User 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… User 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.

FAX with Two Media

The UCM630xA supports Fax re-INVITE with multiple codec negotiation. If a Fax re-INVITE contains both T.38 and PCMA/PCMU codec, UCM630xA will choose T.38 codec over PCMA/PCMU.

Blacklist Configurations

In the UCM630xA, Blacklist is supported for all inbound routes. Users could enable the Blacklist feature and manage the Blacklist by clicking on “Blacklist”.

• Select the checkbox for “Blacklist Enable” to turn on Blacklist feature for all inbound routes. Blacklist is disabled by default.
• Enter a number in “Add Blacklist Number” field and then click ”Add” to add to the list. Anonymous can also be added as a Blacklist Number by typing “Anonymous” in Add Blacklist Number field.
• To remove a number from the Blacklist, select the number in “Blacklist list” and click on
  

  

  
  or click on ”Clear” button to remove all the numbers on the blacklist.
• User can also export the inbound route blacklist by pressing on
  

  

  
  button.

• To add blacklist number in batch, click on “Import” to upload blacklist file in csv format. The supported csv format is as below.

WHATSAPP TRUNK

The WhatsApp Trunk feature allows the UCM Series to integrate directly with Meta WhatsApp Business accounts as a specialized SIP trunk. Once configured, businesses can handle incoming voice calls initiated by customers from WhatsApp, as well as make outbound WhatsApp voice calls directly to authorized clients through Wave applications or registered endpoints.

To create a new WhatsApp Trunk, go to Extension/Trunk→ WhatsApp Trunk and click on Add.

Configure the parameters as described in the table below:

Parameter	Description
Trunk Name	Enter a unique label to identify the WhatsApp Trunk across inbound routes and reporting. Supports 2–64 alphanumeric characters, Chinese characters, and spaces. Special characters ` ' " \ = & $ | < > , ; are not supported.
Disable This Trunk	If selected, the trunk will be disabled. Inbound and outbound WhatsApp calls passing through this trunk will be blocked.
Auto Record	If enabled, all calls handled by this WhatsApp Trunk will automatically be recorded. Recordings can be accessed under CDR → Recording Files.
Phone Number	Enter the business WhatsApp phone number associated with the account. Supports 1–64 characters including digits, letters, and special characters + * # - _ . $ &.
Phone Number ID	Enter the unique Phone Number ID assigned to the business number in the Meta WhatsApp Business API console. Supports 1–64 characters including numbers and the + symbol.
Token	Enter the Meta WhatsApp API authentication Token. Supports 1–256 characters including letters, numbers, and special characters - + _ . * #.

Inbound and Outbound Whatsapp Routing

Inbound Routing (Manual)

Because inbound call matching is strictly tied to the trunk’s unique phone number, each WhatsApp Trunk supports exactly one dedicated Inbound Route and must be configured manually:

1. Navigate to Extension/Trunk→Inbound Routes.
2. In the top-right trunk selection dropdown, choose the target WhatsApp Trunk.
3. Click Add.
4. Enter an Inbound Rule Name and select the desired Default Destination.
5. Click Save and Apply Changes.

Outbound Routing (Automatic Provisioning)

When a WhatsApp Trunk is created on the UCM, the system automatically creates a default outbound route bound to the trunk’s unique phone number.

Manual outbound route configuration is not required. When a Wave user initiates an outbound call to an authorized WhatsApp client number, Wave attaches a SIP header containing the trunk number.

The UCM automatically matches the request header to the default outbound route associated with that WhatsApp Trunk and places the call.

FAX SERVER

The UCM6300A series supports T.30/T.38 Fax and Fax Pass-through. It can convert the received Fax to PDF format and send it to the configured Email address. Fax/T.38 settings can be accessed via Web Advanced Call Features > FAX/T.38. The list of received Fax files will be displayed in the same web page for users to view, retrieve and delete.

Configure Fax/T.38

• Click on “Create New Fax Extension”. In the popped-up window, fill the extension, name and Email address to send the received Fax to.
• Click on “Fax Settings” to configure the Fax parameters.
• Click on
  

  

  
  to edit the Fax extension.
• Click on
  

  

  
  to delete the Fax extension.

FAX/T.38 Settings

Enable Error Correction Mode	Configure to enable Error Correction Mode (ECM) for the Fax.   The default setting is “Yes”.
Maximum Transfer Rate	Configure the maximum transfer rate during the Fax rate negotiation.   The possible values are 2400, 4800, 7200, 9600, 12000 and 14400. The default setting is 14400.
Minimum Transfer Rate	Configure the minimum transfer rate during the Fax rate negotiation. The possible values are 2400, 4800, 7200, 9600, 12000 and 14000. The default setting is 2400.
Max Concurrent Sending Fax	Configure the concurrent fax that can be sent by UCM6300A. Two modes “Only” and “More” are supported.   Only Under this mode, the UCM6300A allows only single user to send fax at a time. More Under this mode, the UCM6300A supports multiple concurrent fax sending by the users. By default, this option is set to “only”.
Fax Queue Length	Configure the maximum length of Fax Queue from 6 to 10.   The default setting is 6.
User Information in Fax Header	If enabled this this will give users the option to send a special header in SIP fax messages.
Fax Header Information	Adds fax header into the fax file.
Default Email Address	Configure the Email address to send the received Fax to if user’s Email address cannot be found.   Note: The extension’s Email address or the Fax’s default Email address needs to be configured in order to receive Fax from Email. If neither of them is configured, Fax will not be received from Email.
Template Variables	Fill in the “Subject:” and “Message:” content, to be used in the Email when sending the Fax to the users.   The template variables are: ${CALLERIDNUM} : Caller ID Number ${CALLERIDNAME} : Caller ID Name ${RECEIVEEXTEN} : The extension to receive the Fax ${FAXPAGES} : Number of pages in the Fax ${VM_DATE} : The date and time when the Fax is received
Send PDF Files Only	If enabled, fax emails will no longer attach TIFF files. Only PDF files will be attached.
Enable Fax Resend	Enables the fax resend option which allow the UCM to keep attempting to send faxes up to a specified amount of times. Additionally, if a fax still fails to send, a Resend button will appear in the File Send Progress list in OtherFeatures🡪Fax Sending to allow manual resending.
Max Resend Attempts	Configures the maximum attempts number to resend the fax.   Default value is set to 5.
Fax Resend Frequency	Configures the Fax Resend Frequency.   Default value is set to 50.

Receiving Fax

Example Configuration to Receive Fax from PSTN Line

The following instructions describe how to use the UCM6300A to receive fax from PSTN line on the Fax machine connected to the UCM6300A FXS port.

1. Connect Fax machine to the UCM6300A FXS port.
2. Connect PSTN line to the UCM6300A FXO port.
3. 
   Go to Web GUI🡪Extension/Trunk page.
4. Create or edit the analog trunk for Fax as below.

Fax Detection: Make sure “Fax Detection” option is set to “NO”.

1. Go to UCM6300A Web GUI🡪Extension/Trunk🡪Extensions page.
2. Create or edit the extension for FXS port.

• Analog Station: Select FXS port to be assigned to the extension. By default, it is set to “None”.
• Once selected, analog related settings for this extension will show up in “Analog Settings” section.

1. Go to Web GUI🡪Extension/Trunk🡪Inbound Routes page.
2. Create an inbound route to use the Fax analog trunk. Select the created extension for Fax machine in step 4 as the default destination.

Now the Fax configuration is done. When there is an incoming Fax calling to the PSTN number for the FXO port, it will send the Fax to the Fax machine.

Example Configuration for Fax-To-Email

The following instructions describe a sample configuration on how to use Fax-to-Email feature on the UCM6300A.

1. Connect PSTN line to the UCM6300A FXO port.
2. Go to UCM6300A Basic Call Feature > Fax/T.38 page. Create a new Fax extension.

3. Go to Extension/Trunk🡪Analog Trunks page. Create a new analog trunk. Please make sure “Fax Detection” is set to “No”.
4. Go to Extension/Trunk🡪Inbound Routes page. Create a new inbound route and set the default destination to the Fax extension.

5. Once successfully configured, the incoming Fax from external Fax machine to the PSTN line number will be converted to PDF+Tiff file and sent to the extension 7200 and email address fax@domain.local as attachment.

FAX Sending

Besides the support of Fax machines, The supports also sending Fax via Web GUI access. This feature can be found on Web GUI🡪Advanced Call Features🡪Fax Sending page. To send fax, pre-setup for analog trunk and outbound route is required. Please refer to [ANALOG TRUNKS], [VOIP TRUNKS] and [Outbound Routes] sections for configuring analog trunk and outbound route.

After making sure analog trunk or VoIP Trunk is setup properly and UCM6300A can reach out to PSTN numbers via the trunk, on Fax Sending page, enter the fax number and upload the file to be faxed. Then click on “Send” to start. The progress of sending fax will be displayed in Web GUI. Users can also view the sending history is in the same web page.

After that you can see the ongoing sending operation on the progress bar.

MEETING

With the UCM you can easily create, schedule, manage, and join meeting calls, from your desktop or laptop computer. UCM conferencing uses WebRTC technology, so all the participants don’t have to download and install any additional software or plugins. UCM conferencing must be enabled by the administrator for the concerned extensions. The meeting configurations can be accessed under Web GUI🡪Basic Call Features🡪Mutimedia Meeting. In this page, users could enable, set the Basic setting, create, edit, view, manage, delete meeting rooms, and edit the Meeting Schedule.

UCM630xA series	Number of meeting room	Participant limit
UCM6300A	3	50
UCM6302A	5	75
UCM6304A	7	120
UCM6308A	9	150

Below are the UCM meeting specifications supported for each model:

Room

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

  

  
  to edit the meeting room.
• Click on
  

  

  
  to delete the meeting room.

Meeting Room Configuration Parameters

Meeting Settings contains the following options:

Meeting Settings

Schedule Meeting

Meeting Schedule can be found under UCM Basic Call Features🡪Mutimedia Meeting🡪Meeting Schedule. Users can create, edit, view, and delete a Meeting Schedule.

• Click on “Add” to add a new Meeting Schedule.
• Click on the scheduled meeting to edit or delete the event.

The Meeting Schedule Parameters are described below:

Once created, at the scheduled meeting time, UCM630xA will send INVITE to the extensions that have been selected for meeting.

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

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

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

Meeting Recordings

The UCM630xA allows users to record the meeting call and retrieve the recording from Web GUI🡪Basic Call Features🡪Meeting🡪Meeting Recordings.

To record the meeting call, when the meeting room is in idle, enable “Record Meeting” 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 as 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.

ONSITE MEETING

For workplaces that require employees to return to physical offices for work, Grandstream UCM offers the Onsite Meeting feature, a new way to stay organized and keep up-to-date with in-person meetings. This feature allows administrators to create and manage onsite meeting rooms, specify meeting room locations, schedule meetings, and add conferencing equipment. The new feature can be found under the Device Management > Onsite Meeting page. The first page that appears is the Scheduled Meetings page and tab page, which provide an overview of all created meeting rooms. It provides information about the rooms’ meeting schedules for the day, their locations, their member capacity, and their equipment.

The Pending Meeting tab and Meeting History tab show detailed information about upcoming meetings and previous meetings respectively. From the Pending Meeting tab, users can delete upcoming meetings and extend the duration of ongoing meetings. The Meeting History tab will display the last 6 months of onsite meeting.

When a meeting is scheduled the invitees will receive an invitation via email which contains all the information regarding the onsite meeting. The email will be attached with an .ics file which can be imported to a calendar to mark your calendar of the onsite meeting, please see the screenshot below.

IVR

Basic Settings

IVR configurations can be accessed under the UCM630xA 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.

IVR Configuration Parameters

Key Pressing Events

Standard Key Event

UCM 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.

The following table describes available options that can be assigned for Standard Key Pressing Events:

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 feature supports the following:

• Up to 100 custom key press events
• Each key combination can contain up to 8 characters (numbers, star (*), and pound (#) depending on End Key settings)
• Supports Time Conditions
• Different custom keys can have the same Destination and Time Condition

The following options are available for Custom Key Pressing Events:

VOICE PROMPTS

The UCM630xA supports multiple languages in Web GUI as well as system voice prompt. Currently, there are 16 languages supported in system voice prompt: British English, Deutsch, English, Spanish, Spanish (Catalonia), Spanish (Spain), Greek, French, Italian, Dutch, Polish, Portuguese (Brazil), Portuguese (Portugal), Russian, Swedish, Turkish, Czech, Ukrainian, Hebrew, Arabic, and Chinese.

English (United States) and Chinese voice prompts are built in with the UCM630xA already. The other languages provided by Grandstream can be downloaded and installed from the UCM630xA Web GUI directly. Additionally, users could customize their own voice prompts, package them and upload to the UCM630xA.

Language settings for voice prompt can be accessed under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Download and Install Voice Prompt Package

To download and install voice prompt package in different languages from UCM630xA Web GUI, click on “Add Voice Prompt Package” button.

A new dialog window of voice prompt package list will be displayed. Users can see the version number (latest version available V.S. current installed version), package size and options to upgrade or download the language.

Click on to download the language to the UCM630xA. The installation will be automatically started once the downloading is finished.

A new language option will be displayed after successfully installed. Users then could select it to apply in the UCM630xA system voice prompt or delete it from the UCM630xA.

Custom Prompt

Record New Custom Prompt

In the UCM630xA PBX Settings→Voice Prompt→Custom Prompt page, click on “Record” and follow the steps below to record new IVR prompt.

1. Specify the file name.
2. Select the format (GSM or WAV) for the prompt file to be recorded.
3. Select the extension to receive the call from the CloudUCM to record the custom prompt.
4. Click the “Record” button. A request will be sent to CloudUCM to call the extension for recording the custom prompt from the phone.
5. Pick up the call from the extension and start the recording following the voice prompt.
6. Once saved, the new custom prompt will be listed on this page. From here, you can manage it by selecting an action:
   
   • Online Playback: listen to the prompt directly from the Web GUI.
   • Playback Using Extension: select an extension to hear the prompt played over a call.
   • Record Again: replace the current recording with a new one.
   • Download: download the custom prompt recorded.
   • Delete: remove the prompt from the system.

Upload Custom Prompt

If the user has a pre-recorded IVR prompt file, click on “Upload” in Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page to upload the file to the UCM630xA. The following are required for the IVR prompt file to be successfully uploaded and used by the UCM630xA:

• PCM encoded.
• 16 bits.
• 8000Hz mono.
• In .mp3 or .wav format; or raw/ulaw/alaw/gsm file with .ulaw or .alaw suffix.
• File size under 5MB.

Click on button, select the file you want to upload, then click on “choose file to upload” to start uploading. Once uploaded, the file will appear in the Custom Prompt web page.

If the file uploaded is mp3, the following prompt will appear.

Tick “Keep Copy of MP3” to keep an mp3 copy of the file which can be diffused with certain GSC speakers. Diffusing the MP3 files result in a highr quality sound played by GCS speakers.

Download All Custom Prompt

On the UCM630xA, the users can download all custom prompts from UCM Web GUI to local PC. To download all custom prompt, log in UCM Web GUI and navigate to PBX Settings🡪Voice Prompt🡪Custom Prompt and click on ”Download All”. The following window will pop up in order to set a name for the downloaded file.

Username Prompt

There are two ways to customize/set new username prompt:

Upload Username Prompt File from Web GUI

1. First, Users should have a pre-recorded file respecting the following format:
   • PCM encoded / 16 bits / 8000Hz mono.
   • In .tar/.tar.gz/.tgz format
   • File size under 30M.
   • Filename must be set as the extension number with 18 characters max. For example, the recorded file name 1000.wav will be used for extension 1000.

2. Go under web GUI PBX Settings 🡪 Voice Prompt 🡪 Username Prompt and click on ”Upload” button.
3. Select the recorded file to upload it and press Save and Apply Settings.
   • Click on

     

     to record again the username prompt.
   • Click on to play recorded username prompt.
   • Select username prompts and press

     

     to delete specific file or select multiple files for deletion using the button ”Delete” .

Record Username via Voicemail Menu

The second option to record username is using voicemail menu, please follow below steps:

• Dial *98 to access the voicemail
• After entering the desired extension and voicemail password, dial “0” to enter the recordings menu and then “3” to record a name.

Another option is that each user can record their own name by following below steps:

• The user dials *97 to access his/her voicemail
• After entering the voicemail password, the user can press “0” to enter the recordings menu and then “3” to record his name.

Call Prompt Tones

SIP Trunk Prompt Tone

Prompt Tone Settings tab on the UCM helps users choose which prompt will be played by the UCM during call failure, the voice message responses can be set to be played for 4XX, 5XX, and 6XX call failures.

The following default prompt names and their associated messages are supported:

Additionally, custom voice messages recorded and uploaded in PBX Settings→Voice Prompt→Custom Prompt can be used for the failure responses instead of the default messages.

General Call Prompt Tones

Moreover, users also have the possibility to customize the prompt for typical call failure reasons like (no permission to allow outbound calls, busy lines, incorrect number dialed …Etc.).

To customize these prompts user could record and upload their own files under PBX Settings > Voice Prompt > Custom Prompts then select each one for specific call failure case under PBX Settings -> Call Prompt Tones > General Call Prompt Tones page as shown on the following figure:

Alert-info Prompt

The user can change upload the ringtone files which can be used with Alert-info to play various ringtones depending on the configured behaviour. In this setting, the user can view all the ringtones uploaded, play them, download them, or delete them. If the user wishes to upload new ringtones, he/she can click on “Upload” and choose the ringtone files to upload.

VOICEMAIL

Configure Voicemail

If voicemail is enabled for UCM630xA extensions, global voicemail configurations can be managed under Basic Call Features → Voicemail → Voicemail.

The table below describes the global voicemail configuration parameters:

Resetting an extension resets the Voicemail Password, Send Voicemail to Email, and Keep Voicemail after Emailing values to defaults. Previous custom voicemail prompts and recorded messages will be deleted.

Access Voicemail

When voicemail is enabled for an extension, users can access voicemail using two distinct feature codes:

• My Voicemail: Dialing the dedicated feature code for the user’s own extension (by default *97) directly accesses the user’s mailbox without requiring the extension number to be entered. The user only needs to enter their voicemail password/PIN.
• Voicemail Access: Dialing the general voicemail feature code (by default *98) allows accessing any extension’s mailbox. Users can dial *98 alone and enter the target extension number when prompted, or dial *98 directly followed by the target extension number (e.g., *981005), followed by the mailbox password/PIN.

After authenticating, both feature codes connect to the same interactive Voicemail IVR menu, which first announces unread voice messages.

The diagram below describes the structure of the Voicemail IVR Menu:

Leaving Voicemail

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

In case if 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 press1 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 UCM630xA can be configured to send the voicemail as attachment to Email. Click on “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

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

Configure Voicemail Group

The UCM630xA supports voicemail groups, allowing all member extensions assigned to a group to receive voicemail messages left for the group extension. Voicemail groups can be configured under Basic Call Features → Voicemail → Voicemail Group. Click on Add to configure a new group.

When creating or editing a Voicemail Group, administrators can configure the parameters detailed in the table below:

RING GROUP

The UCM630xA supports ring group feature with different ring strategies applied to the ring group members. This section describes the ring group configuration on the UCM630xA.

Configure Ring Group

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

• Click on
  

  

  
  to add ring group.
• Click on
  

  

  
  to edit the ring group. The following table shows the ring group configuration parameters.
• Click on
  

  

  
  to delete the ring group.

Remote Extension in Ring Group

Remote extensions from the peer trunk of a remote UCM630xA can be included in the ring group with local extension. An example of Ring Group with peer extensions is presented in the following:

1. Creating SIP Peer Trunk between both UCM630xA _A and UCM630xA _B. SIP Trunk can be found under Web GUI🡪Extension/Trunk🡪VoIP Trunks. Also, please configure their Inbound/Outbound routes accordingly.
2. Click edit button in the menu
   

   

   
   , and check if Sync LDAP Enable is selected, this option will allow UCM630xA_A update remote LDAP server automatically from peer UCM630xA _B. In addition, Sync LDAP Password must match for UCM630xA _A and UCM630xA _B to sync LDAP contact automatically. Port number can be anything between 0~65535, and use the outbound rule created in step 1 for the LDAP Outbound Rule option.

3. In case if LDAP server does not sync automatically, user can manually sync LDAP server. Under VoIP Trunks page, click sync button shown in the following figure to manually sync LDAP contacts from peer UCM630xA.

4. Under Ring Groups setting page, click “Add”. Ring Groups can be found under Web GUI🡪 Basic Call Features 🡪 Ring Groups.
5. If LDAP server is synced correctly, Available LDAP Numbers box will display available remote extensions that can be included in the current ring group. Please also make sure the extensions in the peer UCM630xA can be included into that UCM630xA’s LDAP contact.

RESTRICT CALLS

The Restrict Calls feature allows administrators to control outgoing call permissions for specific internal extensions. When a rule is enabled, users designated as Members are restricted to placing outbound calls only to the extensions listed in the Allowed List.

This section describes the configuration of this feature in the Advanced Call Features → Restrict Calls page.

Configure Restrict Calls

• Click Add to create a new Restrict Calls rule.
• Click the Edit icon to modify an existing rule.
• Click the Delete icon to remove a rule.

When creating or editing a rule, users can configure the following:

PAGING/INTERCOM

Paging and Intercom Group can be used to make an announcement over the speaker on a group of phones. Targeted phones will answer immediately using speaker. The UCM630xA paging and intercom can be used via feature code to a single extension or a paging/intercom group. This section describes the configuration of paging/intercom group under Web GUI🡪Basic Call Features🡪Paging/Intercom.

Paging/Intercom Groups

2-way Intercom

Private Intercom

Private intercom allows the user to initiate an intercom to many endpoints. Whichever endpoint microphone has detected sound input first, only the intercom initiator and the responder will be able to hear each other. Once the first responder has finished talking, the second responder can start talking. To configure private intercom, the user can follow the steps mentioned above in 2-way intercom and while creating the intercom, the user may tick the option “Private Intercom” as indicated in the screenshot below.

1-way Paging

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

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

Multicast Paging

Multicast Community

Multicast community allows creating an extension, which when dialed, can send a preconfigured prompt as a multicast paging to a group of extensions. The user should create first a Paging/Intercom Group with Multicast Paging as the Strategy selected. Please see previous section for more information.

Scheduled Paging/Intercom

Pending Paging/Intercom

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

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

Paging/Intercom Schedule

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

OPERATOR PANEL

Configure Operator Panel

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

The UCM630xA supports operator panel so that UCM extension can be used as admin to manage calls and activities such as extension status, call queue status, transfer, barge-in, hangup and etc. On Grandstream Wave client, it can display the extensions, ring group, voicemail, call queue, call park status under the management of the extension. This section describes how to configure operator panel.

• Click on “Add” to create operator panel.
• Click on
  

  

  
  to edit the operator panel.
• Click on
  

  

  
  to delete the operator panel.

Name	Configure the name for the operator panel created for identification purpose.
Administrator	Assign the administrator for the operator panel. It can be an extension, a extension group or a department. For the selected extension groups and departments, subsequent extensions will automatically become administrators.
Management Module
Extension	The selected extensions will be supervised by the administrator, and you can choose extensions, extension groups, and departments. For the selected extension groups and departments, subsequent extensions will be automatically supervised by the administrator.
Ring Groups	The selected Ring Groups will be supervised by the administrator. If selecting “All”, all Ring Groups and subsequent updates will be automatically supervised by the administrator.
Voicemail Groups	The selected Voicemail Groups will be supervised by the administrator. If selecting “All”, all Voicemail Groups and subsequent updates will be automatically supervised by the administrator.
Call Queue	The selected Call Queue will be supervised by the administrator. If selecting “All”, all Call Queue and subsequent updates will be automatically supervised by the administrator.
Parking Lot	The selected Parking Lot will be supervised by the administrator. If selecting “All”, all Parking Lot and subsequent updates will be automatically supervised by the administrator.

CALL QUEUE

The UCM630X supports call queue by using static agents or dynamic agents. Call Queue system can accept more calls than the available agents. Incoming calls will be held until next representative is available in the system. This section describes the configuration of call queue under Web GUI🡪 Basic Call Features🡪Call Queue.

Add Call Queue

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

• Click on “Add” to add call queue.
• Click on

  

  to edit the call queue.
• Click on

  

  to delete the call queue.

The call queue configuration parameters are listed in the table below.

• Click Add to create a new call queue.
• Click the pencil Edit icon under Options to edit queue settings.
• Click the trash Delete icon under Options to delete a call queue.

The call queue configuration parameters are listed in the table below.

Parameter	Description
Basic Settings
General
Extension	Configure the call queue extension number.
Name	Configure the call queue name to identify the call queue.
Proficiency Routing	If enabled, incoming calls will be routed to the agents with the highest available proficiency level (lower number = higher proficiency). If multiple agents have the same level, the Ring Strategy for Same Proficiency Level will be used to route calls among those agents. Please make sure to assign proficiency levels to the agents under the Agents tab. Note: This option applies to both static and dynamic agents.
Strategy	Select the strategy for the call queue: Ring All: Ring all available Agents simultaneously until one answers. Linear: Ring agents in the specified order. Least Recent: Ring the agent who has been called the least recently. Fewest Calls: Ring the agent with the fewest completed calls. Random: Ring a random agent. Round Robin: Ring the agents in Round Robin scheduling with memory. The default setting is “Ring All”.
Ring Strategy for Same Proficiency Level	Select the strategy for the call queue: Ring All: Ring all available Agents simultaneously until one answers. Linear: Ring agents in the specified order. Least Recent: Ring the agent who has been called the least recently. Fewest Calls: Ring the agent with the fewest completed calls. Random: Ring a random agent. Round Robin: Ring the agents in Round Robin scheduling with memory. The default setting is “Ring All”.
Music On Hold	Select the Music On Hold class for the call queue. Note: Music On Hold classes can be managed from Web GUI > PBX Settings > Music On Hold.
Max Queue Length	Configure the maximum number of calls to be queued at once. This number does not include calls connected with agents, only calls still in queue. When exceeded, callers hear a busy tone and are forwarded to the configured failover destination. Default is 0 (unlimited).
Agent Rest Time (s)	Configure the amount of time in seconds after ending a call where the agent will not receive additional calls. Default is 10 seconds.
Retry Time (s)	Configure the number of seconds to wait before ringing the next agent. Minimum is 1, default is 5 seconds.
Agent Ring Time	Configure the number of seconds to ring an agent. Minimum is 5, default is 30 seconds.
Auto Record	If enabled, calls on the queue will be automatically recorded. Choose to record internal calls, external calls, or all calls. Default is disabled.
Prompt Language	Choose the language of the prompt played to callers. Selecting “default” uses the language defined under PBX Settings > Voice Prompt > Language Settings.
Welcome Prompt
Enable	Enable the welcome prompt.
Custom Prompt	Choose the initial tone that plays when the user dials the queue number. Note: Custom prompts can be uploaded directly in .tar, .tgz, .tar.gz, .mp3, .wav, .gsm, .alaw, or .ulaw formats.
Play Full Welcome Prompt	If enabled, queue agents will not be rung until after the welcome prompt finishes playing. Otherwise, agents are rung while the prompt plays.
Satisfaction Survey Prompt
Custom Prompt	After an agent hangs up, a prompt plays asking the caller to rate satisfaction on a scale of 1 to 5. Survey results can be reviewed under Call Queue Statistics > Overview > Agent Satisfaction Statistics / Queue Satisfaction Statistics.
Max Wait Time
Max Wait Time	Configures the amount of time (in seconds) a caller is kept in queue before automatic routing to the Max Wait Time Destination. Default is 60 seconds.
Destination	The destination where calls are routed if no agent answers before max wait time expires.
Transfer Call Based on Time Conditions
Enable	Toggles time-based routing rules for holidays and non-working hours.
Destination During Holidays	Set the destination type and target destination during official holidays.
Destination During Non-working Hours	Set the destination type and target destination outside defined office time.
Priority Destination	Select whether holiday routing or non-working hours routing takes priority when overlapping.
Play Holiday Prompt During Holiday	Toggles playing the designated holiday voice prompt before transfer. The holiday prompt can be configured in Time Settings → Holiday.
Reset Agent Call Counter
Enable	Specifies the frequency at which the agent call counter is reset. Affects ring strategy counters and switchboard data.
Repeat	Set repeat frequency to One-time, Daily, Weekly, or Monthly.
Date/Time	Set the reset date and time.
Destination Prompt Cycle
Enable	If enabled, callers hear a custom prompt at set intervals while waiting. Pressing “1” routes them immediately to a predefined failover destination.
Destination Prompt Cycle	Configure the voice prompt cycle interval (in seconds).
Custom Prompt	Select or upload the voice prompt file played during the cycle.
Destination	Select the failover destination when callers press “1”.
Advanced Settings
Virtual Queue
Enable Virtual Queue	If enabled, activates a virtual queue allowing users to request an automated callback while retaining their position in line.
Virtual Queue Mode	Select the triggering mode for virtual queue: DTMF mode (Press 2): Callers can manually request a virtual queue callback at any time while waiting in line by pressing key 2 on their telephone keypad. The system will then play a prompt asking the caller to manually enter their preferred callback number. Timeout mode: The system automatically offers the virtual queue callback once the caller’s wait time reaches the threshold configured in Virtual Queue Period (s). Callers will hear a prompt allowing them to choose a callback number, or if Auto-fill Callback Number is enabled, the field automatically populates using their detected Caller ID.The default mode is Timeout mode.
Virtual Queue Period (s)	The amount of time in seconds that must pass before virtual queue is offered in Timeout or Auto mode.
Auto-fill Callback Number	When enabled, automatically populates the callback field with the caller’s Caller ID. Note: This option can only be enabled when Virtual Queue Mode is set strictly to Timeout mode.
Virtual Queue Outbound Prefix	System adds this prefix to dialed numbers when performing virtual queue callbacks.
Enable Virtual Queue Position Announcement	Informs waiting callers of their exact position in line.
Enable Virtual Queue Wait Time Announcement	Periodically announces estimated wait times to callers.
Enable Virtual Queue Callback Timeout	Sets a maximum time limit for agents to answer a virtual queue callback.
Write Timeout	The amount of time in seconds agents have to answer a callback.
Virtual Queue Welcome Prompt	Upload the welcome prompt file for the virtual queue.
Queue Announcement
Enable Position Announcement	Inform callers waiting in queue of their line position.
Enable Wait Time Announcement	Periodically announce estimated wait times (announced if greater than one minute).
Announcement Interval	Configure announcement frequency (20 to 600 seconds, default 20).
Agent ID Announcement	Plays the agent ID prompt to the caller upon connection.
Premium Code	Callers can enter a premium code and # to increase queue priority.
Premium Code Prompt	Shares the announcement cycle with position announcements.
Custom Announcement
Custom Prompt	Periodically play custom audio announcements to queued callers.
Announcement Interval (s)	Interval between custom announcements (20 to 600 seconds, default 60).
Empty Queue
Leave When Empty	Configure whether callers disconnect if no agents remain in queue (Yes, No, or Strict). Default is Strict.
Dial in Empty Queue	Configure whether callers can dial into an empty queue (Yes, No, or Strict). Default is No.
Failover Destination	Choose failover destination (Play Sound, Extension, Voicemail, Queue, Ring Group, Voicemail Group, IVR, External Number) when queue is empty or unstaffed.
CTI
Allow Static Agent Login/Logout	Enabling static agent login restricts dynamic agent access.
Queue Chairman	Assign extensions permitted to operate and monitor the queue via Web UI.
Priority Queue
Enable Priority Queue	Activates weighted call routing for agents who belong to multiple queues simultaneously. When enabled, the UCM prioritizes calls from queues with a higher overall weight.
Weight	Configures the base priority level for the call queue (Range: 1 to 99). Higher weight values take precedence when routing calls to multi-queue agents.
Enable Accelerated Weighting	Prevents excessive waiting times for callers in lower-weight queues. When enabled, the queue weight automatically increments by 1 after callers wait for a specified duration.
Accelerated Weighting Duration (seconds)	Configures the waiting time interval in seconds (Range: 10 to 1800) required for the queue’s overall weight to increase by 1.
Service Level Agreement (SLA)
Enable SLA	Toggles SLA measurement to track call response times against threshold goals.
SLA Time (s)	Maximum target time in seconds for agents to answer calls (1 to 180s, default 20).
SLA Alert Email Notification	Enable SLA breach email notifications.
Alert Threshold (%)	Minimum acceptable SLA answer percentage before triggering alerts (default 80%).
SLA Alert Interval (m)	Minimum time between breach email alerts (1 to 120 minutes, default 120).
SLA Alert Email Template	Select email template used for SLA breach notifications.
Alert Email Recipients	Designate email recipients for SLA breach alerts.
Other Settings
Report Hold Time	If enabled, reports caller hold duration to the agent upon call connection.
Replace Display Name	Replaces caller display name with the Call Queue name on agent devices.
Enable Feature Codes	Enable feature code operations (e.g., *83 for Agent Pause) for this queue.
Autofill	Distributes waiting calls automatically to agents as soon as they become free.
Dynamic Login Password	Requires dynamic agents to enter a PIN when logging in.
Alert-info	Specifies custom Alert-Info SIP header for alternative ringtones.
Call Memory	Prioritizes connecting returning callers to the agent they last spoke with.
Memory Retention Time (Days)	Retention window for Call Memory matching (1 to 30 days, default 7).
Enable RPID Header	Toggles sending Remote-Party-ID SIP header to caller.
Skip Busy Agent	Configures whether the queue will ring agents who are currently on an active call. When enabled, busy agents are skipped during call distribution.
Agents
Dynamic Agents	Assign users authorized to dynamically sign into or out of the queue via feature codes. When Proficiency Routing is enabled, individual skill levels can be configured here.
Static Agents	Assign static agents manually and configure their ringing order within the call queue. When Proficiency Routing is enabled, individual skill levels can be configured here.
Queue Permissions
Queue Chairman Permissions
Queue Board	Grants access to view the Queue Board (Wallboard) metrics, real-time status charts, and agent statistics for this queue.
Queue Operation Panel	Enables access to the interactive Queue Operation Panel interface for this queue.
Change the status of the agent position	Allows the chairman to modify agent login states (Log In, Logout, Pause/Resume) directly from the Queue Operation Panel.
Call Assignment Management	Allows the chairman to assign waiting calls to agents, including manual drag-and-drop call dispatching from the queued list.
Allows for answering or hanging up calls on the agent’s line	Grants remote control capabilities to answer or disconnect active calls on assigned agent lines.
Call monitoring operation	Enables live call supervision capabilities (Listen, Whisper, and Barge-In) for calls handled within this queue.
Call hold operation	Allows the chairman to remotely place active queue calls on hold or resume held calls.
View the CDR of this queue	Grants permission to view and review Call Detail Records (CDR) specifically filtered for this call queue.
Queue Agent Permissions
Queue Board	Grants access to view the Queue Board wallboard for monitoring queue metrics and performance.
Queue Operation Panel	Enables access to the Queue Operation Panel for the agent.
Call Assignment Management	Allows the agent to manage call assignments and answer or retrieve queued calls from the waiting list.
Allows for answering or hanging up calls on the agent’s line	Allows the agent to answer or hang up calls directly through the panel interface.
Call hold operation	Allows the agent to place calls on hold or resume held calls via the panel interface.
View the CDR of this queue	Grants permission to view and review Call Detail Records (CDR) specific to this call queue.

Static and Dynamic Agents

There are two types of agents that can be part of a call queue: static agents, manually added when configuring the queue, and dynamic agents, which can login and log out from the call queue anytime by using preset codes.

Static can be configured on Basic Call Features🡪Call Queue🡪Edit Queue🡪Agents. While dynamic agents only need a Login and Logout suffix to be able to join a queue.

These codes can be found under Basic Call Features🡪Call Queue🡪Global Queue Settings🡪Dynamic Agent Login Setting as shown below:

To guarantee a high level of audio quality with the call queue feature, UCMs will limit the number of agents allowed to be assigned depending on the UCM model used. If the user attempts to configure the number of static agents to be more than the maximum allowed number, a warning message will appear.

The following table lists the maximum number of agents for each UCM model:

Call Queue Feature Codes

Users can leverage feature codes to perform different call queue related actions by accessing Basic Call Features🡪Feature Codes🡪Feature Codes, below is the description of each code along with their default values:

• Agent Pause (*83): Allows agents to pause their activity in all queues for a specific reason. Once the code is dialed, the agent will be prompted to enter a pause reason represented by a digit from 0 to 9. Another way to do this is to dial the feature code and the reason code number together. (e.g. dialing *831 to directly set the pause status as “Lunchtime” (1)).
• Agent Unpause (*84): This code is used by the agent to resume activity in all queues.

• Dynamic Agent Logout (*85): Agents can dial this code to logout from all queues.

Queue Operation Panel

The Queue Operation Panel replaces the legacy Queue Switchboard, providing supervisors and administrators with a real-time interactive workspace for active call management, drag-and-drop call dispatching, live call supervision, and agent status control.

The main workspace is divided into three functional areas:

• Queued Calls: Displays all waiting callers in real time alongside their specific queue position, destination queue, state, wait time, and call type. Supervisors can manually manage waiting calls using available action controls or by clicking a waiting call from the list and dragging and dropping it directly onto an available agent card in the Agents panel to immediately assign and distribute the call.
• Current Call Count: Monitors calls currently in progress across queues, displaying the caller, target agent, queue association, call duration, and real-time call control options (such as call transfer, barge-in, listen, or whisper).
• Agents Panel: Displays agent cards grouped by queue, showing agent extension numbers, agent names, type (Static or Dynamic), current login status dropdown, and individual call count. Supervisors can toggle agent availability directly from this panel or receive drag-and-drop call assignments.

Queue Permissions

When adding a new call queue or editing an existing queue under Basic Call Features → Call Queue → Add/Edit Queue → Queue Permissions, administrators can define specific operational privileges for the Chairman and Agent roles for that queue.

The table below describes the configurable permission options available for each role:

Role	Permission Option	Description
Queue Chairman	Queue Board	Grants access to view the Queue Board (Wallboard) metrics, real-time status charts, and agent statistics for this queue.
	Queue Operation Panel	Enables access to the interactive Queue Operation Panel interface for this queue.
	Change the status of the agent position	Allows the chairman to modify agent login states (Log In, Logout, Pause/Resume) directly from the Queue Operation Panel.
	Call Assignment Management	Allows the chairman to assign waiting calls to agents, including manual drag-and-drop call dispatching from the queued list.
	Allows for answering or hanging up calls on the agent’s line	Grants remote control capabilities to answer or disconnect active calls on assigned agent lines.
	Call monitoring operation	Enables live call supervision capabilities (Listen, Whisper, and Barge-In) for calls handled within this queue.
	Call hold operation	Allows the chairman to remotely place active queue calls on hold or resume held calls.
	View the CDR of this queue	Grants permission to view and review Call Detail Records (CDR) specifically filtered for this call queue.
Queue Agent	Queue Board	Grants access to view the Queue Board wallboard for monitoring queue metrics and performance.
	Queue Operation Panel	Enables access to the Queue Operation Panel for the agent.
	Call Assignment Management	Allows the agent to manage call assignments and answer or retrieve queued calls from the waiting list.
	Allows for answering or hanging up calls on the agent’s line	Allows the agent to answer or hang up calls directly through the panel interface.
	Call hold operation	Allows the agent to place calls on hold or resume held calls via the panel interface.
	View the CDR of this queue	Grants permission to view and review Call Detail Records (CDR) specific to this call queue.

Queue Board

The Queue Board tab provides a real-time wallboard view for monitoring call center metrics, SLA compliance rates, and agent states across queues.

The Wallboard features three analytical views:

• Queue Call Metrics: Displays real-time counts for Queued Calls, Missed Calls, Answered Calls, Cancelled Calls, Transferred Calls, Maximum Wait Time, Average Wait, and Average Talk Duration.
• Queue Performance Metrics: Provides analytical top-5 ranking charts for Average Wait, Caller Cancel Rate, and Answered Calls across queues.
• Queue Agent Status: Displays real-time agent login states, pause/resume timestamps, current call counts, total talk times, and answered call tallies.

/heading

Queue Recordings

Queue recordings are shown under Basic Call Features🡪Call Queue🡪Queue Recordings.

• Click on

to download the recording file in .wav format; and on

to delete the recording file.

• Users can also download recordings in batch either by selecting specific recordings using the checkbox and clicking on

  

  or by downloading all files through

  

  the button.

Call Queue Statistics

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

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

Users can download statistics in CSV format by clicking the button. They can also set up automatic email deliveries of these statistics at regular intervals by clicking on

Additionally, users can clear the statistics using the button.

This section provides a detailed description of each tab on the Basic Call Features🡪Call Queue🡪Call Queue Statistics page.

Overview Tab

The overview page shows shows the following information:

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

By selecting a time interval, administrators can get detailed statistics for agents such as total calls, answered calls etc, as well as queue statistics like abandoned calls, transferred calls and SLA.

A more detailed version of the queue statistics (as shown in image below) can be found under Call Queue🡪Call Queue Statistics🡪Overview🡪Queue Statistics🡪Options🡪Details.

Agent Details Tab

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

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

Login Record Tab

Login Record is a report that shows the timestamps of dynamic agent logins and logouts and calculates the login duration. The following information is available:

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

Pause Log Tab

Pause Log is a report that shows information related to agent pauses. An entry will only be created after an agent unpauses. The following information is available:

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

Global Queue Settings

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

CALL FLOW DESIGN

Call Flow Design allows administrators to visually build, customize, and automate complex call routing logic using an intuitive drag-and-drop workflow canvas. Instead of relying on static, single-destination routing rules, a Call Flow acts as an advanced, dynamic process that evaluates incoming caller interactions, variables, time conditions, and input sequences before directing calls to their final destinations.

Once a Call Flow is built and enabled, it can be assigned as a destination to an inbound route allowing complex and scenario based call routing.

Call Flow System Specifications & Limits

The following table outlines the system capacity and operational limits for Call Flow Design on the UCM Series:

Parameter	Specification/Limit
Maximum Configurable Call Flows	Up to 200 total Call Flows
Maximum Nodes & Branches per Flow	Up to 200 total components (nodes and branches) per Call Flow
Maximum Sequential Steps per Flow	Up to 50 sequential execution steps per individual flow execution

Call Flow Management

To manage and configure Call Flows, navigate to Call Features→Call Flow Design.

The main Call Flow list page displays the following summary information:

Parameter	Description
Name	The name identifier for the call flow.
Enable	Displays whether the call flow is active (Enabled).
Number	The internal extension assigned to trigger or route calls into the flow. The default range reserved for call flow is 8300-8499. Note: users can configure the extension ranges under PBX Settings→Extension Preferences.
Remark	Optional descriptive notes regarding the call flow’s function.
Operations	Quick action icons to Edit (opens the visual editor), Copy (creates a duplicate call flow), or Delete the call flow.

Administrators can also select single or multiple Call Flows and click Export to save configuration files in .json (single) or .zip (multiple) formats.

Creating a Call Flow

Click on Add to create a new Call Flow. The initial setup can be done using four creation methods:

• Manual: Starts with a blank canvas containing only the mandatory Initial Action component (call flow extension).
• Based on Existing Templates: Choose from pre-built workflow templates:
  
  • Call Flow Based on IVR Key Press: Routes callers to different destinations based on single-key DTMF selections.
  • Call Flow Based on IVR Key Press: Routes callers to different destinations based on single-key DTMF selections.
  • Input Matching Authentication Flow: Prompts for DTMF digit entry and matches caller inputs against predefined conditions.
• Based on Existing Flow: Duplicates an existing Call Flow configured on the UCM.
• Import Call Flow File: Uploads a pre-configured Call Flow definition file in .json format.

Call Flow Configuration Options

When users add or edit a call flow, they’re immediately redirected to a separate editor page which provides drag-and-drop components organized by functionality.

The table below describes each of the components supported in the editor.

Call Flow Node	Description
Initial Action(Start Component)	The mandatory entry point for every flow. Configures the assigned extension number, initial welcome audio prompt, and loop settings.
Prompt	Plays selected audio files or AI-generated Text-to-Speech (TTS) announcements to the caller. Users can also play the prompt on loop 3 times.
Time Condition	Evaluates system time conditions (Office Hours, Custom Time, or Custom Time Groups) and branches the routing path accordingly.
IVR Menu	Presents multiple DTMF menu options, stores keystroke entries into variables, and branches calls based on valid inputs, timeouts, or invalid entries.
User Input	The User Input component is used to play an instructional prompt and capture DTMF key sequences (such as PINs or account IDs) entered by the caller and passes the call to the next connected node (typically a Condition node) for expression evaluation.
Language	Overrides the system default voice prompt language for all subsequent components in the call flow path.
Record Audio	Dynamically enables or disables call recording at a specific stage in the flow, with options to play pre-recording notification prompts.
Dial by Number	Directs callers to manually enter numbers to reach allowed extension ranges, predefined internal destinations (IVR, Ring Group, Queue, Conference, Call Flow), or external numbers using a defined pattern or LDAP directory contacts.
Dial by Name	Allows callers to search for and connect to a target user by spelling the letters of the user’s name with their phone keypad. The search directory can be configured using local extensions, LDAP phonebooks, or both.
Transfer	Directs the caller out of the Call Flow to a final internal or external destination (Extension, Voicemail, Ring Group, Queue, IVR, or another Call Flow).
Hang Up	Terminates the active call session.
Condition	Evaluates logical expressions configured under custom branches. Branches evaluate variables (such as ${CF_INPUT_KEY}, ${CALLERID(number)}, or ${CALLFLOW_LANGUAGE}), constants, or built-in functions to determine subsequent call routing. Note: Please refer to the Call Flow Design Configuration Guide for a list the supported expressions.
Loop	Repeats execution of its child components either for a fixed count (Range: 1–99) or until a custom logical condition expression.

PICKUP GROUPS

The UCM630xA supports pickup group feature which allows users to pick up incoming calls for other extensions if they are in the same pickup group, by dialing “Pickup Extension” feature code (by default *8).

Configure Pickup Groups

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

• Click on
  

  

  
  to create a new pickup group.
• Click on
  

  

  
  to edit the pickup group.
• Click on
  

  

  
  to delete the pickup group.

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

Configure Pickup Feature Code

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

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

MUSIC ON HOLD

Music On Hold settings can be accessed via Web GUI🡪PBX Settings🡪Music On Hold. In this page, users could configure music on hold class and upload music files. The “default” Music On Hold class already has 5 audio files defined for users to use.

• Click on “Create New MOH Class” to add a new Music On Hold class.
• Click on
  

  

  
  to configure the MOH class sort method to be “Alpha” or “Random” for the sound files.
• Click on
  

  

  
  next to the selected Music On Hold class to delete this Music On Hold class.
• Click on
  

  

  
  to start uploading. Users can upload:
• Single files with 8KHz Mono Music file, or
• Music on hold files in a compressed package with .tar, .tar.gz and .tgz as the suffix. The file name can only be letters, digits, or special characters -_
• the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire MoH.
• Users could also download all the music on hold files from UCM. In the Music On Hold page, click on
  

   

  
  

  
  and the file will be downloaded to your local PC.

  
  
• Click on
  

  

  
  to disable it from the selected Music On Hold Class.
• Click on
  

  

  
  to enable it from the selected Music On Hold Class.
• Select the sound files and click on
  

  

  
  to delete all selected Music On Hold files.

The UCM630xA allows Users to select the Music On Hold file from Web GUI to play it. The UCM630xA will initiate a call to the selected extension and play this Music On Hold file once the call is answered.

Steps to play the Music On Hold file:

1. Click on the
   

   

   
   button for the Music On Hold file.
2. In the prompted window, select the extension to playback and click
   

   

   
   .

3. The selected extension will ring.
4. Answer the call to listen to the music playback.

Users could also record their own Music On Hold to override an existing custom prompt, this can be done by following those steps:

1. Click on
   

   

   
   .
2. A message of confirmation will pop up, as shown below.

3. Click
   

   

   
   .
4. In the prompted window, select the extension to playback and click
   

   

   
   .

5. Answer the call and start to record your new music on hold.
6. Hang up the call and refresh Music On Hold page then you can listen to the new recorded file.

BUSY CAMP-ON

The UCM630xA supports busy camp-on/call completion feature that allows the PBX to camp on a called party and inform the caller as soon as the called party becomes available given the previous attempted call has failed.

The configuration and instructions on how to use busy camp-on/call completion feature can be found in the following guide:

https://documentation.grandstream.com/knowledge-base/busy-camp-on-2/

PRESENCE

UCM does support SIP presence feature which allows users to advertise their current availability status and willingness to receive calls, this way other users can use their phones in order to monitor the presence status of each user and decide whether to call them or not based on their advertised availability.

This feature is different than BLF which is used to monitor the dialog status for each extension (Ringing, Idle or Busy). Instead, the SIP presence module gives more options for users to choose which state they want to put themselves in.

In order to configure the presence status of an extension from the web GUI, users can access the menu of configuration using one of the two following methods:

• From admin account, go under the menu Extension/Trunk🡪Extensions and choose the desired extension to edit then navigate to the “Features” tab.

OR

• From the User Portal, go under the menu Basic Information🡪Extensions and navigate to the Features tab to have the following options.

Select which status to set from the presence status selection drop list, six options are available and below is a brief description of these states:

Available	The contact is online and can participate in conversations/phone calls.
Away	The contact is currently away (ex: for lunch break).
Chat	The contact has limited conversation flexibility and can only be reached via chat.
Do Not Disturb	The Contact is on DND (Do Not Disturb) mode.
Custom Presence Status	Please enter the presence status for this mode on the Web GUI. Up to 64 characters.
Unavailable	The contact is unreachable for the moment, please try to contact later.

Another option to set the presence status and which is more practical is using the feature code from the user’s phone, one the user dials the feature code (default is *48), a prompt will be played to select which status they want to put themselves in, by pressing the corresponding key.

The feature code can be enabled and customized from the Web GUI🡪Basic Call Features🡪Feature Codes.

When a user does change his/her SIP presence status by making a call using presence feature code, the UCM will create a corresponding CDR entry showing the call as Action type = PRSENCE_STATUS.

FOLLOW ME

Follow Me is a feature on the UCM630xA that allows users to direct calls to other phone numbers and have them ring all at once or one after the other. Calls can be directed to users’ home phone, office phone, mobile and etc. The calls will get to the user no matter where they are. Follow Me option can be found under extension settings page Web GUI🡪Extension/Trunk🡪Extensions.

To configure follow me:

1. Choose the extension and click on
   

   

   
   .
2. Go to the Follow me tab to add destination numbers and enable the feature.

1. Click on
   

   

   
   to add local extensions or external numbers to be called after ringing the extension selected in the first step.
2. Once created, it will be displayed on the follow me list. And you can click on
   

   

   
   to delete the Follow Me.

The following table shows the Follow Me configuration parameters:

Enable	Configure to enable or disable Follow Me for this user.
Skip Trunk Auth	If external number is added in the Follow Me, please make sure this option is enabled or the “Skip Trunk Auth” option of the extension is enabled, otherwise the external Follow Me number cannot be reached.
Music On Hold Class	Configure the Music On Hold class that the caller would hear while tracking users
Confirm When Answering	By default, it is enabled, and user will be asked to press 1 to accept the call or to press 2 to reject the call after answering a Follow Me call.   If it is disabled, the Follow Me call will be established once after the user answers.
Enable Destination	When enabled, the call will be routed to the default destination if no one in the Follow Me extensions answers the call.
Default Destination	Configure the destination if no one in the Follow Me extensions answers the call. The available options are:   Extension Voicemail Queues Ring Group Voicemail Group IVR External Number
Follow Me Numbers	The added numbers are listed here. Click on   to arrange the order. Click on to delete the number. Click on to add new numbers.
New Follow Me Number	Add a new Follow Me number which could be a ‘Local Extension’ or ‘External Number’. The selected dial plan should have permissions to dial the defined external number.
Dialing Order	Select the order in which the Follow Me destinations will be dialed to reach the user: ring all at once or ring one after the other.

Click on “Follow Me Options” under Web GUI🡪Extension/Trunk🡪Extension page to enable or disable the options listed in the following table.

Playback Incoming Status Message	If enabled, the PBX will playback the incoming status message before starting the Follow Me steps.
Record the Caller’s Name	If enabled, the PBX will record the caller’s name from the phone so it can be announced to the callee in each step.
Playback Unreachable Status Message	If enabled, the PBX will playback the unreachable status message to the caller if the callee cannot be reached.

SPEED DIAL

The UCM630xA supports Speed Dial feature that allows users to call a certain destination by pressing one or four digits on the keypad. This creates a system-wide speed dial access for all the extensions on the UCM630xA.

To enable Speed Dial, on the UCM630xA Web GUI, go to page Web GUI🡪Basic Call Features🡪Speed Dial.

User should first click on

. Then decide from one digit up to four digits combination used for Speed Dial and select a dial destination from “Default Destination”. The supported destinations include extension, voicemail, meeting room, voicemail group, IVR, ring group, call queue, page group, DISA, Dial by Name and external number.

The following can be configured as the default destination of a speed dial:

• Extension: Choose the extension to call when dialing the speed dial number.
• Meeting: Choose the meeting room extension to call when dialing the speed dial number.
• Voicemail: Choose the extension which voicemail will be accessed to when dialing the speed dial number.
• Voicemail Group: Choose the voicemail group which will be accessed when dialing the speed dial number.
• IVR: Choose the IVR which will be accessed when dialing the speed dial number.
• Ring Group: Choose the ring group which will be called when dialing the speed dial number
• Queues: Choose the queue which will be called when dialing the speed dial number.
• Paging/Intercom: Choose the paging/intercom group which will be initiated when the speed dial number is dialed.
• Fax: Select the fax extension which will be used as a destination when the speed dial number is dialed.
• DISA: Choose DISA as the destination when the speed dial number is dialed.
• Dial By Name: Choose Dial By Name as the dialing destination.
• Announcement: Choose an announcement as the destination when the speed dial number is dialed.
• External Number: Enter an external number to call when the speed dial number is dialed.
• Custom Number: Choose a custom number to call when the speed dial number is dialed. This can also be a feature code.

Import Speed Dial

The user can import speed dial entries from a csv file, this reduces the amount of configuring the same speed dial entries on different UCMs. To do this, please click on “Import” as the figure below shows.

Then select the csv file of the speed dial entries and click

Export Speed Dial

To export speed dial entries, please click on export as the screenshot below shows, then choose the location where to save the csv file.

DISA

In many situations, the user will find the need to access his own IP PBX resources, but he is not physically near one of his extensions. However, he does have access to his own cell phone. In this case, we can use what is commonly known as DISA (Direct Inward System Access). Under this scenario, the user will be able to call from the outside, whether it is using his cell phone, pay phone, regular PSTN, etc. After calling into UCM630xA, the user can then dial out via the SIP trunk or PSTN trunk connected to UCM630xA as it is an internal extension.

The UCM630xA supports DISA to be used in IVR or inbound route. Before using it, create new DISA under Web GUI🡪Advanced Call Features🡪DISA.

• Click on
  

  

  
  to add a new DISA.
• Click on
  

  

  
  to edit the DISA configuration.
• Click on
  

  

  
  to delete the DISA.

The following table details the parameters to set and configure DISA feature on UCM630xA PBX.

Name	Configure DISA name to identify the DISA.
Password	Configure the password (digit only) required for the user to enter before using DISA to dial out.   Note: The password must be at least 4 digits.
Permission	Configure the permission level for DISA.   The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level. The default setting is “Internal”. If the user tries to dial outbound calls after dialing into the DISA, the UCM630xA will compared the DISA’s permission level with the outbound route’s privilege level. If the DISA’s permission level is higher than (or equal to) the outbound route’s privilege level, the call will be allowed to go through.
Response Timeout	Configure the maximum amount of time the UCM630xA will wait before hanging up if the user dials an incomplete or invalid number.   The default setting is 10 seconds.
Digit Timeout	Configure the maximum amount of time permitted between digits when the user is typing the extension. The default setting is 5 seconds.
Allow Hangup	If enabled, during an active call, users can enter the UCM630xA Hangup feature code (by default it is *0) to disconnect the call or hang up directly. A new dial tone will be heard shortly for the user to make a new call. The default setting is “No”.
Replace Display Name	If enabled, the UCM will replace the caller display name with the DISA name.

Once successfully created, users can configure the inbound route destination as “DISA” or IVR key event as “DISA”. When dialing into DISA, users will be prompted with password first. After entering the correct password, a second dial tone will be heard for the users to dial out.

EMERGENCY

UCM supports configuration and management of numbers to be called in emergency situation, thus bypassing the regular outbound call routing process and allowing users in critical situation to dial out for emergency help with the possibility to have redundant trunks as point of exit in case one of the lines is down.

UCM6xxx series are also now in full compliance with Kari’s Law and Ray Baum’s Act, for more information, please refer to the following links:

https://www.fcc.gov/general/9-1-1-and-e9-1-1-services

https://documentation.grandstream.com/knowledge-base/emergency-calls/

In addition, Emergency calls can be automatically recorded by toggling on the new Auto Record and recordings can be viewed in the new Emergency Recordings tab on the same page. Additionally, users can have these recordings be sent to the configured email address(es).

Email alerts are also supported after enabling the notification for the event under “Maintenance 🡪 System Events”

Emergency Calls

To configure emergency numbers, users need to follow below steps:

1. Navigate on the web GUI under Advanced Call Features 🡪 Emergency Calls
2. Click on
   

   

   
   to add a new emergency number.
3. Configure the required fields “Name, Emergency Number and Trunk(s) to be used to reach the number”.
4. Save and apply the configuration.

The table below gives more description of the configuration Parameters when creating emergency numbers.

Name	Configure the name of the emergency call.   For example, “emergency911″,”emergency211” and etc.
Emergency Number	Config the emergency service number. For example,”911″,”211″ and etc.
Emergency Level	Select the emergency level of the number. Level “3” means the most urgent.
Disable Hunt on Busy	If this option is not enabled, when the lines of trunks which the coming emergency call routes by are completely occupied, the line-grabbing function will automatically cut off a line from all busy lines so that the coming emergency call can seize it for dialing out. This option is not enabled by default.
Custom Prompt	This option sets a custom prompt to be used as an announcement to the person receiving an emergency call. The file can be uploaded from the page “Custom Prompt”. Click “Prompt” to add additional record.
Use Trunks	Select the trunks for the emergency call. Select one trunk at least and select five trunks at most.
Members Notified	Select the members who will be notified when an emergency call occurs.
Strip	Specify the number of digits that will be Stripped from the beginning of the dialed number before the call is placed via the selected trunk.
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.
Auto Record	When enabled, emergency call will be automatically recorded.
Send Recording File	When enabled recording files will be sent to the configured email address.
Email Address	The email address to where the recording files will be sent.

Emergency Recordings

UCM6300 Series allows recording emergency calls and they can be found under WebUI → Call Feature → Emergency Calls → Emergency Recordings

Emergency Location Mapping

In compliance with Kari’s Law and the Ray Baum’s Act, UCM’s Emergency Calls feature supports
emergency location mapping. This will allow users to associate subnets with emergency location
identification numbers (ELINs), which can then be used by E911 service providers for example to determine
the location of callers. The new options can be found under Advanced Call Features→Emergency
Calls→Emergency Location Mapping.

When creating or configuring an Emergency Location Mapping entry, users can set the parameters described in the table below:

CALLBACK

Callback is designed for users who often use their mobile phones to make long distance or international calls which may have high service charges. The callback feature provides an economic solution for reduce the cost from this.

The callback feature works as follows:

1. Configure a new callback on the UCM630xA.
2. On the UCM630xA, configure destination of the inbound route for analog trunk to callback.
3. Save and apply the settings.
4. The user calls the PSTN number of the UCM630xA using the mobile phone, which goes to callback destination as specified in the inbound route.
5. Once the user hears the ringback tone from the mobile phone, hang up the call on the mobile phone.
6. The UCM630xA will call back the user.
7. The user answers the call.
8. The call will be sent to DISA or IVR which directs the user to dial the destination number.
9. The user will be connected to the destination number.

In this way, the calls are placed and connected through trunks on the UCM630xA instead of to the mobile phone directly. Therefore, the user will not be charged on mobile phone services for long distance or international calls.

To configure callback on the UCM630xA, go to Web GUI🡪Advanced Call Features🡪Callback page and click on

. Configuration parameters are listed in the following table.

Callback Configuration Parameters

Name	Configure a name to identify the Callback. (Enter at least two characters)
CallerID Pattern	Configure the pattern of the callers allowed to use this callback. The caller who places the inbound call needs to have the CallerID match this pattern so that the caller can get callback after hanging up the call.   Note: If leaving as blank, all numbers are allowed to use this callback.
Outbound Prepend	Configure the prepend digits to be added at before dialing the outside number. The number with prepended digits will be used to match the outbound route. ‘-’ is the connection character which will be ignored.
Delay Before Callback	Configure the number of seconds to be delayed before calling back the user.
Destination	Configure the destination which the callback will direct the caller to. Two destinations are available:   IVR DISA The caller can then enter the desired number to dial out via UCM630xA trunk.

BLF AND EVENT LIST

BLF

The UCM630xA supports BLF monitoring for extensions, ring group, call queue, meeting room and parking lot. For example, on the user’s phone, configure the parking lot number 701 as the BLF monitored number. When there is a parked call on 701, the LED for this BLF key will light up in red, meaning a call is parked against this parking lot. Pressing this BLF key can pick up the call from this parking lot.

Event List

Besides BLF, users can also configure the phones to monitor event list. In this way, both local extensions on the same UCM630xA and remote extensions on the VOIP trunk can be monitored. The event list setting is under Web GUI🡪Basic Call Features🡪Event List.

• Click on “Add” to add a new event list.
• 
  Sort selected extensions manually in the Eventlist
• Click on
  

  

  
  to edit the event list configuration.
• Click on
  

  

  
  to delete the event list.

URI	Configure the name of this event list (for example, office_event_list). Please note the URI name cannot be the same as the extension name on the UCM630xA. The valid characters are letters, digits, _ and -.
Local Extensions	Select the available extensions/Extension Groups listed on the local UCM630xA to be monitored in the event list.
Remote Extensions	If LDAP sync is enabled between the UCM630xA and the peer UCM630xA, the remote extensions will be listed under “Available Extensions”. If not, manually enter the remote extensions under “Special Extensions” field.
Special Extensions	Manually enter the remote extensions in the peer/register trunk to be monitored in the event list. Valid format: 5000,5001,9000

Remote extension monitoring works on the UCM630xA via event list BLF, among Peer SIP trunks or Register SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the UCM630xA first before using remote BLF feature. Please note the SIP end points need support event list BLF in order to monitor remote extensions.

When an event list is created on the UCM630xA and remote extensions are added to the list, the UCM630xA will send out SIP SUBSCRIBE to the remote UCM630xA to obtain the remote extension status. When the SIP end points register and subscribe to the local UCM630xA event list, it can obtain the remote extension status from this event list. Once successfully configured, the event list page will show the status of total extension and subscribers for each event list. Users can also select the event URI to check the monitored extension’s status and the subscribers’ details.

• To configure LDAP sync, please go to UCM630xA Web GUI🡪Extension/Trunk🡪VoIP Trunk. You will see “Sync LDAP Enable” option. Once enabled, please configure password information for the remote peer UCM630xA to connect to the local UCM630xA. Additional information such as port number, LDAP outbound rule, LDAP Dialed Prefix will also be required. Both the local UCM630xA and remote UCM630xA need enable LDAP sync option with the same password for successful connection and synchronization.
• Currently LDAP sync feature only works between two UCM630xAs.
• (Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is non-UCM630xA PBX. However, it might not work the other way around depending on whether the non-UCM630xA PBX supports event list BLF or remote monitoring feature.

DIAL BY NAME

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

Dial by Name Configuration

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

1. Name
   Enter a Name to identify the Dial by Name group.
2. Extension
   Configure the direct dial extension for the Dial By Name group.
3. Custom Prompt
   This option sets a custom prompt for directory to announce to a caller. The file can be uploaded from the page “Custom Prompt”. Click “Upload Audio File” to add additional record.
4. Available Extensions/Selected Extensions
   Select available extensions from the left side to the right side as the directory for the Dial By Name group. Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group. The extensions here must have a valid first name and last name configured under Web GUI🡪Extension/Trunk🡪Extensions in order to be searchable in Dial By Name directory through IVR. By specifying the extensions here, the administrators can make sure unscreened calls will not reach the company employee if he/she does not want to receive them directly.
5. Prompt Wait Time
   Configure “Prompt Wait Time” for Dial By Name feature. During Dial By Name call, the caller will need to input the first letters of First/Last name before this wait time is reached. Otherwise, timeout will occur, and the call might hang up. The timeout range is between 3 and 60 seconds.
6. Query Type
   Specify the query type. This defines how the caller will need to enter to search the directory.
   By First Name: enter the first 3 digits of the first name to search the directory.
   By Last Name: enter the first 3 digits of the last name to search the directory.
7. Select Type
   Specify the select type on the searching result. The IVR will confirm the name/number for the party the caller would like to reach before dialing out.
   By Order: After the caller enters the digits, the IVR will announce the first matching party’s name and number. The caller can confirm and dial out if it is the destination party, or press * to listen to the next matching result if it is not the desired party to call.
   By Menu: After the caller enters the digits, the IVR will announce 8 matching results. The caller can press number 1 to 8 to select and call or press 9 for results in next page.
   The Dial by Name group can be used as the destination for inbound route and key pressing event for IVR. The group name defined here will show up in the destination list when configuring IVR and inbound route. If Dial by Name is set as a key pressing event for IVR, user could use ‘*’ to exit from Dial by Name, then re-enter IVR and start a new event. The following example shows how to use this option.

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

ACTIVE CALLS AND MONITOR

The active calls on the UCM630xA are displayed in Web GUI🡪System Status🡪Active Calls page. Users can monitor the status, hang up the call as well as barge in the active calls in real time manner.

Active Calls Status

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

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

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.

Hang Up Active Calls

To hang up an active call, click on

icon in the active call dialog. Users can also click on

to hang up all active calls.

Call Monitor

During an active call, click on icon

and the monitor dialog will pop up.

In the “Monitor” dialog, configure the following to monitor an active call:

1. Enter an available extension for “Monitor’s Extension” which will be used to monitor the active call.
2. “Monitored Extension” must be one of the parties in the active call to be monitored.
3. Select spy mode. There are three options in “Spy Mode”.

• Listen

In “Listen” mode, the extension monitoring the call can hear both parties in the active call but the audio of the user on this extension will not be heard by either party in the monitored active call.

• Whisper

In “Whisper” mode, the extension monitoring the call can hear both parties in the active call. The user on this extension can only talk to the selected monitored extension and he/she will not be heard by the other party in the active call. This can be usually used to supervise calls.

• Barge

In “Barge” mode, the extension monitoring the call can talk to both parties in the active call. The call will be established similar to three-way meeting.

1. Enable or disable “Require Confirmation” option. If enabled, the confirmation of the invited monitor’s extension is required before the active call can be monitored. This option can be used to avoid adding participant who has auto-answer configured, or call forwarded to voicemail.
2. Click on “Add”. An INVITE will be sent to the monitor’s extension. The monitor can answer the call and start monitoring. If “Require Confirmation” is enabled, the user will be asked to confirm to monitor the call.

Another way to monitor active calls is to dial the corresponding feature codes from an extension. Please refer to [UCM630xA Feature Codes] and [Call Recording] section for instructions.

CALL FEATURE

The UCM630xA supports call recording, transfer, call forward, call park and other call features via feature code. This section lists all the feature codes in the UCM630xA and describes how to use the call features.

Feature Codes

Feature codes allow users to perform certain actions such as call park or call forwarding, by dialing a configured code. For example, dialing #72 to park a call.

The UCM630xA is by default configured with feature codes for different use cases, but users can change these codes manually by accessing Basic Call Features 🡪 Feature codes.

The UCM630xA also allows user to one click enable / disable specific feature code as shown below:

Parking Lot

User can create parking lots and their related slots under Web GUI🡪 Basic Call Features🡪 Parking Lot. In the Parking Lot page, users can create lots of their own. This allows different groups within an organization to have their own parking lots instead of sharing one large parking lot with others. While creating a new parking lot, users can assign it a range that they think is appropriate for the group that will use the parking lot.

User can create a new Parking lot by clicking on button “Add” :

Parking Lot

Parking Lot Extension	Default Extension: 700 During an active call, initiate blind transfer and then enter this code to park the call.
Parking Lot Name	Set a name to the parking lot
Parked Slots	Default Extension: 701-720 These are the extensions where the calls will be parked, i.e., parking lots that the parked calls can be retrieved.
Use Parklot as Extension	If checked, the parking lot number can be used as extension. The user can transfer the call to the parking lot number to park the call. Please note this parking lot number range might conflict with extension range.
Parking Timeout (s)	Default setting is 300 seconds, and the maximum limit is 99.999 seconds. This is the timeout allowed for a call to be parked. After the timeout, if the call is not picked up, the extension who parks the call will be called back.
Music On Hold Classes	Select the Music on Hold Class.
Failover Destination	Configures a callback failover destination when the extension that is called back is busy. The call will be routed to the destination number and this reduces the chance of dropping parked calls.
Ring All Callback on Timeout	If enabled, all registered endpoints of the extension will ring when callback occurs. Otherwise, only the original endpoint will be called back.
Forward to destination on timeout	If enabled, the call will be routed to the configured destination upon timeout. Otherwise, the call will be routed back to the original caller.
Timeout Destination	This option appears once Forward to Destination on Timeout is enabled. Upon park timeout, the call will be routed to the configured destination.
Parking Lot Timeout Alert-Info	Adds an Alert-Info header to parking lot callbacks after the Parking Timeout has been reached.

Call Park

The UCM630xA provides call park and call pickup features via feature code.

Park a Call

There are two feature codes that can be used to park the call.

• Feature Maps🡪Call Park (Default code #72)

During an active call, press #72 and the call will be parked. Parking lot number (default range 701 to 720) will be announced after parking the call.

• Feature Misc🡪Call Park (Default code 700)

During an active call, initiate blind transfer (default code #1) and then dial 700 to park the call. Parking lot number (default range 701 to 720) will be announced after parking the call.

Retrieve Parked Call

To retrieve the parked call, simply dial the parking lot number and the call will be established. If a parked call is not retrieved after the timeout, the original extension who parks the call will be called back.

Monitor Call Park CID Name Information (GXP21xx, GRP261x Phones Only)

Users can see the CID name information of parked calls. VPK/MPKs must be configured as “Monitored Call Park” with the desired parking lot extension. The display will alternate between displaying the parking lot extension and the call’s CID name. There is no need to configure anything on the UCM.

Call Recording

The UCM630xA allows users to record audio during the call. If “Auto Record” is turned on for an extension, ring group, call queue or trunk, the call will be automatically recorded when there is established call with it. Otherwise, please follow the instructions below to manually record the call.

1. Make sure the feature code for “Start/Stop Call Recording” is configured and enabled.
2. After establishing the call, enter the “Start/Stop Call Recording” feature code (by default it is *3) followed by # or SEND to start recording.
3. To stop the recording, enter the “Start/Stop Call Recording” feature code (by default it is *3) followed by # or SEND again. Or the recording will be stopped once the call hangs up.
4. The recording file can be retrieved under Web GUI🡪CDR. Click on
   

   

   
   to show and play the recording or click on
   

   

   
   to download the recording file.

The above recorded call’s recording files are also listed under the UCM630xA Web GUI🡪CDR🡪Recording Files.

Enable Spy

If “Enable Spy” option is enabled, feature codes for Listen Spy, Whisper Spy and Barge Spy are available for users to dial from any extension to perform the corresponding actions.

Assume a call is on-going between extension A and extension B, user could dial the feature code from extension C to listen on their call (*54 by default), whisper to one side (*55 by default), or barge into the call (*56 by default). Then the user will be asked to enter the number to call, which should be either side of the active call, extension A or B in this example.

Shared Call Appearance (SCA)

Shared Call Appearance (SCA) functionality has been added to the UCM. With SCA, users can assign multiple devices to one extension, configure endpoints to monitor that extension, make actions on behalf of that extension such as viewing call status and placing and receiving calls, and even barging into existing calls. To configure the SCA functionality, please follow the steps below:

1. Users can enable SCA by navigating to the Extensions page, editing the desired extension, and enabling the option SCA.

Note: With SCA enabled, the Concurrent Registrations field can only have a value of 1.

2. After enabling the option, navigate to Advanced Call Features🡪SCA. The newly enabled SCA extension will be listed. Click the “+” button under the Options column to add a number that will share the main extension’s call appearance, which will be called private numbers.

3. Configure the private number as desired.

4. Once the private number has been created, users must now register a device to it. To properly register a device to the private number, use the configured private number as the SIP User ID. Auth ID and Password will be the same as the main extensions. Once registration is complete, SCA is now configured.

5. Next, configure the VPK or MPK to Shared for both the main extension and the private number. SCA is now configured for both endpoint devices.

The following table describe the SCA Number configuration setting:

Add SCA Private Number

Private Number	Configures the private number for the SCA.
Related Shared Line	Display the related shared line.
Enable This Number	Whether enable this private number. If not enabled, this private number is only record in DB, it will not affect other system feature.
Allow Origination from This Number	Enable this option will allow calling from this private number. By default, it is enabled.
Allow Termination to This Number	Enable this option will allows calls to this private number. By default, it is enabled.

The following table describes the options available when editing the SCA number:

Editing the SCA Number

Shared Line Number	While SCA is enabled, this number will be the same as the extension number.
Allow Call Retrieve from Another Location	Allows remote call retrieval. Must be enabled in public hold. By default, it is enabled.
Alert All Appearances for Group Paging Calls	Allows all SCA group members to ring when the SCA shared number is paged. If disabled, only the SCA shared number will ring when paged. By default, it is disabled.
Multiple Call Arrangement	Allows simultaneous calls in an SCA group. By default, it is disabled.
Allow Bridging between Locations	Allows location bridging for SCA group. Must be enabled when using the Barge-In feature. By default, it is disabled.
Bridge Warning Tone	Configures the notification in the bridge when another party join.   None: No notification sound. Barge-In only: Notification sound will play when another party join. Barge-In and Repeat: Notification sound will play when another party joins and repeat every 30 seconds. By default, it is set to “Barge-In Only”.

Time Condition Routing

Time Condition Routing allows the user to create default destinations for calls which are based on time conditions under Advanced Call Features→Time Condition Routing.

ANNOUNCEMENT

The Announcement feature (not to be confused with Announcement Paging and Announcement Center) allows users to set an unskippable audio prompt to play to callers before automatically routing them to a configured destination. Announcements can be assigned as a destination in the Inbound Routes page.

To configure an Announcement, follow the steps below:

1. Navigate on the Web GUI to Advanced Call Features → Announcement.
2. Click on the Add button to create a new announcement entry.
3. Configure the required fields including Name, Prompt, Default Destination, and Call Privileges (when routing to external numbers).
4. Save and apply the configuration.

The table below describes the configuration parameters available when creating or editing an Announcement:

Parameter	Description
Name	Configure a unique name to identify the announcement entry.
Prompt	Select the audio file to play to the caller. Users can choose from the following options via the Add Audio dropdown: Existing Prompt: Select an existing audio file already saved on the PBX. Upload Prompt: Upload a pre-recorded audio file directly from the local computer. TTS Generation: Use the built-in Text-to-Speech (TTS) feature to generate a custom voice audio prompt from entered text.
Default Destination	Select the routing destination after the announcement audio finishes playing. Supported destinations include: Extension: Route to a specific system extension. Multimedia Meeting: Direct the caller into a meeting room. Voicemail / Voicemail Group: Route directly to a voicemail box or group. IVR: Send the call to an Interactive Voice Response system. Ring Group: Direct the call to ring a defined group of extensions. Queues: Place the call into a call queue. Fax: Route to a fax extension. DISA: Direct to Direct Inward System Access. Dial By Name: Route to the directory dial-by-name system. External Number: Route the call to an external phone number over an outbound trunk. Hang-up: Terminate the call immediately after the prompt finishes.
Call Privileges	Configures the permission level required when the Default Destination is set to an External Number. This ensures the outbound call complies with configured outbound route permission restrictions (e.g., Internal, Local, National, International).

PBX SETTINGS

This section describes internal options that have not been mentioned in previous sections yet. The settings in this section can be applied globally to the UCM630xA, including general configurations, jitter buffer, RTP settings, ports config and STUN monitor. The options can be accessed via Web GUI🡪PBX Settings🡪General Settings.

PBX Settings/General Settings

PBX Settings/RTP Settings

RTP Settings

Internal Options/RTP Settings

Payload

The UCM630xA payload type for audio codecs and video codes can be configured here.

Internal Options/Payload

AAL2-G.726	Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword packing). The default setting is 112.
DTMF	Configured payload type for DTMF. The default setting is 101.
G.721 Compatible	Configure to enable/disable G.721 compatible. The default setting is Yes.
G.726	Configure the payload type for G.726 if “G.721 Compatible” is disabled. The default setting is 111.
iLBC	Configure the payload type for iLBC. The default setting is 97.
OPUS	Configure the payload type for OPUS. The default setting is 123.
Audio FEC Payload Type	Configure the Audio FEC Payload Type. The default setting is 127
Audio RED Payload Type	Configure the Audio RED Payload Type. Default setting is 122
H.264	Configure the payload type for H.264. The default setting is 99.
H.263P	Configure the payload type for H.263+. The default setting is 100 103.
VP8	Configure the payload type for VP8. The default setting is 108.
Main Video FEC	Configure the Main Video FEC
RTP FECC	Configure the RTP FECC
RTX	Configure the RTX
G.722.1	G.722.1：Low-complexity coder, 24kbps.
G.722.1C	G.722.1C：Low-complexity coder, 48kbps.

PBX Settings/Voice Prompt Customization

Record New Custom Prompt

In the UCM630xA Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page, click on “Record” and follow the steps below to record new IVR prompt.

1. Specify the IVR file name.
2. Select the format (GSM or WAV) for the IVR prompt file to be recorded.
3. Select the extension to receive the call from the UCM630xA to record the IVR prompt.
4. Click the “Record” button. A request will be sent to the UCM630xA. The UCM630xA will then call the extension for recording the IVR prompt from the phone.
5. Pick up the call from the extension and start the recording following the voice prompt.
6. The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play, or delete the recording.

Upload Custom Prompt

If the user has a pre-recorded IVR prompt file, click on “Upload” in Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page to upload the file to the UCM630xA. The following are required for the IVR prompt file to be successfully uploaded and used by the UCM630xA:

• PCM encoded.
• 16 bits.
• 8000Hz mono.
• In .mp3 or .wav format; or raw/ulaw/alaw/gsm file with .ulaw or .alaw suffix.
• File size under 5M.

Click on “choose file to upload” to start uploading. Once uploaded, the file will appear in the Custom Prompt web page.

If the file uploaded is mp3, the following prompt will appear

Tick “Keep Copy of MP3” to keep an mp3 copy of the file which can be diffused with certain GSC speakers. Diffusing the MP3 files result in a higher quality sound played by GCS speakers.

Download All Custom Prompt

On the UCM630xA, the users can download all custom prompts from UCM Web GUI to local PC. To download all custom prompt, log in UCM Web GUI and navigate to PBX Settings🡪Voice Prompt🡪Custom Prompt and click on ”Download All”. The following window will pop up in order to set a name for the downloaded file.

Note: The downloaded file will have a .tar extension.

PBX Settings/Call Failure Tone Settings

SIP Trunk Prompt Tone

Prompt Tone Settings tab on the UCM helps users choose which prompt will be played by the UCM during call failure, the voice message responses can be set to be played for 4XX, 5XX, and 6XX call failures.

The following default prompt names and their associated messages are supported:

Additionally, custom voice messages recorded and uploaded in PBX Settings→Voice Prompt→Custom Prompt can be used for the failure responses instead of the default messages.

General Call Prompt Tones

Moreover, users also have the possibility to customize the prompt for typical call failure reasons like (no permission to allow outbound calls, busy lines, incorrect number dialed …Etc.).

To customize these prompts user could record and upload their own files under “’PBX Settings 🡪 Voice Prompt 🡪 Custom Prompts” then select each one for specific call failure case under “PBX Settings -> Call Failure Tone Settings 🡪 General Call Prompt Tones” page as shown on the following figure:

File Manager

The File Manager section defines the storage location and automatic storage switching behavior for files generated by the UCM630x Audio system. Storage is configured separately for each file category. This section determines where new files are written and how the system behaves when multiple storage targets are available.

The following file types are supported:

• Recording Files
• IM Files

Each file category allows the administrator to define the storage path used for newly generated files. Available options depend on system configuration and detected storage devices (e.g., Local storage, GDMS Cloud Storage, USB, NAS).

When Enable Auto Change is enabled, the system stores files according to the defined Storage Path Priority. If the active storage device becomes unavailable or reaches capacity, the system automatically switches to the next available device in the priority list.

When the storage path for Recording Files is changed from Local to an external storage device (such as USB or NAS, if available), the Migrate Local Recordings option becomes available. This option allows existing recordings stored in internal memory to be copied to the newly selected external storage device. The administrator can select specific recording categories to migrate as shown below.

Storage Device Management

NAS

The UCM supports adding and backing up recordings to a network-attached storage (NAS) server.

The following table describes the configurable NAS settings:

SFTP

SFTP integration offers synchronizing files between from the UCM to the SFTP server. SFTP is used to synchronize the following data: CDR Records, Recording Files, Voicemail, and Fax.

USB Disk/SD Card File Management

USB Disk/SD Card File Management is a feature which allows to manages the files stored on the USB flash drive attached to the UCM device, or an SD card attached to the system.

Amazon S3

The Storage Device Management module allows you to integrate your UCM63xxA with external cloud storage services. This page provides the option to configure a connection to a third-party Amazon S3 bucket to archive a copy of your critical data, specifically backup files.

This feature is designed for organizations that require an additional, independent copy of their configuration and data for long-term retention, compliance, or disaster recovery purposes, separate from the primary Grandstream cloud storage.

The table below describes the parameters that can be found on the Amazon S3 configuration tab:

SIP SETTINGS

The UCM630xA SIP global settings can be accessed via Web GUI🡪PBX Settings🡪SIP Settings.

SIP Settings/General

SIP Settings/General

Realm For Digest Authentication	Configure the host name or domain name for the UCM630xA. Realms MUST be globally unique according to RFC3261. The default setting is Grandstream.
Bind UDP Port	Configure the UDP port used for SIP. The default setting is 5060.
Bind IPv4 Address	Configure the IPv4 address to bind to. The default setting is 0.0.0.0, which means binding to all addresses.
Bind IPv6 Address	Configure the IPv6 address to bind to. The default is : “[::]” and it means to bind to all IP addresses.
Allow Guest Calls	If enabled, the UCM630xA allows unauthorized INVITE coming into the PBX and the call can be made. The default setting is “No”.   Warning: Please be aware of the potential security risk when enabling “Allow Guest Calls” as this will allow any user with the UCM630xA address to dial into the UCM630xA.
Allow Transfer	If set to “No”, all transfers initiated by the endpoint in the UCM630xA will be disabled (unless enabled in peers or users). The default setting is “Yes”.
MWI From	When sending MWI NOTIFY requests, this value will be used in the “From:” header as the “name” field. If no “From User” is configured, the “user” field of the URI in the “From:” header will be filled with this value.
Enable Diversion Header	If disabled, the UCM will not forward the diversion header.
Block Collect Calls	If enabled, collect calls will be blocked.   Note: Collect calls are indicated by the header “P-Asserted-Service-Info: service-code=Backward Collect Call, P-Asserted-Service-Info: service-code=Collect Call”.

SIP Settings/MISC

Parameter	Description
Outbound SIP Registrations
Register Timeout	Configure the register retry timeout (in seconds). The default setting is 20.
Register Attempts	Configure the number of registration attempts before the UCM630X gives up. The default setting is 0, which means the UCM630X will keep trying until the server side accepts the registration request.
Initial REGISTER Delay Interval (s)	Configures the time window within which to send initial trunk registration requests. Instead of sending out all initial trunk registration requests at once, requests will be randomly sent out within this period. The default setting is 10.
Video
Support SIP Video	Select to enable video support in SIP calls. The default setting is “Yes”.
Security
Reject Non-Matching INVITE	If enabled, when rejecting an incoming INVITE or REGISTER request, the UCM630X will always reject with “401 Unauthorized” instead of notifying the requester whether there is a matching user or peer for the request. This reduces the ability of an attacker to scan for valid SIP usernames. Default setting is “No”.
SDP Attribute Passthrough
Enable Attribute Passthrough	If enabled, and if the service does not know the attribute of FEC/FECC/BFCP, then the attribute will be passed through.
Early Media
Enable Use Final SDP	If enabled, call negotiation will use final response SDP.
Ignore 180 Response	If enabled, ringing indication after 183 response will be ignored.
Transfer
Allow Transfer	If checked, all transfers initiated by endpoints will be enabled.
Allow callback when blind transfer fails	If enabled, the UCM will call back to the transferrer when blind transfer fails (reason of failure includes busy and no answer). Note: This feature takes effect only on internal calls.
Blind transfer timeout	Configure the timeout in (s) for the transferrer waiting for the destination to answer. Default is 60s.
Allow Early Blind Transfer	Enabling this allows callers to perform transfers while the original callee’s phone is still ringing during attended transfer. Note: Requires compatible endpoints such as Grandstream GRP261x on firmware 1.0.15.8 or higher.
Hold
Forward HOLD Requests	Configure the UCM to forward HOLD requests instead of processing holds internally. This serves to meet the standards set by some providers that require HOLD requests to be passed along from endpoint to endpoint. This option is disabled by default. Note: Enabling this option may cause hold retrieval issues and MOH to not be heard.

Transparent Call-info Header

UCM supports transparent call info header in order to integrate GDS door system with GXP21XX/GRP261X phones, the UCM will forward the call-info header to the phone in order to request the live view from GDS door system and give the option to open the door via softkey.

SIP Settings/Session Timer

SIP Settings/Session Timer

Force Timer	If checked, always request, and run session timer.
Timer	If checked, run session timer only when requested by other UA.
Session Expire	Configure the maximum session refresh interval (in seconds). Default is 1800.
Min SE	Configure the minimum session refresh interval (in seconds).   The default setting is 90.

SIP Settings/TCP and TLS

SIP Settings/TCP and TLS

SIP Settings/NAT

SIP Settings/NAT

External Host	Configure a static IP address and port (optional) used in outbound SIP messages if the UCM630xA is behind NAT. If it is a host name, it will only be looked up once.
Use IP address in SDP	If enabled, the SDP connection will use the IP address resolved from the external host.
External UDP Port	Configure externally mapped UDP port when the PBX is behind a static NAT or PAT.
External TCP Port	Configure the externally mapped TCP port when the UCM630xA is behind a static NAT or PAT.
External TLS Port	Configures the externally mapped TLS port when UCM630xA is behind a static NAT or PAT.
Local Network Address	Specify a list of network addresses that are considered inside of the NAT network. Multiple entries are allowed. If not configured, the external IP address will not be set correctly.   A sample configuration could be as follows: 192.168.0.0/16

SIP Settings/ToS

SIP Settings/ToS

ToS for SIP	Configure the Type of Service for SIP packets. The default setting is None.
ToS for RTP Audio	Configure the Type of Service for RTP audio packets. The default setting is None.
ToS for RTP Video	Configure the Type of Service for RTP video packets. The default setting is None.
Default Incoming/Outgoing Registration Time	Configure the default duration (in seconds) of incoming/outgoing registration.   The default setting is 120.
Max Registration/Subscription Time	Configure the maximum duration (in seconds) of incoming registration and subscription allowed by the UCM630xA. The default setting is 3600.
Min Registration/Subscription Time	Configure the minimum duration (in seconds) of incoming registration and subscription allowed by the UCM630xA. The default setting is 60.
Enable Relaxed DTMF	Select to enable relaxed DTMF handling. The default setting is “No”.
DTMF Mode	Select DTMF mode to send DTMF. The default setting is RFC4733. If “Info” is selected, 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. The default setting is “RFC4733”.
RTP Timeout	During an active call, if there is no RTP activity within the timeout (in seconds), the call will be terminated. The default setting is no timeout.   Note: This setting does not apply to calls on hold.
RTP Hold Timeout	When the call is on hold, if there is no RTP activity within the timeout (in seconds), the call will be terminated. This value of RTP Hold Timeout should be larger than RTP Timeout. The default setting is no timeout.
RTP Keep-alive	This feature can be used to avoid abnormal call drop when the remote provider requires RTP traffic during proceeding.   For example, when the call goes into voicemail and there is no RTP traffic sent out from UCM, configuring this option can avoid voicemail drop. When configured, RTP keep-alive packet will be sent to remote party at the configured interval. If set to 0, RTP keep-alive is disabled.
100rel	Configure the 100rel setting on UCM630xA. The default setting is “Yes”.
Trust Remote Party ID	Configure whether the Remote-Party-ID should be trusted. The default setting is “No”.
Send Remote Party ID	Configure whether the Remote-Party-ID should be sent or not. The default setting is “No”.
Generate In-Band Ringing	Configure whether the UCM630xA should generate Inband ringing or not. The default setting is “Never”.   Yes: The UCM630xA will send 180 Ringing followed by 183 Session Progress and in-band audio. No: The UCM630xA will send 180 Ringing if 183 Session Progress has not been sent yet. If audio path is established already with 183 then send in-band ringing. Never: Whenever ringing occurs, the UCM630xA will send 180 Ringing as long as 200OK has not been set yet. Inband ringing will not be generated even the end point device is not working properly.
Server User Agent	Configure the user agent string for the UCM630xA.
Send Compact SIP Headers	If enabled, compact SIP headers will be sent. The default setting is “No”.
Passthrough PAI Header	Passthrough PAI Header

SIP Settings/STIR/SHAKEN

To prevent robocalls, UCM now supports STIR/SHAKE protocols. Related options have been added as a new tab in the SIP Settings page.

Clicking on the Add button will show the following window:

SIP Settings/STIR/SHAKEN – Add Authentication Number Settings

Authentication Number	Configure the Authentication Number.
Device Name	Configure the device name.
Creditability	Configure the attestation level, which is the level of confidence of the carrier that the CID has not been spoofed. The following options are available:   A (Full attestation) – The carrier is associated with the caller and the number. There is high confidence that the CID has not been spoofed. B (Partial attestation) – The carrier is associated with the caller but not the number. There is uncertainty about whether the CID has been spoofed or not. C (Gateway attestation) – The carrier is not associated with the caller and has no confidence at all about the number. Generally used for traceback.

Clicking on the Certificate Settings button will bring up the following window:

SIP Settings/STIR/SHAKEN – Certificate Settings

Certificate Download Time (s)	Configure the public key download timeout period, the default value is 2 seconds.
Signature Valid Time (s)	Configure the validity period of the digital signature, the default value is 15 seconds.
Private Key	Configure the Private key.   Note: The uploaded file must be less than 2MB in file size, only supports the .key format and must be ECC type. This file will automatically be renamed to “private.key”.
Public Key	Configure the Public Key.   Note: The uploaded file must be less than 2MB in file size, only supports the .crt format and must be ECC type. This file will automatically be renamed to “public.crt”.

Transparent Call-Info header

UCM supports transparent call info header in order to integrate GDS door system with GXP21XX/GRP261X phones, the UCM will forward the call-info header to the phone in order to request the live view from GDS door system and give the option to open the door via softkey.

IAX SETTINGS

The UCM630xA IAX global settings can be accessed via Web GUI🡪PBX Settings🡪IAX Settings.

IAX Settings/General

IAX Settings/General

Bind Port	Configure the port number that the IAX2 will be allowed to listen to. The default setting is 4569.
Bind IPv4 Address	Force IAX2 to bind to a specific address instead of all addresses.
Bind IPv6 address	Configure the IPv6 address to bind to. “[::]” means to bind to all IP addresses.
IAX1 Compatibility	Select to configure IAX1 compatibility. The default setting is “No”.
No Checksums	If selected, UDP checksums will be disabled and no checksums will be calculated/checked on systems supporting this feature. The default setting is “No”.
Delay Reject	If enabled, the IAX2 will delay the rejection of calls to avoid DOS. The default setting is “No”.
ADSI	Select to enable ADSI phone compatibility. The default setting is “No”.
Music On Hold Interpret	Specify which Music On Hold class this channel would like to listen to when being put on hold. This music class is only effective if this channel has no music class configured and the bridged channel putting the call on hold has no “Music On Hold Suggest” setting.
Music On Hold Suggest	Specify which Music On Hold class to suggest to the bridged channel when putting the call on hold.
Bandwidth	Configure the bandwidth for IAX settings. The default setting is “Low”.

IAX Settings/Registration

IAX Settings/Registration

IAX Registration Options
Min Reg Expire	Configure the minimum period (in seconds) of registration. Default setting is 60.
Max Reg Expire	Configure the maximum period (in seconds) of registration. Default setting is 3600.
IAX Thread Count	Configure the number of IAX helper threads. The default setting is 10.
IAX Max Thread Count	Configure the maximum number of IAX threads allowed. The default is 100.
Auto Kill	If enabled and no ACK is received for new messages after the specified wait time, the connection will be terminated.
Authentication Debugging	If enabled, authentication traffic in debugging will not show. The default is “No”.
Codec Priority	Configure codec negotiation priority. The default setting is “Reqonly”.   Caller Consider the callers preferred order ahead of the host’s. Host Consider the host’s preferred order ahead of the caller’s. Disabled Disable the consideration of codec preference all together. Reqonly This is the same as “Disabled”, except when the requested format is not available. The call will only be accepted if the requested format is available.
Type of Service	Configure ToS bit for preferred IP routing.
IAX Trunk Options
Trunk Frequency	Configure the frequency of trunk frames (in milliseconds). The default is 20.
Trunk Time Stamps	If enabled, time stamps will be attached to trunk frames. The default is “No”.

IAX Settings/Security

IAX Settings/Static Defense

Call Token Optional	Enter a single IP address (e.g., 1.1.1.1) or a range of IP addresses (1.1.1.1/255.255.255.255) for which call token validation is not required.
Max Call Numbers	Configure the maximum number of calls allowed for a single IP address.
Max Unvalidated Call Numbers	Configure the maximum number of Unvalidated calls for all IP addresses.
Max Call Numbers	Configure to limit the number of calls for a give IP address of IP range.
IP or IP Range	Enter the IP address (1.1.1.1) or a range of IP addresses (1.1.1.1/255.255.255.255) to be considered for call number limits.

INTERFACE SETTINGS

Analog Hardware

The analog hardware (FXS port and FXO port) on the UCM630xA will be listed in this page. Click on

to edit signaling preference for FXS port or configure ACIM settings for FXO port.

Select “Loop Start” or “Kewl Start” for each FXS port. And then click on “Update” to save the change.

For FXO port, users could manually enter the ACIM settings by selecting the value from dropdown list for each port. Or users could click on “Detect” and choose the detection algorithm, two algorithms exist (ERL, Pr) for the UCM630xA to automatically detect the ACIM value. The detecting value will be automatically filled into the settings.

PBX Interface Settings

Tone Region	Select country to set the default tones for dial tone, busy tone, ring tone and etc. to be sent from the FXS port. The default setting is “United States of America (USA)”.
Advanced Settings
FXO Opermode	Select country to set the On-Hook Speed, Ringer Impedance, Ringer Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum Operational Loop Current, and AC Impedance as predefined for your country’s analog line characteristics. The default setting is “United States of America (USA)”.
FXS Opermode	Select country to set the On-Hook Speed, Ringer Impedance, Ringer Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum Operational Loop Current, and AC Impedance as predefined for your country’s analog line characteristics. The default setting is “United States of America (USA)”.
FXS TISS Override	Configure to enable or disable override Two-Wire Impedance Synthesis (TISS). The default setting is No.   If enabled, users can select the impedance value for Two-Wire Impedance Synthesis (TISS) override. The default setting is 600Ω.
PCMA Override	Select the codec to be used for analog lines. North American users should choose PCMU. All other countries, unless already known, should be assumed to be PCMA. The default setting is PCMU.   Note: This option requires system reboot to take effect.
Boost Ringer	Configure whether normal ringing voltage (40V) or maximum ringing voltage (89V) for analog phones attached to the FXS port is required. The default setting is “Normal”.
Fast Ringer	Configure to increase the ringing speed to 25HZ. This option can be used with “Low Power” option. The default setting is “Normal”.
Low Power	Configure the peak voltage up to 50V during “Fast Ringer” operation. This option is used with “Fast Ringer”. The default setting is “Normal”.
Ring Detect	If set to “Full Wave”, false ring detection will be prevented for lines where Caller ID is sent before the first ring and proceeded by a polarity reversal, as in UK. The default setting is “Standard”.
FXS MWI Mode	Configure the type of Message Waiting Indicator on FXS lines. The default setting is “FSK”.   FSK: Frequency Shift Key Indicator NEON: Light Neon Bulb Indicator.
FXO Frequency Tolerance	Allows users to adjust the tolerance of the FXO ringing frequency. 63Hz is considered the standard value and is selected by default.

DAHDI Settings

When users encounter issues such as audio delay in outbound calls using the analog trunk, they can adjust DAHDI settings on the UCM to attempt to lessen or resolve the issues.

For the value of the option such as “32, half”:

The number in the option indicates the number of read/write buffers for TDM (DAHDI).

The “Half”, “Immediate” or “Full” option indicates the strategy when reading/writing data from buffer.

• “Half”: Data will be read/written from buffer when half of the buffer is occupied with data.
• “Immediate”: Read/write from buffer whenever there is data occupying the buffer.
• “Full”: Data will be read/written from buffer when buffer is fully occupied with data.

Normally, DAHDI settings should be kept default and should be adjusted only when users encounter analog trunk/Fax-related issues.

CONTACTS

Address book management is under UCM web UI->Maintenance, and it has two sections “Contact Management” and “Department management”.

Contact Management

Contact management page displays extension contacts and external contacts information.

• Extension contacts

Extension contacts page shows all the extensions that has “Sync Contact” option enabled in extension settings page. The extension contacts here can be edited or deleted individually or in batch. No new extension contact can be added directly from this page. If an extension contact is deleted from this page, “Sync Contact” option is disabled from this extension. This will not delete the extension from UCM.

Click Edit icon to configure name, department, email address and etc for each extension contact.

Extension	Displays extension number.
First Name	Configure first name for the extension contact.
Last Name	Configure last name for the extension contact.
Department	Select department for the extension contact. Department can be created in “Department Management” page.
Department Title	Configure the job title for the extension contact.
Email Address	Configure email address for the extension contact.
Mobile Phone Number	Configure mobile phone number for the extension contact.
Home Number	Configure home number for the extension contact
Fax	Configure Fax for the extension contact.
Same as Department Contact Privileges	When this option is enabled, the contact extension will inherit the same privilege as the department it belongs to.
Contact View Privileges	This option allows configuring privileges for the contact extension.Note: This option will be disabled if “Same as Department Contact Privileges” has been enabled.

• External contacts

On external contacts page, the admin can create single external contact, import contacts in batch, edit contacts, delete contacts and export contacts.

Click on “Export” icon, a CSV format file will be generated with the current external contacts.

Click on “import” icon, then follow the steps below to add external contacts in batch:

• Step 1: For option “On Duplicate External Contacts”, select whether to skip duplicate contact on the imported CSV file or update the duplicate UCM contact with the information in the CSV.
• Step 2: Choose file from local PC to upload.
• Step 3: Click on “Upload”.
• Step 4: Click on “Apply” to complete importing external contacts.

Department Management

Departments are organizational units that allows organizing extensions within groups that specify the specialty of a the extension owners within a company. This makes finding contacts easier within the UCM contact books.

Click on “Add” to create a new department. Configure the department name and select the superior department. By default the superior department is the root directory. If the UCM has cloud IM configured, the root directory will be the deparment in cloud IM.

On the department list:

• Click on

  

  to create sub department.
• Click on

  

  to add member to the department.
• Click on

  

  to edit the department.

Privilege Management

The user can configure custom privileges other than the default ones (All contacts, Departments and sub-departments contacts). These custom privileges allow more flexible ways of allowing contacts to view all or specific contacts from other departments.

UCM admin can add or edit Privilege Management; under UCM web UI🡪Contacts Privilege Management, there are 2 default privileges:

• Visible to all contacts.
• Only the contact person’s department and sub-department contacts are visible.

When Cloud IM is enabled on the UCM, a third privilege becomes available to choose:

• Local Contacts: Restricts the contacts shown to the contacts of the local UCM.

MESSAGING

Live Chat

Live Chat feature allows to create chat channels that can be embedded on your website to enable your client to reach your customer service more easily. The client can contact your agents by text then

Message Broadcast

Message broadcast feature allows the administrator to broadcast a text message to all the endpoints selected by the administrator. The administrator can select departments or individual extensions to boardcast a text message.

DEVICE MANAGEMENT

IPC Devices

The UCM admin can add IPC devices and edit accessible extensions so these extensions can view the surveillance streams for the IPC devices.

Click on “Add” to add IPC device.

Edit the IPC device settings in the table below.

IPC Devices Settings

REMOTECONNECT

An integrated and essential component of Grandstream’s GDMS cloud-based device management service, UCM RemoteConnect runs on Amazon Web Services (AWS) and provides high-availability remote communication and collaboration services.

UCM RemoteConnect enables work from home and mobile users to securely access UCM services using Grandstream Wave softphones, with license-free WebRTC-based communication, automatic NAT traversal, and centralized cloud-based management. In addition, RemoteConnect provides GDMS cloud storage, Cloud IM, AI Services, and other collaboration and management features.

The RemoteConnect service can be configured from the UCM Web GUI under the RemoteConnect tab.

Plan

This section provides an overview of the RemoteConnect service associated with the UCM, including service description, current subscription status, and configurable plan options.

Service Description

After the UCM is added to GDMS, RemoteConnect enables automated SIP extension synchronization, Media NAT Traversal, and basic statistics without requiring manual configuration. These services allow remote users and devices to connect reliably through the cloud.

This tab provides an overview of the UCMRC services and displays the current subscription plan, as well as related links and learning resources for the services supported by RemoteConnect.

My Plan

This page displays detailed information about the currently active RemoteConnect subscription, including plan name, expiration date, concurrent session limits, remote registration limits, call time limits, allocated GDMS cloud storage space, Wave RemoteConnect addresses, and support status for Wave add-ins, Cloud IM Server, and AI features.

Plan Settings

This tab allows administrators to configure how the UCM interacts with UCMRC services, including SIP synchronization, NAT traversal, cloud storage, backups, and remote access behavior. All settings on this page take effect immediately after saving and directly impact how the UCM integrates with GDMS and cloud services.

The table below describes the configurable parameters on this page.

When “SIP Extension Sync” is enabled, the accounts synchronized to GDMS platform can be viewed on the GDMS→VoIP Account→SIP Account page. As shown in the figure below:

AI Services

This section displays AI related service status and usage information for the UCM. It provides visibility into the active AI plan, its validity period, and detailed usage statistics. Users can also jump to configure AI Settings from this page.

Supported AI features include:

• AI Voice Message to Text
• AI Generated Speech (Text-to-Speech)
• AI Call Transcription and Summary
• AI Conference Transcription and Summary

Usage statistics show monthly consumption per feature and total usage versus the available quota, helping administrators monitor AI resource usage.

Integrated Customer Service

This section allows administrators to configure web-based customer service features that integrate communication tools into websites. Under the Integrated Customer Service tab, administrators can access options for WebRTC Trunks, Enable Click2Call, and Live Chat Customer Service.

Through this section, administrators can navigate directly to the configuration pages for WebRTC Trunks and Live Chat Customer Service to manage web-based customer interactions.

Enterprise UI Customization

With a RemoteConnect plan, the Enterprise UI Customization page allows administrators to apply customized branding across the UCM and Wave interfaces. From this page, the company name can be configured and multiple logo images can be uploaded to replace the default branding in supported system interfaces.

The system supports different logo sizes and formats to ensure proper rendering in various locations as described below

• Logo 1 (80×80px): Replaces the Grandstream logo on the top left corner of the Wave Web interface and UCM management interface. The recommended format is PNG, and the maximum file size is 2 MB.
• Logo 2 (256×64px): Replaces the Grandstream banner on the top left corner of the login page and emails. The recommended format is PNG, and the maximum file size is 2 MB.
• Logo 3 (64×64px): Replaces Browser tab icon. The recommended format is .ico, and the maximum file size is 2 MB.

A built-in preview panel is provided on the right side of the page, allowing administrators to review how the configured company name and logos will appear in different interfaces.

Statistics

The Statistics page provides an overview of RemoteConnect usage by displaying historical data on maximum concurrent remote sessions. This information helps administrators monitor RemoteConnect capacity usage and understand how remote access services are consumed over time.

Usage statistics can be viewed by month, week, or day. When viewing data by month, the system displays the highest number of concurrent remote sessions recorded during each month of the selected year.

GDMS Cloud Storage Space

When the correspondent RemoteConnect plan is active, the user can access to a GDMS Cloud Storage Space to get an overview about how the storage space is being used. The data is represented in four categories of file types: CDR Data, Backup Data, Recording Files, and IM Files.

The user is also able to see the names of all the files stored in the GDMS Cloud Storage Space.

On this page the user can obtain the file key by clicking on the following hyperlink.

AI SETTINGS (BETA)

The AI Settings page allows administrators to configure the usage of AI services provided through the RemoteConnect AI service. These features enable automatic speech recognition, transcription, summarization, and text-to-speech processing for calls, conferences, and voice messages.

AI services are a purchaseable add-on and subject to usage limits defined by the active RemoteConnect plan. The administrator can enable each AI feature independently and define whether it applies to all extensions or only selected extensions.

The following AI capabilities are supported on the UCM:

• AI Call Transcription and Summary
• AI Conference Transcription and Summary
• AI Voice Message to Text
• AI Generated Speech

Voice Transcription

The Voice Transcription tab consolidates configuration for all speech-to-text features, including calls, meetings, media recording files, and voicemails.

AI Call Transcription and Summary

This feature enables AI-based transcription and summarization for one-to-one calls. When enabled globally, the administrator defines whether the feature applies to all extensions or only selected extensions. Only extensions included in this scope are eligible for call transcription and summarization.

At the extension level, the Automatic Transcribing and Summarizing option determines whether calls for that extension are processed automatically. When enabled, all calls made or received by the extension are automatically transcribed and summarized.

The recognition language can be configured globally under AI Settings (Beta)→AI Call Transcription and Summary and followed by the extension, or explicitly set per extension.

Generated transcripts and summaries can be viewed by users through the Wave client, as shown below.

AI Meeting Transcription and Summary

This feature enables AI-based transcription and summarization for conference calls and meetings. When enabled, meeting audio is processed by the AI service and a transcript and summary are generated after the meeting ends. The administrator can define whether the feature applies to all extensions or only selected extensions participating in conferences.

When scheduling a meeting, the meeting organizer can explicitly choose whether the conference will be transcribed or not. This selection applies only to the scheduled meeting and does not affect global or extension-level AI settings.

Meeting transcripts and summaries can be accessed by participants through supported Wave clients and user portals.

AI Recording Transcription and Summary

This feature allows audio recording files stored on the UCM to be transcribed and summarized using AI. When enabled globally, the administrator specifies whether the scope applies to all extensions or selected extensions.

Once transcribed, users can review recording transcripts with interactive synchronization. Clicking a text segment automatically seeks media playback progress to that timestamp, and active media playback dynamically highlights the corresponding transcript text.

AI Voicemail to Text

This feature converts voice messages (voicemail) into text using AI speech recognition. When enabled globally, voice messages left for eligible extensions can be transcribed.

At the extension level, the Automatic Transcription option determines whether voice messages for that extension are automatically converted into text.

The recognition language can follow the global configuration or be explicitly specified for the extension.

Transcribed voicemail content can be viewed in supported Wave clients, through the User Portal (where accessing or playing the voicemail marks it as read), and may also be included in voicemail email notifications depending on system configuration.

AI Voice Message to Text

This feature converts voice messages (voicemail) into text using AI speech recognition. When enabled globally, voice messages left for eligible extensions can be transcribed.

At the extension level, the Automatic Transcription option determines whether voice messages for that extension are automatically converted into text.

The recognition language can follow the global configuration or be explicitly specified for the extension.

Transcribed voicemail content can be viewed in supported Wave clients and may also be included in voicemail notifications, depending on system configuration.

AI Translation

The AI Translation feature provides real-time translation capabilities for call transcripts, conference transcripts, and chat messages across Wave endpoints. When enabled globally, the administrator defines the extension scope (All Extensions or Selected Extensions) authorized to use AI Translation.

Once granted permission, users can translate live speech-to-text transcripts during active calls/meetings or translate instant chat messages on-demand.

AI Voice Generation

The Voice Generation tab manages AI Generated Speech capabilities. This feature provides text-to-speech (TTS) processing powered by the AI service, allowing administrators to generate voice prompt audio directly from text. This feature is primarily used for IVR prompts, queue announcements, and other system voice prompts, eliminating the need to manually record or upload audio files.

Before using this feature, AI Generated Speech must be enabled on the Voice Generation tab under AI Settings. Once enabled, administrators can generate AI-based voice prompts from multiple entry points in the system.

The feature can be accessed from PBX Settings→Voice Prompt→Custom Prompt by clicking the AI Generated Speech button, which opens the AI generation window.

It is also accessible while configuring features that require voice prompts (such as call queues). In those cases, selecting AI Generated Speech from the prompt selection dropdown redirects the user to the same AI generation window used in the Voice Prompt page.

When creating an AI-generated voice prompt, the following settings are available:

Name	Defines the name of the generated voice prompt. This name is used to identify the prompt in the system and must be unique. The field supports 2–64 characters, including letters, numbers, and special characters.
Text	Specifies the text content that will be converted into speech. The text length supports up to 5,000 characters.
Language	Specifies the language used for text-to-speech processing.
Voice	Selects the voice profile used to generate the speech. Different voice options may be available depending on the selected language.A preview option is provided, allowing administrators to listen to the voice style before generating the prompt.
Speech Speed	Controls the playback speed of the generated speech. The speed can be adjusted from slower to faster relative to normal speaking speed.
Volume	Adjusts the output volume level of the generated audio. The value can be set from 0 to 100 to accommodate different playback environments.
Pitch	Controls the pitch of the generated voice. Higher values produce a higher-frequency voice, while lower values produce a deeper tone. The pitch range can be set from 0 to 100.

After configuring the parameters, clicking AI Generate produces the audio preview. Once satisfied, the generated prompt can be saved and reused like any other system voice prompt.

AI Voice Assistant

The Voice Assistant tab allows administrators to configure natural voice command controls for endpoints. When enabled globally, extension users within the configured scope can issue spoken commands to perform actions such as making calls, transferring calls, initiating meetings, or sending messages.

Administrators can configure the following settings under AI Settings→Voice Assistant.

• Enable: Toggles the AI Voice Assistant service on or off.
• Extension Range: Controls which extensions are permitted to speak commands to the AI Voice Assistant (e.g., Only allow Extensions 1000-1005 to use the Voice Assistant feature).
• AI Contacts: Controls which contacts/directory entries the AI can look up when executing a voice command (e.g., Allow to search the entire contacts when an extension user asks “Call John”).
• Contact Name Language: Defines the language rules used to recognize and match spoken names in the directory.

API CONFIGURATION

The UCM630xA supports third party billing interface API for external billing software to access CDR and call recordings on the PBX. The API uses HTTPS to request the CDR data and call recording data matching given parameters as configured on the third-party application.

API Configuration Parameters

Before accessing the API, the administrators need enable API and configure the access/authentication information on the UCM630xA first under Integrations 🡪 API Configuration. The API configuration parameters are listed in the table below.

Note: The old version of the API interface only supports cdrapi, recapi and pmsapi functions, and will be removed, please use the new HTTPS API instead.

Configuration Parameters (New)

HTTPS API Settings (New)
Enable	Enable/Disable API. The default setting is enable.
Username	Configure the username for API Authentication.
Password	Configure the password for API Authentication.
Call Control	If enabled, 3rd party applications will be able to manage inbound calls via API actions. acceptCall will accept incoming calls while refuseCall will reject them. If no actions are done within 10 seconds, calls will automatically be accepted.
Permitted IP (s)	Sets an IP address Access Control List (ACL) for addresses that are allowed to authenticate as this user. By default this is not set, meaning all IP addresses will be allowed. The format is: “xxx.xxx.xxx.xxx/255.255.255.255”.

Configuration Parameters (Old)

HTTPS API Settings (Old)
Basic Settings
Enable	Enable/Disable API. The default setting is disabled.
TLS Bind Address	Configure the IP address for TLS server to bind to. “0.0.0.0” means binding to all interfaces. The port number is optional, and the default port number is 8443. The IP address must match the common name (host name) in the certificate so that the TLS socket will not bind to multiple IP addresses.   The default setting is 0.0.0.0:8443.
Username	Configure the username for TLS authentication.
Password	Configure the password for TLS authentication.
Permitted IP(s)	Specify a list of IP addresses permitted to use the API. This creates an API-specific access control list. Multiple entries are allowed.   For example, “192.168.40.3/255.255.255.255” denies access from all IP addresses except 192.168.40.3. By default, this is blank, which indicates that no IP addresses are allowed to use this API.
Other Settings
TLS Private Key	Upload TLS private key. The size of the key file must be under 2MB. This file will be renamed as ‘private.pem’ automatically.
TLS Cert	Upload TLS cert. The size of the certificate must be under 2MB. This is the certificate file (*.pem format only) for TLS connection. This file will be renamed as “certificate.pem” automatically. It contains private key for the client and signed certificate for the server.
API Module
CDR API	Enable/disable CDR API module.
REC API	Enable/disable REC API module.
PMS API	Enable/disable PMS API module.

For more details on CDR API (Access to Call Detail Records), REC API (Access to Call Recording Files) and PMS API, please refer the document in the link here:

• https://documentation.grandstream.com/knowledge-base/cdr-rec-api/
• https://documentation.grandstream.com/knowledge-base/cdr-rec-api/
• PMS API

API Queries Supported

The new API supports the queries listed below which will accomplish certain requests and get data about different modules on UCM630xA.

The table below describes the new API Supported Queries.

API Configuration Parameters

CDR Real-time Output Settings
Enable	Enables real-time CDR output module. This module connects to selected IP addresses and ports and posts CDR strings as soon as it is available.
Server Address	CDR server IP address
Port	CDR server IP port
Upload Prompts User Configuration
Username	Username used to upload prompts.
Password	Password used to upload prompts.

Upload Voice Prompt via API

Customers now can use the “Upload Prompts User Configuration” to upload/replace voice prompt files as an alternative method to the manual upload method on UCM PBX Settings🡪 Voice Prompt🡪 Custom Prompt.

The workflow of the prompt file upload goes as:

An HTTP/HTTPS request is sent to the UCM to upload/replace a voice prompt file, the request should include authentication details to the UCM and the name of the file to be uploaded. Then the UCM will contact an FTP server that should be hosted on the same IP address of the HTTP/HTTPS requester and download the prompt file from the FTP server.

The steps and conditions to upload the voice prompt via API are listed below:

1. Configure the prompt User under Integrations 🡪 API Configuration 🡪 Upload Prompts User Configuration. By default, the username and password for voice prompt user are “Username: uploader; Password: uploader123”.

1. Hash the password of the user configured to an MD5 Encryption format.
2. Set the permission on the FTP server to Anonymous on the local computer hosting the FTP server and make sure that the default FTP port 21 is used.
3. Send an HTTP/HTTPS command to trigger the Prompt file upload on the UCM. If UCM’s HTTP server is set to HTTPS, the example of the request sent to the UCM is:

https://192.168.124.89:8089/cgi?action=uploadprompt&username=uploader&password=9191a6394c21b3aabd779213c7179462&filename=test.mp3

1. If UCM’s HTTP server is set to HTTP, the example of the request sent to the UCM is：

http://192.168.124.89:8089/cgi?action=uploadprompt&username=uploader&password=9191a6394c21b3aabd779213c7179462&filename=test.mp3

CTI SERVER

UCM does support CTI server capabilities which are designed to be a part of the CTI solution suite provided by Grandstream, including GXP21XX and GXP17XX enterprise IP phones along with GS Affinity app.

Mainly the UCM will by default listening on port TCP 8888 for the connections from GS affinity application in order to interact, modify and serve data requests by the application which includes setting call features for the connected extension as call forward and DND.

Users can change the listening port under the menu page, Integrations 🡪 CTI Server as shown on below screenshot:

More information about GS affinity and CTI Support on Grandstream products series please refer to the following link:

https://documentation.grandstream.com/knowledge-base/gs-affinity-user-guide/

ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)

The UCM630xA supports Asterisk Manager Interface (AMI) with restricted access. AMI allows a client program to connect to an Asterisk instance commands or read events over a TCP/IP stream. It is particularly useful when the system admin tries to track the state of a telephony client inside Asterisk.

User could configure AMI parameters on UCM630xA Integrations 🡪 AMI. For details on how to use AMI on UCM630xA, please refer to the following AMI guide:

https://documentation.grandstream.com/knowledge-base/ami-asterisk-management-interface/

CRM

Customer Relationship Management (CRM) refers to practices, strategies, and technologies that companies use to manage and analyze customer interactions and data throughout the customer lifecycle, with the goal of improving business relationships with customers.

The UCM630X supports seamless integration with various CRM systems, allowing users to look up contact information in the Contacts, Leads, and Accounts tables, display contact records directly in the CRM web interface during incoming calls, perform one-click outbound calling, and automatically synchronize call history into the contact’s activity history.

To accommodate different business environments, the UCM630X provides two flexible integration approaches:

• Default Built-in Integration Templates: Pre-configured, out-of-the-box integration solutions for widely used CRM platforms, including SugarCRM, VtigerCRM, Bitrix24CRM, Zoho Telephony, Salesforce, ACT! CRM, and ODOO CRM.
• Custom CRM Templates: Designed for proprietary or third-party CRM platforms not included in the default list. Administrators can create, import, export, or edit custom CRM templates.

The main CRM page serves as the central hub for selecting and managing integrated CRM platforms. Here, administrators can view all available default and custom CRM modules, activate or switch the active CRM instance.

Navigating to Template Management → Default Template displays the list of pre-configured CRM templates built into the system. From this page, administrators can inspect the structure of default templates or download them as JSON configuration files to serve as reference models when building custom integrations.

Clicking the View Template icon on any default template opens a pop-up window displaying its underlying JSON schema. This view reveals the exact REST API endpoints, request methods, authentication structure, token refresh URLs, and parameter definitions used by the platform.

The Template Management → Custom Template page allows administrators to manage user-defined CRM templates. From this tab, administrators can create new templates manually by clicking Add, batch-upload existing JSON template files, export custom templates for deployment across other UCM systems.

When creating or editing a custom CRM template, the Edit Custom Template interface provides sections to upload a custom CRM logo, set general metadata (Name and Remarks), and configure detailed REST API request parameters.

This includes selecting authentication protocols (Basic, Bearer Token, OAuth 2.0), providing certificates or tokens, and defining custom fields that prompt users for required credentials during initial CRM pairing

Sugar CRM

Configuration page of the SugarCRM can be accessed via admin login, on the Integrations 🡪 CRM.

1. Select “Sugar CRM” from the CRM System Dropdown in order to use SugarCRM.

Sugar CRM Settings

CRM System	Select a CRM system from the dropdown menu, four CRM systems are available: SugarCRM, vTigerCRM, ZohoCRM (v1&v2), Salesforce and ACT! CRM.
CRM Server Address	Enter the IP address of the CRM server.
Add Unknown Number	Add the new number to this module if it cannot be found in the selected module.
Contact Lookups	Select from the “Available” list of lookups and press to select where the UCM can perform the lookups on the CRM tables, Leads, Accounts, and Contacts.

Once settings on admin access are configured:

1. Click on
   

   

   
   and
   

   

   
   .
2. Logout from admin access.
3. Login to the UCM as user and navigate under “User Portal🡪Value-added Feature🡪CRM User Settings”.

Click on “Enable CRM” and enter the username/password associated with the CRM account then click on

and

. The status will change from “Logged Out” to “Logged In”. User can start then using SugarCRM features.

Vtiger CRM

Configuration page of the VtigerCRM can be accessed via admin login, on the Integrations 🡪 CRM.

1. Select “Vtiger CRM” from the CRM System Dropdown in order to use Vtiger CRM.

Vtiger CRM Settings

CRM System	Select a CRM system from the dropdown menu, four CRM systems are available: Sugar CRM, Vtiger CRM, Zoho CRM (v2), Salesforce and ACT! CRM.
CRM Server Address	Enter the IP address of the CRM server.
Add Unknown Number	Add the new number to this module if it cannot be found in the selected module.
Contact Lookups	Select from the “Available” list of lookups and press to select where the UCM can perform the lookups on the CRM tables, Leads, Organizations, and Contacts.

Once settings on admin access are configured:

1. Click on
   

   

   
   and
   

   

   
   .
2. Logout from admin access.
3. Login to the UCM as user and navigate under “User Portal🡪Value-added Feature🡪CRM User Settings”.

Click on “Enable CRM” and enter the username/password associated with the CRM account then click on

and

. The status will change from “Logged Out” to “Logged In”. User can start then using SugarCRM features.

Bitrix24 CRM

The UCM6300 Series supports integration with Bitrix24 CRM, allowing the system to query CRM records during calls, display contact information, and automatically create CRM records for unknown numbers based on configuration.

The table below describes the configurable parameters for this integration.

Once the Bitrix24 CRM settings on the admin interface are configured:

1. Click Save and Apply Changes.
2. Log out from the admin account.
3. Log in to the UCM as a user and navigate to User Portal→Other Features→CRM User Settings.
4. Enable CRM and enter the Bitrix24 account credentials information as required.
5. Click Save to complete the login process.

After successful login, the CRM status will change from Logged Out to Logged In, and the user can start using Bitrix24 CRM features during calls.

Salesforce CRM

Configuration page of the Salesforce CRM can be accessed via admin login, on the UCM Web GUI Integrations 🡪 CRM”.

1. Select “Salesforce” from the CRM System Dropdown in order to use Salesforce CRM.

Once settings are configured:

2. Click on 

   

    and 

   

    .
3. Logout from admin access.
4. Login to the UCM as user and navigate under “User Portal🡪Other Features🡪CRM User Settings”.

Click on “Enable CRM” and enter the username, password and Security Token associated with the CRM account then click on  and  . The status will change from “Logged Out” to “Logged In”. User can start then using Salesforce CRM features.

ACT! CRM

Configuration page of the ACT! CRM can be accessed via admin login, on the UCM Web GUI Integrations🡪CRM”.

The configuration steps of the ACT! CRM are as follows:

1. Navigate to Integrations 🡪 CRM and select the “ACT! CRM” option.

2. Log into the UCM as a regular user and navigate to Other Features🡪CRM User Settings and check “Enable CRM” option and enter the username and password, which will be the ACT! CRM account’s API Key and Developer Key, respectively. To obtain these, please refer to the ACT! CRM API developer’s guide here: https://www.act.com/

Odoo CRM

The UCM6300 Series supports integration with Odoo CRM. To enable Odoo CRM, please access the UCM’s web UI then navigate to Integration > CRM.

Click “Save” then “Apply Changes”.

Once Odoo CRM is enabled, the user can log into the user portal by access the UCM web UI and entering their username and password, then they can navigate to Other Features > CRM User Settings, then they configure their account using username and password created on Odoo platform.

PMS INTEGRATION

UCM630xA supports Property Management System PMS, including check-in/check-out services, wakeup calls, room status, Do Not Disturb which provide an ease of management for hotel applications. This feature can be found on Web GUI→Integrations🡪PMS.

The PMS module built-in the UCM supports the following features based on each solution:

Char pmslink

In this mode, the system leverages the char pmslink cloud middleware to integrate the UCM with a wide variety of Property Management System (PMS) software.

The communication flow is divided into three parts:

• PMS (Property Management System): Your hotel’s management software.
• charpmslink (Middleware): Acts as the universal interface, translating messages between the PMS and the PBX.
• PBX (UCM630xA): The Grandstream UCM, which receives commands from the PMS via charpmslink to execute hospitality features.

This integration enables a large variety of hotel features such as: Check-in, Check-out, set Room Status, Wake-up call, and more.

The following figure illustrates the communication flow between the UCM and PMS software, which is done through a middleware system (char pmslink) acting as an interface between both parties.

HSC PMS

In this mode, the system can be divided into two parts:

• PMS (Property Management System)
• PBX

Grandstream UCM6XXX series has integrated HSC PMS providing the following features:

• Changing Display Name
• Set Station Restriction
• Call forwarding
• DND
• Name Change
• MWI

Notes:

1. Added support for receiving HTTP GET keep-alive messages from HSC PMS. This will allow the PMS to be aware of its connection to the UCM and take the appropriate actions such as raising alarms, sending notifications, etc.
2. Added support for HTTP GET requests from HSC PMS to retrieve UCM extension information. UCM can provide the following information:

• extension – UCM extension number
• name – extension display name / CID name
• mwi – MWI state
• permission – permission level of the extension
• cfwt – call forwarding always number
• dnd – DND state
• language – display language of the extension in ISO 639-1 format

The UCM should respond with either 200 OK or 404 responses.

3. Added HTTPS support
   

The following figure illustrates the communication flow between the PBX (Grandstream UCM630xA Series) and PMS software (HSC). The communication between both parties is direct with no middleware.

Mitel PMS

In this mode, the system can be divided into two parts:

• PMS (Property Management System)
• PBX

Grandstream UCM630xA series has integrated Mitel PMS providing the following hospitality features: Check-in, Check-out, set Room Status, Wake-up call, and more.

The following figure illustrates the communication flow between the PBX (Grandstream UCM630xA Series) and PMS software (Mitel). The communication between both parties is direct with no middleware.

IDS PMS

In this mode, the system can be divided into two parts:

• PMS Clients – These are the hotel management applications such as the front desk system, reservation software, or any client that normally communicates with a PMS server.
• UCM PBX – Instead of acting as a client or middleware, the UCM directly emulates an IDS PMS server.

With this setup, PMS clients connect directly to the UCM through its PMS port. The UCM listens for incoming requests and responds as if it were the IDS PMS server. This allows hotel staff to perform operations such as setting room status (clean, occupied, available), posting Mini Bar charges, scheduling and managing wake-up calls, or enabling and disabling dialing permissions.

By integrating IDS PMS in this way, the UCM simplifies deployment since no external IDS PMS server or additional middleware is required—the PBX itself provides the PMS server functionality.

PMS AP

The PMS API allows users to use their middleware to work with PMS systems instead of currently supported integrations.

Additionally, this API allows access to read and modify certain UCM parameters that current supported PMS integrations cannot. To use this, users must first enable and configure the HTTPS API settings.

For more details, please refer to the online https://documentation.grandstream.com/knowledge-base/https-api/, Pmsapi section.

Oracle Hospitality OPERA

The UM603xA series integrates directly with Oracle Hospitality OPERA PMS to manage room statuses and extensions from within the OPERA interface. This connection is established using the IFC_PBX interfacing protocol.

To configure the integration, first select the pre-built interface, “Oracle Hospitality OPERA TMS Interface for UCM Series PBX by Grandstream Networks,” within the OPERA software. Once enabled, you can complete the setup from the CloudUCM admin portal.

Data exchange between the UCM and the OPERA server is handled by the Fidelio Interface Application Specification (FIAS) protocol. FIAS uses specific attributes to define the data being transmitted. For example:

• “RN” represents the Room Number.
• “RS” represents the Room Status.
• “RU” is used as an update indicator to initiate a room status change.

A typical FIAS message to update a room’s status would be formatted as follows: RU|RN211|RS3

In this example, room 211 is updated to status code 3.

In addition, The UCM630xA supports Oracle OPERA’s Youth Hostel Mode. This feature allows a single room to be partitioned into multiple independent units, enabling management of separate groups with distinct extensions and individual billing records within the same physical room.

Local PMS

The UCM6300 series includes an integrated on-premise Property Management System (PMS), providing essential hospitality management capabilities directly from the Web GUI under Integrations → PMS without requiring external software licenses.

The local PMS solution manages core hotel operations including Room Management, Reservation and Check-in/Check-out management, Hourly Room rentals, Minibar consumption tracking, Wake-up Services, and more.

Key highlights and features supported by the Local PMS include:

• Custom Hotel Branding: Configure hotel logos, contact phone numbers, website, and a QR code to customize printed bills and automated self-check-in/check-out emails.
• Self Check-in and Check-out: Guests can check in and check out using emails automatically sent by the UCM for reservations.
• Room & Calendar Management: Real-time Room Availability Calendar, itemized bill printing (breakdowns for taxes, room charges, and minibar items), and hourly room options.
• Guest List: Stored database of guests which helps with future bookings and retains guest information for quick reference.

For detailed step-by-step configuration guides and operational workflows, refer to the Local PMS Guide.

https://documentation.grandstream.com/knowledge-base/ucm63xx-local-pms-guide

The configuration parameters available under Local PMS Basic Settings are explained in the table below:

Parameter	Description
General
PMS Module	Selects the Local PMS module functionality.
Hotel Name	Configures the official hotel name to be displayed on guest bills, and emails.
Address	Configures the hotel physical address displayed on bills and emails.
Hotel Phone Number	Configures the contact phone number printed on bills and emails.
Hotel Email	Configures the official hotel email address used for bills and emails.
Hotel Website	Configures the hotel website URL printed on guest bills and emails.
Logo	Uploads a custom hotel logo for printed bills and automated emails. Supports JPEG, JPG, and PNG formats up to 5 MB.
QR Code	Uploads a custom QR code (e.g., for online hotel guides) displayed on printed bills and automated emails. Supports JPEG, JPG, and PNG formats up to 5 MB.
Prompt
Wakeup Prompt	A customized prompts that can be played when the wakeup call is answered. To customize it please navigate to PBX Settings→Voice Prompt→Custom Prompt.
Room Status Update Prompt	Choose a previously uploaded prompt or upload the prompt which will be played when the room status is changed. If the room status codes have been change, please update the Room Status accordingly.
Automatically Clear/Reset
Automatically Clear Phone Call History	Configures whether or not the call history of phones will be automatically cleared upon check-in or check-out. Currently only supported on Grandstream phones. None: Call history will not be deleted after checking-in or checking-out. Check Out: Call history will be delete when the guest checks-out. Check In: Call history will be delete when a new guest checks-in.
Automatically Clear Wakeup Calls	Scheduled wakeup calls for rooms can be cleared upong checking in or checking out. None: The wakeup calls won’t be automatically cleared. Check out: The wake up calls assigned to the guest will be cleared when they check out. Check In: The wake up calls assigned to a guest will be cleared when a new client checks in.
Automatically Clear Wave Chat History	If enabled, room Wave chat history will be automatically cleared upon check-in or check-out.
Automatically Reset User/Wave Password	If enabled, the User/Wave password of the room extension will be automatically reset to a random password upon check-out.
Self-service Settings
Self-Check-In Via Email	Enables the system to automatically send a self check-in email with a link prior to the guest’s arrival.
Check-in Email Template	Selects the email template used for sending automated self check-in notifications.
Send in Advance (hrs)	Defines how many hours before the scheduled check-in time the self check-in email will be sent. (Range: 0 – 6 hours).
Self-Check-Out Via Email	Allows the system to automatically send a self check-out email with a link prior to departure.
Check-out Email Template	Selects the email template used for sending automated self check-out notifications.
Send in Advance (hrs)	Defines how many hours before the scheduled check-out time the email will be delivered. (Range: 0 – 6 hours).
Allow Check-out with Unsettled Bill	Controls whether guests are allowed to complete self check-out even if their bill is not fully paid. Enabled: The guest may check out before completing the payment. Disabled: Payment must be settled before self-check-out is allowed. This option is disabled by default.
Other
Back Up Voicemail Recordings	Configures automatic backup of voicemail recordings to external storage after check-out. When this feature is enabled, users can specify the storage destination.
Storage Location	Select the storage location for voicemail recordings. The available options are: SFTP Server NAS USB SD Card Note: To configure storage devices, go to PBX Settings→Storage Device Management.
Email Address	Enter the destination email address where the system will deliver backup files.
Sync Guest Name to Phone	Provisions the name of the checked-in guests to endpoints via Zero Config. Requires endpoints to be discoverable and provisinable by Zero Config.
Review Bill at Check-Out	If enabled, a pop-up window with all the charges will appear during guest check-out for reviewing purposes.
Call Billing
Call Rate	Configures call charge rates for outbound calls made from guest room extensions.
Currency Unit	The currency unit for the call rate. These are the supported options which you can choose from: American Dollar Euro Sterling Pound Yen Won Hong Kong Dollar Australian Dollar Canadian Dollar Baht Singapore Dollar Swiss Franc Swedish Krona Danish Krone Norwegian Krone New Zealand Dollar South African Rand Brazilian Rial Indian Rupee Russian Ruble Vietnamese Dong Polish Zolty Czech Koruna Turkish Lira Custom: Enter the currency unit.
Periodically Send Records to Email	If enabled, new recordings will be automatically sent to the specified email inbox periodically at the specified intervals.
Type	Configures the type of recordings that should be sent periodically: Wakeup Service Check-in/Check-out History CDR Records
Period	Specifies the automatic download period. The default option is “By Day”. At the specific time of each day, week or month, the last cycle of new records will be sent to the user. By Day By Week By Month
Email	Specifies the automatic download period. The default option is “By Day”. At the specific time of each day, week or month, the last cycle of new records will be sent to the user.
Guest Email
Customer Email	Enables the system to send automated emails to guests. This includes reservation updates and billing information. Note: Email notifications will not be sent if the customer’s email address is missing.
Reservation Notification	When enabled, the guest will automatically receive an email whenever a reservation is created, modified, or cancelled.
Billing Notification	If enabled, billing-related emails will be sent automatically to the guest.
Email Template	Selects or edits custom email templates used for reservation and billing notifications.

INTEGRATION OF COLLABORATION PLATFORMS

The UCM6300/A Series supports integration with enterprise collaboration and identity platforms to centralize user management, authentication, and directory synchronization. These integrations allow administrators to synchronize user identity information from external platforms into the UCM, associate users with extensions or system accounts, and enable Single Sign-On for simplified and secure access to the UCM web interface.

Supported platforms include:

• Microsoft Entra ID (Azure AD)
• Windows Active Directory
• Google Workspace

Each integration provides configurable options for user synchronization scope, user linking behavior, LDAP directory integration, and authentication methods. The following sections describe the configuration and usage of each platform.

Microsoft Entra ID (Azure AD)

Microsoft Entra (formerly Azure Active Directory) is a cloud platform which provides access management and identity provision. When integrated with the UCM6300 Series, it allows synchronizing the identity information with the UCM which can be used to log into the UCM’s user interface.

The Microsoft Entra ID configuration page consists of the following tabs:

• Basic Settings
• User Settings
• Single Sign-On (SSO)

Each tab is described below.

Basic Settings

This tab defines the connection between the UCM and Microsoft Entra ID.

User Settings

This tab controls how Microsoft users are synchronized to the UCM.

The table below describes the parameters available on this tab

Single Sign-on (SSO)

The SSO tab allows users to log into the UCM Web UI and Wave using Microsoft Entra authentication.

Windows AD

Integrating AD with the UCM allows the system to fetch and manage user data directly from the AD server, making it easier to maintain consistent user information across platforms. Contacts can be imported into the UCM’s LDAP phonebook, and AD users can be linked to UCM extensions or system accounts for unified user administration. The integration also supports automatic and scheduled synchronization, reducing manual workload and keeping user records up to date.

Additionally, enabling Single Sign-On (SSO) allows AD users to log in to the UCM web interface using their domain credentials, enhancing security and simplifying password management.

Google Workspace Integration

The Google Workspace Integration allows the UCM Series to connect with a Google Workspace domain in order to synchronize user identity information and enable Single Sign-On (SSO). When enabled, Google users can be associated with UCM system users or extensions, optionally synchronized into the UCM LDAP directory, and used for authentication when accessing the UCM web interface.

This feature is configured under Integrations→Integration of Collaborative Office Platforms→Google Workspace.

The Google Workspace integration module consists of three tabs:

• Basic Settings
• User Settings
• Single Sign-on (SSO)
• Google Calendar

Basic Settings

The Basic Settings tab is used to configure OAuth authorization between the UCM and Google Workspace. This step is mandatory before any user synchronization or SSO functionality can be enabled.

The table below describes the parameters of this page:

User Settings

The User Settings tab controls how Google Workspace users are retrieved, synchronized, and linked to UCM users or extensions. The settings on this page are grouped into three categories:

1. Google User
2. Linked User Settings
3. Google User Management

The sections below describe each of the Google Workspace user settings tab categories.

Google User

This section defines how Google Workspace users are retrieved from the domain and whether they are synchronized into the UCM system.

Linked User Settings

This section controls how Google Workspace users are associated with UCM users or extensions.

Google User List

This section displays the list of retrieved Google Workspace users and provides controls for synchronization and user linking.

The Sync Settings page is used to define how Google Workspace user data is synchronized with the UCM and how linked extensions are handled during synchronization events. This page is accessed by clicking Sync Settings from the Google User List section.

The Extensions section controls automatic cleanup behavior for UCM extensions based on Google Workspace user status:

• Deleted Google Workspace users: Automatically deletes the linked UCM extension when the corresponding Google Workspace user is deleted.
• Paused Google Workspace users: Automatically deletes the linked UCM extension when the corresponding Google Workspace user is suspended or paused.
• Unlinked Users: Automatically deletes UCM extensions that are no longer linked to any Google Workspace user.

The Google User Data Sync section controls how and when Google Workspace user data is synchronized. When Scheduled Sync is enabled, the UCM automatically retrieves the latest Google Workspace users and updates the association between Google users and UCM users or extensions according to the current configuration.

The Sync History section displays historical records of Google Workspace synchronization activities. The following information is shown for each synchronization event:

• Sync Time: Date and time when the synchronization was executed.
• Sync Mode: Indicates whether the sync was triggered manually or by schedule.
• Results: Displays the synchronization result (Success, Partial Success, or Failed).
• Options: Provides access to detailed synchronization logs for troubleshooting purposes.

Single Sign-On (SSO)

The Single Sign-On (SSO) tab controls Google-based authentication for accessing the UCM web interface.

This page is used to enable or disable Google Workspace SSO and define how Google-authenticated users are mapped to UCM system users.

If SSO is enabled, the Sign in with Google option will appear on the UCM login page.

Google Calendar

The Google Calendar tab allows administrators to synchronize Google Calendar schedules with PBX extension presence statuses, enabling automated call routing policies based on a user’s calendar availability.

The table below describes the parameters available on the Google Calendar tab:

Parameter	Description
Google Calendar Sync	Enables or disables Google Calendar synchronization. When enabled, the schedules of associated users are retrieved and synced to PBX extensions’ presence status to apply dynamic call policies.
Key File	Upload or select the service account JSON key file generated in Google Cloud Console for API authentication.
Scheduled Sync	Configures the periodic sync schedule (e.g., Every Day at a specified time) to fetch calendar updates.
Real-time Update via Webhook	Enables real-time event updates via Google push notifications using a Webhook URL.
WebHook URL	Specifies the Webhook URL used to receive real-time calendar updates. Clicking the Auto Obtain button automatically populates it with the UCMRC domain URL. Note: Using the Webhook URL requires domain verification in Google Console.
Enable Calendar Sync for Extensions	Select extensions from the Available list and move them to the Selected list to enable calendar-based presence sync for specific extensions.

SAML SSO

SAML Single Sign-On (SSO) is an enterprise-grade identity authentication solution that allows the UCM6300 series to integrate with external Identity Providers (IdP) such as Microsoft Entra ID, Okta, OneLogin, or AD FS. Once configured, users can authenticate via their organization’s centralized identity provider to access the UCM Web UI without managing separate local credentials.

This feature is configured under Integrations → Integration of Collaborative Office Platforms → SAML SSO.

SAML SSO Configuration

To enable SAML SSO, configure the Identity Provider credentials, Service Provider details, and user mapping rules under the SAML SSO tab.

The table below describes the parameters available on the SAML SSO page:

Parameter	Description
SAML SSO	Enables or disables SAML Single Sign-On integration on the UCM.
IdP Entity ID	Enter the unique entity identifier (Issuer URL) provided by your Identity Provider. Used as a mechanism to validate SAML responses.
X.509 Cert SHA1 Fingerprint	Enter the SHA1 fingerprint of the IdP’s X.509 signing certificate for response validation.
SSO Login URL	Enter the Single Sign-On service endpoint URL provided by your IdP where UCM will redirect authentication requests.
User Mapping Attribute Name	Specifies the attribute name on the IdP side used for user mapping. The value of this attribute for the IdP user must have a one-to-one correspondence with the UCM user/extension (it is recommended that the value be unique on the IdP side). Example: userName, email.
Reply URL	Use Click to copy to copy the Assertion Consumer Service (ACS) URL of the UCM and paste it into the SAML Service Provider configuration on the IdP side.
Identifier ID	Use Click to copy to copy the SP Entity ID of the UCM and paste it into the SAML Service Provider configuration on the IdP side.

Account Linking

Before users can log in via SAML SSO, their enterprise IdP attribute value must be mapped to a corresponding UCM user account.

To associate accounts:

1. Click Add to open the Account Linking dialog.
2. Select the target UCM User from the drop-down menu.
3. In the IdP User field, enter the exact attribute value (e.g., user email address or username) corresponding to the configured User Mapping Attribute Name.
4. Click OK to save the association.

Logging in with SAML SSO

Once SAML SSO is enabled and user accounts are linked, a Log in with SAML option will appear on the UCM Web UI login page. Clicking this option redirects the user to the configured Identity Provider login page. Upon successful authentication, the user is redirected back to the UCM Web UI and logged in automatically.

SCHEDULED CALL

The Wake Up service can be used to schedule a reminder or wake up calls to any valid destination. This service is available on the UCM630xA as a separated module.

There are three ways to set up Wakeup Service:

• Using admin login
• Using user portal
• Using feature code

Wake Up Service using Admin Login

1. Login to the UCM as admin.
2. Wake Up service can be found under Web GUI🡪Advanced Call Features🡪Scheduled Call, click on ”Add” to create a new scheduled call . The following window will pop up.

1. Fill out the required fields and select the members to add to the wakeup group.

Enable Wakeup Service	Enable Wakeup service.
Name	Enter a name (up to 64 characters) to identify the wakeup service.
Prompt	Select the prompt to play for that extension.
Custom Date	If disabled, users can select a specific date and time.   If enabled users can select multiple days of the week to perform the wakeup.
Date	Select the date or dates when to performs the wakeup call.
Time	Select the time when to play the wakeup call.
Members	Select the members involved within the wakeup service group.

1. 
   Click
   

   

   
   and
   

   

   
   to apply the changes.

A wakeup service entry is created. The UCM will send a wakeup call to every extension in the member list at the scheduled date and time.

Note: the wakeup service has the following limitation on how many members can be added depending on UCM model.

UCM Model	Max members in a Wakeup Service
UCM6300A	50
UCM6302A	100
UCM6304A	150
UCM6308A	200

Wake Up Service from User Portal

1. Login to the user portal on the UCM630xA.
2. Wake Up service can be found under “Other Features🡪Wakeup Service”, click on ”Add” to create a new wakeup service.
3. Configures the Name, Prompt, Date and Time for the user to make the wakeup to.
4. Click
   

   

   
   and
   

   

   
   to apply the changes.

Wake Up Service using Feature Code

1. Login to the UCM as admin.
2. Enable “Wakeup Service” from the WebGUI under Basic Call Features🡪Feature Codes.

1. Click
   

   

   
   and
   

   

   
   to apply the changes.
2. Dial “*36” which is the feature code by default to access to the UCM wakeup service to add, update, activate or deactivate UCM wakeup service.

ANNOUNCEMENT CENTER

The UCM630xA supports Announcement Center feature which allows users to pre-record and store voice message into UCM630xA with a specified code. The users can also create group with specified extensions. When the code and the group number are dialed together in the combination of code + group number, the specified voice message is sent to all group members and only extensions in the group will hear the voice message.

Announcement Center Settings

Group Settings

Group Settings

Announcements Center feature can be found under Web GUI > Advanced Call Features > Announcement Center. The following example demonstrates the usage of this feature.

1. Click
   

   

   
   to add new group.
2. Give a name to the newly created group.
3. Create a group number which is used with code to send voice message.
4. Select the extensions to be included in the group, who will receive the voice message.

In this example, group “Test” has number 666. Extension 1000, 1001 and 1002 are in this group.

1. Click
   

   

   
   to create a new Announcement Center.
2. Give a name to the newly created Announcement Center.
3. Specify the code which will be used with group number to send the voice message to.
4. Select the message that will be used by the code from the Custom Prompt drop down menu. To create a new Prompt, please click “Prompt” link and follow the instructions in that page.

Code and Group number are used together to direct specified message to the target group. All extensions in the group will receive the message. For example, we can send code 55 to group 666 by dialing 55666 from any extension registered to the UCM630xA. All the members in group 666 which are extension 1000, 1001 and 1002 will receive this voice message after they pick up the call.

QUEUEMETRICS

QueueMetrics docking tool provides an interface for UCM system and QM docking. Pass the UCM call queue report to QueueMetrics in a richer form. QueueMetrics is a call center control platform that supports login and logout of frequently used agents in the call center, provides call reports, real-time queue monitoring and other functions.

Queue Metrics configuration parameters

3RD PARTY DATABASE INTEGRATION

The Third-Party Database Integration feature allows the UCM to connect to external databases (such as MySQL) to query contact information when calls are received. When a matching record is found, the caller name is displayed. Optionally, contacts can be synchronized into the UCM LDAP phonebook for centralized management.

For comprehensive details on this integration, its workflows, and configuration steps, please refer to the following guide.

Third-Party SQL Database Integration Guide

This feature is configured under Integrations→3rd Party Database Integration.

The Basic Settings tab is used to configure and enable the database connection. The table below describes the parameters on this page:

After the database connection is successfully established, an additional Contact Synchronization tab becomes available, which enables synchronization of database contacts into the UCM LDAP phonebook.

The configurable parameters on this page are described in the table below:

STATUS AND REPORTING

PBX Status

The UCM630xA monitors the status for Trunks, Extensions, Queues, Meeting Rooms, Interfaces and Parking lot. It presents administrators the real-time status in different sections under Web GUI🡪System Status🡪Dashboard.

The administrator can also use the dashboard to view information about the different services. Such as the number of meetings, the number of extensions (registered and unregistered), Storage Health status, active security modules (Dynamic Defense, Fail2Ban), scheduled system tasks, etc…

Trunks

Users could see all the configured trunk status in this section.

Status	Display trunk status.   Analog trunk status: Available Busy Unavailable Unknown Error SIP Peer trunk status: Unreachable: The hostname cannot be reached. Unmonitored: Heartbeat feature is not turned on to be monitored. Reachable: The hostname can be reached. SIP Register trunk status: Registered Unrecognized Trunk
Trunks	Display trunk name
Type	Display trunk Type:   Analog SIP IAX
Username	Display username for this trunk.
Port/Hostname/IP	Display Port for analog trunk, or Hostname/IP for VoIP (SIP/IAX) trunk.

Extensions

Extensions Status can be seen from the same configuration page, users can go under Web GUI🡪Extension/Trunk🡪Extensions and following page will be displayed listing the extensions and their status information.

Status	Display extension number (including feature code). The color indicator has the following definitions.   Green: Free Blue: Ringing Yellow: In Use Grey: Unavailable
Presence Status	Display the presence status of the extension.
Extension	Display the extension number.
Name	First name and last name of the extension.
IP and Port	Display the IP and port number of the registered device.
Email	Display Email Notification status for the extension.   When notification is waiting for be sent, shows and once sent it will display
Terminal Type	Displays extension type.   SIP User IAX User Analog User Ring Groups Voicemail Groups

Interfaces Status

This section displays interface/port connection status on the UCM630xA. The following example shows the interface status for UCM6304A with USB, WAN port, FXS1, FXS2 and FXO1 connected.

	USB connected.
	USB disconnected.
	SD Card connected.
	SD Card disconnected.
	LAN/WAN connected.
	LAN/WAN not configured.
	LAN/WAN disconnected.
	FXS/FXO connected.
	FXS/FXO waiting.
	FXS/FXO busy.
	FXS/FXO not configured.
	FXS/FXO disconnected.

System Status

The UCM630xA system status can be accessed via Web GUI🡪Status🡪System Status, which displays the following system information.

General

Under Web GUI🡪System Status🡪System Information🡪General, users could check the hardware and software information for the UCM630xA. Please see details in the following table.

System Status 🡪System Information🡪General
Model	Product model.
Part Number	Product part number.
System Time	Current system time. The current system time is also available on the upper right of each web page.
Up Time	System up time since the last reboot.
Boot	Boot version.
Core	Core version.
Base	Base version.
Program	Program version. This is the main software release version.
Recovery	Recovery version.
Lang	Lang version
Wave	Grandstream Wave version

Network

Under Web GUI🡪System Status🡪System Information🡪Network, users could check the network information for the UCM630xA. Please see details in the following table.

System Status🡪System Status🡪Network
MAC Address	Global unique ID of device, in HEX format. The MAC address can be found on the label coming with original box and on the label located on the bottom of the device.
IPv4 Address	IPv4 address.
IPv6 Address Link	IPv6 address
Gateway	Default gateway address.
Subnet Mask	Subnet mask address.
DNS Server	DNS Server address.
Duplex Mode	Duplex Mode
Speed	Speed

Remark

The UCM admin could add remark on UCM web UI->System Status->System Information->Remark to log any necessary information for the UCM such as location, technical contacts, important topology information and etc. This could be useful for UCM admin especially when there are multiple UCMs to be managed.

If this UCM has UCMRC service, the remark will also be sync up to GDMS. If this information is edited on GDMS, it will also be updated to the UCM web UI.

Storage Usage

Users could access the storage usage information from Web GUI🡪System Status🡪Dashboard🡪Storage Usage. It shows the available and used space for Space Usage and Inode Usage.

Space Usage includes:

• Configuration partition

This partition contains PBX system configuration files and service configuration files.

• Data partition

Voicemail, recording files, IVR file, Music on Hold files etc.

• USB disk

USB disk will display if connected.

• SD Card

SD Card will display if connected.

Inode Usage includes:

• Configuration partition
• Data partition

Note:

Inode is the pointer used for file reference in the system. The system usually has limited resources of pointers

Resource Usage

When configuring and managing the UCM630xA, users could access resource usage information to estimate the current usage and allocate the resources accordingly. Under Web GUI 🡪 System Status 🡪 Dashboard 🡪 Resource Usage, the current CPU usage and Memory usage are shown in the pie chart.

System Events

The UCM630xA can monitor important system events, log the alerts, and send Email notifications to the system administrator.

Alert Events List

The system alert events list can be found under Maintenance → System Events → Alert Events List. The UCM630X supports various system event types that generate alerts and/or email notifications when triggered.

Click on the Edit icon next to any event entry to configure its notification parameters. The table below describes the available configuration options for alert events.

Alert Log

Under Web GUI🡪Maintenance🡪System Events🡪Alert Log, system messages from triggered system events are listed as alert logs. The following screenshot shows system reboot alert logs.

User could also filter alert logs by selecting a certain event category, type of alert log, and/or specifying a certain time period. The matching results will be displayed after clicking on

. Alert logs are classified into two types by the system:

1. Generate Alert: Generated when alert events happen, for example, alert logs for disk usage exceeding the alert threshold.
2. Restore to Normal: Generated when alert events being cleared, for example, logs for disk usage dropping back below the alert threshold.

User could filter out alert logs of “Generate Alert” or “Restore to Normal” by specifying the type according to need. The following figure shows an example of filtering out alert logs of type of “Restore to Normal”.

Alert Contact

This feature allows the administrator to be notified when one of the Alert events mentioned above happens. Users could add administrator’s Email address under Web GUI🡪Maintenance🡪System Events🡪Alert Contact to send the alert notification to an email (Up to 10 Email addresses can be added) or also specify an HTTP server where to send this alert.

Super Admin Email	Configure the email addresses to send alert notifications to.   Up to 10 email addresses can be added.
Admin Email	Configure the email addresses to send alert notifications to.   Up to 10 email addresses can be added.
Email Template	Please refer to section Email Templates
Protocol	Protocol used to communicate with the server. HTTP or HTTPS.   Default one is HTTP.
HTTP Server	The IP address or FQDN of the HTTP/HTTPS server.
HTTP Server Port	HTTP/HTTPS port
Warning Template	Customize the template used for system warnings.   By default: {“action”:”${ACTION}”,”mac”:”${MAC}”,”content”:”${WARNING_MSG}”}
Notification Template	Customize the notification template to receive relevant alert information.   By default: {“action”:”${ACTION}”,”cpu”:”${CPU_USED}”,”memery”:”${MEM_USED}”,”disk”:”${DISK_USED}”,”external_disk”:”${EXTERNAL_DISK_USED}”} Note: The notification message with “action:0” will be sent periodically if Notification Interval is set.
Notification Interval	Modifies the frequency at which notifications are sent in seconds.   No notifications will be sent if the value is “0”. Default value: 20
Template Variables	${MAC} : MAC Address   ${WARNING_MSG} : Warning message ${TIME} : Current System Time ${CPU_USED} : CPU Usage ${MEM_USED} : Memory Usage ${ACTION} : Message Type. Refer to [Alert Events] ${DISK_USED} : Disk Usage ${EXTERNAL_DISK_USED} : Disk Usage

Task Management

The Task Management feature allows users to schedule various tasks to be performed at a configured date and time. Each task can be set to run either once or repeatedly, depending on user requirements.

The tasks that can be scheduled include:

• Paging/Intercom: Schedule automated paging or intercom announcements at specific times.
• Backup: Schedule automatic backups of system data to USB, SD card, SFTP server, NAS, or Amazon S3. Users can configure multiple backup tasks, each with its own schedule and storage location.
• Data Sync: Schedule synchronization of system data with remote servers to ensure data consistency.
• Cleaner: Schedule cleaning tasks for system data, such as CDR records, reports, IM messages, and other files, to maintain storage efficiency.
• Detect New Firmware: Schedule automatic checks for new firmware versions.
• Update Country/Region: Schedule updates for system country/region settings, which are used for Geo-IP Access Control.
• Update Time Zone: Schedule updates for the system time zone configuration.

Task List

In the Task List, users can schedule tasks to be performed either once or repeatedly all of the above task types can be scheduled with specific parameters.

To schedule a task, please follow the steps below:

1. Navigate to Maintenance→Task Management→Task List→Pending Task
2. Click on the “Add” button.

3. Select the type of task from the drop-down menu.
4. Click Next and configure the required parameters for the task, such as date, time, frequency, and additional options specific to the task type.

The Pending Tasks section displays all tasks that have been scheduled but not yet executed. Users can review, edit, or delete tasks from this list before execution.

The Task History section logs all tasks that have been performed, including execution time, task type, and status. Users can use this log to verify completed tasks and troubleshoot any issues.

Scheduled Paging/Intercom

The user can configure a paging or intercom to be initiated automatically according to a schedule. Refer the configuration explained in the table below to be able to configure this feature successfully.

Scheduled Backup

Scheduled backup task management allows the user to schedule regular backups with a specific interval. This provides an efficient and reliable archiving process which can prevent any accidental loss of data.

Scheduled Data Sync

Scheduling Data Sync allows the administrator to automate the task of synchronizing the data. When this feature is configured, the UCM will synchronize data to an SFTP server in an incremental way.

Scheduled Cleaner

Scheduled Cleaner allows the user to automate the process of cleaning the files stored in the UCM as well as the attached storage of the UCM.

The following is the list of the types of the cleaner which can be configured for scheduling.

• Scheduled Cleaner (CDR)
• Scheduled Cleaner (Report)
• Scheduled Cleaner (IM Data)
• Scheduled Cleaner (File)

CDR Cleaner

Report Cleaner

IM Data Cleaner

File Cleaner

Scheduled Detect New Firmware

Task scheduling for detecting new firmware allows the user to automate the process of verifying the availability of a new firmware update. Using this feature can ensure that the administrator is notified as soon as there is a new update of the UCM6300 Series software.

Scheduled Update of Country/Region List

Scheduling a regular update of the Country/Region List ensures keeping the list for Geo-IP Access Control up to date with any emergence of new countries or name-change of existing countries. This also allows receiving the update without having to wait for a new firmware release.

Schedule Update Time Zone

Scheduling update for timezones ensures that the list can be updated regularly and without the need to wait for a new firmware release.

Task Schedule

To get an overview about all the tasks which have been scheduled, the user can click on Task Schedule tab to view the full schedule.

CDR

CDR (Call Detail Record) is a data record generated by the PBX that contains attributes specific to a single instance of phone call handled by the PBX. It has several data fields to provide detailed description for the call, such as phone number of the calling party, phone number of the receiving party, start time, call duration, etc.

On the UCM630xA, the CDR can be accessed under Web GUI🡪CDR🡪CDR. Users could filter the call report by specifying the date range and criteria, depending on how the users would like to include the logs to the report. Click on “Filter” button to display the generated report.

Administrators and authorized users can categorize call records by attaching pre-configured labels or adding custom comments by clicking the Edit icon under Options to open the Edit Call Note window.

The dialog allows selecting a pre-configured tag from the drop-down menu (configured under CDR → Call Note) or entering a custom note regarding the call (up to 512 characters; special characters $ & | : > ` are restricted).

Clicking the recording icon on a CDR entry opens the Recordings pop-up modal, allowing users to preview, manage, or transcribe call recording files. The pop-up dialog provides interactive operations including:

• Playback (using the built-in audio player controls to play, pause, or seek)
• Download (saving the recording file in .wav format to the local computer)
• AI Recording Transcription (initiating AI speech-to-text transcription after selecting the desired language)
• Delete (deleting the recording file without removing the CDR call entry).

Transferred calls are represented in the CDR log as consolidated entries. Accessing CDR entries under Web GUI → CDR → CDR allows administrators to review full details for individual call sessions.

Click the Option icon next to a specific call record entry to view detailed call flow information, including primary caller IDs, call transfer steps, and timing details.

CDR Automatic Download

The UCM630x Audio series can be configured to automatically export and transmit CDR records at scheduled intervals. Administrators can deliver CDR files via email or automatically sync them to Google Sheets by navigating to CDR → CDR and clicking on Automatic Download Settings.

The table below describes the parameters available in the Automatic Download configuration dialog:

Downloaded CDR File

The downloaded CDR (.csv file) has different format from the Web GUI CDR. Here are some descriptions.

• Caller number, Callee number

“Caller number”: the caller ID.

“Callee number”: the callee ID.

If the “Source Channel” contains “DAHDI”, this means the call is from FXO/PSTN line.

• Context

There are different context values that might show up in the downloaded CDR file. The actual value can vary case by case. Here are some sample values and their descriptions.

from-internal: internal extension makes outbound calls.

ext-did-XXXXX: inbound calls. It starts with “ext-did”, and “XXXXX” content varies case by case, which also relate to the order when the trunk is created.

ext-local: internal calls between local extensions.

• Source Channel, Dest Channel

Sample 1:

• DAHDI means it is an analog call, FXO or FXS.
• For UCM6302A, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports.
• For UCM6304A, DAHDI/(1-4) are FXO ports, and DAHDI(5-6) are FXS ports.
• For UCM6308A, DAHDI/(1-8) are FXO ports, and DAHDI(9-10) are FXS ports.

Sample 2:

• “SIP” means it is a SIP call. There are three format:
• (a) PJSIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string and can be ignored.
• (c) PJSIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out through the trunk.
• (c) PJSIP/trunk_X-XXXXXX, where trunk_X is the internal trunk name and it is an inbound call from this trunk. The last XXXXX is a random string and can be ignored.

There are some other values, but these values are the application name which are used by the dialplan.

IAX2/NUM-XXXXXXX: it means this is an IAX call.

Local/@from-internal-XXXXX: it is used internally to do some special feature procedure. We can simply ignore it.

Hangup: the call is hung up from the dialplan. This indicates there are some errors or it has run into abnormal cases.

Playback: play some prompts to you, such as 183 response or run into an IVR.

ReadExten: collect numbers from user. It may occur when you input PIN codes or run into DISA

Note: The language of column titles in exported CDR reports and statistics reports will be based on the UCM’s display language

CDR Export Customization

Users can select the data they want to see in exported CDR reports by first clicking on the Filter button on the CDR page under CDR🡪CDR and selecting the desired information in the Export File Data field.

CDR in GDMS Cloud

Cloud Storage for CDR Record which can be displayed under CDR 🡪 CDR in GDMS Cloud.

Call Note

Under CDR → Call Note, administrators can enable the Call Note feature and define customized call labels (tags) that users can attach to call detail records and call logs across the User Portal and Wave application.

The table below describes the parameters available on the Call Note configuration page:

To create a new call label tag, click the Add button to open the Add Call Label window.

Configure the following fields in the Add Call Label dialog:

• Label: Enter a unique name for the call tag (1–128 characters; supports numbers, letters, special characters _ -, and Chinese characters).
• Description: Enter an optional summary or explanation for the label (0–128 characters; special characters $ & | : > ` are restricted).

Call Rate

In Call Rate, the user can set call billing for outbound calls. The user can choose to include the call rate onfigured in Least Call Rates or configure rates for calls which are identified through a prefix.

To add specific rates, click on “Add Rate”.

Statistics

The UCM provides a comprehensive CDR analytics framework for monitoring, analyzing, and reporting voice traffic across the system. This module is designed to give administrators operational visibility into call patterns, resource utilization, user behavior, and cost distribution.

CDR Statistics

UCM supports the function of concurrent call statistics. This function provides users with statistics on the number of concurrent calls of all VoIP trunks (SIP trunks and IAX trunks). Users can set search criteria to generate custom charts. Select the trunk and time to view the chart of the maximum number of concurrent calls corresponding to the trunk in a certain day or month.

Concurrent Call Statistics

In this tab, the administrator can view the statistics regarding the concurrent calls. The administrator can choose to view the statistics per day, month, and year. The user can also view all calls, inbound calls, or outbound calls; or a combination of all of them together.

Call Charges Statistics

In call Charges Statistics tab, the administrator can view the charges applied to the calls represented in a graph. The administrator can search by time interval, they can also filter per extension, extension group, by department. Once the administrator sets the intended parameters in the filter, they can display and download the filtered statistics.

Call Reports

The Call Reports section provides analytics based on different operations within the PBX. These reports allow administrators to analyze how calls are distributed, handled, and processed across extensions, trunks, and system components.

Extension Call Statistics

This report displays insights into call volume associated with selected:

• Extensions
• Extension groups
• Departments

Administrators can filter by time period and call type (inbound, outbound, internal, etc.) to analyze how traffic is distributed across users or organizational units.

Extension Call Activity

This report focuses on how calls and selected call types were handled by the specified extensions/extension groups/departments over the specified time periods.

PBX Call Activity

This report provides a system-wide view of call activity across selected trunks, which include or exclude internal calls (extension-to-extension) and can be filtered by call direction.

This enables comparison between internal PBX traffic and external trunk usage, helping identify overall system load distribution and trunk utilization patterns.

Outbound Caller ID Activity

This report provides visibility into outbound call usage based on the selected trunk and caller ID configuration.

It displays how outbound calls are distributed across:

• Trunks
• Caller IDs

IVR Report

This option provides analysis of IVR interactions over a selected time period and supports two reporting modes:

• Standard Mode: Captures key press events (0–9, *, #), including invalid inputs and timeout events, providing detailed per-key interaction statistics.
• Custom Mode: Supports multi-digit or combined key inputs. Reports aggregate IVR activity, including total call entries, invalid inputs, and timeouts.

The report can be exported in two formats:

• Basic List: Lists all IVRs and the trigger counts of their key press events.
• Detailed List: Includes the caller information, triggered key press events, the time of the events, and how long the callers were in the IVRs. In addition to the information in the Basic report

Ring Group Statistics

This report provides performance metrics for ring groups, including total calls received, answered calls, and answer rate.

It enables evaluation of call distribution efficiency and group responsiveness across configured ring groups.

Call Queue Reports

The Call Queue Reports section provides comprehensive, dedicated analytics for call queue performance, agent activities, and queue response metrics across the UCM. Administrators can use these reports to analyze call flow efficiency, monitor agent productivity, and track queue responsiveness over selected time periods.

To access these reports, please navigate to CDR → Statistics → Call Queue Report and select a report type from the drop-down menu to view its detailed metrics.

Agent Call Summary

Measures individual agent call statistics throughout a specified date range and queue. It displays total calls handled, answered volume, total talk time, average wait time, average talk time, and average handling time per agent.

Agent Login Report

Lists individual agent login activity throughout a selected date range for a specific call queue, allowing administrators to track attendance, active duration, and login/logout timestamps.

Agent Unanswered Report

Displays individual agent unanswered calls throughout a date range. It provides visibility into the number of missed calls, attempts, and caller number.

Agent Pause Report

Lists individual agent pausing activity throughout a date range, tracking pause durations, reasons, and total offline time during operational hours.

Agent Performance

Measures individual agent performance listed and grouped directly under their respective queues throughout a date range, enabling direct performance comparisons within specific teams.

Queue Average Wait Time & Talk Time

Displays a graph showing caller wait time and talk time hourly, daily, or monthly for selected queues, averaging the data across the chosen time frame to help optimize queue staffing levels.

Queue Callback Report

Lists individual queue callback activity throughout a date range and specified queue, detailing requested callbacks, execution timestamps, and call results.

Queue Callback Summary

Displays a graphical representation of queue callback statistics over a specified date range. It provides visual insights into overall callback performance, comparing successful connections against failed callback attempts across the selected queues.

Queue Performance

Measures individual queue performance throughout a date range, focusing on total wait duration, average handling time, and overall service levels per queue.

Queue Performance Activity

Provides a time-based breakdown (hourly, daily, or monthly) of call queue activity. Administrators can use this report to evaluate traffic distribution, monitor answer rates and average wait times, and track SLA performance over a selected period.

Agent Activity Report

Provides a comprehensive performance breakdown for each agent across selected queues over a specified date range. It details total call volume, answer rates, unanswered calls, wait times, talk duration, and average handling times per agent.

Queue Record Report

Lists every individual call made to the selected queues throughout a specified date range. It details the caller number, queue status, number of call attempts, answering agent, call timestamp, total wait duration, and total talk time.

Agent Inbound Queue Record Report

Displays a granular call log for each agent within selected queues over a date range, tracking individual interactions, ring duration, talk time, wait time, and whether the agent answered or missed the call.

Queue Analysis Report

Sorts and measures call statistics on an hourly, daily, or monthly basis, displaying a dedicated row for each individual queue during every time interval. This provides per-queue trends, unlike Queue Performance Activity, which summarizes total system traffic.

Queue Statistics Report

Generates a satisfaction breakdown for selected queues within a date range, categorizing caller rating scores into levels 1 to 5 alongside counts for non-surveyed calls.

Scheduled Report

The Scheduled Report tab allows administrators to create automated report delivery schedules and have reports sent via email at regular intervals.

To configure an automated report schedule:

1. Navigate to CDR → Statistics → Scheduled Report.
2. Click Add to open the creation window (or click Email Template to customize the notification layout).

3. Under Report Data, select the Report Type, Time frame, target Queues, Agents, and customize the List Fields to include in the output.

4. Under Schedule Information, enter the schedule Name, recipient Email, preferred Export Type (XLSX, CSV, PDF), and execution recurrence pattern (Repeat interval and time).

Once configured, all active schedules are listed on this page for quick editing or management.

Schedule Records

The Schedule Records tab provides full historical tracking and audit logging for all automated email report transmissions. On this page users can:

• Review historical entries including the schedule Name, transmission Send Status (e.g., Sent successfully), and exact Send Time.
• Click the Info icon (i) under the Options column to preview the sent report in a different tab.
• Select individual or multiple log entries to delete historical records as needed.

Recordings

This page lists all the recording files recorded by “Auto Record” per extension/ring group/call queue/trunk, or via feature code “Audio Mix Record”. If external storage device is plugged in, for example, SD card or USB drive, the files are stored on the external storage. Otherwise, internal storage will be used on the UCM630xA.

• Click on “Download” to batch-download the selected recording files.
• Click on “Download All” to download all the recording files.
• Click on “Delete” to batch-delete the selected recording files.
• Click on “Clear” to delete all the recording files.
• Click on “Scan” to retrieve the file information and display all the recording files on external storage. The UCM automatically retrieves the info of the first 5000 files from external storage already. This button can be used when the number of files stored on the external storage exceeds 5000 files and it requires manual file scanning.
• Select either “USB Disk” or “Local” to show recording files stored on external or internal storage, depending on the selected storage space.
• Select whether to show call recordings, queue recordings or conference recordings.
• Click on

  

  to download the recording file in .wav format.
• Click on

  

  to delete the recording file.
• To sort the recording file, click on the title “Caller”, “Callee” or “Call Time” for the corresponding column. Click on the title again can switch the sorting mode between ascending order or descending order.

USER PORTAL

Users could log into their web GUI portal using the extension number and user password. When an extension is created in the UCM630xA, the corresponding user account for the extension is automatically created. The user portal allows access to a variety of features which include user information, extension configuration and CDR as well as settings and managing other features like Call Queue, Wakeup Service and CRM.

Users also can access their personal data files (call recordings, Voicemail Prompts …).

The login credentials are configured by Super Admin. The following figure shows the dialog of editing the account information by Super Admin. The Username must be the extension number and it is not configurable, and the password is set on “User Password” field and it should not be confused with the SIP extension password.

The following screenshot shows an example of login page using extension number 1000 as the username.

After login, the Web GUI display is shown as below.

After successful login, the user has the following three configuration tabs:

Basic Information

Under this menu, the user can configure and change his/her personal information including (first name, last name, password, email address, department…). And they can also set and activate their extension features (presence status, call forward, DND ….) to be reflected on the UCM.

Also, the user can see from this menu the Call Details Records and search for specific ones along with the possibility to download the records on CSV format for later usage.

Personal Data

Under this section, the user can access and manage their personal data files which includes (voicemail files, call recordings …) along with the possibility to set Follow me feature to without requesting the Super admin to set the feature from admin account.

Other Features

On this section, the user has access to manage and use all rich features which includes.

+ If user is a member of call queue, they can check the queue’s activity from the “Call Queue” section.

+ Create and enable Wake Up service.

+ Enable and configure CRM connection to either SugarCRM or Salesforce.

For the configuration parameter information in each page, please refer to [User Management🡪Create New User] for options in User Portal🡪Basic Information🡪User Information page; please refer to [EXTENSIONS] for options in User Portal🡪Basic Information🡪Extension page; please refer to [CDR] for User Portal🡪Basic Information🡪CDR page.

MAINTENANCE

User Management

User management is on Web GUI🡪Maintenance🡪User Management page. User could create multiple accounts for different administrators to log in the UCM630xA Web GUI. Additionally, the system will automatically create user accounts along with creating new extensions for extension users to login to the Web GUI using their extension number and password. All existing user accounts for Web GUI login will be displayed on User Management page as shown in the following figure.

User Information

When logged in as Super Admin, click on ”Add” to create a new account for Web GUI user. The following dialog will prompt. Configure the parameters as shown in below table.

Username	Configure a username to identify the user which will be required in Web GUI login. Letters, digits, and underscore are allowed in the username.
User Password	Configure a password for this user which will be required in Web GUI login. English input is allowed without space,’ and “.
Privilege	This is the role of the Web GUI user. When super admin creates new user, “Administrator” or customized privilege can be selected.
Multi-Factor Authentication	If this authentication is enabled, the user account needs to be verified with an MFA code every time it logs in to enhance the security of the product.
Email Address	Configure the email address for the user. This is optional.

Once created, the Super Admin can edit the users by clicking on

or delete the user by clicking on

.

Multi-Factor Authentication

To enhance the security for UCM, super admin and admin can select to use multi-factor authentication method for login to protect the login information. Virtual MFA and hardware MFA are supported and can be selected. Once enabled, the user must use MFA code to verify before login.

Notes:

• The user cannot enable or disable MFA for another different user.
• Super admin can edit user settings for admin but cannot edit Multi-Factor Authentication option. MFA option is only viewable for super admin when super admin edits other users.
• When the user sees MFA enabled, only this user can disable or enable it again.
• Email address and email settings are required before enabling Multi-Factor Authentication. Please ensure email setting has “Client” type configured. Otherwise, MFA cannot be enabled.

Please refer to MFA how to guide here for more information.

Custom Privilege

By default, three levels are supported: Super Administrator, Administrator, Wave Administrator:

Super Administrator

• This is the highest privilege. Super Admin can access all pages on UCM630xA Web GUI, change configuration for all options and execute all the operations.
• Super Admin can create, edit, and delete one or more users with the “Admin” privilege
• Super Admin can edit and delete one or more users with the “Consumer” privilege
• Super Admin can view operation logs generated by all users.
• By default, the user account “admin” is configured with the “Super Admin” privilege and it is the only user with the “Super Admin” privilege. The Username and Privilege level cannot be changed or deleted.
• Super Admin could change its own login password on Maintenance→Login Settings page.
• Super Admin could view operations done by all the users in Maintenance→User Management→Operation Log
• The Super Admin can allow administrators to view SFTP, NAS and Email Account passwords.

Administrator

• Users with “Admin” privilege can only be created by “Super Admin” user.
• “Admin” privilege users cannot create new users for login.
• “Admin” privilege users are by default not allowed to access the following pages:

Maintenance 🡪 Backup

Maintenance 🡪 Upgrade

RemoteConnect

Settings🡪User Management🡪Operation Log

Note: By default, administrator accounts are not allowed to access backup menu, but this can be assigned to them by editing the option “Maintenance→User Management→Custom Privilege” then press the “Edit” icon to edit the “Admin” account and include backup operation permission for these types of users.

Wave Administrator

• This permission type does not support editing or deletion.
• This includes management of Wave-related function settings only and does not involve access to the UCM module.
• Users can set the Wave Admin privilege for specific extension under Extension/Trunk→Extensions→Edit Extension.

For more information, please refer to the Wave Administrator Guide guide.

Add Custom Privilege

To create a custom privilege role, click Add and select the accessible modules from the table below:

Module	Custom Privileges
System Status	System Status
Extension/Trunk	Extensions, Extension Groups, Outbound Routes, Inbound Routes
Basic Call Features	Meeting, IVR, Voicemail, Ring Groups, Paging/Intercom, Call Queue, Queue Recordings, Queue Statistics, Queue Board, Queue Operation Panel, Speed Dial, Event List, Feature Codes, Parking Lot
Advanced Call Features	Pickup Groups, Dial By Name, DISA, Callback, Scheduled Call, Fax Sending, Fax/T.38, Announcement, SCA, Emergency Calls, Restrict Calls
PBX Settings	Music on Hold, Voice Prompt
System Settings	Time Settings, LDAP Server
Contacts	Contacts
Device Management	Zero Config
Maintenance	System Events, Backup
CDR	CDR Recordings, CDR Records, Call Note, CDR Statistics, Call Rate
RemoteConnect	UCM RemoteConnect
Integrations	AI Settings, API Configuration, PMS – Basic Settings, PMS – Room Management, PMS – Check-in Management, PMS – Wakeup Service, PMS – Mini Bar, PMS – Housekeeper

When enabling specific privileges such as CDR Records or CDR Recordings, granular sub-options will appear (such as Allow Deletion of CDR and Allow Deletion of CDR Recordings), providing precise control over data retention permissions.

To assign custom privilege to a sub-admin, navigate to UCM Web GUI→Maintenance→User Management→User Information→Create New User/Edit Users, select the custom privilege from “Privilege” option.

Concurrent Multi-User Login

When there are multiple Web GUI users created, concurrent multi-user login is supported on the UCM630xA. Multiple users could edit options and have configurations take effect simultaneously. However, if different users are editing the same option or making the same operation (by clicking on “Apply Changes”), a prompt will pop up as shown in the following figure.

User Portal/Wave Privileges

The administrator can create customized privileges related to an extension’s User Portal and Wave. The created privilege can be affected to the extensions to limit or allow them to use certain functionalities related to Wave and the User Portal.

After the creation of the privilege, the administrator can assign the privilege to a specific extensions by navigating to Extension/Trunk > Extensions > Basic Settings > User Portal/Wave Privileges.

Wave Permissions

Under Wave Permissions category, the user can configure the various permissions for the basic features of Wave such as using the chat, making audio/video calls, meeting, use application store, etc…

• Chat: Toggles ability to use the Wave Chat feature.
  • Delete Chat Session: Toggles support for Wave to delete chats and chat history. This data will only be deleted on the Wave client side.
  • Delete Message: Controls whether Wave users will be allowed to delete messages that have been sent. Deleted messages will not be visible to anyone.
  • Send File: Toggles file/image sending support in Wave chat. If disabled, users will still be able to download, view and forward chat files.
  • Download Chat Logs: If enabled, chat logs will be downloadable from the Wave client, including chat logs from Wave/WhatsApp/Telegram/LiveChat sessions.
• End-to-End Encrypted Chat*: Toggles ability to use the Wave End-to-End Encrypted Chat feature.
• Wave App Screenshot and Screen Recording: Controls whether users are allowed to take screenshots and perform screen recordings in the Wave mobile app.
• Meeting: Toggles ability to use the Wave Meeting feature.
• Online Status: Toggles ability to set Wave online status such as “Busy”, “Appear Away”, “Do Not Disturb”, “Appear Offline”, etc. If unchecked, the status will be displayed as only either “Online” or “Idle”.
• Login Settings
  • Multi-client Login Settings: If enabled, users will be allowed to enable or disable the option “Allow Simultaneous Logins from the Same Client Type”.
• Remote Logout: If enabled, Wave users will be able to log out of their accounts from other logged-in devices.
• Recording Status: Controls whether the recording status is displayed in all calls/conferences of the extensions. Note: If the call recording prompt is enabled, it will be played as this setting affects the indicator that shows that a call is being recorded on Wave.
• Clear Recent Call History: Toggles ability to delete recent call history entries and entire recent call history on Wave.
• Application: Toggles ability to access the “Applications” page under Wave Desktop and Wave Web.
• Smart Devices: Toggling off privileges will hide the corresponding pages and options in Wave.
  • Door System
  • Monitor
  • Call Device (CTI)
• 3rd Party Applications
  • App Store: Toggles ability to install applications from Wave application store. If unchecked, the application store will be hidden, but installed apps can still be used.

User Portal/Wave Privileges

Under User Portal/Wave Privileges, the administrator can configure access privileges to specific settings and features, such as adjusting account settings, changing SIP/IAX passwords and AuthIDs, setting DND, or managing personal data (e.g., CDRs, Follow Me, Voicemail, recordings, etc.).

The table below describes the privilege options available under the User Portal/Wave Privileges tab:

*: Features which are marked by an asterisk are a part of RemoteConnect plans.

Administration Privileges

Under Administrator Privileges, the administrator can configure enabling certain extensions to view a designated team’s CDR.

• View Team CDRs and Recordings: Define whether certain extensions can view each other’s CDR. Click on “Team” to designate the extensions.
  • Download Team Recording: When this option is enabled, the extensions selected in the team will be allowed to access each other’s CDR and call recording files.

Pre-intalled Plugins

The administrator can set the plug-ins which will be installed when the user logs into Wave Desktop.

To configure the pre-installed plugins, follow the steps below.

1. Navigate to Maintenance > User Management > User Portal/Wave Privileges then click “Pre-installed Plug-ins”.

2. Click on “Add Add-in” button.

Extension Login Management

Extension Login Management allows the administrator to review the logged-in sessions of SIP devices and Wave.

For Wave sessions, the administrator can click on to terminate a Wave session. SIP sessions cannot be logged out.

User Endpoint Access History

The User Endpoint Access History tab allows the administrator to view the access history of all extensions, the time on which the access has occurred, the IP addresses from which the extensions were accessed, and whether they were accessed from the User Portal, Wave Web/Desktop, or mobile. Extension access from the SIP endpoints won’t be logged in this page.

Login Settings

Change Password

After logging in the UCM630xA Web GUI for the first time, it is highly recommended for users to change the default password to a more complicated password for security purpose. Follow the steps below to change the Web GUI access password.

1. Go to Web GUI🡪Maintenance🡪Login Settings🡪Change Password / Email page.
2. Enter the old password first.
3. Enter the new password and re-type the new password to confirm. The new password has to be at least 4 characters. The maximum length of the password is 30 characters.
4. Configure the Email Address that is used when login credential is lost.
5. Click on “Save” and the user will be automatically logged out.
6. Once the web page comes back to the login page again, enter the username “admin” and the new password to login.

Enter Old Password	Enter the Old Password for UCM630xA
Change Password	Enable Change Password
Enter New Password	Enter the New Password for UCM630xA
Re-enter New Password	Retype the New Password for UCM630xA
Change Username	Enable Change Username
Please enter the username	Enter the Username
Email Address	The Email address is the User Email Address. It is used for receiving password information if the user forgets his password.

Change Username

UCM630xA allows users now to change Super Administrator username.

Change binding Email

UCM630xA allows user to configure binding email in case login password is lost. UCM630xA login credential will be sent to the designated email address. The feature can be found under Web GUI🡪 Maintenance🡪Login Settings🡪Change Password / Email

Email Address

Email Address is used to retrieve password when password is lost

Login Security

After the user logs in the UCM630X Web GUI, the user will be automatically logged out after certain timeout, or he/she can be banned for a specific period if the login timeout is exceeded. Those values can be specified under UCM630X web GUI🡪Maintenance🡪Login Settings🡪Login Security page.

The “User Login Timeout” value is in minute and the default setting is 10 minutes. If the user does not make any operation on Web GUI within the timeout, the user will be logged out automatically. After that, the Web GUI will be redirected to the login page and the user will need to enter username and password to log in.

Remote Login

This feature allows the user to manage trusted login locations, also, verifying where login sessions were initiated from, this is very important since, in this type of scenario, the UCM6300 would be directly connected to the Internet, and the public IP address would be used for the remote login. This feature adds a layer of visibility and control, thus enhacing the security of the UCM.

In this tab there are two types of lists of locations:

• Trusted Login Locations: These are the trusted login locations that are added manually by the admin. Any added trusted login location will not generate any remote login alert upon the first time login.
• Other Login Location: This list will show all the remote login locations that are not trusted, logging in for the first time from an untrusted login location will generate an alert, but the subsequent remote logins from the same location will not generate alerts.

To add a trusted login location, the user must click on

Then add the public IP address of the location, click on “Check Location” to verify if it’s the correct location then click “OK“.

Operation Log

Super Admin has the authority to view operation logs on UCM630xA Web GUI🡪Settings🡪User Management🡪Operation Log page. Operation logs list operations done by all the Web GUI users, for example, Web GUI login, creating trunk, creating outbound rule and etc. There are 7 columns to record the operation details “Date”, “Username”, “IP Address”, “Results”, “Page Operation”, “Specific Operation” and “Remark”.

The operation log can be sorted and filtered for easy access. Click on

or

at the top of each column to sort. For example, clicking on

for “Date” will sort the logs according to newer operation date and time. Clicking on

for “Date” will reverse the order.

Operation Log Column Header

User could also filter the operation logs by time condition, IP address and/or username. Configure these conditions and then click on ”Display Filter”.

The above figure shows an example that operations made by user “support” on device with IP 192.168.40.173 from 2014-11-01 00:00 to 2014-11-06 15:38 are filtered out and displayed.

To delete operation logs, users can perform filtering first and then click on

to delete the filtered result of operation logs. Or users can click on

to delete all operation logs at once.

Upgrade

During its lifetime, the UCM6300 Audio Series models will receive regular updates. These updates include new features and functionalities, bug fixes, security patches, and general improvements to the security and performance of the device. It is highly recommended that you keep the firmware of your UCM up to date.

Before proceeding with the firmware upgrade, please visit https://www.grandstream.com/support/firmware and download the latest firmware available for the UCM. Once the file has been download, please decompress the file, you will get a .bin file, that’s the file which will uploaded to the UCM.

To upgrade the firmware of the UCM6300 Series, please access the web UI of the UCM, then navigate to Maintenance > Upgrade

On this page, the user can view the firmware version which is installed on the UCM. Clicking “Detect” will initiate firmware version verification, if there is a newer version, a prompt will appear to confirm downloading and installing the firmware.

The user can also schedule detection of the firmware version by clicking on “Schedule Task”. For more information regarding scheduling tasks, please refer to [Task Management] section.

1. Click on “Choose File to Upload” field in front of “System Firmware (.bin)”
2. Use the window to navigate to where the firmware file is located then select it.
3. Once the firmware file has been selected, the file will be uploaded to the UCM and the upgrade will begin.
4. A prompt will appear to reboot the device to finalize the upgrade.

Backup

The UCM630xA configuration can be backed up locally or via network. The backup file will be used to restore the configuration on UCM630xA when necessary.

Backup/Restore

Click on the “Upload” to upload backup file from the local device to UCM630xA. The uploaded backup file will also be displayed in the web page and can be used to restore the UCM630xA.

The Scheduled Backup feature enables UCM to automatically back up selected data according to defined schedules. Users can configure multiple backup tasks, each with its own schedule and storage destination, while ensuring that no two backups run simultaneously. Supported storage options include USB, SD card, SFTP server, NAS, and Amazon S3 storage.

To configure a scheduled backup, please refer to the steps below:

1. Access the Schedule Backup feature either by clicking the “Schedule Backup” button on the Maintenance→Backup page or by creating a new task in the Maintenance→Task Management page.

2. Check the Schedule Backup option checkbox to activate backup scheduling.
3. Enter a name for the task (should be between 2 and 64 characters. Names can include alphanumeric characters, Chinese characters, spaces, and the special characters _ and –).
4. Select the desired storage destination, such as USB, SD card, SFTP server, NAS, or Amazon S3. The same location can be used for multiple tasks as long as each task is scheduled at a different time.
5. Specify the date and time for the backup. Scheduled backup times must be at least one hour apart.

6. Choose how often the backup should occur: Every Day, Weekly, Monthly, or Custom.
7. Incremental Backup (Optional): When a supported target storage device (USB, SD Card, or NAS) is selected, users can enable the Incremental Backup option. When enabled, the system automatically backs up data added or modified during the previous cycle (specifically CDR Records, Recording Files and Queue Statistics Reports) rather than creating a full system copy.

   • Example: For a monthly incremental backup task running on the 28th, the backup data range will cover from the 28th of the previous month (00:00:00) to the 27th of the current month (23:59:59). If a scheduled day (e.g., 29th, 30th, or 31st) does not exist in a given month, the task automatically executes on the last day of that month.

8. Choose which data to include in the backup. Options include: All, Config File, IM Message, IM Files, CDR Records, Recording Files, Fax Files, Voicemail, Voice Prompt Files, Queue Statistics Report, ZeroConfig Storage, Meeting Report, and GDS Manager Data.
9. Users can view both scheduled backup tasks and completed backup tasks in the Maintenance→Task Management page. Within this page, the Task List displays all the pending and completed tasks, while the Task Schedule shows the timing and recurrence of upcoming backups.

Data Sync

Besides local backup, users could backup the voice records/voice mails/CDR in a daily basis to a remote server via SFTP protocol automatically under Web GUI🡪Maintenance🡪Backup🡪Data Sync.

The client account supports special characters such as @ or “.” Allowing the use email address as SFTP accounts. It allows users as well to specify the destination directory on SFTP server for backup file. If the directory does not exist on the destination, UCM630xA will create the directory automatically

Data Sync Configuration

Enable Data Sync	Enable the auto data sync function. The default setting is “No”.
Account	Enter the Account name on the SFTP backup server.
Password	Enter the Password associate with the Account on the SFTP backup server.
Server Address	Enter the SFTP server address.
Destination Directory	Specify the directory in SFTP server to keep the backup file. Format: ‘xxx/xxx/xxx’, If this directory does not exist, UCM will create this directory automatically.
Sync Time	Enter 0-23 to specify the backup hour of the day.

Before saving the configuration, users could click on

. The UCM630xA will then try connecting the server to make sure the server is up and accessible for the UCM630xA. Save the changes and all the backup logs will be listed on the web page. After data sync is configured, users could also manually synchronize all data by clicking on

instead of waiting for the backup time interval to come.

Restore Configuration from Backup File

To restore the configuration on the UCM630xA from a backup file, users could go to Web GUI 🡪 Maintenance 🡪 Backup 🡪 Backup/Restore.

• A list of previous configuration backups is displayed on the web page. Users could click on
  

  

  
  of the desired backup file and it will be restored to the UCM630xA.
• If the backup was stored on GDMS, it will be displayed under Backups GDMS Cloud Storage, that can be restored by clicking on
  

  

• If users have other backup files on PC to restore on the UCM630xA, click on “Upload Backup File” first and select it from local PC to upload on the UCM630xA. Once the uploading is done, this backup file will be displayed in the list of previous configuration backups for restore purpose. Click on
  

  

  
  to restore from the backup file.

System Cleanup/Reset

Reset and Reboot

Users could perform reset and reboot under Web GUI🡪Maintenance🡪System Cleanup/Reset🡪Reset and Reboot.

• To reboot the device, click on reboot icon.
• To factory reset the device, click on reset icon, then all the configurations and data will be reset to factory default.

• User can also verify UCM certificate under the same path.

Cleaner

Users can configure the system to clean Call Detail Records (CDRs), voice recordings, voicemails, reports, chat data, meeting history, fax files, backup files, and other stored data either manually or automatically. The Cleaner feature is available under Maintenance→System Cleanup/Reset→Cleaner.

UCM provides the ability to manually clean a variety of data types, including CDRs, report data, chat messages and shared files, recording files, meeting history records, voicemails, fax files, and backup files.

The following screenshot illustrates the settings and parameters available for configuring the Manual Cleaning feature on UCM630xA.

Automatic cleanup is not enabled by default. When configured, automatic cleanup supports scheduled deletion of file data based on parameters such as cleanup time, cleanup conditions, and cleanup intervals.

Users can configure Automatic Cleaning by navigating to Maintenance→System Cleanup/Reset→Cleaner→Automatic Cleaning, or by creating an automated cleaning task under Maintenance→Task Management.

The following screenshot shows some available parameters for configuring automatic cleanup on UCM630xA.

The table below describes all the configurable options that can be configured for automatic cleaning:

When automatic cleaning tasks run, the system logs the cleaning results, allowing users to review each task results, including the total size cleaned and number of items removed, by checking either the Cleaner page or the Task Management sections (Task History).

USB/SD Card Files Cleanup

Users could configure to clean or download the Call Detail Report/Voice Records/Voice Mails automatically under Web GUI🡪Maintenance🡪System Cleanup/Reset🡪USB / SD Card Files Cleanup.

USB/SD Card Files Cleanup

Current Path	Displays the current path.
Directory	Select the directory user want to clean.
Delete Selected File	Select multiple entries to delete from USB or SD card.

System Recovery

In some cases (for example after wrong upgrading procedure where the user doesn’t follow the correct steps to perform an upgrade) the system may go into some hardware/software issues where the web UI access is lost as well as SSH, in this case the only solution would be to perform a full system recovery in order to reset or update the software version of the device in order to use it again.

1. To access recovery mode on UCM, please follow below steps:
2. Remove the power from the unit and keep the network cable connected.
3. Press using a PIN the reset button and keep holding.
4. Plug back the power supply while maintaining the reset button pressed.
5. Wait for couple of seconds until you hear a click sound.
6. Release the reset button, and the system should display on the LCD a message “Recovery Mode” along with an IP address.

Once at this stage, the administrator can access the recovery mode web portal by typing in either the IP0 address (typically WAN) or IP1 address (typically LAN) into a browser address bar. The following page should appear:

Make sure to enter the correct admin password, and press login to access the recovery mode page :

From here, the user can either upload a firmware file, factory reset or just reboot the device.

Syslog

On the UCM630xA, users could dump the syslog information to a remote server under Web GUI🡪Maintenance🡪Syslog. Enter the syslog server hostname or IP address and select the module/level for the syslog information as well as Process Log Level.

The default syslog level for all modules is “error”, which is recommended in your UCM630xA settings because it can be helpful to locate the issues when errors happen.

Some typical modules for UCM630xA functions are as follows and users can turn on “NOTICE” and “VERBOSE” levels besides “error” level.

• pbx: This module is related to general PBX functions.
• pjsip: This module is related to SIP calls.
• chan_dahdi: This module is related to analog calls (FXO/FXS).

Network Troubleshooting

On the UCM630xA, users could capture traces, ping remote host and traceroute remote host for troubleshooting purpose under Web GUI🡪Maintenance🡪Network Troubleshooting.

The following sections shows the steps to capture different types of traffic traces for analysis purposes.

Ethernet Capture

Ethernet Capture allows capturing the traffic of the UCM for troubleshooting purposes. To access Ethernet Capture feature, please navigate to Maintenance → Network Troubleshooting → Ethernet Capture

The capture packets can be stored locally and downloaded for analysis. However, if the user is diagnosing a randomly-occurring issue, he/she can run a continuous packet capture which can be limited by the size of the packet capture and the number of packet capture instances

The output result is in .pcap format. Therefore, users could specify the capture filter as used in general network traffic capture tool (host, src, dst, net, protocol, port, port range) before starting to capture the trace.

IP Ping

Enter the target host in host name or IP address. Then press “Start” button. The output result will dynamically display in the window below.

Traceroute

Enter the target host in host name or IP address. Then press “Start” button. The output result will dynamically display in the window below.

Record Meeting for Diagnosis

Enter the target meeting, support the ongoing meeting, and then click the “Start” button to capture the recording diagnosis of the meeting members in progress. The output result will be automatically displayed below, click the “download” button to download to the local. After the download is complete, immediately click the “Delete” button to clear the system content.

Signaling Troubleshooting

Analog Record Trace

• Analog Record Trace

Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM630xA user has caller ID issue for incoming call from Analog trunk. Users can access analog record trance under Web GUI🡪Maintenance🡪Signal Troubleshooting.

Here is the step to capture trace:

1. Select FXO or FXS for “Record Ports”. If the issue happens on FXO 1, select FXO port 1 to record the trace.
2. Click on “Start”.
3. Make a call via the analog port that has the issue.
4. Once done, click on “Stop”.
5. Click on “Download” to download the analog record trace.

• A key Dial-up FXO

Users can directly set a PSTN number on the “External Extension” text box to troubleshoot issues related to the analog trunk easily, the following steps shows how to use this feature:

1. Configure analog trunk on UCM, including outbound route.
2. Enter a reachable external number in “External Extension”.
3. Press “Start” button. The call will be initiated to the external number.
4. Answer and finish the call before pressing “Stop” button.

The trace will be available for analysis to download after output result shows “Done! Click on Download to download the captured packets”.

Note: When using a Key Dial-up FXO feature the outbound trunk for the analog trunk need to have internal permission. As well as it should be the trunk with the highest outbound route priority.

1. After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream
2. Technical support in the following link for further assistance if the issue is not resolved. https://www.grandstream.com/support

Service Check

Enable Service Check to periodically check UCM630xA. Check Cycle is configurable in seconds and the default setting is 60 sec. Check Times is the maximum number of failed checks before restart the UCM630xA. The default setting is 3. If there is no response from UCM630xA after 3 attempts (default) to check, current status will be stored and the internal service in UCM630xA will be restarted.

Network Status

In UCM630xA Web GUI🡪System Status🡪Network Status, the users can view active Internet connections. This information can be used to troubleshoot connection issue between UCM630xA and other services.

APPENDIX A: RFC STANDARDS USED IN THE UCM6300 AUDIO SERIES

• RFC 3261 SIP: Session Initiation Protocol
• RFC 3262 Reliability of Provisional Responses in SIP
• RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers
• RFC 3264 An Offer/Answer Model with the Session Description Protocol
• RFC 3515 The Session Initiation Protocol (SIP) Refer Method
• RFC 3311 The Session Initiation Protocol (SIP) UPDATE Method
• RFC 4028 Session Timers in the Session Initiation Protocol (SIP)
• RFC 2976 The SIP INFO Method
• RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)
• RFC 3892 The Session Initiation Protocol (SIP) Referred-By Mechanism
• RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging
• RFC 4733 RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals
• RFC 4566 SDP: Session Description Protocol
• RFC 2617 HTTP Authentication; Basic and Digest Access Authentication
• RFC 3856 A Presence Event Package for the Session Initiation Protocol (SIP)
• RFC 3711 The Secure Real-time Transport Protocol (SRTP)
• RFC 5245 Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols
• RFC 5389 Session Traversal Utilities for NAT (STUN)
• RFC 5766 Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)
• RFC 6347 Datagram Transport Layer Security Version 1.2
• RFC 6455 The WebSocket Protocol
• RFC 8860 Sending Multiple Types of Media in a Single RTP Session
• RFC 4734 Definition of Events for Modem, Fax, and Text Telephony Signals
• RFC 3665 Session Initiation Protocol (SIP) Basic Call Flow Examples
• RFC 3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)
• RFC 3550 RTP: A Transport Protocol for Real-Time Applications

CHANGELOG

This section documents significant changes from previous versions of the UCM630xA user manuals. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here.

Firmware Version 1.0.35.8

• Call Flow Design
  • Added Visual Call Flow Design (Call Flow Design) to build complex call routing using drag-and-drop components. [Call Flow Design]
• WhatsApp Trunk
  • Added support for WhatsApp SIP Trunk integration and Wave WhatsApp calling support. [WhatsApp Trunk]
• AI Features
  • Added AI Translation feature for real-time translation of call/meeting transcripts and chat messages. [AI Translation]
  • Added AI Voice Assistant (Beta) support for controlling endpoints via voice commands. [AI Voice Assistant]
  • Added option to allow meeting summaries to be shared with all participants. [Schedule Meeting]
  • Added AI Recordings Transcription and Summary support for audio recording files. [AI Recording Transcription and Summary]
  • Added interactive synchronization between recording playback progress and AI transcript text. [AI Recording Transcription and Summary]
  • Added automated and manual cleanup options as well as backup and restore support for AI transcript files. [Cleaner][Backup/Restore]
  • Added feature code to start or stop AI call transcription during an active call. [Feature Codes]
  • Added User Portal support for playing and accessing AI transcriptions of voicemails. [AI Voicemail to Text]
• Collaborative Office Integration
  • Added support for automatically syncing CDR CSV files to Google Docs via Google Workspace integration. [Google User][CDR Automatic Download]
  • Added support for SAML SSO authentication. [SAML SSO]
  • Added Google Workspace Calendar integration to dynamically map user calendar events to extension status. [Google Calendar]
• Messaging
  • Added support for Voxtelesys SMS provider integration. [SMS Configuration]
  • Added support for VoiceMeUp SMS provider integration. [SMS Configuration]
  • Added Smart Call SMS Link Service to trigger automatic SMS messages with links via IVR key presses. [SMS Template]
  • Added Message Edit/Delete Period option to the IM settings page. [IM Settings]
• Call Queue
  • Added Wallboard feature to display real-time call queue metrics and agent status. [Queue Board]
  • Added Queue Panel for interactive call management, agent status changes, and real-time monitoring. [Queue Operation Panel]
  • Added dedicated Queue Permissions configuration to restrict control operations for chairmen and agents. [Queue Permissions]
  • Added Queue CDR management with support for viewing, calling back, and tracking missed call processing status. [Queue Permissions]
  • Added Priority Queue feature to dynamically assign call weights and route high-priority calls first. [Call Queue]
  • Added time-based condition routing for call queues during out-of-office hours or holidays. [Call Queue]
  • Optimized virtual queue timeout triggering logic and updated user interface. [Call Queue]
  • Added “Skip Busy Agents” option to allow or prevent ringing call queue agents who are currently in a call. [Call Queue]
  • Removed Queue Switchboard to be replaced with Queue Operation Panel. [Queue Operation Panel]
  • Added custom user permissions for Queue Wallboard and Queue Operation Panel. [Add Custom Privilege]
• PMS
  • Added Local PMS unique order ID generation for reservation tracking across lists and invoices. [Local PMS]
  • Added Local PMS read-only historical reservation views to prevent accidental data changes. [Local PMS]
  • Added Local PMS overdue reservation filtering to identify guests who failed to check in on time. [Local PMS]
  • Added global Local PMS reservation time settings to configure default daily check-in/check-out times and set allowed thresholds for earliest check-in and latest check-out. [Local PMS]
  • Updated Local PMS multi-room reservations to preserve checked-in room rates when reservation prices are modified. [Local PMS]
  • Added a dedicated refresh button to the Local PMS Room Availability Calendar for real-time updates. [Local PMS]
  • Added tax-inclusive rate displays for active bookings on the Local PMS Room Availability Calendar. [Local PMS]
  • Added Local PMS hour-room management rules restricting room changes after check-in. [Local PMS]
  • Added mandatory base price requirement when enabling Local PMS hourly rental option. [Local PMS]
  • Added customizable hotel branding logo, phone numbers, and websites to Local PMS bill printing. [Local PMS]
  • Added Local PMS auto-assign room button for automated room selection during reservation creation. [Local PMS]
  • Added itemized tax, room charge, and minibar breakdowns to Local PMS bill printing. [Local PMS]
  • Added Local PMS support for custom hotel branding (logo, phone number, website, and QR code) in self-check-in and self-check-out emails. [Local PMS]
  • Added custom permission controls for PMS Basic Settings and Minibar management. [Add Custom Privilege]
• Zero Config
  • Added support for batch local firmware upgrades via Zero Config. [Firmware]
  • Added Zero Config support for Avaya IP phones. [Zero Config Supported Devices]
  • Added Zero Config multi-account configuration support for Poly Edge B Series IP phones. [Zero Config Supported Devices]
  • Added Zero Config support for Grandstream HT and GXW gateways. [Zero Config Supported Devices]
  • Added support for Cisco phone DND and Conference features. [Zero Config Supported Devices]
  • Added support for Cisco CP-6921 and CP-3905 IP phones. [Zero Config Supported Devices]
• CRM
  • Added Custom CRM Integration interface, custom templates, OAuth2/Bearer Token options, and import/export capabilities. [CRM]
• VoIP Trunk
  • Added option to select whether to use DOD numbers when “Keep Original CID” is enabled on SIP trunks. [VoIP Trunks]
  • Added “Not Passthrough” option and refined configuration for Diversion headers on SIP trunks. [VoIP Trunks]
• Inbound Routes
  • Added support for custom Inbound Mode names and audio prompts for mode switching. [Inbound Route: Multiple Mode]
• Outbound Routes
  • Added Diversion ID Manipulation settings (strip and prepend) for outbound routes. [Configuring Outbound Routes]
• User Management
  • Added global configuration option to restrict users from modifying their extension email addresses. [User Portal/Wave Privileges]
  • Added Multi-Factor Authentication (MFA) login support for User Portal. [Create New SIP Extension]
• Upgrade
  • Added interactive feature highlights guide on new system updates for administrators. [New Feature Guide]
• Backup
  • Added support for incremental backup tasks for CDR, queue statistics, and recording files. [Backup/Restore]
• Storage Device Management
  • Added support for SMB 3.0 protocol for network disk storage. [NAS]
• System Cleanup/Reset
  • Increased maximum automatic cleanup retention period to 730 days. [Cleaner]
• System Events
  • Added alert notifications when outbound call duration thresholds are exceeded on outbound routes. [Alert Events List]
• Voicemail
  • Optimized voicemail audio encoding format terminology in the web interface. [Configure Voicemail]
• Fax/T.38
  • Added support for configuring a unified administrator email address for failed Email-to-Fax notifications. [Email Settings]
• Meeting
  • Added support for including UCM Station IDs in random meeting room numbers for cross-PBX routing. [General Settings][Meeting Settings]
• CDR
  • Added Call Notes and Labels feature for tagging and adding comments to call records. [CDR][Call Note]
• IVR
  • Added support for configurable IVR end keys (including #). [Key Pressing Events]
• Emergency Calls
  • Added option to configure whether Geolocation-Routing headers are included in emergency call location mapping. [Emergency Location Mapping]
• Voicemail
  • Added support for feature code *99 to record and modify Voicemail Group prompts. [Feature Codes][Configure Voicemail Group]
• Announcement
  • Added call permission configuration for Announcements when the destination is set to an external number. [Announcement]
• Time Settings
  • Added custom prompt support for holiday events in Time Settings. [Holiday]
• SIP Settings
  • Added option to enable blind transfers while the callee’s phone is still ringing. [Misc]
• WebRTC Trunk
  • Added WebRTC Trunk one-click dialing mode and script-based integration code. [WebRTC Trunks]
• Dashboard
  • Added Storage Health display to dashboard. [PBX Status]
• RemoteConnect
  • Retired Click2Call option while preserving existing configurations for legacy setups. [Integrated Customer Service]
• PBX Settings
  • Moved Enterprise Contact Number field to PBX Settings→General Settings. [General Settings]

Firmware Version 1.0.33.42

• Added support for call queue reports. [Call Queue Reports]
• Added support for scheduling call queue reports. [Scheduled Report]

Firmware Version 1.0.33.32

• No major changes.

Firmware Version 1.0.33.30

• Added AI configuration tab to SIP extensions, allowing per-extension control of Automatic AI Transcription. [Create a New SIP Extension]
• Added scheduled meeting configuration with AI transcription option, allowing organizers to enable or disable AI transcription per meeting. [Schedule Meeting]
• Added AI Settings (Beta) page to centrally manage AI services. [AI Settings (Beta)]
• Added RemoteConnect AI Services page to display AI service status, plan validity, and detailed usage statistics [AI Services]
• Added support for AI Call Transcription & Summary. [AI Call Transcription and Summary]
• Added support for AI Conference Transcription & Summary. [AI Meeting Transcription and Summary]
• Added support for AI Voice Message to Text. [AI Voice Message to Text]
• Added AI Generated Speech (TTS) feature for creating voice prompts directly from text. [AI Generated Speech]

Firmware Version 1.0.33.22

• No major changes.

Firmware Version 1.0.33.21 (BETA Only)

• Added several new call report types for various call scenarios [Call Reports]
• Added support for Cisco CP69xx/78xx/88xx endpoints DND. [Extensions]

Firmware Version 1.0.33.7 (BETA Only)

• Added Strict Defend Mode option to Geo-IP Access Control [Geo-IP Access Control]
• Added unified Integration tab for collaboration and identity platforms. [Integration of Collaboration Platforms]
• Added support for Google Workspace integration to synchronize user identity information between Google Workspace and the UCM Series. [Google Workspace Integration]
• Added Google Workspace Single Sign-On (SSO) support for UCM web interface authentication. [Google Workspace Integration]
• Added support for configuring Microsoft Entra ID user synchronization scope. [Microsoft Entra ID (Azure AD)]
• Added support for configuring holidays by date or by week. [Holiday]
• Increased the maximum size limit for backup files uploaded to the UCM via the Web GUI. [Restore Configuration from Backup File]
• Added Zero Config support for Yealink IP phones. [Zero Config Supported Devices]
• Updated the Local PMS wake-up service prompt to dynamically use the language configured during room check-in. [Local PMS]
• Optimized Local PMS check-in and check-out actions to take effect immediately without requiring “Apply Changes”. [Local PMS]
• Added option to control Wave app Screenshot and Screen Recording permission. [Wave Permissions]
• Added support for configuring whether the Wave external host address is automatically synchronized from UCMRC. [Plan Settings][HTTP Server]
• Added support for Bitrix24 CRM integration. [Bitrix24 CRM]
• Added support for network abnormality HA switchover. [HA Settings]

Firmware Version 1.0.31.7

• Added support for third-party SQL database integration for phonebook lookup and synchronization. [3rd Party Database Integration]
• Updated the Inbound Routes→3rd Party Database Search page to align with the new third-party SQL database integration feature. [Inbound Routes: 3rd Party Database Search]
• Added support for enabling local HA while operating in Dual network mode. [HA Settings]
• Added third-party SIP Phone support (Cisco/Poly). [Zero Config Supported Devices]
• Added vendor and model search capability when downloading model templates. [Model Update]
• Increased maximum Zero Config firmware upload size to support larger GXV series firmware packages. [Firmware]
• Added configurable room types with support for day-based pricing, holiday pricing, tax rates, and hourly pricing. [Local PMS]
• Added reservation management. [Local PMS]
• Added a guest list for centralized customer profile management, including search, import/export. [Local PMS]
• Added email templates for reservation notifications, cancellation notices, and billing emails. [Email Template]
• Added multiple check-in workflows, supporting check-in from reservation list, from room list (immediate check-in), and hourly or full-day room selection. [Local PMS]
• Added room change (house switch) functionality to move checked-in guests to another available room while preserving guest and billing data. [Local PMS]
• Added a room availability calendar with visual status indicators, booking and check-in actions, hover-based detail display. [Local PMS]
• Added bill printing with detailed room charges, minibar costs, call charges, tax breakdowns, and prepayment deductions. [Local PMS]
• Added wake-up service management with support for room group based scheduling. [Local PMS]
• Added support for self-service check-in and chek-out via email for guests with valid reservations. [Local PMS]
• Added configurable email delivery timing for self-service check-in and check-out messages relative to arrival and departure times. [Local PMS]
• Added tracking of check-in and check-out methods, distinguishing between front-desk and self-service operations in history records. [Local PMS]
• Added automatic room assignment during self-service check-in when reservations are not pre-assigned to specific rooms. [Local PMS]
• Added support for DOD numbers batch deletion. [Direct Outward Dialing (DOD)][Outbound Routes DOD]
• Added support for inbound and outbound routes batch deletion. [Configuring Inbound Routes][Configuring Outbound Routes]
• Added outbound route total call duration limit setting. [Configuring Outbound Routes]
• Added display of trunk status and current call count display on the VoIP Trunk page [VoIP Trunks]
• Added Caller ID Transform feature on outbound routes [Configuring Outbound Routes]
• Added support for data backup and archiving to third-party cloud storage (Amazon S3). [Amazon S3]
• Add the ability to play/hear the uploaded Prompts directly from the WebGUI. [Custom Prompt]
• Added support for manual addition of Fail2Ban blocklist entries. [Fail2Ban]
• Added Cleaner Record and Details [Cleaner]
• Added the ability to have multiple Scheduled Backup tasks on different storage locations. [Scheduled Backup][Backup/Restore]
• Added ability to modify LDAP phonebooks through API. [API Queries Supported]
• Added support for adding LDAP contacts as members of paging/intercom groups to 2-way intercom and 1 way paging. [LDAP Members]
• Added support for proficiency-based routing as a routing strategy in the call queue. [Proficiency Routing]
• Added selecting the dynamic agents which can be added to a specific call queue. [Dynamic Agents]
• Added recording file and CDR access permission. [View Team CDRs and Recordings]
• Added Wave recording indicator appearance permission. [Recording Status]
• Added support for configuring the pre-installed Wave add-ins. [Pre-intalled Plugins]
• Added support for multifactor authentication using email for Wave. [Multifactor Authetication]

Firmware Version 1.0.29.21

• Some general call failure prompt tones (e.g., no-answer, user-busy, no-permission, wrong-number, general-failed) can now be used for SIP trunk call failure messages. [SIP Trunk Prompt Tone]

Firmware Version 1.0.29.11

• Added Least Cost Routing policy to Outbound Routes [Carrier]
• Added support for call billing [Call Rate]
• The administrator can choose to prioritize call pattern matching or outbound route priority order as the criteria for call routing [Outgoing Routing Priority]
• Added a configuration field to enter maximum concurrent calls allowed by a SIP trunk. [VoIP Trunk Configuration]
• Added ability to select the error message to display when the number of maximum current calls is achieved. [VoIP Trunk Configuration]
• Added support for enabling/disabling sending the diversion header on a particular trunk. [VoIP Trunk Configuration]
• Added support for Microsoft Entera ID integration, this allows synchronizing SIP accounts from LDAP or Active Directory for single sign-on. There is potential synchronization of additional information other than authentication information. [Microsoft Entra ID (Azure AD)]
• Added support for Active Directory to synchronize SIP accounts (user portal and Wave login only) and admin accounts. [Windows AD]
• Strong password requirements have changed, now administrator password and user portal password requires at least 10 characters, with at least one upper letter and one lower letter, one number, and one special character.
• Added support for detecting new firmware directly from the web UI, regular firmware version checking can be scheduled on the webUI. [Upgrade]
• Added country-specific restriction in ACL [Country/Region IP Defense]
• Added support for automatic update of the timezone list [Automatic Date and Time]
• Now some of the services can have their ports modified, while this was not possible before []
• Added a link to the Open Source Software License Agreement
• Administrator password, user password, and SIP password now require at least 10 characters [Create New SIP Extension][User Management]
• Added TLS compression for LDAP server [LDAP Server Configurations]
• Backup allows now excluding instant messages [Backup]
• The dashboard now displays the physical port name [PBX Status]
• Added Call Policy [Create New SIP Extension]
• Added name configuration in Speed Dial [Speed Dial]
• Added “First Answer Termination” in Paging configuration [Paging/Intercom Groups]
• Added Call Privilege to Announcement Center [Announcement Center]
• Increased voicemail group members number to 150
• CDR now displays the name of the callee
• Added prompt language configuration in Call Queue [Add Call Queue]
• Call Queue Chairman can now view the CDR
• Queue Chairman codes can now be configured
• Added option to disable RPID in the Call Queue
• ELIN field in Emergency calls now can be configured with more character, letters and numbers [Emergency Location Mapping]
• Add the ability to support Emergency Location Mapping based on a single Extension approach for the Emergency calls feature [Emergency Location Mapping]
• Added default emergency callID field for Extension’s autopopulate [Emergency Location Mapping]
• Added ability to customize PAI for Emergency Calls in multi-site environment [Emergency Location Mapping]
• Added Oracle Hospitality OPERA PMS integration [PMS]
• PMS parameters have been added to list of privileges for admins [Custom Privilege]
• Added ability to use key pressing events when receiving a wake up call. [Wakeup Service]
• Improved “Move” function in PMS, now more information are transferred along with the guest
• Added support for configuring the allowed API modules when adding a user [API Configuration Parameters]
• The administrator can now configure up to 10 subnets on Zero Config whitelist [Zero Config Settings]
• Administrator can now use MAC OUI to filter search results when searching for devices on Zero Config [Discovery]
• Updated email template for “new messages to inactive user notification” [Email Template]
• Added “Delete message” Wave permission. [User Portal/Wave Privileges]
• Added “Allow Simultaneous Logins from the Same Client Type” Wave permission [User Portal/Wave Privileges]
• Added “Play Joining/Leaving Meeting Announcement” which enables/disables the sound played when a user joins/leave [Schedule Meeting]
• Add permission for the meeting host to play, delete, and download meeting recordings
• Added end time for scheduled meetings [Schedule Meeting]
• Meeting will now end when a GDS or GSC extensions are the only ones left in the meeting
• Added event alert in case synchronizing files to cloud storage fails [Alert Events List]
• GDMS cloud storage security key can now be obtained [GDMS Cloud Storage Space]
• Added compatibility for GDMS configuration template for the UCM6300 Series
• Import and export functions now support “.xlsx” files.
• Backup packages now use .tgz format by default instead of .tar
• Added support for triggering webhooks via an IVR [Configure IVR]
• Added Background Sound to Multicast Community [Multicast Community]
• Added Linked ID to CDR AMI event reporting for improved call tracking and management
• Added Premium Code and and Premium Code Prompt option to Advanced Settings page. The user can use the premium code to increase their priority in the queue. [Add Call Queue]

Firmware Version 1.0.27.23

• Hmobile PMS option has been named to char pmslink.

Firmware Version 1.0.27.20

• Added support for Microsoft Outlook SMTP OAuth2 authentication. [Email Settings]

Firmware Version 1.0.27.15

• No major change

Firmware Version 1.0.27.10

• Added QR code to email template for quick SIP registration [Email Templates]
• Added Account SIP Trunk [Account SIP Trunk]
• Added VoIP VLAN [Network Settings]
• Added support for searching CallerID in a third-party MySQL database [Inbound Route: Third-party Database Search]
• Added support for integration with Don’t Call Me database [Don’t Call Me Blacklist Integration]
• Added support for ZRTP on the extensions level [Create New SIP Extension]
• Added support for ZRTP on the trunk level [VoIP Trunk Configuration]
• Added support for enabling multiple Wave session on the same platform [Create New SIP Extension]
• Added support for verifying ACS [TR-069]
• Added support for Tx and Rx Noise Cancellation in Analog Trunks [Analog Trunk Configuration]
• Added Task Management feature [Task Management]
• Added support for search function in LDAP server phonebooks [LDAP Phonebook]
• Added support for data encryption on the local storage and attached storage [Data/File Encryption]
• Added Merge Same Call Recordings [PBX Settings]
• Added support for adding suffix to Call Forward Enable/Disable feature [Feature Codes]
• Added icons next to feature codes to indicate whether they can be nested by other feature codes [Feature Codes]
• Added support for Time Condition Routing [TIME CONDITION ROUTING]
• Added support for Custom Time Groups [Custom Time Groups]
• Added support for custom announcement for Call Queue [Configure Call Queue]
• Added Reset Agent Call Counter to Call Queue [Configure Call Queue]
• Added number and percentage of transferred calls in the Call Queue Switchboard [Configure Call Queue]
• Added Call Memory in Call Queue [Configure Call Queue]
• Updated Wake-up Service by adding call failure notification [Wakeup Service]
• Added Wake-Up Call Service Failure email template. [Email Templates]
• Room status can now be modified using the API
• Odoo CRM has been added [Odoo CRM]
• Added custom repeating setting when scheduling meetings [Schedule Meeting]
• Added Extension Login Management module. [Extension Login Management]
• Added Wave Administrator privilege [Create New SIP Extension][Custom Privilege]
• Added Download Chat Logs privilege [User Portal/Wave Privileges]
• Added Remote Logout Wave Privilege [User Portal/Wave Privileges]
• Added API support to retrieve recording filename
• Added Password Visibility Toggle [Super Administrator]
• Added enable/disable setting for Paging/Intercom [Paging/Intercom]
• Added support for displaying the voicemail group name on the voicemail page on Wave
• Added support for configuring custom agent pause reason with a custom prompt [Global Queue Settings]
• Added TLS key download in Ethernet Capture diagnosis tool [Ethernet Capture]
• Login Security options have been improved [Login Security]
• External Host address will be automatically pushed to Wave clients when the UCM is deployed in a Remote Disaster Recovery setup. [HTTP Server]
• UCM reboot event alerts and logs will now contain the reboot reason and the name of the user who initiated the reboot (if applicable) [System Reboot]
• Increased members notified for Emergency Calls from 10 to 30 [Emergency Calls]

Firmware Version 1.0.25.9

• Add Portugal Portuguese voice prompt support. [LANGUAGE SETTINGS FOR VOICE PROMPT]
• Add Wi-Fi related settings to GHP-W series template.

Firmware Version 1.0.25.7

• Multicast Community has been added. [Multicast Community]
• Scheduled Paging UI has been reworked. [Scheduled Paging/Intercom]
• Added support for playing MP3 files over GSC35XX devices. [Upload Custom Prompt]
• Added support for PMS billing system for Local PMS. [Call Rate]
• Mini bar has been improved. [Mini Bar]
• Added ability to set a Feature Code as a the Default Destination. [Speed Dial]
• Added ability to add External Number and LDAP numbers on Announcement Center. [Announcement]
• Added Fax Sending privilege when creating custom privileges. [Custom Privilege]
• Added enabling TURN Relay for a specific extension. [Create New SIP Extension]
• HA Data Sync Failure alert event has been added to the Alert Events List. [Alert Events List]
• SIP endpoints firmware can now be updated using the Zero Config page. [Managing Discovered Devices]
• Clearing the call history upon guest check-in or check-out has been added. [Local PMS]
• Onsite Meeting email invitations will include now an ics file. [Onsite Meeting]
• “Send File” Wave privilege has been added. [User Portal/Wave Privilege]

Firmware Version 1.0.23.17

• No major change.

Firmware Version 1.0.23.15

• Increased Fail2Ban list limit to 50 entries. [Fail2ban]
• NAS password field now supports special characters “&$’^`>|, [PBX Settings/NAS]
• Speed Dial limit has been increased to 1000. [Speed Dial]

Firmware Version 1.0.23.12

• Added support for limiting chat history viewing for newly added participants. [Room]
• Added support for including company number when scheduling a meeting. [Meeting Schedule]
• Added support for tracking outbound calls for QueueMetrics.
• Added support for displaying call queue names on GRP phones.
• Added support for auto answer on inbound route level to detect fax call. [Inbound Rule Configurations]
• Added support for updating the call forwarding status when enabled via feature code. [Feature Codes]
• Added support for controlling diversion header. [SIP Settings]
• Added “Extension Group” as a custom privilege. [Custom Privilege]
• Added ability to retrieve recording files saved on NAS/GDMS cloud through API.
• Added support for manually backing up the UCM configuration to SFTP. [Backup/Restore]
• Added disaster recovery for High Availability. [HA]
• Added feature code to blacklist last called number. [Feature Codes]
• Added ability to share voicemail to group members. [VOICEMAIL]
• Added ability to delete voicemails for all voicemail group from one member. [VOICEMAIL]
• Added ability to monitor voicemail groups with voicemail key on endpoints. [VOICEMAIL]
• Added “Ignore Safe Operational Flow” to Network Port Traffic Control. [Network Settings]
• Added support for domains in API ACL.
• Added support for configuring chat filesize limit. [IM Settings]
• Added support for “announce message caller-ID” in Annoucement Center. [Announcement Center Settings]
• Added support for preventing Wave chat deletion. [User Portal/Wave Privilege]
• Added “Priority Call” feature code. [Feature Codes]
• Added support for updating time zones under Time Settings. [Time Settings]
• Added Live Chat feature. [Live Chat]
• Added Message Broadcast feature. [Message Broadcast]

Firmware Version 1.0.21.9

• Added SMS service support. [SMS Settings]
• Added support for creating shared departments. [Department Management]
• Added support for RADIUS login. [RADIUS]
• Added support for private intercom. [Configure Private Intercom]
• Added support for setting an extension as the default destination in Click2call. [Integrated Customer Service]
• Added support for SIP TLS cipher suite. [SIP Settings/TCP and TLS]
• Added support for continuous packet capture when using USB/SD card storage or SFTP. [Ethernet Capture]
• Added support for resetting TLS certifcates to the default ones. [SIP Settings/TCP and TLS]
• Added support for SSH token. [SSH Access]
• Added support for subscribing to a voicemail group. [Configure Voicemail Group]
• Added support for setting separate call forwarding conditions for external and internal calls. [Create New SIP Extension]
• Added support for forwarding calls to a custom prompt. [Create New SIP Extension]
• Added support for external numbers to opt out of being recorded when calling into the UCM. [PBX Settings]
• Added support for caller name look up. [Inbound Routes]
• The name of the agent will now be displayed in the switchboard. [Switchboard]
• Agent pause can now be performed quickly by dialing the respective feature code and the corresponding pause reason without having to interact with the IVR. [Configure Call Queue]
• Agent pause reason will now be displayed in the switchboard. [Switchboard]
• Added support for enabling the welcome prompt to be played simultaneously with background music while the agent phone is ringing. [Configure Call Queue]

Firmware Version 1.0.19.10

• Specific Time configuration is now included in the extension exports

Firmware Version 1.0.19.9

• Optimized various system processes
• Added Onsite Meeting feature. [Onsite Meeting]
• Added ability to customize extension call waiting tone [General Call Prompt Tones]
• Updated Zoho CRM authentication process [Zoho Telephony]
• {VM_DATE} date value format has been changed to MM/dd/yyyy hh:mm:ss from DDD yyyy MMM hh:mm:ss. [Configure Fax/T.38] [Voicemail Email Settings]
• Added Device Name ${DEVICE_NAME} variable to Alert Events and Emergency Calls email templates
• Added Geolocation header support [Emergency Location Mapping]
• Added P-Called-Party-ID header option to the Add/Edit Extension -> Features page [Create New SIP Extension]
• Added Allow Operator Panel Monitoring extension option to toggle whether the Operator Panel can monitor the extension. [Create New SIP Extension]
• Added Basic extension export option [Export Extensions]
• Allow Call-barging Extension List option changed to Call Monitoring Whitelist [Create New SIP Extensions] [Create New IAX Extension]
• Added Silence Suppression option to Extensions/VoIP Trunks page [Create New SIP Extension] [VoIP Trunk Configuration]
• If a storage device is full, the UCM will mark it as unavailable and automatically change file storage path to the next available location based on the Storage Path Priority. Previously, UCM would change the file storage path to its own local storage if external storage was full. [File Manager]
• Added new commands related to call queue and Wave [API Configuration Parameters]
• Added support for multiple API (new) users [API Configuration Parameters]
• Added the Default Certificate Auto Renewal option. If enabled, the default browser certificate will be automatically renewed after 398 days (the max certificate validity period of Chrome, Firefox, and Safari browsers). User-defined certificates are not affected. [HTTP Server]
• Added ability to sync local IM data to Cloud IM [Cloud IM Service]
• Added ability create custom IVR key presses [Custom Key Event]
• Added Chat Data from Endpoint option to the Maintenance -> System Cleanup/Reset -> Cleaner page. If enabled, this option will clean out chat data from Wave clients at the same time as the UCM’s server-side automatic/manual cleaning of chat data. [Cleaner]
• Added support for meeting room passwords. However, meetings cannot be scheduled for rooms with meeting password enabled. [Room]
• Meeting kick warning interval has been changed from 30 mins to 20 mins. Note: This kick warning will only play when there is only 1 person in a meeting room, and if they do not opt to stay in the meeting room after the warning, they will be removed from the meeting room after 5 minutes.
• ARP will now be used instead of ping to check NAS connectivity.
• Upgrade logs will now contain firmware version information.
• Queue chairmen can now log out dynamic agents
• Added option to automatically reset user/Wave password upon check-in/check-out. [Local PMS]
• Added option to clear Wave chat history automatically upon check-in or check-out. [Local PMS]
• Added Local PMS functionality [Local PMS]
• Check-out will no longer reset the “Skip Voicemail Password Verification” extension setting
• Added ability to assign two extensions to a room [Room Management]
• Added option to clear scheduled wakeup calls on both check-in and check-out. [Local PMS]
• Added the ability to change the default call privilege of a room. A room’s privilege will be reset to this value after a guest checks out of it.
• Added support for stereo audio recording [General Settings]
• Added option to route calls based on a caller’s Diversion header value [Inbound Rule Configuration]
• Added ability to control whether to use failover trunks based on the call response codes [Failover Trunk Toggles]
• Added support for H.264 with multiple payload types in SDP
• When receiving an INVITE with no SDP, following INVITEs with SDP will offer H.264 1080p resolution by default.
• In the scenario where an inbound external call is forwarded from an extension to an external number, the Contact header will now use the CID of the forwarding extension instead of the caller’s CID.
• Removed External Device Usage Threshold option. If a connected NAS has only 1GB remaining available storage space, it will be considered unavailable and trigger the external disk usage alert. [NAS]
• Added User Endpoint Access History page [User Endpoint Access History]
• Added User Portal/Wave privilege control [User Portal/Wave Privilege]
• Dial Trunk option has been renamed to Dial External Number and moved to the Dial Other Extensions section
• The Wave Welcome email will now use the port number configured in System Settings->HTTP Server->Wave Settings->Port if the Wave Settings->External Host value is not a RemoteConnect address or does not contain a port number.
• Added links to relevant online documentation to various pages of the UCM webUI.
• Added Phonebook VMPK mode to GRP261x template
• Added Firmware tab for improved firmware management [Firmware]
• Added ability delete downloaded base model templates in the Model Update page
• Added ability to search for templates via the device model name
• Added ability to select either LAN1 or LAN2 to scan for devices on when using dual network method

Firmware Version 1.0.17.11

• Several system process optimizations

Firmware Version 1.0.17.8

• Updated python version to 3.8 and related processes.
• Improved speed of applying changes
• Updated lighttpd version to 1.4.61.
• The Privilege Name field now supports parenthesis ().
• The Contacts page has been moved to its own category in the sidebar.
• Added https://www.zohoapis.in option to CRM Server Address list.
• Added Channel Path option for accessing specific IP camera channels via URL. [Device Management]
• Improved alert email sending process.
• Updated Remote Registration email template.
• Emergency calls will no longer be restricted by the RemoteConnect call limit.
• The default highest priority codec is now G.722.
• Added Remote Extension Privilege Update feature code and Remote Extension Privilege Update Whitelist field to allow specified users to remotely change extensions’ privileges. [Feature Codes]
• Automatic file migration after file storage path failure to the next storage location in the storage priority path.
• Users can now customize the storage path priority for recordings, and IM files. [File Manager]
• FXO FSK CID detection now uses spandsp.
• If SIP extensions synced from UCMs are deleted on GDMS, they will no longer be synced again.
• If HA is enabled, the HA cluster IP address will now be provisioned as the config server to endpoints instead of the active UCM’s IP address.
• Added support for configuring inbound route blacklist via HTTPS API.
• If dialing into a Dial by Name directory, the call will end automatically after failing 3 times.
• Added Server Type option to the LDAP Server→LDAP Phonebook→Phonebook Download Configurations page. Users can select between LDAP and Active Directory. [LDAP Server]
• Added Department field to LDAP phonebook contacts. [LDAP Server]
• Added meeting room extensions to LDAP phonebooks. [LDAP Server]
• External Contacts created from the Contacts page will now be added to the system’s internal LDAP phonebook. [LDAP Server]
• Added Remote Login tab to the Maintenance→Login Settings page. [Maintenance]
• Created new Meetings Settings page under the Meetings page and moved several meeting-related options to it. [Meeting]
• Regular meeting participants can now invite other members to join the meeting by dialing 1 if “Allow User Invite” is enabled. [Meeting]
• Meetings will become “Pending” after rescheduling.
• Pending meetings are now sorted by start time by default.
• Added the Allowed to Override Host Mute option to the Edit Meeting Room and Schedule Meeting pages to allow participants to unmute themselves even after the meeting host mutes them. [Meeting]
• Added support for user authentication. [OpenVPN]
• Operation Log entries will now contain the IP address and location information from which the operation originated. [Operation Log]
• Added option Automatically Clear Wakeup Calls for deleting scheduled wakeup calls after either guest check-in and check-out. [PMS Features]
• Users can now dial the Update PMS Room Status feature code, the maid code, and the room status code all at once to change room status. [PMS Features]
• Added a Scan button to manually retrieve the list of recordings on external storage. The UCM automatically displays up to 5000 recordings on attached external storage, but pressing this button will allow the UCM to display more. [Recording Files]
• Added ability to batch delete cloud storage files.
• If IP endpoints cannot connect to the GDMS TURN server via UDP, UCM will use TCP to connect them.
• Added Trunk Registration Period (s) option to SIP Settings->Misc. [SIP Settings]
• Added option Special Attributes to the Extension/Trunk→VoIP Trunks→Edit SIP Trunk→Advanced Settings page. If enabled, the following attributes will be included in the SIP SDP: ssrc, msid, mid, ct, as, tias, record. Enabling this may cause compatibility issues with non-Grandstream devices. [VoIP Trunk Configuration]
• Added CEI msid for audio calls.
• Added trickle-ice param to SIP OPTION’s 200 OK.
• profile-level-id will be added to 200 OK when receiving INVITEs without SDP.
• Added ability to import/export speed dials. [Speed Dial]
• Improved processes to avoid duplicate alerts for the following events: Registered SIP Trunk failed, Local Disk Usage and External Disk Usage.
• Separated Allow Deletion of CDR and Recordings option to Allow Deletion of CDR and Allow Deletion of CDR Recordings. [User Management]
• Added the Call Waiting option to the User Portal.
• Added support for voicemail message seeking. When listening to voicemail, users can press star (*) to rewind 3 seconds or pound (#) to fast forward 3 seconds. [Voicemail]
• Added Line Selection Strategy option for Trunk Groups. [VoIP Trunks]
• Changed register trunk Username field name to Trunk Registration Number. [VoIP Trunks]
• [Web] General web UI improvements
• [Web] Added Help option under the username dropdown menu that will redirect to the UCM6300 Series FAQ.
• [Web] Updated some tooltips.
• [Web] Optimized search functionality
• [Zero Config] Added GMT+2:00 (Israel) option to Time Zone drop down list in all Zero Config pages.
• [ZeroConfig] Added support for WP22 and WP825 model templates.

Firmware Version 1.0.15.13

• Added SNMP monitoring feature. [SNMP]
• Added support to configure the time of a holiday. [Holiday]
• Added ability to determine the maximum total call duration per trunk for outbound calls. [VoIP Trunk Configuration]
• Added contact viewing privilege (independent from Department Contact Privilege). [Contact Management]
• Added support for agent ID announcement. [Configure Call Queue]
• Added support for Service Level Agreement for Call Queue. [Service Level Agreement]
• Added support for changing the Meeting room’s name. [Room]
• Added WebRTC Trunk feature. [WebRTC Trunks]
• Extension data cleaning has been improved. [Search and Edit an Extension]
• Added support for SRTP Crypto Suite. [VoIP Trunk Configuration]
• STIR/SHAKEN has been improved. [STIR/SHAKEN]
• Added support for displaying the extension that initiated an emergency call in the emergency email notification. [EMERGENCY]
• Added support for collecting ICE candidates when an RTP connection is requested. [RTP Settings]
• Flood Attacks and Network Traffic Storm alerts have been added to the Alert Events List. [Alert Events List]
• Added support for Network Port Traffic Control for the ports of the UCM63xx Audio Series. [Network Settings]
• Added Support for limiting the frequency of calls that can be made in a period of time. [Create New SIP Extension]
• Added support for storing the local chat files in the GDMS. [File Manager]
• UCM RemoteConnect plan expiry screen has been improved.
• GDMS Cloud Storage Space details can now be viewed in the RemoteConnect menu. [GDMS Cloud Storage Space]

Firmware Version 1.0.13.9

• Added option to enable/disable DND status remotely for an extension. [CALL FEATURES]
• Added local proxy in IM settings. [Cloud IM Service]
• Added support for enabling/disabling auto audio recording for meeting. [Auto Record]
• Added privilege management for contacts. [Privilege Management]
• Improved fail2ban blacklist display. [Fail2ban]
• Improved email template for scheduling meeting. [Email Templates]
• Contacts sync-up between UCM and end points (wave/IP phones). [LDAP Settings]

• Added ability to specify DOD number based on outbound route. [Outbound Routes DOD]
• Fixed an issue where updating model templates will result in deleting the existing ones.
• Added support to use TURN Relay as an option to allow hosts behind NAT firewalls to communicate. [VoIP Trunk Configuration]

Firmware Version 1.0.11.10

• Added Operator Panel. [OPERATOR PANEL]
• Added time condition support for IVR key events. [Key Press Event]
• Added cloud IM abnormal alert event. [Alert Events List]
• Support setting to choose whether to play Follow Me.
• Add Fail2Ban whitelist comment information. [Fail2ban]
• Support Call Flip feature code. [Feature Codes]
• Added Multi-Factor Authentication for UCM login. [Multi-Factor Authentication]
• Added support for IoT device management. [DEVICE MANAGEMENT]
• Added option to enable and disable password-less remote access. [UCM RemoteConnect Plan Settings]
• Added option “Stop Ringing”. [Stop Ringing]
• Add option “Email Missed Call Log”. [Email Missed Call Log]
• Added remark for UCM system status. [Remark]
• Added option to enable and disable virtual queue call back keys settings. [Virtual Queue Callback Key Setting]
• Added Contacts section. [Contacts]
• Add option “Security Mode” for NAS settings. [Security Mode]
• Removed display for consumer users in user management page. [User Management]
• Support custom ignoring 180 response after 183 response. [SIP Settings/MISC]
• Added option to enable IPv6 for HA settings. [Enable IPv6]

Firmware Version 1.0.9.10

• Added Support for import/export Zero Config. [Global Policy]
• Added support for enable Wave and Sync Contact under the extension. [Create New SIP Extension]
• Added support for Custom time supplement time conditions. [Create New SIP Extension]
• Added support for Call Restriction. [RESTRICT CALLS]
• Added support for Queue Metrics. [QUEUE METRICS]
• Added support for CDR API add whitelist. [Permitted IP (s)]
• Added support for Call queue satisfaction survey. [Queue Statistics].
• The old API Configuration is reopened for use. [HTTPS API Settings (Old)]
• Custom permissions support the function of deleting CDR and recording files. [Custom Privilege]
• Added support to adjust recording file storage path. [File Manager]
• Added support to High Availability feature on UCM6300A series. [HA]
• Paging/Intercom supports delayed paging. [Configure Paging/Intercom]
• UCMRC remote service diagnosis. [Remote Diagnosis]
• Support LDAP to automatically update the phone book. [LDAP Automatic Update Cycle]
• Support meeting room automatic gain control. [Meeting AGC]

Firmware Version 1.0.7.12

• 
  Added support for email reminder when editing the time of a scheduled meeting. [Email Reminder (m)]
• Improved extension status syncing process to the IM server.

Firmware Version 1.0.7.9

• This is the initial version.