Welcome
Thank you for purchasing Grandstream UCM6200 series IP PBX appliance. The UCM6200 series IP PBX appliance is designed to bring enterprise-grade voice, video, data, and mobility features to small-to-medium businesses (SMBs) in an easy-to-manage fashion. This IP PBX series allows businesses to unify multiple communication technologies, such comprehensive voice, video calling, video conferencing, video surveillance, data tools and facility access management onto one common network that that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprise-grade features without any licensing fees, costs-per-feature or recurring fees.
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 UCM6200 series.
Table 1: Technical Specifications
Interfaces | |
Analog Telephone FXS Ports | 2x RJ11 ports with lifeline support Each port supports 2 REN |
PSTN Line FXO Ports |
|
Network Interfaces | UCM6202/6204/6208: 1x Gigabit WAN port 1x Gigabit LAN port with PoE Plus (IEEE 802.3at-2009) |
NAT Router | Yes |
Peripheral Ports | USB 2.0, SD |
LED Indicators | Power/Ready, Network, PSTN Line, USB, SD |
LCD Display | 128×32 graphic LCD with DOWN and OK button |
Reset Switch | Yes |
Voice/Video Capabilities | |
Voice-over-Packet Capabilities | 128ms tail-length carrier-grade Line Echo Cancellation with NLP Packetized Voice Protocol Unit, dynamic jitter buffer, modem detection, and auto-switch to G.711. |
Voice and Fax Codecs | G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC (30ms only), GSM, AAL2-G.726-32, RTX, ADPCM; T.38 |
Video Codecs | H.264, H, H.263, H.263+, VP8 |
QoS | Layer 3 QoS, Layer 2 QoS |
Signaling and Control | |
DTMF Methods | Inband, RFC4733, and SIP INFO |
Provisioning Protocol and Plug-and-Play | TFTP/HTTP/HTTPS, auto-discovery and auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66/multicast SIP SUBSCRIBE/mDNS), Eventlist between local and remote trunk |
Network Protocols | TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS, LDAP/LDAPS |
Disconnect Methods | Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect, Busy Tone |
Security | |
Media | SRTP, TLS1.2, HTTPS, SSH |
Advanced Defense | Fail2ban, alert events, whitelist, blacklist and strong password requirement. |
Physical | |
Universal Power Supply |
|
Dimensions |
|
Environmental |
|
Mounting |
|
Weight |
|
Additional Features | |
Multi-language Support | English/Simplified Chinese/Traditional Chinese/Spanish/French/ Portuguese/German/Russian/Italian/Polish/Czech for Web GUI; Customizable IVR/voice prompts for English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic; Customizable language pack to support any other languages |
Caller ID | Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT |
Polarity Reversal/ Wink | Yes, with enable/disable option upon call establishment and termination |
Call Center | Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability busy level, in-queue announcement |
Customizable Auto Attendant | Up to 5 layers of IVR (Interactive Voice Response) |
Concurrent audio calls up to 50, concurrent WebRTC calls up to 25.
Concurrent audio calls up to 75, concurrent WebRTC calls up to 35.
Concurrent audio calls up to 100, concurrent WebRTC calls up to 50. Or up to 66% performance if calls are SRTP encrypted | |
SIP Devices |
|
Conference Rooms |
|
Call Features | Call park, call forward, call transfer, DND, ring/hunt group, paging/intercom and etc. |
Compliance |
|
Installation
Before deploying and configuring the UCM6200 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 UCM6200 series.
Equipment Packaging
Main Case | 1x |
Power Adaptor | 1x |
Ethernet Cable | 1x |
Quick Installation Guide | 1x |
GPL License | 1x |
Connect Your UCM6200
Connect the UCM6202
To set up the UCM6202, follow the steps below:
- Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6202.
- Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
- Connect the 12V DC power adapter into the power jacks located on the back of the UCM6202. It is highly recommended to connect the other end of the plug to a surge protected power outlet.
- Wait for the UCM6202 to boot up. The LCD in the front will show the device hardware information when the boot process is done.
- Once the UCM6202 is successfully connected to network, the LED indicator for WAN in the front will be solid green and the LCD will display the IP address.
- (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax) to the FXS ports.
Connect the UCM6204
To set up the UCM6204, follow the steps below:
- Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6204.
- Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
- Connect the 12V DC power adapter into the power jacks located on the back of the UCM6204. It is highly recommended to connect the other end of the plug to a surge protected power outlet.
- Wait for the UCM6204 to boot up. The LCD in the front will show the device hardware information when the boot process is done.
- Once the UCM6204 is successfully connected to network, the LED indicator for WAN in the front will be solid green and the LCD will display the IP address.
- (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax) to the FXS ports.
Connect the UCM6208
To set up the UCM6208, follow the steps below:
- Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6208.
- Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
- Connect the 12V DC power adapter into the power jacks located on the back of the UCM6208. It is highly recommended to connect the other end of the plug to a surge protected power outlet.
- Wait for the UCM6208 to boot up. The LCD in the front will show the device hardware information when the boot process is done.
- Once the UCM6208 is successfully connected to network, the LED indicator for NETWORK in the front will be solid green and the LCD will display the IP address.
- (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax) to the FXS ports.
Getting Started
To get started with the UCM6200 setup process, use the following available interfaces: LCD display, LED indicators, and web portal.
- The LCD display shows hardware, software, and network information and can be navigated via the DOWN and OK buttons next to the display. From here, users can configure basic network settings, run diagnostic tests, and factory reset.
- The LED indicators at the front of the device provides interface connection and activity status.
- 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 (e.g., V1.4A), and IP address. Upon menu key press timeout (15 seconds), the screen will default back to this information. Pressing the DOWN button will show the system time.
- Menu
Pressing the OK button will show the main menu. All available menu options are found in [Table 3: LCD Menu Options].
- Menu Navigation
Pressing the DOWN button will scroll 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 UCM.
View Events |
|
Device Info |
|
Network Info |
|
Network Menu |
|
Factory Menu |
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.
Select Auto or On.
All On, All Off, and Blinking are the available options. Selecting Back in the
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.
Select Test SVIP to verify hardware connections within the device. The result will display on the LCD when the test is complete. |
Web Info |
|
SSH Switch |
SSH access is disabled by default |
Use the LED Indicators
The UCM6200 has LED indicators in the front to display connection status. The following table shows the status definitions.
LED Indicator | LED Status |
LAN WAN USB SD FXS (Phone/Fax) FXO (Telco Line) | Solid: Connected Flashing: Data Transferring OFF: Not Connected |
LED | LED Status |
NETWORK | Solid: Connected OFF: Not Connected |
ACT USB SD Phone (FXS) Line (FXO) | Solid: Connected Flashing: Data Transferring 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:
- Make sure your computer is on the same network as the UCM.
- Make sure that the UCM’s IP address is displayed on its LCD.
- Enter the UCM’s IP address into a web browsers’ address bar. The login page should appear (please see the above image).
- Enter default administrator username “admin” and password.
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.
Main Settings
There are 8 main sections in the web portal to manage various features of the UCM.
- System Status: Displays the dashboard, system information, current active calls, and network status.
- Extensions/Trunks: Manages extensions, trunks, and routing rules.
- Call Features: Manages various features of the UCM such as the IVR and voicemail.
- PBX Settings: Manages the settings related to PBX functionality such as SIP settings and interface settings.
- System Settings: Manages the settings related to the UCM system itself such as network and security settings.
- CDR: Contains the call detail records, statistics, and audio recordings of calls processed by the UCM.
- Value-Added Features: Manages the settings of features unrelated to core PBX functionality such as Zero Config provisioning and CRM/PMS integrations.
- Maintenance: Manages settings and logs related to system management and maintenance such as
user management, activity logs, backup settings, upgrade settings and troubleshooting tools.
Web GUI Languages
Currently the UCM6200 series Web GUI supports English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, Russian, Italian, Polish, German, Turkish and Czech.
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 appears to finalize the changes. If a modification requires a reboot, a prompt will appear asking to reboot the device.
Setting Up an Extension
Power on the UCM6200 and your SIP endpoint. Connect both devices to the same network and follow the steps below to set up an extension.
- Log into the UCM web portal and navigate to Extension/Trunk->Extensions
- 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.
- 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.
- 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.
- 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 UCM62xx. This includes settings for the following items: HTTP server, network, OpenVPN, DDNS, LDAP server and email server.
HTTP Server
The UCM6200’s embedded web server responds to HTTP/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.
Redirect From Port 80 | Toggles automatic redirection to UCM’s web portal from port 80. If disabled, users will need to manually add the UCM’s configured HTTP/HTTPS port to the server address when accessing the UCM web portal via browser. Default is “Enabled”. |
Protocol Type | Select either HTTP or HTTPS as the protocol to access the UCM’s HTTP server. This will also determine what is used when endpoints download config files from the UCM via Zero Config. Default is “HTTPS”. |
Port | Specifies the port number used to access the UCM HTTP server. Default is “8089”. |
If enabled, only the server addresses in whitelist will be able to access the UCM’s web portal. It is highly recommended to add the IP address currently used to access the UCM web page before enabling this option. Default is “Disabled”. | |
Permitted IP(s) | List of addresses that can access the UCM web portal. Ex: 192.168.6.233 / 255.255.255.255 |
Selects the method of acquiring SSL certificates for the UCM web server. Two methods are currently available: – Upload Certificate: Upload the appropriate files from one’s own PC. – Request Certificate: Enter the domain for which to request a certificate for from Let’s Encrypt. | |
TLS Private Key | Uploads the private key for the HTTP server. Note: Key file must be under 2MB in file size and in *.pem format. File name will automatically be changed to “private.pem”. |
TLS Cert | Uploads the certificate for the HTTP server. Note: Certificate must be under 2MB in file size and in *.pem format. This will be used for TLS connections and contains private key for the client and signed certificate for the server. |
Domain |
Enter the domain to request the certificate for and click on
|
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 UCM6200 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.
- UCM6200 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 UCM6202, UCM6204 and UCM6208, respectively.
Table 7: UCM6200 Network Settings🡪Basic Settings
Method | Select “Route”, “Switch” or “Dual” mode on the network interface of UCM6200. The default setting is “Route”.
WAN port will be used for uplink connection. LAN port will function similarly to a regular router port.
WAN port will be used for uplink connection. LAN port will be used as a bridge for connections.
Both WAN and LAN ports will be used for uplink connections labeled as LAN2 and LAN1, respectively. The port selected as the Default Interface will need to have a gateway IP address configured if it is using a static IP. |
Specifies the maximum transmission unit value. Default is 1500. | |
IPv4 Address | |
Preferred DNS Server | If configured, this will be used as the Primary DNS server. |
WAN (when “Method” is set to “Route”) | |
IP Method | Select DHCP, Static IP, or PPPoE. The default setting is DHCP. |
IP Address | Enter the IP address for static IP settings. The default setting is 192.168.0.160. |
Subnet Mask | Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0. |
Gateway IP | Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0. |
DNS Server 1 | Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0. |
DNS Server 2 | Enter the DNS server 2 address for static IP settings. |
Username | Enter the username to connect via PPPoE. |
Password | Enter the password to connect via PPPoE. |
Layer 2 QoS 802.1Q/VLAN Tag | Assign the VLAN tag of the layer 2 QoS packets for WAN port. The default value is 0. |
Layer 2 QoS 802.1p Priority Value | Assign the priority value of the layer 2 QoS packets for WAN port. The default value is 0. |
LAN (when Method is set to “Route”) | |
IP Address | Enter the IP address assigned to LAN port. The default setting is 192.168.2.1. |
Subnet Mask | Enter the subnet mask. The default setting is 255.255.255.0. |
DHCP Server Enable | Enable or disable DHCP server capability. The default setting is “Yes”. |
DNS Server 1 | Enter DNS server address 1. The default setting is 8.8.8.8. |
DNS Server 2 | Enter DNS server address 2. The default setting is 208.67.222.222. |
Allow IP Address From | Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100. |
Allow IP Address To | Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254. |
Default IP Lease Time | Enter the IP lease time (in seconds). The default setting is 43200. |
LAN (when Method is set to “Switch”) | |
IP Method | Select DHCP, Static IP, or PPPoE. The default setting is DHCP. |
IP Address | Enter the IP address for static IP settings. The default setting is 192.168.0.160. |
Subnet Mask | Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0. |
Gateway IP | Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0. |
DNS Server 1 | Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0. |
DNS Server 2 | Enter the DNS server 2 address for static IP settings. |
Username | Enter the username to connect via PPPoE. |
Password | Enter the password to connect via PPPoE. |
Layer 2 QoS 802.1Q/VLAN Tag | Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 0. |
Layer 2 QoS 802.1p Priority Value | Assign the priority value of the layer 2 QoS packets for LAN port. The default value is 0. |
LAN 1 / LAN 2 (when Method is set to “Dual”) | |
Default Interface | If “Dual” is selected as the Method, users must select either LAN 1 (WAN port) or LAN 2 (LAN port) to be used as the default interface. Default setting is LAN 2. |
IP Method | Select DHCP, Static IP, or PPPoE. The default setting is DHCP. |
IP Address | Enter the IP address for static IP settings. The default setting is 192.168.0.160. |
Subnet Mask | Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0. |
Gateway IP | Enter the gateway IP address for static IP settings when the port is assigned as default interface. The default setting is 0.0.0.0. |
DNS Server 1 | Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0. |
DNS Server 2 | Enter the DNS server 2 address for static IP settings. |
Username | Enter the username to connect via PPPoE. |
Password | Enter the password to connect via PPPoE. |
Layer 2 QoS 802.1Q/VLAN Tag | Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 0. |
Layer 2 QoS 802.1p Priority Value | Assign the priority value of the layer 2 QoS packets for LAN port. The default value is 0. |
WAN (when “Method” is set to “Route”) | |
IP Method | Select Auto or Static. The default setting is Auto |
IP Address | Enter the IP address for static IP settings. |
IP Prefixlen | Enter the Prefix length for static settings. Default is 64 |
DNS Server 1 | Enter the DNS server 1 address for static settings. |
DNS Server 2 | Enter the DNS server 2 address for static settings. |
LAN (when Method is set to “Route”) | |
DHCP Server | Select Disable, Auto or DHCPv6. Disable: Disables the DHCPv6 server. Auto: Stateless address auto configuration using NDP protocol. DHCPv6: Stateful address auto configuration using DHCPv6 protocol. |
DHCP Prefix | Enter DHCP prefix. (Default is 2001:db8:2:2::) |
DHCP prefixlen | Enter the Prefix length for static settings. Default is 64 |
DNS Server 1 | Enter the DNS server 1 address for static settings. Default is (2001:4860:4860::8888 ) |
DNS Server 2 | Enter the DNS server 2 address for static settings. Default is (2001:4860:4860::8844 ) |
Allow IP Address From | Configure starting IP address assigned by the DHCP prefix and DHCP prefixlen. |
Allow IP Address To | Configure the ending IP address assigned by the DHCP Prefix and DHCP prefixlen. |
Default IP Lease Time | Configure the lease time (in second) of the IP address. |
LAN (when Method is set to “Switch”) | |
IP Method | Select Auto or Static. The default setting is Auto |
IP Address | Enter the IP address for static IP settings. |
IP Prefixlen | Enter the Prefix length for static settings. Default is 64 |
DNS Server 1 | Enter the DNS server 1 address for static settings. |
DNS Server 2 | Enter the DNS server 2 address for static settings. |
LAN 1 / LAN 2 (when Method is set to “Dual”) | |
Default Interface | Users must select either LAN 1 (WAN port) or LAN 2 (LAN port) to be used as the default interface. Default setting is LAN 2. |
IP Method | Select Auto or Static. The default setting is Auto |
IP Address | Enter the IP address for static IP settings. |
IP Prefixlen | Enter the Prefix length for static settings. Default is 64 |
DNS Server 1 | Enter the DNS server 1 address for static settings. |
DNS Server 2 | Enter the DNS server 2 address for static settings. |
- Method: Route
When the UCM6200 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 the 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.
DHCP Client List
This page lists all the detected devices on the LAN and the IP addresses that were provided to them. Additionally, users can manually link a MAC address to an IP address.
When devices receive IP addresses from the UCM’s DHCP server, they will be listed in the System Settings->Network Settings->DHCP Client List page as shown below.
To manually add and bind a MAC address to an IP address, click on
. The following menu will then be displayed.
Enter the device’s MAC address and the IP address to bind it to. This IP address must be in the UCM’s DHCP range.
To bind multiple existing MAC addresses that are in the list to their respective IP addresses, check the boxes next to each MAC address and click on the
button. A confirmation message will appear on the screen. Click
to bind the addresses.
The Bind Status for the selected MAC addresses should now be changed from “Unbound” to “Bound”.
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 UCM6200 supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure show UCM6200 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 UCM6200. 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 as the 802.1X mode, users will also need to upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated from the RADIUS server.
Table 8: UCM6200 Network Settings🡪802.1X
802.1X Mode | Select 802.1X mode. The default setting is “Disable”. The supported 802.1X modes are:
|
Identity | Enter 802.1X mode Identity information. |
MD5 Password | Enter 802.1X mode MD5 password information. |
802.1X 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 UCM6200 provides users static routing capability that allows the device to use manually configured routes, instead of dynamically assigned routes or the default gateway. Static routes can be used to define a route when no others are available or can serve as complementary routes alongside existing routes as failover routes, with existing routing on the UCM6200 as a failover backup, etc.
-
Click on
to create a new IPv4 static route or click on
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.
Table 9: UCM6200 Network Settings🡪Static Routes
Destination | Configure the destination IPv4 address or the destination IPv6 subnet for the UCM6200 to reach using the static route. Example: IPv4 address – 192.168.66.4 IPv6 subnet – 2001:740:D::1/64 |
Netmask | 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 UCM6200 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 UCM6200 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 UCM6204.
The network topology of the above diagram is as below:
- Network 192.168.69.0 has IP phones registered to UCM6204 LAN 1 address
- Network 192.168.40.0 has IP phones registered to UCM6204 LAN 2 address
- Network 192.168.66.0 has IP phones registered to UCM6204 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 UCM6204. Therefore, we need configure a static route on the UCM6204 so that the phones in isolated networks can make calls between each other.
Port Forwarding
The UCM network interface supports router functionality that allows users to configure port forwarding. If the UCM is set to “Route” in System Settings->Network Settings->Basic Settings->Method, the Port Forwarding tab will be available.
WAN Port | Specify the WAN port number or a range of WAN ports. Unlimited number of ports can be configured. Note: When set to a range, both the WAN port and LAN port must be configured with the same range (e.g., WAN port: 1000-1005, LAN port: 1000-1005). Access from the WAN port will be forwarded to the LAN port with the same port number. |
LAN IP | Specify the LAN IP address. |
LAN Port | Specify the LAN port number or a range of LAN ports. Please see the note for WAN Port. |
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”. |
Here is an example of an environment and how to configure port forwarding to an endpoint’s web portal.
- Configure the UCM as a router by selecting “Route” in System Settings->Network Settings->Basic Settings->Method.
- WAN port is connected to an uplink switch with a public IP address configured (e.g., 1.1.1.1).
- LAN port provides a DHCP pool for devices on the LAN network (gateway address is 192.168.2.1 by default).
- Connect a GXP2160 to the UCM’s LAN network. It should obtain an IP address from UCM’s DHCP pool.
- While still on the Network Settings page, navigate to the Port Forwarding tab.
-
Click on button to start setting up port forwarding.
WAN Port: Enter the port that will be opened on the WAN side to allow access.
LAN IP: Enter the GXP2160’s IP address.
LAN Port: Enter the port that will be opened on the GXP2160 to allow access.
Protocol Type: Select the protocol to use for this port forwarding.
This will allow users to access the GXP2160 web portal from an external network.
OpenVPN®
The UCM can be configured as an OpenVPN client.
Table 11: UCM6200 System Settings🡪Network Settings🡪OpenVPN®
DDNS Settings
Configuring UCM DDNS settings will allow the UCM to be accessed via domain name instead of IP address. UCM supports the following DDNS services:
- dydns.org
- noip.com
- freedns.afraid.org
- zoneedit.com
- oray.net
Here is an example of using noip.com for DDNS.
- The UCM must be publicly accessible in order to work with the service provider.
- Navigate to System Settings->Network Settings->DDNS Settings, check the Enable DDNS option, select your service provider, and configure all the displayed fields.
- You can now use the configured domain to access the UCM web portal.
Security Settings
The UCM offers several methods of protection against malicious attacks and unauthorized access such as firewall rules, connection thresholds, and Fail2ban.
To get started on configuring security settings, navigate to the System Settings🡪Security Settings page.
Static Defense
On the Static Defense page, users can configure firewall rules and view the ports used by various UCM services and processes.
The following table shows a few examples of the information available on the Static Defense page.
Table 12: UCM6200 Firewall🡪Static Defense🡪Current Service
Port | Process | Type | Protocol or Service |
7777 | Asterisk | TCP/IPv4 | SIP |
389 | Slapd | TCP/IPv4 | LDAP |
2000 | Asterisk | TCP/IPv4 | SCCP |
22 | Dropbear | TCP/IPv4 | SSH |
80 | Lighthttpd | TCP/IPv4 | HTTP |
8089 | Lighthttpd | TCP/IPv4 | HTTPS |
69 | Opentftpd | UDP/IPv4 | TFTP |
9090 | Asterisk | UDP/IPv4 | SIP |
6060 | zero_config | UDP/IPv4 | UCM6200 zero_config service |
5060 | Asterisk | UDP/IPv4 | SIP |
4569 | Asterisk | UDP/IPv4 | SIP |
5353 | zero_config | UDP/IPv4 | UCM6200 zero_config service |
37435 | Syslogd | UDP/IPv4 | Syslog |
For general firewall defense mechanisms, UCM supports Ping Defense, SYN-Flood Defense, and Ping-of-Death Defense. These can be configured separately for the WAN interface and LAN interface.
Table 13: Typical Firewall Settings
Ping Defense Enable | If enabled, the UCM will not send ICMP messages in response to ping requests. |
If enabled, the UCM will limit the amount of SYN packets accepted from one source to 10 packets per second, preventing the UCM web portal from becoming inaccessible. Excess packets will be discarded. There is no need to mention the WAN and LAN parts since it is already mentioned in sentence before the table. | |
Ping-of-Death Defense Enable | If enabled, the UCM will not be affected by ping of death attacks. |
In the Custom Firewall Settings section, users can create rules to accept, reject, or drop specific traffic going through the UCM. To create a new rule, click on the
button.
The following menu will appear. Here is an example firewall rule created to reject SSH access to the UCM from the WAN interface.
Rule Name | Enter a name for the firewall rule. |
Action | Select the action for the Firewall to perform.
|
Type | Select the traffic type.
If selected, the Interface field will appear. Users must specify the interface that the inbound rule will be applied to.
|
Service | Select the connection type the rule will apply to.
If selected, users must configure the Source IP and Port,Destination IP Address and Port, and the Protocol fields that appear. If the source and destination are left blank, the “Anywhere” values will 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
On the Dynamic Defense page, users can configure the UCM to monitor incoming TCP connections and prevent excessive traffic from hosts. The UCM must have “Route” configured in the System Settings->Network Settings->Basic Settings page. The blacklist on this page is automatically updated.
The following options are available:
Table 15: UCM6200 Firewall Dynamic Defense
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 UCM6200 within 1 minute, it will be added into UCM6200 blacklist.
- This host 192.168.5.7 will be blocked by the UCM6200 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 UCM6200 within 1 minute, it will not be added into UCM6200 blacklist. It can still establish TCP connection with the UCM6200.
Fail2ban
Fail2Ban feature on the UCM6200 provides intrusion detection and prevention for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE and prevents SIP brute force attacks on the PBX system.
Once an IP address exceeds the allowed number of login or SIP authentication attempts within the configured “Max Retry Duration” period, all SIP and HTTP requests from that IP address will be dropped, forbidding web access and blocking further authentication attempts.
Global Settings | |
Enable Fail2Ban | Enable Fail2Ban. The default setting is disabled. Please make sure both “Enable Fail2Ban” and “Asterisk Service” are turned on to use Fail2Ban for SIP authentication on the UCM6200. |
Banned Duration | Configure the duration (in seconds) for the detected host to be banned. The default setting is 600. If set to 0, the host will be always banned. |
Max Retry Duration | If a host exceeds the maximum allowed number attempts configured for Max Retry within the configured Max Retry Duration window, the host will be banned. The default setting is 600 seconds. |
MaxRetry | Configures the maximum number of allowed authentication failures within the configured Max Retry Duration window. The default setting is 5. |
Configures the IP addresses, CIDR masks, and DNS hosts in the Fail2Ban whitelist. Whitelisted entries will not be banned by Fail2Ban even after exceeding the allowed number of authentication failures. Up to 20 addresses can be added. | |
Local Settings | |
Asterisk Service | Enable Asterisk service for Fail2Ban. The default setting is disabled. Please make sure both “Enable Fail2Ban” and “Asterisk Service” are turned on to use Fail2Ban for SIP authentication on the UCM6200. |
Listening Port Number | Configure the listening port number for the service. By default, port 5060 will be used for UDP and TCP, and port 5061 will be used for TLS. |
MaxRetry | Configures the maximum number of authentication failures before the host is banned. The default setting is 10. Please note that this will override the Global Settings->MaxRetry setting. |
Enables defense against excessive login attacks to the UCM’s web GUI. The default setting is disabled. | |
Listening Port Number | This is the Web GUI listening port number which is configured under System Settings🡪HTTP Server🡪Port. The default is 8089. |
MaxRetry | Configures the maximum allowed number of failed login attempts from an IP address before it is added to the Fail2Ban blacklist. |
Blacklist | |
Users will be able to view the IPs that have been blocked by UCM. |
TLS Security
TLS security to specify minimum and maximum TLS versions supported by the UCM.
Please log in UCM62xx web interface and go to System Settings🡪Security Settings🡪TLS Security.
By default, minimum TLS version is set to TLS1.1, and maximum TLS version is set to TLS1.2.
Supported versions are 1.0, 1.1 and 1.2
SSH Access
SSH access can be toggled from the UCM’s webUI and physical LCD screen. The webUI option can be found under System Settings->Security Settings->SSH Access. SSH access is disabled by default and should only be turned on for troubleshooting and debugging.
LDAP Server
The UCM6200 has an embedded LDAP/LDAPS server for users to manage corporate phonebooks in a centralized manner.
- The LDAP server automatically generates an initial phonebook with PBX DN “ou=pbx,dc=pbx,dc=com” based on the UCM6200’s existing extensions.
- 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 UCM6200 LDAP server have the same Base DN “dc=pbx,dc=com”.
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 UCM6200, 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 UCM6200. If the UCM6200 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 UCM6200 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 UCM6200.
The UCM6200 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” (limited with 64 and 32 characters respectively) 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.
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 first phonebook with default phonebook dn “ou=pbx,dc=pbx,dc=com” displayed on the LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly. The contacts information will need to be modified via Web GUI🡪Extension/Trunk🡪Extensions first. The default LDAP phonebook will then be updated automatically.
- 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 an example of CSV file. Please note “Account Number” and “Phonebook DN” fields are required. Users can export a phonebook file from the UCM’s LDAP phonebook section and use it as a template.
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 UCM6200 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 UCM6200 LDAP Phonebook, the contacts in the CSV file will be added into the existing phonebook. If the phonebook DN does not exist on the UCM6200 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 UCM6200.
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 that have “pbx” in the Phonebook DN field of the CSV file.
- Export phonebook to your computer from UCM6200 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 UCM6200 again.
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:
- LDAP Server: Enter a name for the remote LDAP server
- Server Address: Enter the IP address or domain name for remote LDAP server.
- Base DN: dc=pbx,dc=com
- Username: Enter username if authentication is required. This field cannot exceed 64 characters and can contain space.
- Password: Enter password if authentication is required.
- Filter: Enter the filter. Ex: (|(CallerIDName=%)(AccountNumber=%))
- Port: Enter the port number. Ex:389
- LDAP Name Attributes: Enter the name attributes for remote server
- LDAP Number Attributes: Enter the number attributes for remote server
- Client type: Protocol of LDAP or LDAPS.
- LDAP Client CA cert: Upload LDAP client CA certificate, The following file types are supported: .crt .der and .pem.
- LDAP Client Private Key: Upload LDAP client private key.
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 UCM6200
- 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 33: LDAP Server Configurations].
Time settings
Automatic Date and Time
The current system time can be found under System Status->Dashboard and System Information.
To configure the UCM to automatically update time, navigate to System Settings->Time Settings->Automatic Date and Time.
Note:
The configurations under Web GUI🡪Settings🡪Time Settings🡪Automatic Date and Time page require reboot to take effect. Please consider configuring auto time updating related changes when setting up the UCM6200 for the first time to avoid service interrupt after installation and deployment in production.
Specify the server address of the NTP server for the UCM6510 to sync date and time with. The default NTP server is pool.ntp.org. | |
Enable DHCP Option 2 | If set to “Yes”, the UCM’s time zone can be provisioned via DHCP Option 2 from a local server automatically. |
Enable DHCP Option 42 | If set to “Yes”, the UCM’s NTP server can be provisioned via DHCP Option 42 from a local server automatically. This will override the manually configured NTP server. The default setting is “Yes”. |
Select the time zone for the UCM to use. If “Self-Defined Tome Zone” is selected, please specify the time zone parameters in “Self-Defined Time Zone” field as described in below option. | |
Self-Defined Time Zone | If “Self-Defined Time Zone” is selected in “Time Zone” option, users will need define their own time zone following the format below.
The syntax is: std offset dst [offset], start [/time], end [/time] Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0 MTZ+6MDT+5 This indicates a time zone with 6 hours offset and 1 hour ahead for DST, which is U.S central time. If it is positive (+), the local time zone is west of the Prime Meridian (A.K.A: International or Greenwich Meridian); If it is negative (-), the local time zone is east. M4.1.0,M11.1.0 The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec). The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd Tuesday…). Normally 1, 2, 3, 4 are used. If 5 is used, it means the last iteration of the weekday. The 3rd number indicates weekday: 0,1,2,..,6 ( for Sun, Mon, Tues, … ,Sat). Therefore, this example is the DST which starts from the First Sunday of April to the 1st Sunday of November. |
Set Time Manually
To manually set the time on the UCM6200, go to Web GUI🡪System Settings🡪Time Settings🡪Set Date and Time. The format is YYYY-MM-DD HH:MM:SS.
Note:
Manually setup time will take effect immediately after saving and applying change in the Web GUI. If users would like to reboot the UCM6200 and keep the manually setup time setting, please make sure “Remote NTP Server”, “Enable DHCP Option 2” and “Enable DHCP Option 42” options under Web GUI🡪Settings🡪Time Settings🡪Auto Time Updating page are unchecked or set to empty. Otherwise, time auto updating settings in this page will take effect after reboot.
NTP Server
The UCM can act as an NTP server for clients to sync their system times with. To configure this, navigate to System Settings->Time Settings->NTP Server and set Enable NTP Server to “Yes”. On the client side, use the UCM’s IP address or hostname as the NTP server address.
Office Time
The system administrator can define office hours which can be used as conditions for call forwarding and inbound routing. To configure this, navigate to System Settings->Time Settings->Office Time and click on the Add button to create office hours.
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 Selected Office Times” to delete multiple selected office times at once.
Holiday
System administrators can define holidays which can be used as conditions for call forwarding and inbound routing. To configure this, navigate to System Settings->Time Settings->Holiday and click on the Add button to create a new holiday.

Enter holiday “Name” and “Holiday Memo” for the new holiday. Then select “Month” and “Day”. The system administrator can also define days in one week as advanced options. Once done, click on “Save” and then “Apply Change” for the holiday to take effect. The holiday will be listed in the web page as the figure shows below.
-
Click on
to edit the holiday.
-
Click on
to delete the holiday.
- Click on “Delete Selected Holidays” to delete multiple selected holidays at once.
Note:
For more details on how to use office time and holiday, please refer to the link below:
https://documentation.grandstream.com/knowledge-base/managing-office-time-holidays/
Email Settings
Email settings
The UCM’s email module can send out alert event emails, fax (Fax-to-Email), voicemail (Voicemail-to-Email), etc. Email settings can be configured in System Settings->Email Settings.
TLS Enable | Enable or disable TLS during transferring/submitting your Email to another SMTP server. The default setting is “Yes”. |
Type | Select Email type.
|
Domain | Specify the domain name to be used in the Email when using type “MTA”. |
Server | Specify the SMTP server when using type “Client”. |
Username is required when using type “Client”. This is typically the email address. | |
Password for the entered Username. | |
Toggles the Email-to-Fax feature. If enabled, the UCM will monitor the configured email inbox (using provided [Username] and [Password]) for emails with the subject “SendFaxMail To XXX” or “XXX”. Subject example: SendFaxMail To 7200 Or 7200. The UCM will extract the attachments of detected emails and send it to the XXX extension by fax. The attachment must be in PDF/TIF/TIFF format. Note: This field will appear when using Type “Client”. | |
Enables the Email to fax Blacklist/Whitelist functionality. | |
Specify the Email subject format for fax sending, the subject can be either “SendFaxMail To XXX” or “XXX” with XXX the fax number. | |
Internal Blacklist/Whitelist | Specify the Email address blacklist/whitelist for local extensions. This feature prevents faxing from unauthorized email addresses. The internal list includes only contacts with local extensions. |
External Blacklist/Whitelist | Specify the Email address blacklist/whitelist for non-local contacts. This feature prevents faxing from unauthorized email addresses. The external list is for non-local contacts. Note: Multiple addresses can be separated with semicolon (;) i.e. ” XXX;YYY “. |
POP/POP3 Server Address |
Configure the POP/POP3 server address for the configured username
|
POP/POP3 Server Port |
Configure the POP/POP3 server port for the configured username
|
Display Name | Specify the display name in the FROM header in the Email. |
Sender | Specify the sender’s Email address. For example: pbx@example.mycompany.com. |
The following figure shows example email settings, where 192.168.6.202 is the SMTP server.
Once configuration is complete, click on the Save button first and then the Test button to verify that the settings work.
Email Templates
The UCM provides various email templates for different email notifications. Email templates can be accessed from System Settings->Email Settings->Email Templates.
Press on
to upload pictures to be used on email templates.
Press
to reset all email templates to default ones.
To configure the email template, click the
button under Options column, and edit the template as desired.
-
Users have the ability to preview mail sample by clicking on
.
-
Click on
in order to restore the default email template.
-
Finally, users can click on
to upload a custom picture to the email template to display their own logo in the sent mails for example
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:
|
Email logs will be shown on bottom of the “Email Send Log” page, as shown on the following figure.
Recordings Storage
The UCM supports both automatic and manual call recordings. Recordings can be saved to UCM’s local storage, external storage (SD/USB disks), and NAS.
- If Enable Auto Change is selected, the recording storage location will automatically change to NAS, USB Disk, or SD card if they are available. If all storage location types are available, the priority will be NAS->USB Disk->SD Card->Local storage.
- If “Local” is selected, the recordings will be stored in UCM62xx internal storage.
- If “USB Disk” or “SD Card” is selected, the recordings will be stored in the corresponding plugged in external storage device. Please note the options “USB Disk” and “SD Card” will be displayed only if they are plugged into the UCM62xx.
Once “USB Disk” or “SD Card” is selected, click on “OK”. The user will be prompted to confirm to copy the local files to the external storage device.
Click on “OK” to continue. The users will be prompted a new dialog to select the categories for the files to be copied over.
On the UCM62xx, recording files are generated and exist in 3 categories: normal call recording files, conference recording files, and call queue recording files. Therefore, users have the following options when select the categories to copy the files to the external device:
- Recording Files: Copy the normal recording files to the external device.
- Conference: Copy the conference recording files to the external device.
- Queue: Copy the call queue recording files to the external device.
- All: Copy all recording files to the external device.
Provisioning
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 UCM6200 provides a Plug and Play mechanism to auto-provision the Grandstream SIP devices in a zero-configuration manner by generating XML config file and having the phone to download it within LAN area. This allows users to finish the installation with ease and properly manage SIP devices.
To provision a phone, three steps are involved, i.e., discovery, configuration and provisioning. This section explains how Zero Config works on the UCM6200. The settings for this feature can be accessed via Web GUI🡪Value-added Features🡪Zero Config.
Configuration Architecture for End Point Device
Started from firmware version 1.0.7.10, the end point device configuration in zero config 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 Grandstream 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 individual device here.
Each layer also has its own structure in different levels. Please see figure below. The details for each layer are explained in sections [Global configuration], [Model configuration] and [Device Configuration].

The configuration options in model layer and device layer have all the option in global layers already, i.e., the options in global layer is a subset of the options in model layer and device layer. If an option is set in all three layers with different values, the highest layer value will override the value in lower layer. For example, if the user selects English for Language setting in Global Policy and Spanish for Language setting in Default Model Template, the language setting on the device to be provisioned will use Spanish as model layer has higher priority than global layer. To sum up, configurations in higher layer will always override the configurations for the same options/fields in the lower layer when presented at the same time.
After understanding the zero-config configuration architecture, users could configure the available options for end point devices to be provisioned by the UCM6200 by going through the three layers. This configuration architecture allows users to set up and manage the Grandstream end point devices in the same LAN area in a centralized way.
Auto Provisioning Settings
By default, the Zero Config feature is enabled on the UCM6200 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 UCM6200 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 UCM6200 and take the new configuration.
- DHCP OPTION 66
Route mode need 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 UCM6200 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 UCM6200.
- mDNS
When the phone boots up, it sends out mDNS query to get the TFTP server address. The UCM6200 will respond with its own address. The phone will then send TFTP request to download the XML config file from the UCM6200.
To start the auto provisioning process, under Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config Settings, fill in the auto provision information.
Table 22: Auto Provision Settings
Enable Zero Config | Enable or disable the zero-config feature on the PBX. The default setting is enabled. |
Enable Automatic Configuration Assignment | By default, this is disabled. If disabled, when SIP device boots up, the UCM6200 will not send the SIP device the URL to download the config file and therefore the SIP device will not be automatically provisioned by the UCM6200. Note: When disabled, SIP devices can still be provisioned by manually sending NOTIFY from the UCM6200 which will include the XML config file URL for the SIP device to download. |
Automatically Assign Extension | If enabled, when the device is discovered, the PBX will automatically assign an extension within the range defined in “Zero Config Extension Segment” to the device. The default setting is disabled. |
Zero Config Extension Segment | Click on the link “Zero Config Extension Segment” to specify the extension range to be assigned if “Automatically Assign Extension” is enabled. The default range is 5000-6299. Zero Config Extension Segment range can be defined in Web GUI🡪PBX Settings🡪General Settings🡪General page🡪Extension Preference section: “Auto Provision Extensions”. |
Enable Pick Extension | If enabled, the extension list will be sent out to the device after receiving the device’s request. This feature is for the GXP series phones that support selecting extension to be provisioned via phone’s LCD. The default setting is disabled. |
Pick Extension Segment | Click on the link “Pick Extension Segment” to specify the extension list to be sent to the device. The default range is 4000 to 4999. Pick Extension Segment range can be defined in Web GUI🡪PBX Settings🡪General Settings🡪General page🡪Extension Preference section: “Pick Extensions”. |
Pick Extension Period (hour) | Specify the number of minutes to allow the phones being provisioned to pick extensions. |
This feature allows the UCM to provision devices in different subnets other than UCM network. Enter subnets IP addresses to allow devices within these subnets to be provisioned. The syntax is <IP>/<CIDR>. Examples: 10.0.0.1/8 192.168.6.0/24 Note: Only private IP ranges (10.0.0.0 | 172.16.0.0 | 192.168.0.0) are supported. |
Please make sure an extension is manually assigned to the phone or “Automatically Assign Extension” is enabled during provisioning. After the configuration on the UCM6200 Web GUI, click on “Save” and “Apply Changes”. Once the phone boots up and picks up the config file from the UCM6200, it will immediately apply the configuration.
Discovery
Grandstream endpoints are automatically discovered after bootup. Users could also manually discover device by specifying the IP address or scanning the entire LAN network. Three methods are supported to scan the devices.
- PING
- ARP
- SIP Message (NOTIFY)
Click on “Auto Discover” under Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config, fill in the “Scan Method” and “Scan IP”. The IP address segment will be automatically filled in based on the network mask detected on the UCM6200. If users need scan the entire network segment, enter 255 (for example, 192.168.40.255) instead of a specific IP address. Then 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 UCM6200 Web GUI🡪Value-added Features🡪Zero Config🡪Auto Provisioning Settings.
The following figure shows a list of discovered phones. The MAC address, IP Address, Extension (if assigned), Version, Vendor, Model, Connection Status, Create Config, Options (Edit /Delete /Update /Reboot /Access Device WebGUI) are displayed in the list.
Auto Discover can also search for devices located on other subnets, in condition for the subnet to be added under Zero Config Settings> Subnet Whitelist. The method allowed to auto discover other subnets then the UCM’s is SIP-Message like shown below.
Uploading Devices List
Besides the built-in discovery method on the UCM, users could prepare a list of devices on .CSV file and upload it by clicking on the button
, 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
- 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.
- All: Display all discovered devices.
- Scan Results: Display only manually discovered devices. [Discovery]
- IP Address: Enter device IP and press Search button.
- MAC Address: Enter device MAC and press Search button.
- Model: Enter a model name and press Search button. Example: GXP2130.
From the main menu of zero config, users can perform the following operations:
-
Click on
in order to access to the discovery menu as shown on [Discovery] section.
-
Click on
to add a new device to zero config database using its MAC address.
-
Click on
to delete selected devices from the zero-config database.
-
Click on
to modify selected devices.
-
Click on
to batch update a list of devices, the UCM on this case will send SIP NOTIFY message to all selected devices in order to update them at once.
-
Click on
to reboot selected devices (the selected devices, should have been provisioned with extensions since the phone will authenticate the server which is trying to send it reboot command).
-
Click on
to clear all devices configurations.
-
Click on
to upload CSV file containing list of devices.
-
Click on
to copy configuration from one device to another. This can be useful for easily replace devices and note that this feature works only between devices of same model.
All these operations will be detailed on the next sections.
Global configuration
Global policy
Global configuration will apply to all the connected Grandstream SIP end point devices in the same LAN with the UCM6200 no matter what the Grandstream device model it is. It is divided into two levels:
- Web GUI🡪Value-added Features🡪Zero Config🡪Global Policy
- Web GUI🡪Value-added Features🡪Zero Config🡪Global Templates.
- Global Templates configuration has higher priority to Global Policy configuration.
Global Policy can be accessed in Web GUI🡪Value-added Features🡪Zero Config🡪Global Policy page. On the top of the configuration table, users can select category in the “Options” dropdown list to quickly navigate to the category. The categories are:
- Localization: configure display language, data and time.
- Phone Settings: configure dial plan, call features, NAT, call progress tones and etc.
- Contact List: configure LDAP and XML phonebook download.
- Maintenance: configure upgrading, web access, Telnet/SSH access and syslog.
- Network Settings: configure 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.
Language settings | |
Language | Select the LCD display language on the SIP end device. |
Date and Time | |
Date Format | Configure the date display format on the SIP end device’s LCD. |
Time Format | Configure the time display in 12-hour or 24-hour format on the SIP end device’s LCD. |
NTP Server | Configure the URL or IP address of the NTP server. The SIP end device may obtain the date and time from the server. |
Time Zone | Configure the time zone used on the SIP end device. |
Default Call Settings | |
Dial Plan | Configure the default dial plan rule. For syntax and examples, please refer to user manual of the SIP devices to be provisioned for more details. |
Enable Call Features | When enabled, “Do Not Disturb”, “Call Forward” and other call features can be used via the local feature code on the phone. Otherwise, the ITSP feature code will be used. |
Use # as Dial Key | If set to “Yes”, pressing the number key “#” will immediately dial out the input digits. |
Auto Answer by Call-info | If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep, based on the SIP Call-Info header sent from the server/proxy. The default setting is enabled. |
NAT Traversal | Configure if NAT traversal mechanism is activated. |
User Random Port | If set to “Yes”, this parameter will force random generation of both the local SIP and RTP ports. |
General Settings | |
Call Progress Tones | Configure call progress tones including ring tone, dial tone, second dial tone, message waiting tone, ring back tone, call waiting tone, busy tone and reorder tone using the following syntax: f1=val, f2=val[, c=on1/ off1[- on2/ off2[- on3/ off3]]];
|
HEADSET Key Mode | Select “Default Mode” or “Toggle Headset/Speaker” for the Headset key. Please refer to user manual of the SIP devices to be provisioned for more details. |
Table 25: Global Policy Parameters – Contact List
LDAP Phonebook | |
Source | Select “Manual” or “PBX” as the LDAP configuration source.
|
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:
|
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 user manual for more details. |
Name Filter | Configure the filter used for name lookups. Please refer to 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 which are returned in the LDAP search result. Example: gn cn sn description |
Number Attribute | Specify the “number” attributes of each record which are returned in the LDAP search result. Example: telephoneNumber telephoneNumber Mobile |
Display Name | Configure the entry information to be shown on 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. 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 client waits for server to return. Valid range is 0 to 180. Default value is 30. |
Sort Results | Specify whether the searching result is sorted or not. 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 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 phonebook XML downloading.
Once selected, users need specify downloading protocol HTTP, HTTPS or TFTP and the server path to download the phonebook XML file. The server path could be IP address or URL, with up to 256 characters.
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 Minute). If set to 0, automatic download will be disabled. Valid range is 5 to 720. |
Remove manually-edited entries on download | If set to “Yes”, when XML phonebook is downloaded, manually added entries will be removed. |
Table 26: Global Policy Parameters – Maintenance
Upgrade and Provision | |
Firmware Source | UCM can provision the firmware source to devices. The following options are available: Select a source to get the firmware file:
If selected, complete the configuration for the following four parameters: “Upgrade Via”, “Server Path”, “File Prefix” and “File Postfix”.
Firmware can be uploaded to the UCM6200 internal storage for firmware upgrade. If selected, click on “Manage Storage” icon next to “Directory” option, upload firmware file and select directory for the end device to retrieve the firmware file.
If selected, the USB storage device needs to be plugged into the UCM6200 and the firmware file must be put under a folder named “ZC_firmware” in the USB storage root directory.
If selected, an SD card needs to be plugged into the UCM6200 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 | When URL is selected as firmware source, configure the firmware file prefix. If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone, if URL is selected as firmware source. |
File Postfix | 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.
Once selected, specify the day of the week to check HTTP/TFTP server for firmware upgrades or configuration files changes.
Once selected, specify the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes.
Once selected, specify the interval X that the SIP end device will request for new firmware every X minutes. |
Firmware Upgrade Rule | Specifies how firmware upgrade and provision requests will be sent. |
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. |
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 Server | Configure the URL/IP address for the syslog server. |
Syslog Level | Select the level of logging for syslog. |
Send SIP Log | Configures whether SIP messages will be included in the syslog. |
Table 27: Global Policy Parameters – Network Settings
Basic Settings | |
IP Address | Configures how the SIP endpoint will obtain IP addresses. The following options are available:
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.
Once selected, users need specify the Account ID, Password and Service Name for PPPoE. |
Advanced Setting | |
Layer 3 QoS | Define the Layer 3 QoS parameter. This value is used for IP Precedence, Diff-Serv or MPLS. Valid range is 0-63. |
Layer 2 QoS Tag | Assign the VLAN Tag of the Layer 2 QoS packets. Valid range is 0 -4095. |
Layer 2 QoS Priority Value | Assign the priority value of the Layer 2 QoS packets. 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 Interval | Specify how often the phone will send a blank UDP packet to the SIP server in order to keep the “ping hole” on the NAT router to open. Valid range is 10-160. |
Table 28: Global Policy Parameters – Customization
Wallpaper | |
Screen Resolution 1024 x 600 | Check this option if the SIP endpoint uses a 1024×600 resolution wallpaper.
Configure the location where wallpapers are stored.
If “URL” is selected as source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as source, click to upload wallpaper file to the UCM6200. |
Screen Resolution 800 x 400 | Check this option if the SIP endpoint uses a 800×400 resolution wallpaper.
Configure the location where wallpapers are stored.
If “URL” is selected as source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as source, click to upload wallpaper file to the UCM6200. |
Screen Resolution 480 x 272 | Check this option if the SIP endpoint uses a 480×272 resolution wallpaper.
Configure the location where wallpapers are stored.
If “URL” is selected as source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as source, click to upload wallpaper file to the UCM6200. |
Screen Resolution 320 x 240 | Check this option if the SIP endpoint uses a 320×240 resolution wallpaper.
Configure the location where wallpapers are stored.
If “URL” is selected as source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as source, click to upload wallpaper file to the UCM6200. |
Table 29: Global Policy Parameters – Communication Settings
Email Settings | |
SMTP Settings | Check this option to configure the email settings that will be sent to the provisioned phones:
IP address of the SMTP server
SMTP server port
Email address
Username of the sender
Email where recovered password will be sent
Email address where alarms notifications will be sent
Email address where alarms notifications will be sent
Enable SSL protocol for SMTP |
FTP | |
FTP | Check this option to configure the FTP settings that will be sent to the provisioned phones:
Either FTP or Central Storage
FTP server address
FTP port to be used
FTP username
FTP Directory path |
Global Templates
Global Templates can be accessed in Web GUI🡪Value-added Features🡪Zero Config🡪Global Templates. Users can create multiple global templates with different sets of configurations and save the templates. Later on, when the user configures the device in Edit Device dialog🡪Advanced Settings, the user 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 global template, users can select the categories and the parameters under each category to be used in the template. The global policy and the selected global template will both take effect when generating the config file. However, the selected global template has higher priority to the global policy when it comes to the same setting option/field. If the same option/field has different value configured in the global policy and the selected global template, the value for this option/field in the selected global template will override the value in global policy.
Click on “Create New Template” to add a global template. Users will see the following configurations.
Template Name | Create a name to identify this global template. |
Description | Provide a description for the global template. This is optional. |
Active | Check this option to enable the global template. |
-
Click on
to edit the global template.
The window for editing global template is shown in the following figure. In the “Options” field, after entering the option name key word, the options containing the key word will be listed. Users could then select the options to be modified and click on “Add Option” to add it into the global template.
The added options will show in the list. Users can then enter or select value for each option to be used in the global template. On the left side of each added option, users can click on
to remove this option from the template. On the right side of each option, users can click on
to reset the option value to the default value.
Click on “Save” to save this global template.
-
The created global templates will show in the Web GUI🡪Value-added Features🡪Zero Config🡪Global Templates page. Users can click on
to delete the global template or click on “Delete Selected Templates” to delete multiple selected templates at once.
- Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected templates.
Model configuration
Model templates
Model layer configuration allows users to apply model-specific configurations to different devices. Users could create/edit/delete a model template by accessing Web GUI, page Value-added Features🡪Zero Config🡪Model Templates. If multiple model templates are created and enabled, when the user configures the device in Edit Device dialog🡪Advanced Settings, the user can select to use one of the model templates for the device. Please refer to section [Manage Devices] for more details on using the model template.
For each created model template, users can assign it as default model template. If assigned as default model template, the values in this model template will be applied to all the devices of this model. There is always only one default model template that can be assigned at one time on the UCM6200.
The selected model template and the default model template will both take effect when generating the config file for the device. However, the model template has higher priority to default model template when it comes to the same setting option/field. If the same option/field has different value configured in the default model template and the selected model template, the value for this option/field in the selected model template will override the value in default model template.
- Click on “Create New Template” to add a model template.
Table 31: Create New Model Template
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 default model template will be overwritten if other selected model template has a different value for the same option. |
Active | Check this option to enable the model template. |
-
Click on
to edit the model template.
The editing window for model template is shown in the following figure. In the “Options” field, enter the option name key word, the option that contains the key word will be listed. User could then select the option and click on “Add Option” to add it into the model template.
Once added, the option will be shown in the list below. On the left side of each option, users can click on
to remove this option from the model template. On the right side of each option, users can click on
to reset the option to the default value.
User could also click on “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. For P value information of different models, please refer to configuration template here https://content.grandstream.com/hubfs/Grandstream_Feb_2021/Zip%20File/config-template.zip
Note: Some devices use Alias and not P-values.
- Click on Save when done. The model template will be displayed on Web GUI🡪Value-added Features🡪Zero Config🡪Model Templates page.
-
Click on
to delete the model template or click on “Delete Selected Templates” to delete multiple selected templates at once.
- Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected model templates.
Model Update
UCM6200 zero config feature supports provisioning all models of Grandstream SIP end devices including OEM device models.
OEM Models
Users can associate OEM device models with their original Grandstream-branded models, allowing these OEM devices to be provisioned appropriately.
-
Click on
button.
- In the Source Model field, select the Grandstream device that the OEM model is based on from the dropdown list.
- For the Destination Model and Destination Vendor field, enter the custom OEM model name and vendor name.
- 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 UCM6200 already. Templates for GS Wave and Grandstream surveillance products require users to download and install under Web GUI🡪Value-added Features🡪Zero Config🡪Model Update first before they are available in the UCM6200 for selection. After downloading and installing the model template to the UCM6200, it will show in the dropdown list for “Model” selection when editing the model template.
-
Click on
to download the template.
-
Click on
to upgrade the model template. Users will see this icon available if the device model has template updated in the UCM6200.
Upload Model Template Package
In case the UCM6200 is placed in the private network and Internet access is restricted, users will not be able to get packages by downloading and installing from the remote server. Model template packages can be manually uploaded to the UCM.. Please contact Grandstream customer support if the model package is needed for manual uploading.
Device Configuration
On Web GUI, page Value-added Features🡪Zero Config🡪Zero Config, users could create new device, delete existing device(s), make special configuration for a single device, or send NOTIFY to existing device(s).
Create New Device
Besides configuring the device after the device is discovered, users could also directly create a new device and configure basic settings before the device is discovered by the UCM6200. 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 “Create New Device” and the following dialog will show. Follow the steps below to create the configurations for the new device.
- 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.
- 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.
- Click on “Create New Device” to save the configuration for this device.
Manage Devices
The device manually created or discovered from Auto Discover will be listed in the Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config page. Users can see the devices with their MAC address, IP address, vendor, model etc.

-
Click on
to access the Web GUI of the phone.
-
Click on
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 above figure. There are five levels configurations as described in (1) (2) (3) (4) (5) below, with priority from the lowest to the highest. The configurations in all levels will take effect for the device. If there are same options existing in different level configurations with different value configured, the higher-level configuration will override the lower level configuration.
- Global Policy
This is the lowest level configuration. The global policy configured in Web GUI🡪Value-added Features🡪Zero Config🡪Global Policy will be applied here. Clicking on “Modify Global Policy” to redirect to page Value-added Features🡪Zero Config🡪Global Policy.
- Global Templates
Select a global template to be used for the device and click on
to add. Multiple global templates can be selected, and users can arrange the priority by adjusting orders via
and
. All the selected global templates will take effect. If the same option exists on multiple selected global templates, the value in the template with higher priority will override the one in the template with lower priority. Click on
to remove the global template from the selected list.
- Default Model Template
Default Model Template will be applied to the devices of this model. Default model template can be configured in model template under Web GUI🡪Value-added Features🡪Zero Config🡪Model Templates page. Please see default model template option in [Table 31: Create New Model Template].
- Model Templates
Select a model template to be used for the device and click on
to add. Multiple global templates can be selected, and users can arrange the priority by adjusting orders via
and
. All the selected model templates will take effect. If the same option exists on multiple selected model templates, the value in the template with higher priority will override the one in the template with lower priority. Click on
to remove the model template from the selected list.
- Customize Device Settings
This is the highest-level configuration for the device. Click on “Modify Customize Device Settings” and following dialog will show.
Scroll down in the dialog to view and edit the device-specific options. If the users would like to add more options which are not in the pre-defined list, click on “Add New Field” to add a P value number and the value to the configuration. The above figure shows setting P value “P1362” to “en”, which means the display language on the LCD is set to English. The warning information on right tells that the option matching the P value number exists and clicking on it will lead to the matching option. For P value information of different models, please refer to configuration template here http://www.grandstream.com/sites/default/files/Resources/config-template.zip.
-
Select multiple devices that need to be modified and then click on
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
to view more devices of other models. Users are only allowed to make modifications in Global Templates and Global Policy level.
Note:
Performing batch operation will override all the existing device configuration on the page.
After the above configurations, save the changes and go back to Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config page. Users could then click on
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.
In this web page, users can also click on “Reset All Extensions” to reset the extensions of all the devices.
Example Application
Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1 GXV3370 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 provisioning the office phones for the customer support team.
- Go to Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config Settings, select “Enable Zero Config”.
- Go to Web GUI🡪Value-added Features🡪Zero Config🡪Global Policy, configure Date Format, Time Format and Firmware Source as follows.
- Go to Web GUI🡪Value-added Features🡪Zero Config🡪Model Templates, create a new model template “English Support Template” for GXP2140. Add option “Language” and set it to “English”. Then select the option “Default Model Template” to make it the default model template.
- Go to Web GUI🡪Value-added Features🡪Zero Config🡪Model Templates, create another model template “Spanish Support Template” for GXP2140. Add option “Language” and set it to “Español”.
- After 9 devices are powered up and connected to the LAN network, use “Auto Discover” function or “Create New Device” function to add the devices to the device list on Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config.
-
On Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config page, users could identify the devices by their MAC addresses or IP addresses displayed on the list. Click on
to edit the device settings.
- 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 “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.
-
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 “Advanced” settings tab to bring up the following dialog.
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 default model template.
- For the GXV3370 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 GXV3370.
- Click on “Apply Changes” to apply saved changes.
-
On the Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config page, click on
to send NOTIFY to trigger the device to download config file from UCM6200.
Now all the 9 phones in the network will be provisioned with a unique extension registered on the UCM6200. 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 GXV3370 used by the supervisor will be provisioned to use the default language on LCD display since it is not specified in the global policy.
Extensions
Create New User
Create New SIP Extension
To create a new SIP extension, navigate to Extension/Trunk->Extensions and click on the Add button. The following window will appear:
Extension options are divided into four categories:
- Basic Settings
- Media
- Features
- Specific Time
- Follow me
Select first which type of extension: SIP Extension, IAX Extension or FXS Extension. The configuration parameters are as follows.
Table 32: SIP Extension Configuration Parameters – Basic Settings
General | |
Extension | The extension number associated with the user. Note: This field supports (+) sign. |
CallerID Number | Configure the CallerID Number that would be applied for outbound calls from this user. Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider. |
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 using this rule. |
SIP/IAX Password | Configure the password for the user. A random secure password will be automatically generated. It is recommended to use this password for security purpose. |
Auth ID | Configure the authentication ID for the user. If not configured, the extension number will be used for authentication. Note: This filed supports (+) sign. |
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). |
Voicemail Password | Configure the password to access the extension's voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported. |
Skip Voicemail Password Verification | If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default. |
Send Voicemail Email Notification |
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Send Voicemail to Email | If enabled, the voicemail will be attached to the email notification that is sent when a voicemail is received. Note: When set to "Default", the global settings in Call Features->Voicemail->Voicemail Email Settings will be used. |
Keep Voicemail after Emailing | Configure whether or not to retain the voicemail in local storage after sending the voicemail attachment in the email notification. |
Enable Keep-alive | If enabled, empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is "Yes". |
Keep-alive Frequency | Configure the Keep-alive interval (in seconds) to check if the host is up. The default setting is 60 seconds. |
Disable This Extension | If selected, this extension will be disabled on the UCM6200. Note: The disabled extension still exists on the PBX but cannot be used on the end device. |
User Settings | |
First Name | Configure the user's first name. This field supports alphanumeric characters, underscores (_), and periods. |
Last Name | Configures the user's last name. This field supports alphanumeric characters, underscores (_), and periods. |
Email Address | Configure the user's email address. Email notifications will be sent to this address. |
User Password | Configure the password for user portal access. A random password is automatically generated by default and is highly recommended for security. |
Language | Select the voice prompt language that will be used for this extension. By default, the selected voice prompt language under PBX Settings->Voice Prompt->Language Settings will be used. To add more supported languages, please download the voice prompt language packages in the same page. |
Concurrent Registrations | The maximum allowed number of endpoint registrations to this extension. Default value is 1. |
Mobile Phone Number | Configure a phone number for the extension. Users can dial the Direct Dial Mobile Phone Prefix feature code (*88 by default) + extension number to directly dial this number. Example: Dial * |
Table 33: SIP Extension Configuration Parameters – Media
SIP Settings | |
NAT | Use NAT when the UCM6200 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 UCM6200 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 UCM6200 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, 64-kbit PCMU and PCMA are required. When “Auto” is selected, RFC4733 will be used if offered, otherwise “Inband” will be used. |
TEL URI | If the phone has an assigned PSTN telephone number, this field should be set to “User=Phone”. “User=Phone” parameter will be attached to the Request-Line and “TO” header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel” will be used instead of “SIP” in the SIP request. |
Alert-Info | Configure the Alert-Info, when UCM6200 receives an INVITE request, the Alert-Info header field specifies an alternative ring tone to the UAS. |
Enable T.38 UDPTL | Enable or disable T.38 UDPTL support. |
SRTP | Enable SRTP for the call. The default setting is disabled. |
Fax Mode | Select Fax mode. The default setting is “None”.
|
| |
Access Control List manages the IP addresses that can register to this extension.
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Specifies allowed IP address or networks from where the extension can be registered. Up to 10 entries are allowed. Format: “xxx.xxx.xxx.xxx”, “xxx.xxx.xxx.xxx/32”, “[::]” or “[::]/128”. | |
Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, OPUS, iLBC, ADPCM, H.264, H.263, H.263p and RTX. |
Table 34: SIP Extension Configuration Parameters – Features
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”. |
Call Forward Unconditional | Enable and configure the Call Forward Unconditional target number. Available options for target number are:
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”. Notes:
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Call Forward No Answer | Configure the Call Forward No Answer target number. Available options for target number are:
|
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:
|
Call Forward Busy | Configure the Call Forward Busy target number. Available options for target number are:
|
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:
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Do Not Disturb | If enabled the extension will ignore all incoming calls |
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:
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DND Whitelist | If DND is enabled, all calls to this extension will be rejected except the numbers listed on this list. Note: The maximum number on the Whitelist is 10. |
FWD Whitelist | If call forward is enabled, all calls to this extension will be forwarded except the calls coming from the specified numbers on this list. Note: The Maximum number on the whitelist is 10. |
CC Settings | |
Enable CC | If enabled, UCM6200 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:
The default setting is “Normal”. Note: The number of CC agents (for “Normal” mode) will now be limited by the extensions’ allowed number of concurrent registrations. |
CC Max Agents | Configure the maximum number of CCSS agents which may be allocated for this channel (when CC Mode is set to “For Trunk”). In other words, this number serves as the maximum number of CC requests this channel is allowed to make. Min. value is 1. |
CC Max Monitors | Configure the maximum number of monitor structures which may be created for this device (when CC Mode is set to “For Trunk”). 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 based on this time condition. |
Use callee DOD on FWD or Ring Simultaneously | Use the DOD number when calls are being diverted/forwarded to external destinations or when ring simultaneous is configured. |
Monitor privilege control | |
Allow 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 UCM6200, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪General Settings: 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 recording files can be accessed under Web GUI🡪CDR🡪Recording Files. |
Skip Trunk Auth |
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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. |
Support Hot-Desking Mode | If enabled, SIP Password will accept only alphabet characters and digits. Auth ID will be changed to the same as Extension. |
Enable LDAP | If enabled, the extension will be added to LDAP Phonebook PBX list. |
Enable WebRTC Support | Enable registration and call from Web RTC. |
Music On Hold | Specify which Music On Hold playlist to suggest to the bridged channel when putting them on hold. |
Enable Seamless Transfer | Enable the seamless transfer for this extension. |
Permission | Set the permission for this extension when using the seamless transfer |
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. |
Custom Call-info for Auto Answer | If enabled, when a call is sent to this extension from UCM, the SIP INVITE message will contain a Call-info header indicating auto answer. |
Enable Call Waiting | If disabled, UCM will not ring the extension if it is already in a call and will give the caller a busy response. |
Email Missed Calls | When enabled, the UCM sends a log of missed calls to the extension's email address. |
Table 34: SIP Extension Configuration Parameters – Features
Specific Time | |
Time Condition | Click to add Time Condition to configure specific time for this extension. |
Create New IAX Extension
The UCM6200 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.
Table 36: IAX Extension Configuration Parameters – Basic Settings
General | |
Extension | The extension number associated with the user. |
CallerID Number | Configure the CallerID Number that would be applied for outbound calls from this user. Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider. |
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 using this rule. |
SIP/IAX Password | Configure the password for the user. A random secure password will be automatically generated. It is recommended to use this password for security purpose. |
Voicemail | Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”.
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Voicemail Password | Configure the password to access the extension’s voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported. |
Skip Voicemail Password Verification | If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default. |
Disable This Extension | If selected, this extension will be disabled on the UCM6200. Note: The disabled extension still exists on the PBX but cannot be used on the end device. |
User Settings | |
First Name | Configure the user’s first name. This field supports alphanumeric characters, underscores (_), and periods. |
Last Name | Configures the user’s last name. This field supports alphanumeric characters, underscores (_), and periods. |
Email Address | Configure the user’s email address. Email notifications will be sent to this address. |
User Password | Configure the password for user portal access. A random password is automatically generated by default and is highly recommended for security. |
Language | Select the voice prompt language that will be used for this extension. By default, the selected voice prompt language under PBX Settings->Voice Prompt->Language Settings will be used. To add more supported languages, please download the voice prompt language packages in the same page. |
Table 37: IAX Extension Configuration Parameters – Media
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 SRTP for the call. The default setting is disabled. |
Fax Mode | Select Fax Mode. The default setting is “None”.
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ACL Policy | Access Control List manages the IP addresses that can register to this extension.
Local Network Address: Only IP addresses in the configured network segments can register to this extension. |
Codec Preference | Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p, RTX and VP8. |
Table 38: IAX Extension Configuration Parameters – Features
Call Transfer | |
Call Forward Unconditional | Configure the Call Forward Unconditional target number. If not configured, |
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”. Notes:
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Call Forward No Answer | Configure the Call Forward No Answer target number. Available options for target number are: |
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:
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Call Forward Busy | Configure the Call Forward Busy target number. Available options for target number are:
|
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:
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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 based on this time condition. |
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 UCM6200, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪General Settings: 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 recording files can be accessed under Web GUI🡪CDR🡪Recording Files. |
Skip Trunk Auth |
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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 playlist to suggest to the bridged channel when putting them on hold. |
Call Duration Limit | The maximum duration of call-blocking. |
Enable Call Waiting | If disabled, UCM will not ring the extension if it is already in a call and will give the caller a busy response. |
Email Missed Calls | When enabled, the UCM sends a log of missed calls to the extension's email address. |
Table 38: IAX Extension Configuration Parameters – Features
Specific Time | |
Time Condition | Click to add Time Condition to configure specific time for this extension. |
Create New FXS Extension
The UCM6200 supports Foreign eXchange Subscriber (FXS) interface. FXS is used when user needs to connect analog phone lines or FAX machines to the UCM6200.
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.
Table 40: FXS Extension Configuration Parameters – Basic Settings
General | |
Extension | The extension number associated with the user. |
Analog Station | Select the FXS port to be assigned for this extension. |
Caller ID Number | Configure the CallerID Number that would be applied for outbound calls from this user. Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider. |
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 using this rule. |
Voicemail | Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”. |
Voicemail Password | Configure the password to access the extension’s voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported. |
Skip Voicemail Password Verification | If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default. |
Disable This Extension | If selected, this extension will be disabled on the UCM6200. Note: The disabled extension still exists on the PBX but cannot be used on the end device. |
User Settings | |
First Name | Configure the user’s first name. This field supports alphanumeric characters, underscores (_), and periods. |
Last Name | Configures the user’s last name. This field supports alphanumeric characters, underscores (_), and periods. |
Email Address | Configure the user’s email address. Email notifications will be sent to this address. |
User Password | Configure the password for user portal access. A random password is automatically generated by default and is highly recommended for security. |
Language | Select the voice prompt language that will be used for this extension. By default, the selected voice prompt language under PBX Settings->Voice Prompt->Language Settings will be used. To add more supported languages, please download the voice prompt language packages in the same page. |
Table 41: FXS Extension Configuration Parameters – Media
Analog Settings | |
Call Waiting | Configure to enable/disable call waiting feature. The default setting is “No”. |
User ‘#’ as SEND | If configured, the # key can be used as SEND key after dialing the number on the analog phone. The default setting is “Yes”. |
RX Gain | Configure the RX gain for the receiving channel of analog FXS port. The valid range is -30dB to +6dB. The default setting is 0. |
TX Gain | Configure the TX gain for the transmitting channel of analog FXS port. The valid range is -30dB to +6dB. The default setting is 0. |
MIN RX Flash | Configure the minimum period of time (in milliseconds) that the hook-flash must remain unpressed for the PBX to consider the event as a valid flash event. The valid range is 30ms to 1000ms. The default setting is 200ms. |
MAX RX Flash | Configure the maximum period of time (in milliseconds) that the hook-flash must remain unpressed for the PBX to consider the event as a valid flash event. The minimum period of time is 256ms and it cannot be modified. The default setting is 1250ms. |
Enable Polarity Reversal | If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as Hangup on a polarity reversal. The default setting is “Yes”. |
Echo Cancellation | Specify “ON”, “OFF” or a value (the power of 2) from 32 to 1024 as the number of taps of cancellation. Note: When configuring the number of taps, the number 256 is not translated into 256ms of echo cancellation. Instead, 256 taps mean 256/8 = 32 ms. The default setting is “ON”, which is 128 taps. |
3-Way Calling | Configure to enable/disable 3-way calling feature on the user. The default setting is enabled. |
Send CallerID After | Configure the number of rings before sending CID. Default setting is 1. |
Fax Mode | For FXS extension, there are three options available in Fax Mode. The default setting is “None”.
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Table 42: FXS Extension Configuration Parameters – Features
Call Transfer | |
Call Forward Unconditional | Configure the Call Forward Unconditional target number. If not configured, |
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”. Notes:
|
Call Forward No Answer | Configure the Call Forward No Answer target number. If not configured, |
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:
|
Call Forward Busy | Configure the Call Forward Busy target number. If not configured, the Call |
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:
|
CC Settings | |
Enable CC | If enabled, UCM6200 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. |
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 based on this time condition. |
Hotline | |
Enable Hotline | If enabled, hotline dialing plan will be activated, a pre-configured number |
Hotline Number | Configure the Hotline Number |
Hotline Type | Configure the Hotline Type:
|
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 UCM6200, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪General Settings: 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 recording files can be accessed under Web GUI🡪CDR🡪Recording Files. |
Skip Trunk Auth |
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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 playlist to suggest to the bridged channel when putting them on hold. |
Call Duration Limit | The maximum duration of call-blocking. |
Email Missed Calls | When enabled, the UCM sends a log of missed calls to the extension's email address. |
Table 43: FXS Extension Configuration Parameters – Specific Time
Specific Time | |
Time Condition | Click to add Time Condition to configure specific time for this extension. |
Batch Add Extensions
Batch Add SIP Extensions
To create multiple SIP extensions quickly, users can select “Batch” for the Select Add Method option during the extension creation process.
Table 44: Batch Add SIP Extension Parameters
General | |
Create Number | Specify the number of extensions to be added. The default setting is 5. |
Specify how much to increment each additional extension (e.g., if the value is 2, the extensions will be 1000, 1002, 1004, etc.). Note: Supports 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”.
|
Enable WebRTC Support | Enable WebRTC support. |
SIP/IAX Password | Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions.
A random secure password will be automatically generated. It is recommended to use this password for security purpose.
|
Voicemail Password | Configure the password to access the extension’s voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported. |
Configure CallerID Number when adding Batch Extensions.
Users can choose to use the extension number as CallerID
Users can choose to set a specific number instead of using the extension number. | |
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 UCM6200, 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 recording files can be accessed under Web GUI🡪CDR🡪Recording Files. |
Skip Voicemail Password Verification | If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default. |
Music On Hold | Select which Music On Hold class to suggest to extensions when putting them on hold. |
Enable LDAP | If enabled, the batch added extensions will be added to LDAP Phonebook PBX list; if disabled, the batch added extensions will be skipped when creating LDAP Phonebook. |
Enable WebRTC Support | If enabled, extensions will be able to login to user portal and use Web RTC features. |
Call Duration Limit | Configure the maximum duration of call-blocking. |
SIP Settings | |
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, 64-kbit codec PCMU and PCMA are required. When “Auto” is selected, RFC4733 will be used if offered, otherwise “Inband” will be used. |
Enable Keep-alive | If enabled, empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is “Yes”. |
Keep-alive Frequency | Configure the number of seconds for the host to be up for Keep-alive. The default setting is 60 seconds. |
TEL URI | If the end device/phone has an assigned PSTN telephone number, this field should be set to “User=Phone”. Then a “User=Phone” parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is disabled. |
Concurrent Registrations | The maximum allowed number of endpoint registrations to this extension. Default value is 1. |
Other Settings | |
SRTP | Enable SRTP for the call. The default setting is “No”. |
Fax Mode | Select Fax mode for this user. The default setting is “None”.
|
ACL Policy | Access Control List manages the IP addresses that can register to this extension.
Local Network Address: Only IP addresses in the configured network segments can register to this extension. |
Enable T.38 UDPTL | Enable or disable T.38 UDPTL Support. |
Skip Trunk Auth | If enable “All”, users do not need to enter password when making an outbound call. If enable “Follow Me”, the user can dial out via follow me without password. |
Codec Preference | Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, iLBC, ADPCM, LPC10, H.264, H.263, H.263p and VP8. |
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.
Table 45: Batch Add IAX Extension Parameters
General | |
Start Extension | Configure the starting extension number of the batch of extensions to be added. |
Create Number | Specify the number of extensions to be added. The default setting is 5. |
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. |
Voicemail | Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”.
|
SIP/IAX Password | Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions.
A random secure password will be automatically generated. It is recommended to use this password for security purpose.
|
Voicemail Password | Configure the password to access the extension’s voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported. |
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 UCM6200, 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 recording files can be accessed under Web GUI🡪CDR🡪Recording Files. |
Skip Voicemail Password Verification | If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default. |
Music On Hold | Select which Music On Hold class to suggest to extensions when putting them on hold. |
Enable LDAP | If enabled, the batch added extensions will be added to LDAP Phonebook PBX list; if disabled, the batch added extensions will be skipped when creating LDAP Phonebook. |
Call Duration Limit | Configure the maximum duration of call-blocking. |
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”. |
Other Settings | |
SRTP | Enable SRTP for the call. The default setting is “No”. |
Fax Mode | Select Fax Mode for this user. The default setting is “None”.
|
ACL Policy | Access Control List manages the IP addresses that can register to this extension.
Local Network Address: Only IP addresses in the configured network segments can register to this extension. |
Skip Trunk Auth | If enable “All”, users do not need to enter password when making an outbound call. If enable “Follow Me”, the call can dial out via follow me without password. |
Codec Preference | Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, iLBC, ADPCM, LPC10, H.264, H.263, H.263p and VP8. |
Batch-Edit Extensions
To edit many extensions at once, please navigate to the web GUI of the UCM62xx then choose Extensions / Trunk → Extensions then click on
General | |
Permission | Please select the permission level for outgoing calls. |
Voicemail | The default option is "Local Voicemail".
|
SIP/IAX Password | The password for the user to register SIP/IAX account. |
Voicemail Password | Password to access the user's voicemail. |
CallerID Number | Configure the caller ID number displayed in dialing calls from this user. |
Ring Timeout | Number of seconds to ring the extension before forwarding the call to voicemail. |
Auto Record |
|
Skip Voicemail Password Verification | When user dials My Voicemail, the password verification IVR is skipped. |
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. |
Music On Hold | Specify which Music On Hold class to suggest to the bridged channel when putting them on hold. |
Disable This Extention | Check this box to disable this extension. |
Enable LDAP | If enabled, the extension will be added to LDAP Phonebook PBX lists. |
Enable WebRTC Support | Enable registration and call from WebRTC. |
Call Duration Limit | Block calls for the configured duration. |
Language | Select voice prompt language for this extension. If set to "Default", the global setting for voice prompt language will be used. |
Send Voicemail to Email | Whether to attach voicemail messages to email. |
Keep Voicemail after Emailing | The voicemail recording will be kept even if it has been sent through email. Only applies if extension-level or global Send Voicemail to Email is enabled. |
Call Waiting | Allows calls to the extension even when it is already in a call. This only works if the caller is directly dialing the extension. If disabled, the CC service will not work. |
SIP Settings | |
NAT | Enable this setting if the UCM is using public IP and communicating with devices behind NAT. Note 1: This setting will overwrite the Contact header of received messages, which may affect the ability to establish calls when behind NAT. Consider changing setttings in PBX Settings->SIP Settings->NAT instead. Note 2: If one is experiencing one-way audio issues, please check the NAT configuration and SIP/RTP ports in the firewall. |
Enable Direct Media | By default, the PBX will route the media streams from SIP endpoints through itself. If this option is enabled, the PBX will attempt to redirect the RTP media streams to bypass the PBX and to go directly between caller and callee. Note: It is not always possible for the PBX to negotiate endpoint-to-endpoint media routing. |
DTMF Mode | Configures the mode for sending DTMF.
|
TEL URI | If "Enabled" option is selected, TEL URI and Remove OBP from Route cannot be enabled at the same time. If the phone has an assigned PSTN telephone number, this field should be set to "User=Phone". A "User=Phone" parameter will then be attached to the Request-Line and "TO" header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. |
Concurrent Registrations | The maximum endpoints which can be registered to this extension. |
CC Settings | |
Enable CC | Check this box to allow the system to automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason. |
Monitor Privilege Control | |
Allow Call-barging | Members of this list can spy on this extension via feature codes. |
Seamless Transfer Privilege Control | |
Allowed to seamless transfer | Members of the list can seamless transfer via feature code |
Other Settings | |
SRTP | Enable/disable SRTP for RTP stream encryption. |
Fax Mode | Check this box to allow the system to automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason.
|
ACL Policy | Access Control List manages the IP addresses that can register to this extension.
|
Skip Trunk Auth | If set to "Yes", users do not need to enter a password when making outbound calls. If set to "Bytime", users do not need to enter a password when making outbound calls that matches the time condition. |
Codec Preference | Configure the codecs to be used. |
Batch Extension Resetting Functionality
To reset multiple extensions to default settings, select the extensions on the main Extensions/Trunks->Extensions page and click on the Reset button. Once reset, all settings except for Concurrent Registrations will be reverted to default. User voicemail password will be randomized again, and user voicemail prompts and recordings will be deleted. The settings for the extension user in the User Management page will also be reverted to default except for usernames and custom privileges.
Search and Edit Extension
All the UCM6200 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 Selected Extensions” or “Delete Selected Extensions”. Also, options “Edit”
, “Reboot”
, “Delete”
and reset 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.
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 UCM6200 extension already registered. To successfully reboot the user, “Zero Config” needs to be enabled on the UCM6200 Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config Settings.
- Delete selected extensions
Click on the
button to delete a single extension. Users can also select multiple extensions and click on the Delete button to delete several extensions at once.
- Modify selected extensions
Select the checkbox for the extension(s). Then click on “Edit” to edit the extensions in a batch.
Export Extensions
UCM extensions can be exported to a CSV file. Different extension types (SIP, IAX, FXS) cannot be exported to the same CSV file. To export extensions, click on the Export button and select the extension type to export.
Users can also use the exported CSV file to use as a template to manually edit extension information.
Import Extensions
The capability to import extensions to the UCM6200 provides users flexibility to batch add extensions with similar or different configuration quickly into the PBX system.
- Export extension csv file from the UCM6200 by clicking on “Export Extensions” button.
- Fill up the extension information you would like in the exported csv template.
- Click on “Import Extensions” button. The following dialog will be prompted.
- 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.
- Click on “Choose file to upload” to select csv file from local directory in the PC.
- Click on “Apply Changes” to apply the imported file on the UCM6200.
Example of file to import:
Table 46: SIP extensions Imported File Example
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 |
Fax Mode | None/Fax Detection |
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 WebRTC | 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 |
Table 47: IAX extensions Imported File Example
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 |
Fax Mode | None/Fax Detection |
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 |
Table 48: FXS Extensions Imported File Example
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 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 UCM6200 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
Users can request for their SIP acccount information via email. The system administrator can select the extensions to send account information emails to and press the Email Notification button in the Extensions page.
When click on “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. After clicking the button, the following prompt will appear:
The selected users will receive emails containing account registration information, LDAP configuration, and QR codes for quick setup on mobile apps.
Multiple Registrations per Extension
UCM6200 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 1 for security purposes. Maximum is 10.
SMS Message Support
The UCM6200 provides built-in SIP SMS message support. For SIP end devices such as Grandstream GXP or GXV phones that supports SIP message, after an UCM6200 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.
SMS Message support is a new feature added since firmware 1.0.10.x which is built with Asterisk 13.
Extension Groups
The UCM6200 extension group feature allows users to assign and categorize extensions in different groups to better manage the configurations on the UCM6200. 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.
Table 49: Analog Trunk Configuration Parameters
Channels | Select the channel for the analog trunk.
|
Trunk Name | Specify a unique label to identify the trunk when listed in outbound rules, incoming rules and etc. |
Advanced Options | |
SLA Mode | Enable this option to satisfy two primary use cases, which include emulating a simple key system and creating shared extensions on a PBX. Enable SLA Mode will disable polarity reversal. |
Barge Allowed | The barge option specifies whether other stations can join a call in progress on this trunk. If enabled, the other stations can press the line button to join the call. The default setting is Yes. |
Hold Access | The hold option specifies hold permissions for this trunk. If set to “Open”, any station can place this trunk on hold and any other station is allowed to retrieve the call. If set to “Private”, only the station that places the call on hold can retrieve the call. The default setting is Yes. |
Enable Polarity Reversal | If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as “Hangup” on a polarity reversal. The default setting is “No”. |
Polarity on Answer Delay | When FXO port answers the call, FXS may send a Polarity Reversal. If this interval is shorter than the value of “Polarity on Answer Delay”, the Polarity Reversal will be ignored. Otherwise, the FXO will Onhook to disconnect the call. The default setting is 600ms. |
Current Disconnect Threshold (ms) | This is the periodic time (in ms) that the UCM6200 will use to check on a voltage drop in the line. The default setting is 200. The valid range is 50 to 3000. |
Ring Timeout | Configure the ring timeout (in ms). Trunk (FXO) devices must have a timeout to determine if there was a Hangup before the line is answered. This value can be used to configure how long it takes before the UCM6200 considers a non-ringing line with Hangup activity. The default setting is 8000. |
RX Gain | Configure the RX gain for the receiving channel of analog FXO port. The valid range is from -12.0 (dB) to + 12.0 (dB). The default setting is 0. |
TX Gain | Configure the TX gain for the transmitting channel of analog FXO port. The valid range is from -12.0 (dB) to + 12.0 (dB). The default setting is 0. |
Use CallerID | Configure to enable CallerID detection. The default setting is “Yes”. |
Caller ID Scheme | Select the Caller ID scheme for this trunk.
If you are not sure which scheme to choose, please select “Auto Detect”. The default setting is “Bellcore/Telcordia”. |
Fax Mode | Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38. The default setting is “No”. Note: If enabled, Fax Pass-through cannot be used. |
FXO Dial Delay(ms) | Configure the time interval between off-hook and first dialed digit for outbound calls. |
Auto Record | Enable automatic recording for the calls using this trunk. The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files. |
Disable This Trunk | If selected, the trunk will be disabled and incoming/Outgoing calls via this trunk will not be possible. |
DAHDI Out Line Selection | This is to implement analog trunk outbound line selection strategy. Three options are available:
When the call goes out from this analog trunk, it will always try to use the first idle FXO port. The port order that the call will use to go out would be port 1🡪port 2🡪port 10🡪port 16. Every time it will start with port 1 (if it is idle).
When the call goes out from this analog trunk, it will use the port that is not used last time. And it will always use the port in the order of port 1🡪2🡪10🡪16🡪1🡪2🡪10🡪16🡪1🡪2🡪10🡪16…, following the last port being used.
When the call goes out from this analog trunk, it will always try to use the last idle FXO port. The port order that the call will use to go out would be port 16🡪port 10🡪port 2🡪port 1. Every time it will start with port 16 (if it is idle). The default setting is “Ascend” mode. |
Echo Cancellation Mode | The Non-Linear Processing (NLP) in echo cancellation helps to remove/suppress residual echo components that could not be removed by the LEC (Line Echo Canceller). Following modes are supported:
|
Direct Callback | Allows external numbers the option to get directed to the extension that last called them. For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination. |
Tone Settings | |
Busy Detection | Busy Detection is used to detect far end Hangup or for detecting busy signal. The default setting is “Yes”. |
Busy Tone Count | If “Busy Detection” is enabled, users can specify the number of busy tones to be played before hanging up. The default setting is 2. Better results might be achieved if set to 4, 6 or even 8. Please note that the higher the number is, the more time is needed to Hangup the channel. However, this might lower the probability to get random Hangup. |
Congestion Detection | Congestion detection is used to detect far end congestion signal. The default setting is “Yes”. |
Congestion Count | If “Congestion Detection” is enabled, users can specify the number of congestion tones to wait for. The default setting is 2. |
Tone Country | Select the country for tone settings. If “Custom” is selected, users could manually configure the values for Busy Tone and Congestion Tone. The default setting is “United States of America (USA)”. |
Busy Tone | Syntax: f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]]; Frequencies are in Hz and cadence on and off are in ms. Frequencies Range: [0, 4000) Busy Level Range: (-300, 0) Cadence Range: [0, 16383]. Select Tone Country “Custom” to manually configure Busy Tone value. Default value: f1=480@-50,f2=620@-50,c=500/500 |
Congestion Tone | Syntax: f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]]; Frequencies are in Hz and cadence on and off are in ms. Frequencies Range: [0, 4000) Busy Level Range: (-300, 0) Cadence Range: [0, 16383]. Select Tone Country “Custom” to manually configure Busy Tone value. Default value: f1=480@-50,f2=620@-50,c=250/250 |
PSTN Detection | Click on “Detect” to detect the busy tone, Polarity Reversal and Current Disconnect by PSTN. Before the detecting, please make sure there is more than one channel configured and working properly. If the detection has busy tone, the “Tone Country” option will be set as “Custom”. |
PSTN Detection
The UCM6200 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 UCM6200 settings.
- Go to UCM6200 Web GUI🡪Extension/Trunk🡪Analog Trunks page.
- Click to edit the analog trunk created for the FXO port.
- In the dialog 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.
- 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.
- 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 UCM6200 Web GUI will display prompt to notify the user to answer or hang up the call to finish the detecting process.
- Once done, the detected result will show. Users could save the detecting result as the current UCM6200 settings.
Table 50: PSTN Detection for Analog Trunk
Detect Model | Select “Auto Detect” or “Semi-auto Detect” for PSTN detection.
Please make sure two or more channels are connected to the UCM6200 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 UCM6200 will control the call to be established and hang up between caller and callee to finish the detection.
Semi-auto detection requires answering or hanging up the call manually. Please make sure one channel is connected to the UCM6200 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. |
Note:
- The PSTN detection process will keep the call up for about 1 minute.
- If “Semi-auto Detect’ is used, please pick up the call only after informed from the Web GUI prompt.
- Once the detection is successful, the detected parameters “Busy Tone”, “Polarity Reversal” and “Current Disconnect by PSTN” will be filled into the corresponding fields in the analog trunk configuration.
VoIP Trunks
VoIP Trunk Configuration
VoIP trunks can be configured in UCM6200 under Web GUI🡪Extension/Trunk🡪VoIP Trunks. Once created, the VoIP trunks will be listed with Provider Name, Type, Hostname/IP, Username and Options to edit/detect the trunk.
Note: UCM6200 currently supports up to 200 VoIP trunks.
- Click on “Create New SIP Trunk” or “Create New IAX Trunk” to add a new VoIP trunk.
-
Click on
to configure detailed parameters for the VoIP trunk.
-
Click on
to configure Direct Outward Dialing (DOD) for the SIP Trunk.
-
Click on
to start LDAP Sync.
-
Click on
to delete the VoIP trunk.
For VoIP trunk example, please refer to the document in the following link:
https://documentation.grandstream.com/knowledge-base/sip-trunks-guide/
The VoIP trunk options are listed in the table below.
Table 51: Create New SIP Trunk
Type | Select the VoIP trunk type.
|
Configure a unique label (up to 64 character) to identify this trunk when listed in outbound rules, inbound rules etc. | |
Host Name | Configure the IP address or URL for the VoIP provider’s server of the trunk. |
Keep Original CID | Keep the CID from the inbound call when dialing out. This setting will override “Keep Trunk CID” option. Please ensure that the remote peer PBX supports matching user entry via the “username” field from authentication line. |
Keep Trunk CID | If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”. |
NAT | Turn on this setting when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it is related to NAT configuration or SIP/RTP port support on the firewall. |
Disable This Trunk | If checked, the trunk will be disabled. Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider. |
TEL URI | If the trunk has an assigned PSTN telephone number, this field should be set to “User=Phone”. Then a “User=Phone” parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is disabled. |
Caller ID | Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored. Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call: From user (Register Trunk Only) 🡪 CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID (Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled) 🡪 Global Outbound CID. |
Need Registration | Select whether the trunk needs to register on the external server or not when “Register SIP Trunk” type is selected. The default setting is No. |
Username | Enter the username to register to the trunk from the provider when “Register SIP Trunk” type is selected. |
Password | Enter the password to register to the trunk from the provider when “Register SIP Trunk” is selected. |
Auth ID | Enter the Authentication ID for “Register SIP Trunk” type. |
Auto Record | Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files. |
Allows external numbers the option to get directed to the extension that last called them. For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination. |
Table 52: SIP Register Trunk Configuration Parameters
Basic Settings | |
Provider Name | Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc. |
Host Name | Configure the IP address or URL for the VoIP provider’s server of the trunk. |
Transport | Configure the SIP transport protocol to be used in this trunk. The default setting is “UDP”.
|
When TLS is selected as Transport for register trunk, users can select between SIP and SIPS URI scheme | |
Keep Original CID | Keep the CID from the inbound call when dialing out. This setting will override “Keep Trunk CID” option. Please ensure that the remote peer PBX supports matching user entry via the “username” field from authentication line. |
Keep Trunk CID | If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”. |
NAT | Turn on this option when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it is related to NAT configuration or SIP/RTP port configuration on the firewall. |
Disable This Trunk | If selected, the trunk will be disabled. Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider. |
TEL URI | If the trunk has an assigned PSTN telephone number, this field should be set to “User=Phone”. Then a “User=Phone” parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is disabled. |
Need Registration | Select whether the trunk needs to register on the external server or not when “Register SIP Trunk” type is selected. The default setting is No. |
If enabled, outbound calls can still be made even if registration to the trunk fails. Note: If Need Registration is disabled, this option will be ignored. | |
CallerID Name | Configure the new name of the caller when the extension has no CallerID Name configured. |
Configure the domain name of the SIP user account. This will overwrite the domain in the “From” header. Example: If the user account is “1234567”, setting “trunk.UCM6510.provider.com” as the From Domain value will result in the following From header: “sip:1234567@trunk.UCM6510.provider.com”. | |
From User | Configure the username of the account. This will overwrite the user value in the From header. This is useful for scenarios where a single user account has multiple DIDs. Example: If the domain is “trunk.UCM6510.provider.com”, setting “1234567” as the From User value will result in the following From header: “sip:1234567@trunk.UCM6510.provider.com”. |
Username | Enter the username to register to the trunk from the provider when “Register SIP Trunk” type is selected. |
Password | Enter the password to register to the trunk when “Register SIP Trunk” is selected. |
Auth ID | Enter the Authentication ID for “Register SIP Trunk” type. |
Auth Trunk | If enabled, the UCM will send 401 response to the incoming call to authenticate the trunk. |
Auto Record | Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files. |
Allows external numbers the option to get directed to the extension that last called them. For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination. | |
Advanced Settings | |
Codec Preference | Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p, RTX, OPUS and VP8. |
Send PPI Header | If enabled, the SIP INVITE message sent to the trunk will contain PPI (P-Preferred-Identity) header. The default setting is “No”. Note: “Send PPI Header” and “Send PAI Header” cannot be enabled at the same time. Only one of the two headers can be contained in SIP INVITE message. |
| |
Send PAI Header | If enabled, the SIP INVITE message sent to the trunk will contain PAI (P-Asserted-Identity) header including configured PAI Header. The default setting is “No”. Note: “Send PPI Header” and “Send PAI Header” cannot be enabled at the same time. Only one of the two headers can be contained in the SIP INVITE message. |
If “Send PAI Header” is enabled and “PAI Header” is configured as “123456” for instance, the PAI header in the SIP message sent from the UCM will contain “123456”. If “Send PAI Header” is enabled and “PAI Header” is configured as “empty”, the PAI header in the SIP message sent from the UCM will contain the original CID. Note: “Send PAI Header” needs to be enabled in order to use this feature. Only alphanumeric characters are allowed and/or special characters #*-_+. with a limit of 64 characters. | |
DOD As From Name | If enabled and “From User” is configured, the INVITE’s From header will contain the DOD number. |
If checked and option “Send PAI Header” not checked, the PAI header will be passthrough from one side to the other side. | |
Outbound Proxy Support | Select to enable outbound proxy in this trunk. The default setting is “No”. |
Outbound Proxy | When outbound proxy support is enabled, enter the IP address or URL of the outbound proxy. |
Remove OBP from Route | It is used to set if the phone system will remove outbound proxy URI from the route header. If is set to “Yes”, it will remove the route header from SIP requests. The default setting is “No”. |
DID Mode | Configure where to get the destination ID of an incoming SIP call, from SIP Request-line or To-header. The default is set to “Request-line”. |
DTMF Mode | Configure the default DTMF mode when sending DTMF on this trunk.
|
Enable Heartbeat Detection | If enabled, the UCM6200 will regularly send SIP OPTIONS to the trunk to check connection status. The default setting is “No”. |
Heartbeat Frequency | When “Enable Heartbeat Detection” option is set to “Yes”, configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds. |
Maximum Number of Call Lines | The maximum number of concurrent calls using the trunk. The default settings 0, which means no limit. |
Fax Mode | Select Fax mode. The default setting is “None”.
|
SRTP | Enable SRTP for the VoIP trunk. The default setting is “No”. |
CC Settings | |
Enable CC | If enabled, the system will automatically alert the user when a called party is available, given that a previous call to that party failed for some reason. |
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 is allowed to 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. |
Table 53: SIP Peer Trunk Configuration Parameters
Basic Settings | |
Provider Name | Configure a unique label to identify this trunk when listed in outbound rules, inbound rules and etc. |
Host Name | Configure the IP address or URL for the VoIP provider’s server of the trunk. |
Auto Record | Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files. |
Keep Original CID | Keep the CID from the inbound call when dialing out, this setting will override “Keep Trunk CID” option. Please ensure that the remote peer PBX supports matching user entry via the “username” field from authentication line. |
Keep Trunk CID | If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”. |
NAT | Turn on this option when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it is related to NAT configuration or SIP/RTP port configuration on the firewall. |
Disable This Trunk | If selected, the trunk will be disabled. Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider. |
TEL URI | If the trunk has an assigned PSTN telephone number, this field should be set to “User=Phone”. Then a “User=Phone” parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is disabled. |
Caller ID | Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored. Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call :
|
CallerID Name | Configure the name of the caller to be displayed when the extension has no CallerID Name configured. |
Transport | Configure the SIP transport protocol to be used in this trunk. The default setting is “UDP”.
|
Allows external numbers the option to get directed to the extension that last called them. For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination. | |
Advanced Settings | |
Codec Preference | Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p,RTX, OPUS and VP8. |
DID Mode | Configure where to get the destination ID of an incoming SIP call, from SIP Request-line or To-header. The default is set to “Request-line”. |
DTMF Mode | Configure the default DTMF mode when sending DTMF on this trunk.
|
Enable Heartbeat Detection | If enabled, the UCM6200 will regularly send SIP OPTIONS to the trunk to check connection status. The default setting is “No”. |
Heartbeat Frequency | When “Enable Heartbeat Detection” option is set to “Yes”, configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds. |
Maximum Number of Call Lines | The maximum number of concurrent calls using the trunk. The default settings 0, which means no limit. |
Fax Mode | Select Fax mode. The default setting is “None”.
|
SRTP | Enable SRTP for the VoIP trunk. The default setting is “No”. |
If enabled, it will allow SDP passthrough to Grandstream IPVideoTalk therefore it will allow calls between the UCM and IPVideoTalk. The default setting is disabled. | |
Sync LDAP Enable | If enabled, the local UCM6200 will automatically provide and update the local LDAP contacts to the remote UCM6200 SIP peer trunk. In order to ensure successful synchronization, the remote UCM6200 peer also needs to enable this option on the SIP peer trunk. The default setting is “No”. |
Sync LDAP Password | This is the password used for LDAP contact file encryption and decryption during the LDAP sync process. The password must be the same on both UCM6200 peers to ensure successful synchronization. |
Sync LDAP Port | Configure the TCP port used LDAP sync feature between two peer UCM6200. |
LDAP Outbound Rule | Specify an outbound rule for LDAP sync feature. The UCM6200 will automatically modify the remote contacts by adding prefix parsed from this rule. |
LDAP Dialed Prefix | Specify the prefix for LDAP sync feature. The UCM6200 will automatically modify the remote contacts by adding this prefix. |
This option allows to toggle T.38 support for individual trunks. The default setting is enabled. | |
CC Settings | |
Enable CC | If enabled, the system will automatically alert the user when a called party is available, given that a previous call to that party failed for some reason. |
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. |
Table 54: Create New IAX Trunk
Type | Select the VoIP trunk type.
|
Provider Name | Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc. |
Host Name | Configure the IP address or URL for the VoIP provider’s server of the trunk. |
Keep Trunk CID | If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”. |
Username | Enter the username to register to the trunk from the provider when “Register IAX Trunk” type is selected. |
Password | Enter the password to register to the trunk from the provider when “Register IAX Trunk” type is selected. |
Disable This Trunk | If selected, the trunk will be disabled. |
Table 55: IAX Register Trunk Configuration Parameters
Basic Settings | |
Provider Name | Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc. |
Host Name | Configure the IP address or URL for the VoIP provider’s server of the trunk. |
Keep Trunk CID | If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”. |
Disable This Trunk | If selected, the trunk will be disabled. |
Caller ID | Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored. Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call : From user (Register Trunk Only) 🡪 CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID (Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled) 🡪 Global Outbound CID. |
CallerID Name | Configure the name of the caller to be displayed when the extension has no CallerID Name configured. |
Username | Enter the username to register to the trunk from the provider. |
Password | Enter the password to register to the trunk from the provider. |
Advanced Settings | |
Codec Preference | Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p and VP8. |
Enable Heartbeat Detection | If enabled, the UCM6200 will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is “No”. |
Heartbeat Frequency | When “Enable Heartbeat Detection” option is set to “Yes”, configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds. |
Maximum Number of Call Lines | The maximum number of concurrent calls using the trunk. The default settings 0, which means no limited. |
Fax Mode | Select Fax mode. The default setting is “None”.
|
Table 56: IAX Peer Trunk Configuration Parameters
Basic Settings | |
Provider Name | Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc. |
Host Name | Configure the IP address or URL for the VoIP provider’s server of the trunk. |
Keep Trunk CID | If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”. |
Disable This Trunk | If selected, the trunk will be disabled. |
Caller ID | Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored. Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call : CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID (Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled) 🡪 Global Outbound CID.. |
CallerID Name | Configure the name of the caller to be displayed when the extension has no CallerID Name configured. |
Advanced Settings | |
Codec Preference | Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p and VP8. |
Enable Heartbeat Detection | If enabled, the UCM6200 will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is “No”. |
Heartbeat Frequency | When “Enable Heartbeat Detection” option is set to “Yes”, configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds. |
Maximum Number of Call Lines | The maximum number of concurrent calls using the trunk. The default settings 0, which means no limited. |
Fax Mode | Select Fax mode. The default setting is “None”.
|
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.
Direct Outward Dialing (DOD)
The UCM6200 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 UCM6200:
- To setup DOD go to UCM6200 Web GUI🡪Extension/Trunk🡪VoIP Trunks page.
-
Click
to access the DOD options for the selected SIP Trunk.
- Click “Add DOD” to begin your DOD setup
- For “DOD Number” enter one of the numbers (DIDs) from your SIP trunk provider. In the example above Company ABC received 4 DIDs from their provider. ABC will enter in the number for the CEO’s direct line. Note: DOD number cannot exceed 32 characters.
-
Set the DOD name and If extension number need to be appended to the DID number click on “Add Extension”.
Note: DOD name cannot exceed 32 characters. -
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.

- Click “Save” at the bottom.
Once completed, the user will return to the Edit DOD page that shows all the extensions that are associated to a particular DOD.
DOD can also be assigned to the UCM’s Fax Sending feature. To do this, select “VFAX” from the extension list when creating or editing a DOD number like shown below.

UCM62xx now supports to import/export DOD CSV file for an easy configuration.
- To Import DOD CSV file, click on “Import”.
Note: the file must contain and respect the following columns: DOD Number, DOD Name, Add extension, Local Member, LDAP Members.

- To export the DOD file, users can click on “Export”.
SLA Station
The UCM6200 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 UCM6200 is like BLF but SLA is used to monitor external line i.e., analog trunk on the UCM6200. 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 UCM6200.
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.
Table 57: 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. |
Selected SLA Trunks |
Select a trunk for this SLA from the Available SLA Trunks list. Click on
|
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
- On the UCM6200, 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.
- Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.
- On the UCM6200, 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”.
- On the SIP phone 1, configure to register UCM6200 extension 1002. Configure the MPK as BLF mode and the value must be set to “extension_trunkname”, which is 1002_fxo1 in this case.
- On the SIP phone 2, configure to register UCM6200 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
In the following sections, we will discuss the steps and parameters used to configure and manage outbound rules in UCM6200, these rules are the regulating points for all external outgoing calls initiated by the UCM through all types of trunks: SIP, Analog and Digital.
Configuring Outbound Routes
In the UCM6200, an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern. This allows different patterns to be dialed through different trunks (e.g., “Local” 7-digit dials through an FXO while “Long distance” 10-digit dials through a low-cost SIP trunk). Users can also set up a failover trunk to be used when the primary trunk fails.
Go to Web GUI🡪Extension/Trunk🡪Outbound Routes to add and edit outbound rules.
-
Click on
to add a new outbound route.
- Click on to edit the outbound route.
- Click on
to add a DOD to the outbound route
-
Click on to delete the outbound route.
On the UCM6200, the outbound route priority is based on “Best matching pattern”. For example, the UCM6200 has outbound route A with pattern 1xxx and outbound route B with pattern 10xx configured. When dialing 1000 for outbound call, outbound route B will always be used first. This is because pattern 10xx is a better match than pattern 1xxx. Only when there are multiple outbound routes with the same pattern configured.
Table 58: Outbound Route Configuration Parameters
Calling Rule Name | Configure the name of the calling rule (e.g., local, long_distance, and etc.). Letters, digits, _ and – are allowed. |
X: Any Digit from 0-9. Z: Any Digit from 1-9. N: Any Digit from 2-9. “.“: Wildcard. Match one or more characters. “!“: Wildcard. Match zero or more characters immediately. Example: [12345–9] – Any digit from 1 to 9. Notes:
_X. _NNXXNXXXXX /* 10-digit long distance */ _818X. /* Any number with leading 818 */ | |
After creating the outbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore. However, the route settings will remain in UCM. Users can enable it again when it is needed. | |
Password | Configure the password for users to use this rule when making outbound calls. |
Call Duration Limit | Once call duration limit is enabled, it will set the maximum duration of call-blocking. |
Maximum Call Duration | User can customize the maximum call duration (in seconds) that is allowed for the outbound call. By default, this value is set to 0 means there is no limit for the call duration. |
Warning Time | This option will give caller warning when call duration is approaching to its limit. If the warning time is set to ‘y’, the warning tone will be played to caller when y seconds is left to end the call by UCM. |
Warning Repeat Interval | Once this option is set to ‘z’, it will repeatedly be warning caller every z seconds after the first warning. |
PIN Groups | If selected, the Password, Privilege Level and Enable Filter on Source Caller ID will not take effect. For more details, refer to [PIN Groups] section. |
If enabled and PIN Groups are used, Privilege Levels and Filter on Source Caller ID will also be applied. | |
Password | Configure the password for users to use this rule when making outbound calls. |
Privilege Level | Select privilege level for the outbound rule.
The lowest level required. All users can use this rule.
Users with Local, National, or International level are allowed to use this rule.
Users with National or International level are allowed to use this rule.
The highest level required. Only users with international level can use this rule. The default setting is “Disable”. Please be aware of the potential security risks when using “Internal” level, which means all users can use this outbound rule to dial out from the trunk. |
Enable Filter on Source Caller ID | When enabled, users could specify extensions allowed to use this outbound route. “Privilege Level” is automatically disabled if using “Enable Filter on Source Caller ID”. The following two methods can be used at the same time to define the extensions as the source caller ID.
X: Any Digit from 0-9. Z: Any Digit from 1-9. N: Any Digit from 2-9. “.“: Wildcard. Match one or more characters. “!“: Wildcard. Match zero or more characters immediately. Example: [12345–9] – Any digit from 1 to 9. Note: Multiple patterns can be used. Patterns should be separated by comma “,”. Example: _X. , _NNXXNXXXXX , _818X. |
Attempt to use the configured outbound route CID. This CID will not be used if DOD is configured. It is formatted as “name<number>” or “<number>” or “number”. | |
Send This Call Through Trunk | |
Use Trunk | Select the trunk for this outbound rule. |
Strip | Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk. Example: The users will dial 9 as the first digit of a long-distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In this case, 1 digit should be stripped before the call is placed. |
Prepend | Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped. |
Failover Trunk | Failover trunks can be used to ensure calls can still be made in the scenario that the main trunk is congested or down. UCM6200 support up to 10 failover trunks. Example: The user’s primary trunk is a VoIP trunk and the user would like to use the PSTN when the VoIP trunk is not available. The PSTN trunk can be configured as the failover trunk of the VoIP trunk. |
Strip | Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk. Example: The users will dial 9 as the first digit of a long-distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In this case, 1 digit should be stripped before the call is placed. |
Prepend | Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped. |
Outbound routes can be configured to use different trunks based on time conditions such as holidays, office times, specific times, etc. |
Outbound Blacklist
The UCM6200 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.
Blacklist Manage | |
Add Blacklist Rule | Allows to define a rule based on number(s) or pattern(s) as blacklist entry. Pattern rules: N : Any digit from 2-9 X : Any digit from 0-9 Z : Any digit from 1-9 . : Wildcard, matching one or more characters ! : Wildcard, matching zero or more characters immediately – : Hyphens are used mainly to improve readability and are not involved in pattern matching. |
Note: Users can export outbound route blacklists and delete all blacklist entries. Additionally, users can also import blacklists for outbound routes.

PIN Groups
The UCM6200 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.
Name | Specify the name of the group |
Record In CDR | Specify Whether or not to display the used PIN group and name in a call’s CDR entry. |
PIN Number | Specify the PIN that will be required to dial out. |
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.
User can also import PIN Groups by uploading CSV files for each group. To do this:
- Navigate to Extension/Trunk🡪Outbound Routes🡪PIN Groups and click on the “Choose file to upload” button.
- Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in error messages such as the one below:

- 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.
- Exporting PIN Groups from CSV files:
Press
button under “Options” to download/export PIN Group settings.
Inbound Routes
Inbound routes can be configured via Web GUI🡪Extension/Trunk🡪Inbound Routes.
Note: UCM6200 currently supports up to 5000 Inbound route.
-
Click on
to add a new inbound route.
- Click on “Blacklist” to configure blacklist for all inbound routes.
-
Click on
to edit the inbound route.
-
Click on
to delete the inbound route.
Inbound Rule Configurations
Trunks | Select the trunk to configure the inbound rule. |
Pattern | – All patterns are prefixed with the "_". – Special characters:
Example: [12345-9] - Any digit from 1 to 9. – There are two pattern fields: Pattern and CallerID:
Notes:
Example: |
Disable This Route | After creating the inbound route, users can choose to enable and disable it. |
Allowed to seamless transfer | Allows the selected extension to use this function. If an extension is busy, |
Alert-Info | Configure the Alert-Info, when UCM6200 receives an INVITE request, the |
Special Ringing Tone | This option will allow users to choose the custom ring back tone to play |
Fax Detection | If enabled, fax signals from the trunk during a call will be detected. |
Fax Destination | Configures the destination of faxes.
Note: Please make sure the sending email |
Prepend Trunk Name | If enabled, the trunk name will be added to the caller id name as the displayed caller id name. |
Block Collect Calls | If enabled, collect calls will be blocked. Note 1: Collect calls are indicated by the header “P-Asserted-Service-Info: service-code=Backward Collect Call, P-Asserted-Service-Info: service-code=Collect Call”. Note 2: There is also a global setting for this option in the SIP Settings -> General Settings page. |
Set Caller ID Info | Manipulates Caller ID (CID) name and/or number within the call flow to help identify who is calling. When enabled two field will show allowing to manipulate the CallerID Number and the Caller ID Name. |
CallerID Number | Configures the pattern-matching format to manipulate the numbers of incoming callers or to set a fixed CallerID number for calls that go through this inbound route.
|
CallerID Name | Default string is ${CALLERID(name)} which means the name of an incoming caller, it is a pattern-matching syntax format. A${CALLERID(name)}B means Prepend a character ‘A’ and suffix a character ‘B’ to ${CALLERID(name)}. Not using pattern-matching syntax means setting fix name to incoming caller. |
Enable Route-Level Inbound | Gives uses the ability to configure inbound mode per individual route. When enabled two fields will show allowing to set the Inbound mode and the Inbound mode Suffix. Note: Global inbound mode must be enabled before users can configure route-level inbound mode |
Inbound Mode | Choose the inbound mode for this route. Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If all routes have the option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be affected by the global inbound mode change. |
Inbound Mode Suffix | Dial “Global Inbound Mode feature code + Inbound Mode Suffix” or a route’s assigned suffix to toggle the route’s inbound mode. The BLF subscribed to the inbound mode suffix can monitor the current inbound mode. |
Inbound Multiple Mode | Multiple mode allows user to switch between destinations of the inbound rule by feature codes. Configure related feature codes as described in [Inbound Route: Multiple Mode]. If this option is enabled, user can use feature code to switch between different modes/destinations. |
Dial Trunk | This option shows up only when “By DID” is selected. If enabled, the external users dialing in to the trunk via this inbound route can dial outbound call using the UCM6200’s trunk. |
Privilege Level | This option shows up only when “By DID” is selected.
|
Allowed DID Destination | This option shows up only when “By DID” is selected. This controls the destination that can be reached by the external caller via the inbound route. The DID destination are:
|
Default Destination | Select the default destination for the inbound call.
When “By DID” is used, the UCM6200 will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR, voicemail groups and Fax extension as configured in “DID destination”. If the dialed number matches the DID pattern, the call will be allowed to go through.
|
Strip | Specify the number of digits to strip from the beginning of the DID. This is used when “By DID” is selected in “Default Destination”. |
Prepend | Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped. |
Time Condition | |
Start Time | Select the start time “hour:minute” for the trunk to use the inbound rule. |
End Time | Select the end time “hour:minute” for the trunk to use the inbound rule. |
Frequency | Select either “By Week” or “By Month” for the time condition. If “By Week” is selected, select the weekdays that the time condition will be used. |
Destination | Select the destination for the inbound call under the defined time condition.
When “By DID” is used, the UCM6200 will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR, voicemail groups and Fax extension as configured in “DID destination”. If the dialed number matches the DID pattern, the call will be allowed to go through. Configure the number of digits to be stripped in “Strip” option.
|
Table 60: Inbound Rule Configuration Parameters
Inbound Route: Prepend Example
UCM6200 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:
- If Trunk provides a DID pattern of 18005251163.
- If Strip is set to 8, UCM6200 will strip the first 8 digits.
- If Prepend is set to 2, UCM6200 will then prepend a 2 to the stripped number, now the number become 2163.
- UCM6200 will now forward the incoming call to extension 2163.
Inbound Route: Multiple Mode
In the UCM6200, the user can configure 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 a “Mode 1” destination for all routes. By default, the call coming into the inbound routes will be routed to the default destination.
SIP end devices that have registered on the UCM6200 can dial feature code *62 to switch to inbound route “Mode 1” and dial feature code *61 to switch back to “Default Destination”. Switching between different mode can be easily done without Web GUI login.
For example, the customer service hotline destination has to be set to a different IVR after 7PM. The user can dial *62 to switch to “Mode 1” with that IVR set as the destination before off work.
To customize feature codes for “Default Mode” and “Mode 1”, click on
under “Inbound Routes” page, check “Enable Inbound Multiple Mode” option and change “Inbound Default Mode” and “Inbound Mode 1” values (By default, *61 and *62 respectively).

Inbound Route: Route-Level Mode
In the UCM6200, users can enable Route-Level Inbound Mode to switch between different destinations for each individual inbound route. The inbound Route-Level mode can be enabled under Inbound Route settings.

Figure 126: Inbound Route – Route-Level Mode
Global inbound mode must be enabled before configuring Route-Level Inbound Mode. Additionally, the Mode 1 must be configured as well.
When Route-Level Inbound Mode is enabled, the user can configure a “Default Destination” and a “Mode 1” destination for each specific route. By default, the call coming into this specific inbound route will be routed to the default destination.
Users can toggle the route’s inbound mode by dialing “Global Inbound Mode feature code + Inbound Mode Suffix” and the current inbound route can be monitored by subscribing a BLF to the Inbound Mode Suffix.
For example, Inbound Default Mode feature code is set to *61 and the Inbound Mode suffix for route 1 is set to 1010. To switch the mode of route 1 to Default Mode, users can dial *611010.
Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If all routes have the option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be affected by the global inbound mode change.
Inbound Route: Inbound Mode BLF Monitoring
Users can assign MPKs and VPKs to monitor and toggle the current global inbound mode of the UCM.
To do this, please refer to the following steps:
- Access the UCM web GUI and navigate to Extension/Trunk🡪Inbound Routes.
-
Click on the
button and enable Inbound Multiple Mode.
- Edit the subscribe number field to the desired BLF value.

- 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”.

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

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]
- Fax Detection: No, Yes.
- Fax Type: Extension, Fax to Email.
- Fax Destination: [Extension_number] or [Email address]
- 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 Detection
The UCM can automatically detect fax signaling from FXO ports and forward these calls to specified destinations.
FAX with Two Media
Since UCM’s system is now based on Asterisk 13, fax re-invites that negotiate with multiple codecs are now supported. If a re-invite contains both T.38 and PCMU/PCMA codecs, T.38 will be prioritized.
Blacklist Configurations
In the UCM6200, 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
to add to the list. Anonymous can also be added as a Blacklist Number.
-
To remove a number from the Blacklist, select the number in “Blacklist list” and click on
or click on
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 “choose file to upload” to upload blacklist file in csv format. The supported csv format is as below.

Note:
Users could also add a number to the Blacklist or remove a number from the Blacklist by dialing the feature code for “Blacklist Add’ (default: *40) and “Blacklist Remove” (default: *41) from an extension. The feature code can be configured under Web GUI🡪Call Features🡪Feature Codes.
Conference
The UCM6200 supports conference room allowing multiple rooms used at the same time:
- UCM6202/6204 supports up to 3 conference rooms allowing up to 25 simultaneous PSTN or IP participants.
- UCM6208 supports up to 6 conference rooms allowing up to 32 simultaneous PSTN or IP participants.
The conference room configurations can be accessed under Web GUI🡪Call Features🡪Conference. In this page, users could create, edit, view, invite, manage the participants and delete conference rooms. The conference room status and conference call recordings (if recording is enabled) will be displayed in this web page as well.
Conference Room Configurations
- Click on “Create New Conference Room” to add a new conference room.
-
Click on
to edit the conference room.
-
Click on
to delete the conference room.
Table 61: Conference Room Configuration Parameters
Configure the conference number for the users to dial into the conference. Note: The conference extension number can contain up to 64 characters. | |
When configured, the users who would like to join the conference call must enter this password before accessing the conference room. Notes:
| |
Host Password | Configure the password to join the conference room as administrator. Conference administrator can manage the conference call via IVR (if “Enable Caller Menu” is enabled) as well as invite other parties to join the conference by dialing “0” (permission required from the invited party) or “1” (permission not required from the invited party) during the conference call. Notes:
|
Enable Caller Menu | If enabled, conference participant could press the * key to access the conference room menu. The default setting is “No”. |
Record Conference | If enabled, the calls in this conference room will be recorded automatically in a .wav format file. All the recording files will be displayed and can be downloaded in the conference web page. The default setting is “No”. |
Quiet Mode | If enabled, if there are users joining or leaving the conference, voice prompt or notification tone will not be played. The default setting is “No”. Note: “Quiet Mode” and “Announce Callers” cannot be enabled at the same time. |
If there is only one participant in a conference room, a kick warning prompt will play at the configured interval. If no input from the participant is received after the prompt, he will be automatically kicked out of the conference. The valid range is 1-60 minutes. | |
Wait For Admin | If enabled, the participants will not hear each other until the conference administrator joins the conference. The default setting is “No”. Note: If “Quiet Mode” is enabled, the voice prompt for “Wait For Admin” will not be announced. |
Allow User Invite | If enabled, users could press 0 to invite other users (with the users’ permission) or press 1 to invite other users (without the user’s permission) to join the conference. The default setting is “No”. Note: Conference administrator can always invite other users without enabling this option. |
If enabled, the caller will be announced to all conference participants when there the caller joins the conference. The default setting is “No”. Notes:
| |
Public Mode | If enabled, no authentication will be required when joining the conference call. The default setting is “Yes”. |
Play Hold Music | If enabled, the UCM6200 will play Hold music when there is only one user in the conference. The default setting is “No”. |
Music On Hold | Select the music on hold playlist to be played in conference call. Music On Hold playlist can be set up under Web GUI🡪PBX Settings🡪Music On Hold. |
Custom Music On Hold | Select a custom Music On Hold. |
Skip Authentication | If enabled, the invitation from Web GUI for a conference room with password will skip the authentication for the invited users. The default setting is “No”. |
Conference Settings contains the following options:
Enable Talk detection | If enabled, the AMI will send the corresponding event when a user starts or ends talking. |
DSP Talking Threshold | The time in milliseconds of sound above what the dsp has established as base line silence for a user before a user is considered to be talking. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 128. |
DSP Silence Threshold | The time in milliseconds of sound falling within the what the dsp has established as base line silence before a user is considered to be silent. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 2500. |
Enable Neteq | If enabled, conference audio quality may improve, but system performance will be lowered. |
Users can check the talking Caller IDs in conference control page (UCM WebUI🡪Call Features🡪Conference). The image will move up and down when the user is talking.

Conference Call Operations
Join a Conference Call
Users could dial the conference room extension to join the conference. If password is required, enter the password to join the conference as a normal user, or enter the admin password to join the conference as administrator.
Invite Other Parties to Join Conference
When using the UCM6200 conference room., there are two ways to invite other parties to join the conference.
- Invite from Web GUI.
For each conference room in UCM6200 Web GUI🡪Call Features🡪Conference, there is an icon
for option “Invite a participant”. Click on it and enter the number of the party you would like to invite. Then click on “Add”. A call will be sent to this number to invite it to the conference.

- Invite by dialing 0 or 1 during conference call.
A conference participant can invite other parties to the conference by dialing during the conference call. Please make sure “Allow User Invite” is turned on for the conference room first. Enter 0 or 1 during the conference call. Follow the voice prompt to input the number of the party you would like to invite. A call will be sent to this number to join it into the conference.
If 0 is entered, once the invited party picks up the invitation call, the system will ask the party to accept or reject the invitation.
If 1 is entered, once the invited party picks up the call, they will automatically be brought into the conference.
Note:
Conference administrator can always invite other parties from the phone during the call by entering 0 or 1. To join a conference room as administrator, enter the admin password when joining the conference. A conference room can have multiple administrators.
During The Conference
During the conference call, users can manage the conference from Web GUI or IVR.
- Manage the conference call from Web GUI.
Log in UCM6200 Web GUI during the conference call, the participants in each conference room will be listed.
-
Click on
to kick a participant from the conference.
-
Click on
to mute the participant.
-
Click on
to lock this conference room so that other users cannot join it anymore.
-
Click on
to invite other users into the conference room.
- Manage the conference call from IVR.
If “Enable Caller Menu” is enabled, conference participant can input * to enter the IVR menu for the conference. Please see options listed in the table below.
Table 63: Conference Caller IVR Menu
Conference Administrator IVR Menu | |
1 | Mute/unmute yourself. |
2 | Lock/unlock the conference room. |
3 | Kick the last joined user from the conference. |
4 | Decrease the volume of the conference call. |
5 | Decrease your volume. |
6 | Increase the volume of the conference call. |
7 | Increase your volume. |
8 | More options.
|
Conference User IVR Menu | |
1 | Mute/unmute yourself. |
4 | Decrease the volume of the conference call. |
5 | Decrease your volume. |
6 | Increase the volume of the conference call. |
7 | Increase your volume. |
8 | Exit the caller menu and return to the conference. |
Note:
When there is participant in the conference, the conference room configuration cannot be modified.
Google Service Settings Support
UCM6200 now supports Google OAuth 2.0 authentication. This feature is used for supporting UCM6200 conference scheduling system. Once OAuth 2.0 is enabled, UCM6200 conference system can access Google calendar to schedule or update conference.
Google Service Settings can be found under Web GUI🡪Call Features🡪Conference🡪Google Service Settings🡪Google Service Settings.

If you already have OAuth2.0 project set up on Google Developers web page, please use your existing login credential for “OAuth2.0 Client ID” and “OAuth2.0 Client Secret” in the above figure for the UCM6200 to access Google Service.
If you do not have OAuth2.0 project set up yet, please following the steps below to create new project and obtain credentials:
- Go to Google Developers page https://console.developers.google.com/start Create a New Project in Google Developers page.

- Enable Calendar API from API Library.
- Click “Credentials” on the left drop down menu to create new OAuth2.0 login credentials.

- Use the newly created login credential to fill in “OAuth2.0 Client ID” and “OAuth2.0 Client Secret”.
- Click “Get Authentication Code” to obtain authentication code from Google Service.
- Now UCM6200 is connected with Google Service.
You can also configure the Status update, which automatically refresh your Google Calendar with the configured time (m). Note: Zero means disable.
Conference Schedule
Conference Schedule can be found under UCM6200 Web GUI 🡪 Call Features 🡪 Conference🡪 Conference Schedule. Users can create, edit, view and delete a Conference Schedule.
- Click on “Create New Conference Schedule” to add a new Conference Schedule.
- Click on the scheduled conference to edit or delete the event.
After the user configures UCM6200 with Google Service Settings [Google Service Settings Support] and enables Google Calendar for Conference Schedule, the conference schedule on the UCM6200 can be synchronized with Google Calendar for authorized Google account.
Table 64: Conference Schedule Parameters
Schedule Options | |
Conference Subject | Configure the topic of the scheduled conference. Letters, digits, _ and – are allowed. |
Conference Room | Select a conference room for this scheduled conference. |
Conference Password | Conference login password. |
Host Password | Host Password. |
Kick Time(m) | Set kick time before conference starts. When kick time is reached, a warning prompt will be played for all attendees in the conference room. After 5 minutes, this conference room will be cleared and locked for the scheduled conference to begin. Note: Kick Time cannot be less than 6 minutes in order to clear the conference room. |
Wait for Host |
If enabled, conference participants will not hear each other until the host joins the conference.
|
Description | The description of scheduled conference. |
Repeat | Repeat interval of scheduled conference. By default, set to single event. |
Schedule Time | Configure the beginning date and duration of scheduled conference. Note: Please pay attention to avoid time conflict on schedules in the same conference room. |
Duration of the conference meeting. Note: The maximum allowed meeting duration that can be set is 8 hour(s). | |
Enable Google Calendar | Select this option to synchronize scheduled conference with Google Calendar. Note: Google Service Setting OAuth2.0 must be configured on the UCM6200. Please refer to section [Google Service Settings Support]. |
Send email notification | Sends Email notification to the extension. |
Conference Administrator | Select the administrator of scheduled conference from selected extensions. Note: “Public Mode” must be disabled from Conference Room Options tab. |
Local Extension | Select available extensions from the list to attend scheduled conference. |
Remote Extension | Select available extensions from the remote peer PBX. Note: “LDAP Sync” must be enabled on the UCM6200 in order to view remote extensions here. |
Special Extension | Add extensions that are not in the list (both local and remote list). If the user wishes to add the special extension, please make sure that the UCM’s outbound routes will allow calling that special extension. |
Remote Conference | Invite a remote conference. |
Conference Room Options | |
Password | Configure conference room password. Please note that if “Public Mode” is enabled, this option is automatically disabled. |
Host Password | Configure the password to join as conference administrator. Please note that if “Public Mode” is enabled, this option is automatically disabled. |
Enable Caller Menu | If this option is enabled, conference participants will be able to access conference room menu by pressing the * key. |
Record Conference | If this option is enabled, conference call will be recorded in .wav format. The recorded file can be found from Conference page. |
Quiet Mode | If this option is enabled, the notification tone or voice prompt for joining or leaving the conference will not be played. Note: Option “Quiet Mode” and option “Announce Caller” cannot be enabled at the same time. |
Wait For Admin | If this option is enabled, the participants in the conference will not be able to hear each other until conference administrator joins the conference. Note: If “Quiet Mode” is enabled, voice prompt for this option will not be played. |
Allow User Invite | If this option is enabled, the user can:
Note: Conference Administrator is always allowed to access this menu. |
Announce Callers | If this option is enabled, when a participant joins the conference room, participant’s name will be announced to all members in the conference room. Note: Option “Quiet Mode” and option “Announce Caller” cannot be enabled at the same time. |
Public Mode | If this option is enabled, no authentication is required for entering the conference room. Note: Please be aware of the potential security risks when turning on this option. |
Play Hold Music | If this option is enabled, UCM6200 will play Hold Music while there is only one participant in the conference room, or the conference is not yet started. |
Skip Authentication | If this option is enabled, the invitation from Web GUI via a trunk with password will not require authentication. Note: Please be aware of the potential security risks when turning on this option. |
Cleaner Options
Cleaner Options | |
Enable Conference Schedules Cleaner | If this option is enabled, conference schedules will be automatically cleaned as configured. |
Conference Schedules Clean Time | Enter the clean time (in hours). The valid range is from 0 to 23. |
Clean Interval | Enter the clean interval (in days). The valid range is from 1 to 30. |
After configuring the scheduled conference, it will be shown under Conference Schedule page as below:

Once the conference room is scheduled, at the kick time, all users will be removed from conference room and no extension can join the conference room anymore. At the scheduled conference time, UCM6200 will send INVITE to the extensions that have been selected for conference.
Notes:
- Please make sure that outbound route is properly configured for remote extensions to join the conference.
- Once Kick Time is reached, Conference Schedule is locked and cannot be modified.
Contact Group
Users can now quickly invite multiple participants at once to a conference via conference contact groups. Up to 5 contact groups can be created. The maximum allowed number of contacts per group is based on the UCM model’s conference participant limit: 25 for 6202/6204, 32 for 6208, 64 for 6510.
Each contact group must have a password configured, which will be required when inviting the specified contact group to a conference. Additionally, an audio file can be uploaded to each group to be used to announce the contact group name such as “Sales” or “Marketing”. The default announcement for each group is “Conference Contact Group 1”, “Conference Contact Group 2”, etc.

Contact Group Configurations
- Click on “Create New Contact Group” to add a new Contact Group.
-
Click on
to edit the Contact Group.
-
Click on
to delete the Contact Group.
Table 65: Contact Group Parameters
Name | Name associated to the contact group. |
Password | Password required to invite the specified contact group to a conference. |
Prompt | Audio file that can be uploaded to the group to announce the contact group name such as “Sales” or “Marketing”. The default announcement for each group is “Conference Contact Group 1”, “Conference Contact Group 2”, etc. |
Members | Contacts that needs to be added in each group. |
Type | Type of the members to be added, it can be either Extensions or a self-defined number. |
Conference Recordings
The UCM6200 allows users to record the conference call and retrieve the recording from Web GUI🡪Call Features🡪Conference🡪Record Conference.
To record the conference call, when the conference room is in idle, enable “Record Conference” from the conference room configuration dialog. Save the setting and apply the change. When the conference 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 “Delate All Recording Files” or delete multiple recording files at once by clicking on “Delete Selected Recording Files” after selecting the recording files.
Conference Call Statistics
Conference reports will now be generated after every conference. These reports can be exported to a .CSV file for offline viewing. The conference report page can be accessed by clicking on the Call Statistics button on the main Conference page.

Video Conference
With the UCM you can easily create, schedule, manage, and join video conference calls, from your desktop or laptop computer. UCM Video conferencing uses WebRTC technology, so all the participants don’t have to download and install any additional software or plugins. If upgrading from firmware that does not have this functionality, system administrators must first toggle on the Enable WebRTC option for extensions that want to use the UCM’s WebRTC video conferencing feature. The video conference configurations can be accessed under Web GUI🡪Call Features🡪Video Conference. Here, users can create and manage video conference rooms and schedule video conferences.
Video Conference Room Configurations
- Click on “Create New Conference Room” to add a new conference room.
-
Click on
to edit the conference room.
-
Click on
to delete the conference room.
Table 66: Video Conference room Configuration Parameters
Extension | Configure the conference number for the users to dial into the conference. Note: Up to 64 characters. |
Password | When configured, the users who would like to join the conference call must enter this password before accessing the conference room. Note:
|
Conference Settings

Basic Settings | |
Video Conferencing | This option should be enabled in order to activate the Video Conference feature. |
Bind UDP Port | Configure the UDP port number for MCM. The standard UDP port for MCM is 5062. |
Packet Loss Retransmission | Configure to enable Packet Loss Retransmission. |
FEC | If enabled, the Forward Error Correction (FEC) will be activated. The default setting is “No”. |
Enable Talk Detection | If enabled, the AMI will send the corresponding event when a user starts or stops talking. |
DSP Talking Threshold | The amount of time (ms) that sound exceeds what the DSP has established as the baseline for silence before a user is considered to be talking. This value affects several operations and should not be changed unless the impact on call quality is fully understood. |
DSP Silence Threshold | The amount of time(ms) that sound falls within what the DSP has established as the baseline for silence before a user is considered be silent. This value affects several operations and should not be changed unless the impact on call quality is fully understood. |
Conference Schedule
Conference Schedule can be found under UCM Web GUI 🡪 Call Features🡪Video Conference 🡪 Conference Schedule. Users can create, edit, view and delete a Conference Schedule.
- Click on “Schedule New Conference” to add a new Conference Schedule.
- Click on the scheduled conference to edit or delete the event.
Table 67: Video Conference Schedule Parameters
Schedule Options | |
Conference Subject | Configure the name of the scheduled conference. Letters, digits, _ and – are allowed. |
Conference Room | Select a conference room for this scheduled conference. |
Conference Password | Configure conference room password. Please note that if “Public Mode” is enabled, this option is automatically disabled. |
Kick Time(m) | Configure the time before the scheduled conference. When this time is reached, a warning prompt will be played, and all attendees currently in the scheduled conference room will be kicked after 5 mins. The conference room will be locked until the scheduled conference begins. Default value is 10 min. |
Start Time | Configure the beginning date and duration of scheduled conference. Note: Please be mindful to avoid schedule conflicts for the same conference room. |
Host | Set the admin of this scheduled conference from the following list of members. |
Repeat | Choose when to repeat a scheduled conference. |
Local Extension | Select the extensions from the list to attend this scheduled conference. |
Remote Extension | The remote extension in the peer PBX connected to the local PBX via LDAP sync. |
Special Extension | Add extensions that are not in the list (both local and remote list). If the user wishes to add the special extension, please match the pattern on the outbound route. |
Description | Set a description of scheduled conference. |
Time Zone | Configure the conference Time Zone. |
Once created, the Web GUI will display scheduled conference in Conference Schedule.
Please see figure below:

Once the conference room is scheduled, at the kick time, all users will be removed from conference room and no extension is allowed to join the conference room anymore. At the scheduled conference time, UCM will send INVITE to the extensions that have been selected for conference.
Notes:
- Video conferencing can be resource-intensive and may cause performance issues with the UCM when used.
- To ensure the best experience, please use Google Chrome (v67 or higher) or Mozilla Firefox (v60).
Wave WebRTC Video Calling & Conferencing
Web audio and video calls and conferencing can now be achieved through the UCM’s new WebRTC page. To get started with this new feature, please make sure to:
- Navigate to Value-Added Features 🡪 WebRTC and enable WebRTC support.

- Enable the WebRTC on the extensions that would use this feature under Extension / Trunk 🡪 Extensions.
The UCM offers the possibility to log in to an extension via Grandstream Wave Portal using user portal password in addition to SIP registration password, where it offers a sleek interface to host conferences, receive email reminders for scheduled conferences, manage contacts, initiate calls, call transfer, chat functionality and more.
Access the page by adding “/gswave” after the UCM’s server address and port. (e.g. https://my.ucm.com:8089/gswave).
Note: Starting with 1.0.19.27, the registration limit was increased to 300 for UCM62xx. The limit is the same regardless of whether the user is making voice calls or video calls
For more details about the WebRTC feature, please refer to the following guide:
https://documentation.grandstream.com/knowledge-base/wave-video-calling-conferencing-guide/
IPVideoTalk Meetings
UCM extensions can now dial into IPVT (IPVideoTalk) meetings by creating a peer trunk to an IPVT server. However, users must make sure that the IPVT server they are peering to also has a peer trunk to their UCM configured. This setting can be found in Admin Center🡪SIP Trunk Configuration.

Next, users must create a peer trunk on the UCM to the IPVT server. Enter one of the following addresses based on the desired connection protocol:
- TCP: pro.ipvideotalk.com:20000
- TLS: pro.ipvideotalk.com:20001

Make sure that the Transport field is either “TCP” or “TLS”. Save and apply changes to create the trunk.
Next, edit the newly created trunk and click on the Advanced Settings tab. Make sure the IPVT Mode option is checked. Otherwise, you may experience audio issues when dialing into IPVT.

Finally, create an outbound route for this trunk. This route will be used to dial IPVT meeting IDs. Due to IPVT meeting IDs having a random assortment of numbers, it is recommended to use a unique code to precede the meeting ID so that UCM can direct calls to the IPVT trunk without fail (e.g., *99). In the below image, “x.” would be the meeting ID.

However, if a unique code is used, users must also configure the Strip field to remove the unique code from the meeting ID before the call is sent to IPVT.

In this example, the Strip field has “3” configured to remove the example unique code *99 from the dialed number before the call is routed out to the IPVT server. Once this outbound route has been created, users can now use a UCM extension to dial IPVT meeting rooms.
Note: An IPVT account can have only 1 SIP trunk peered to it.
IVR
Configure IVR
IVR configurations can be accessed under the UCM6200 Web GUI🡪Call Features🡪IVR. Users could create, edit, view and delete an IVR.
Note: UCM6200 currently supports up to 500 IVR entry.
- Click on “Create New IVR” to add a new IVR.
-
Click on
to edit the IVR configuration.
-
Click on
to delete the IVR.

Table 68: IVR Configuration Parameters
Basic Settings | |
Name | Configure the name of the IVR. Letters, digits, _ and – are allowed. |
Extension | Enter the extension number for users to access the IVR. |
Dial other Extensions | If enabled, the caller is allowed to dial extensions other than the ones explicitly defined. |
DID Destination | This option shows up only when “By DID” is selected. This controls the destination that can be reached by the external caller via the inbound route. The DID destination are:
|
Dial Trunk | If enabled, all callers to the IVR is allowed to use trunk. The permission must be configured for the users to use the trunk first. The default setting is “No”. |
Permission | Assign permission level for outbound calls if “Dial Trunk” is enabled. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level. The default setting is “Internal”. If the user tries to dial outbound calls after dialing into the IVR, the UCM6200 will compared the IVR’s permission level with the outbound route’s privilege level. If the IVR’s permission level is higher than (or equal to) the outbound route’s privilege level, the call will be allowed to go through. |
If enabled, the UCM will replace the caller display name with the IVR name the caller know whether the call is incoming from a direct extension or an IVR. | |
Alert-Info | When present in an INVITE request, the alert-Info header field specifies and alternative ring tone to the UAS. |
Select an audio file to play as the welcome prompt for the IVR. Click on “Prompt” to add additional audio file under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt. Note: Up to 5 welcome prompts can be used. | |
Digit Timeout | Configure the timeout between digit entries. After the user enters a digit, the user needs to enter the next digit within the timeout. If no digit is detected within the timeout, the UCM6200 will consider the entries complete. The default timeout is 3 seconds. |
Response Timeout | After playing the prompts in the IVR, the UCM6200 will wait for the DTMF entry within the timeout (in seconds). If no DTMF entry is detected within the timeout, a timeout prompt will be played. The default setting is 10 seconds. |
Response Timeout Prompt | Select the prompt message to be played when timeout occurs. |
Invalid Input Prompt | Select the prompt message to be played when an invalid extension is pressed. |
Response Timeout Prompt Repeats | Configure the number of times to repeat the prompt if no DTMF input is detected. When the loop ends, it will go to the timeout destination if configured, or hang up. The default setting is 3. |
Invalid Input Prompt Repeats | Configure the number of times to repeat the prompt if the DTMF input is invalid. When the loop ends, it will go to the invalid destination if configured, or hang up. The default setting is 3. |
Language | Select the voice prompt language to be used for this IVR. The default setting is “Default” which is the selected voice prompt language under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. The dropdown list shows all the current available voice prompt languages on the UCM6200. To add more languages in the list, please download voice prompt package by selecting “Check Prompt List” under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. |
Key Pressing Events | |
Key Press Event: Press 0 Press 1 Press 2 Press 3 Press 4 Press 5 Press 6 Press 7 Press 8 Press 9 Press * Timeout Invalid | Select the event for each key pressing for 0-9, *, Timeout and Invalid. The event options are:
|

IVR Black/Whitelist
In some scenarios, the IPPBX administrator needs to restrict the extensions that can be reached from IVR.
For example, the company CEO and directors prefer only receiving calls transferred by the secretary, some special extensions are used on IP surveillance end points which shouldn’t be reached from external calls via IVR for privacy reason. UCM has now added blacklist and whitelist in IVR settings for users to manage this.
Note: up to 500 extensions are allowed on the black/white list.
To use this feature, log in UCM Web GUI and navigate to Call Features🡪IVR🡪Create/Edit IVR: IVR
Black/White List.
- If the user selects “Blacklist Enable” and adds extension in the list, the extensions in the list will not be allowed to be reached via IVR.
- If the user selects “Whitelist Enable” and adds extension in the list, only the extensions in the list can be allowed to be reached via IVR.
Create Custom Prompt
To record new IVR prompt or upload IVR prompt to be used in IVR, click on “Prompt” next to the “Welcome Prompt” option and the users will be redirected to Custom Prompt page. Or users could go to Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page directly.

Once the IVR prompt file is successfully added to the UCM6200, it will be added into the prompt list options for users to select in different IVR scenarios.
Voice Prompt
The UCM6200 supports multiple languages in Web GUI as well as system voice prompt. The following languages are currently supported in system voice prompt:
English (United States), Arabic, Chinese, Dutch, English (United Kingdom), French, German, Greek, Hebrew, Italian, Polish, Portuguese, Russian, Spanish, Catalan, Swedish, Czech and Turkish.
English (United States) and Chinese voice prompts are built in with the UCM6200 already. The other languages provided by Grandstream can be downloaded and installed from the UCM6200 Web GUI directly. Additionally, users could customize their own voice prompts, package them and upload to the UCM6200.
Language settings for voice prompt can be accessed under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. Additionally, UCM6200 allows to customize specific prompt instead of full language package, and it provides ability to upload greeting files for extensions.
Language Settings
Download and Install Voice Prompt Package
To download and install voice prompt package in different languages from UCM6200 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 UCM6200. 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 UCM6200 system voice prompt or delete it from the UCM6200.
Upload Language Package
On the UCM6200, if the user needs to replace some specific customized prompt, the user can upload a single specific customized prompt from Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings instead of the entire language pack.

The package file should follow below requirements:
Each file uploaded must be under 50MB.
Package structure:
[Package]
├ [voice prompt dir]
│ ├ [... dir]
│ └ [... files]
└ info.txt (containing the language name for display, in UTF8)
- Language dir name format:
- Custom dir name format: language_xxx;
For example: If there is a Chinese custom directory named zh_xxx, the custom voice prompt in zh_xxx would be used first, then the Chinese voice prompt zh, then use the default language prompt (en); If not named the format as above, then the custom prompt will be used first, then use the default language prompt (en).
For more details, please refer to:
https://www.grandstream.com/sites/default/files/Resources/ucm_voiceprompt_customization_guide.zip
Custom Prompt
Record New Custom Prompt
In the UCM6200 Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page, click on “Record New IVR Prompt” and follow the steps below to record new IVR prompt.

- Specify the IVR file name.
- Select the format (GSM or WAV) for the IVR prompt file to be recorded.
- Select the extension to receive the call from the UCM6200 to record the IVR prompt.
- Click the “Record” button. A request will be sent to the UCM6200. The UCM6200 will then call the extension for recording the IVR prompt from the phone.
- Pick up the call from the extension and start the recording following the voice prompt.
- 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 IVR Prompt” in Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page to upload the file to the UCM6200. The following are required for the IVR prompt file to be successfully uploaded and used by the UCM6200:
- 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.
- Filename should not exceed 100 characters.

Click on “choose file to upload” to select audio file from local PC and to start uploading. Once uploaded, the file will appear in the IVR Prompt web page.
Download All Custom Prompt
On the UCM6200, 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
. 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.
Username Prompt Customization
There are two ways to customize/set new username prompt:
Upload Username Prompt File from Web GUI
- First, users should have a pre-recorded file respecting the following format:
- PCM encoded / 16 bits / 8000Hz mono.
- In “.GSM” or “.WAV” format.
- File size under 5MB.
- 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.
-
Go under web GUI PBX Settings 🡪 Voice Prompt 🡪 Username Prompt and click on
button.
- Select the recorded file to upload it and press Save and Apply Settings.
-
To delete username prompts, either click on the button
to delete a single file or select multiple files and click on Delete button to delete multiple prompts at once.
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.
Voicemail
Configure Voicemail
If the voicemail is enabled for UCM6200 extensions, the configurations of the voicemail can be globally set up and managed under Web GUI🡪Call Features🡪Voicemail.

Max Greeting | Configure the maximum number of seconds for the voicemail greeting. The default setting is 60 seconds. |
Dial ‘0’ For Operator | If enabled, the caller can press 0 to exit the voicemail application and connect to the configured operator’s extension. |
Operator Extension | Select the operator extension, which will be dialed when users press 0 to exit voicemail application. The operator extension can also be used in IVR. |
Max Messages Per Folder | Configure the maximum number of messages per folder in users’ voicemail. The valid range 10 to 1000. The default setting is 50. |
Max Message Time | Select the maximum duration of the voicemail message. The message will not be recorded if the duration exceeds the max message time. The default setting is 15 minutes. The available options are:
|
Min Effective Message Time | Configure the minimum duration (in seconds) of a voicemail message. Messages will be automatically deleted if the duration is shorter than the Min Message Time. The default setting is 3 seconds. The available options are:
Note: Silence and noise duration are not counted in message time. |
Announce Message Caller-ID | If enabled, the caller ID of the user who has left the message will be announced at the beginning of the voicemail message. The default setting is “No”. |
Announce Message Duration | If enabled, the message duration will be announced at the beginning of the voicemail message. The default setting is “No”. |
Play Envelope | If enabled, a brief introduction (received time, received from, and etc.) of each message will be played when accessed from the voicemail application. The default setting is “Yes”. |
Play from Last | If enabled, UCM will play from the voice message left most recently; if disabled, UCM will play from the earliest left voice message |
Allow User Review | If enabled, users can review the message following the IVR before sending. |
If enabled, external callers routed by DID and reaching VM will be prompted by the UCM with 2 options:
To leave a message for the extension reached by DID.
This will allow caller to access any extension VM after entering extension number and its VM password. Note: This option applies to inbound call routed by DID only. The default setting is “Disabled”. | |
Enables the forwarding of voicemail to remote extensions on peered SIP trunks. The default setting is “Disabled”. | |
Configures the default voicemail password that will be used when an extension is reset. |
Note: Resetting an extension will reset Voicemail Password, Send Voicemail to Email, and Keep Voicemail after Emailing values to default. Previous custom voicemail prompts and messages will be deleted.
Access Voicemail
If the voicemail is enabled for UCM6200 extensions, the users can dial the voicemail access number (by default *97) to access their extension’s voicemail. The users will be prompted to enter the voicemail password and then can enter digits from the phone keypad to navigate in the IVR menu for different options.
Otherwise, the user can dial the voicemail access code (by default *98) followed by the extension number and password in order to access to that specific extension’s voicemail.
Main Menu | Sub Menu 1 | Sub Menu 2 |
1 – New messages | 3 - Advanced options | 1 - Send a reply |
2 - Call the person who sent this message | ||
3 - Hear the message envelop | ||
4 - Leave a message | ||
* - Return to the main menu | ||
5 - Repeat the current message | ||
7 - Delete this message | ||
8 - Forward the message to another user | ||
9 – Save | ||
* - Help | ||
# - Exit | ||
2 – Change folders | 0 - New messages | |
1 - Old messages | ||
2 - Work messages | ||
3 - Family messages | ||
4 - Friend messages | ||
# - Cancel | ||
3 – Advanced options | 1 - Send a reply | |
2 - Call the person who sent this message | ||
3 - Hear the message envelop | ||
4 - Leave a message | ||
* - Return to the main menu | ||
0 – Mailbox options | 1 - Record your unavailable message | 1 - Accept this recording |
2 - Listen to it | ||
3 - Re-record your message | ||
2 - Record your busy message | 1 - Accept this recording | |
2 - Listen to it | ||
3 - Re-record your message | ||
3 - Record your name | 1 - Accept this recording | |
2 - Listen to it | ||
3 - Re-record your message | ||
4 - Record temporary greeting | 1 - Accept this recording | |
2 - Listen to it | ||
3 - Re-record your message | ||
5 - Change your password | ||
* - Return to the main menu |
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 UCM6200 can be configured to send the voicemail as attachment to Email. Click on “Voicemail Email Settings” button to configure the Email attributes and content.
Table 71: Voicemail Email Settings
Attach Recordings to | If enabled, voicemails will be sent to user’s Email address. The default setting is “Yes”. |
Keep Recordings | If enabled, voicemail will be stored in the UCM6200 after the email is sent. The default setting is “Yes”. |
Template for Voicemail Emails | Fill in the “Subject:” and “Message:” content, to be used in the Email when sending to the user. The template variables are:
|

Click on “Load Default Settings” button to view the default template as an example.
Configure Voicemail Group
The UCM6200 supports voicemail group and all the extensions added in the group will receive the voicemail to the group extension. The voicemail group can be configured under Web GUI 🡪 Call Features 🡪 Voicemail 🡪 Voicemail Group. Click on “Create New Voicemail Group” to configure the group.
Table 72: Voicemail Group Settings
Voicemail Group Extension | Enter the Voicemail Group Extension. The voicemail messages left to this extension will be forwarded to all the voicemail group members. |
Name | Configure the Name to identify the voicemail group. Letters, digits, _ and – are allowed. |
Voicemail Password | The Voicemail password for the user to check Voicemail messages. |
Email Address | The Email address of current user. |
Select available extensions from the left list and add them to the right list. The extensions need to have voicemail enabled to be listed in available mailboxes list. Note: Members selected cannot exceed 30 extensions. |
Ring Group
The UCM6200 supports ring group feature with different ring strategies applied to the ring group members. This section describes the ring group configuration on the UCM6200.
Configure Ring Group
Ring group settings can be accessed via Web GUI🡪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.
Table 73: Ring Group Parameters
Ring Group Name | Configure ring group name to identify the ring group. Letters, digits, _ and – are allowed. |
Extension | Configure the ring group extension. |
Members |
Select available users from the left side to the ring group member list on the right side. Click on
|
LDAP Phonebook |
Select available remote users from the left side to the ring group member list on the right side. Click on
|
Ring Strategy | Select the ring strategy. The default setting is “Ring in order”.
Ring all the members at the same time when there is incoming call to the ring group extension. If any of the member answers the call, it will stop ringing.
Ring the members with the order configured in ring group list. If the first member does not answer the call, it will stop ringing the first member and start ringing the second member. |
Select the “Music On Hold” Class of this Ring Group, “Music On Hold” can be managed from the “Music On Hold” panel on the left. | |
This option is to set a custom prompt for a ring group to announce to caller. Click on ‘Prompt’, it will direct the users to upload the customized voice prompts. Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom Prompt, where they could record new prompt or upload prompt files. | |
Ring Timeout for Each Member | Configure the number of seconds to ring each member. If set to 0, it will keep ringing. The default setting is 30 seconds. Note: The actual ring timeout might be overridden by users if the phone has ring timeout settings as well. |
Auto Record | If enabled, calls on this ring group will be automatically recorded. The default setting is No. The recording files can be accessed from Web GUI🡪CDR🡪Recording Files. |
This allows the UCM to work with endpoint-configured call forwarding settings to redirect calls to ring group. For example, if a member wants to receive calls to the ring group on his mobile phone, he would have to set his endpoint’s call forwarding settings to his mobile number. By default, it is disabled. However, this feature has the following limitations:
| |
If enabled, the UCM will replace the caller display name with the Ring Group name the caller know whether the call is incoming from a direct extension or a Ring Group. | |
If enabled calls to busy agents will be skipped and sent to the following available ones. Default is enabled. | |
If enabled, users could select extension, voicemail, ring group, IVR, call queue, voicemail group as the destination if the call to the ring group has no answer. Note: Voicemail Password and Email address (limited by 128 characters) are required if voicemail is selected as the destination. Voicemail system will mention if a voicemail is from a ring group. | |
Default Destination | The call would be routed to this destination if no one in this ring group answers the call. |
Remote Extension in Ring Group
Remote extensions from the peer trunk of a remote UCM6200 can be included in the ring group alongside local extensions. An example of Ring Group with peer extensions is presented in the following:
- Create SIP trunks from UCM A to UCM B and vice-versa on the Extension/Trunk->VoIP Trunks page. Additionally, please create the appropriate outbound and inbound routes for these SIP trunks.
-
Click edit button in the menu
, and check if Sync LDAP Enable is toggled on.

- In the scenario that the LDAP is not synced automatically, users can manually sync LDAP phonebooks by clicking on the LDAP sync button that’s highlighted in the following image.
- Under Ring Groups setting page, click “Add”. Ring Groups can be found under Web GUI🡪Call Features🡪Ring Groups.
- If the LDAP phonebook has synced correctly, the remote UCM’s extensions should appear in the Create/Edit Ring Group->LDAP Phonebook list.
Restrict Calls
Restrict calls is a feature that can be used to restrict calls between internal extensions besides those in the Allowed List.
This section describes the configuration of this feature in the Call Features->Restrict Calls page.

Configure Restrict Calls
- Click on “Add” to add a rule for restrict calls.
-
Click on
to edit the rule of restrict calls.
-
Click on
to delete the rule of restrict calls.
Table 74: Restrict Calls Configuration Parameters
Name | Configure Restrict call’s name. |
Restrict Calls between extensions | When enabled, members of the group cannot dial other extension, only the numbers in the Allowed List. |
Members | Configure the members that will not be able to call any extensions besides those in the Allowed List. |
Allowed list | Select the extensions that the Members list can be able to call. |
Paging and Intercom Group
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 UCM6200 paging and intercom can be used via feature code to a single extension or a paging/intercom group. This section describes the configuration of paging/intercom group under Web GUI🡪Call Features🡪Paging/Intercom.
Configure Paging/Intercom Group
- Click on “Add” to add paging/intercom group.
-
Click on
to edit the paging/intercom group.
-
Click on
to delete the paging/intercom group.
- Click on “Paging/Intercom Group Settings” to edit Alert-Info Header. This header will be included in the SIP INVITE message sent to the callee in paging/intercom call.
Configure Multicast Paging

Table 75: Multicast Paging Configuration Parameters
Name | Configure paging/intercom group name. |
Type | Select “Multicast Paging”. |
Extension | Configure the paging/intercom group extension. |
Multicast IP Address | The allowed multicast IP address range is 224.0.1.0 – 238.255.255.255. Note: This field appears only when “Type” is set to “Multicast Paging”. |
Maximum Call Duration | Specify the maximum call duration in seconds. The default value 0 means no limit. |
Custom Prompt | This option is to configure a custom to play to paging callees. Users can now directly upload custom prompts from this page. They do not need to be redirected anymore. |
Multicast IP Address | Configure the multicast IP address that endpoints will listen to. |
Port | Configure the port for multicast paging. Note: This field appears only when “Type” is set to “Multicast Paging”. |
Configure 2-way Intercom
Table 76: 2-way Intercom Configuration Parameters
Name | Configure paging/intercom group name. |
Type | Select “2-way Intercom”. |
Extension | Configure the paging/intercom group extension. |
Replace Display Name | If enabled, the UCM will replace the caller display name with Paging/Intercom name. |
Maximum Call Duration | Specify the maximum call duration in seconds. The default value 0 means no limit. |
Custom Prompt | This option is to set a custom prompt for a paging/intercom group to announce to caller. Users can now directly upload custom prompts from this page. They do not need to be redirected anymore. |
Members | Select available users from the left side to the paging/intercom group member list on the right. |
Paging/Intercom Whitelist | Select which extensions are allowed to use the paging/intercom feature for this paging group. |
Configure 1-way Paging
Table 77: 1-way Paging Configuration Parameters
Name | Configure paging/intercom group name. |
Type | Select “1-way Paging”. |
Extension | Configure the paging/intercom group extension. |
Replace Display Name | If enabled, the UCM will replace the caller display name with Paging/Intercom name. |
Maximum Call Duration | Specify the maximum call duration in seconds. The default value 0 means no limit. |
Custom Prompt | This option is to set a custom prompt for a paging/intercom group to announce to caller. Users can now directly upload custom prompts from this page. They do not need to be redirected anymore.. |
Members | Select available users from the left side to the paging/intercom group member list on the right. |
Paging/Intercom Whitelist | Select which extensions are allowed to use the paging/intercom feature for this paging group. |
Configure Announcement Paging

Table 78: Announcement Paging Configuration Parameters
Enable/Disable Announcement Paging. | |
Name | Configure paging/intercom group name. |
Type | Select “Announcement Paging” |
Custom Prompt | This option is to set a custom prompt for a paging/intercom group to announce to caller. Users can now directly upload custom prompts from this page. They do not need to be redirected anymore. |
Repeat | If enabled, the announcement page will be repeated for the selected weekdays. |
Date | |
Include Holidays | If enabled, Announcement Paging will run during holidays. |
Time | Configure Announcement Paging Time. |
Transmission Method | Configure Announcement Paging transmission method. Unicast: Depending on members selection Multicast: Depending on Multicast IP address and Port |
Members | Select available users from the left side to the paging/intercom group member list on the right. |
Configure Private Intercom
Private Intercom is a new paging type that is meant to be used with Grandstream GSC3510.
http://www.grandstream.com/products/facility-management/intercoms-paging/product/gsc3510
In a private intercom:
- The initiator can be heard by all parties
- The initiator can hear only one of the intercom members, which is determined by whose audio is initially detected. Audio from other members cannot be heard until the first responder is done talking.
- Intercom members can hear only the initiator’s audio and not other intercom members
Table 79: Private Intercom Configuration Parameters
Name | Configure paging/intercom group name. |
Type | Select “Private Intercom”. |
Extension | Configure the paging/intercom group extension. |
Replace Display Name | If enabled, the UCM will replace the caller display name with Paging/Intercom name. |
Maximum Call Duration | Specify the maximum call duration in seconds. The default value 0 means no limit. |
Custom Prompt | This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct the users to upload the customized voice prompts. Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom Prompt, where they could record new prompt or upload prompt files. |
Members | Select available users from the left side to the paging/intercom group member list on the right. |
Paging/Intercom Whitelist | Select which extensions are allowed to use the paging/intercom feature for this paging group. |
Paging/Intercom Group Settings

The UCM6200 has pre-configured paging/intercom feature code. By default, the Paging Prefix is *81 and the Intercom Prefix is *80. To edit page/intercom feature code, click on “Feature Codes” in the “Paging/Intercom Group Settings” dialog. Or users could go to Web GUI🡪Call Features🡪Feature Codes directly.
Configure a Scheduled Paging/Intercom
Users can schedule paging/intercom calls by using the Schedule Paging/Intercom page. To schedule, click the Add button on the new page and configure the caller, the group to use, and the time to call out.
Table 80: Schedule Paging / Intercom Settings
Caller | Configure the caller ID for the paging / intercom group. |
Paging/Intercom Group | Select the paging / intercom group from the list of the available groups. |
Start Time | Configure the start time of the scheduled paging / intercom call. |
Type | Select the type for the scheduled paging / intercom call. The available types are: Single time or Daily, Weekly basis. Default is “Single”. |
If enabled Paging/Intercom will run during holidays. | |
Action Status | Display the action status of the scheduled paging / intercom call. |

Call Queue
The UCM provides call center functionality through its Call Queue feature. Static and dynamic agents can be assigned to queue groups to handle varying loads of call traffic. This section describes the configuration of call queues in the Call Features->Call Queue page.
Configure Call Queue
Call queue settings can be accessed via Web GUI🡪Call Features🡪Call Queue.
UCM6200 supports custom prompt feature in call queue. This custom prompt will active after the caller waits for a period of time in the Queue. Then caller could choose to leave a message/ transfer to default extension or keep waiting in the queue.
To configure this feature, please go to UCM Web GUI🡪Call Features🡪Call Queue🡪Create New Queue/Edit Queue🡪Queue Options🡪set Enable Destination to Enter Destination with Voice Prompt. Users could configure the wait time with Voice Prompt Cycle.
- Click on “Create New Queue” to add call queue.
-
Click on
to edit the call queue. The call queue configuration parameters are listed in the table below.
-
Click on
to delete the call queue.
Table 81: Call Queue Configuration Parameters
To guarantee a high level of audio quality with the call queue feature, UCMs will limit the number of static 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 static agents for each UCM model:
Table 82: Static Agent Limitation
UCM Model | Max Static Agents in Call Queue |
UCM6202 | 22 |
UCM6204 | 23 |
UCM6208 | 34 |
UCM6510 | 75 |
Click on “Global Queue Settings” to configure Agent Login Extension Postfix and Agent Logout Extension Postfix. Once configured, users could log in the call queue as dynamic agent.

For example, if the call queue extension is 6500, Agent Login Extension Postfix is * and Agent Logout Extension Postfix is **, users could dial 6500* to login to the call queue as dynamic agent and dial 6500** to logout from the call queue. Dynamic agent does not need to be listed as static agent and can log in/log out at any time.
- Call queue feature code “Agent Pause” and “Agent Unpause” can be configured under Web GUI🡪Call Features🡪Feature Codes. The default feature code is *83 for “Agent Pause” and *84 for “Agent Unpause”.
-
Queue recordings are shown on the Call Queue page under “Queue Recordings” Tab. Click on
to download the recording file in .wav format; click on
to delete the recording file. To delete multiple recording files by one click, select several recording files to be deleted and click on “Delete Selected Recording Files” or click on “Delete All Recording Files” to delete all recording files.
Global Queue Settings | |
Dynamic Agent Login Settings |
|
|
Call Center Settings and Enhancements
UCM supports additional call center features such as virtual call queue and position announcement, allowing callers to know their current position in the queue and providing them the option to either stay in the queue or request for callback once an agent is available.
To configure these, click on the
button for a queue group and click on the Advanced Settings tab. The following options are available:
Table 83: Call Center Parameters
Enable Virtual Queue | Enable virtual queue to activate call center features. |
Virtual Queue Period | Configure the amount of seconds a caller must wait before the virtual queue prompt is played to them. Default is 20 seconds. |
Virtual Queue Mode | Timeout Mode: After the virtual queue period passes, the caller will enter the virtual call queue and be presented with a menu to choose an option, the choices are summarized below :
Note: These options are customizable from the Global Queue Settings page. DTMF Mode: In this mode, the callers can activate the virtual queue by pressing 2, then they will be presented with the menu to choose an option as below:
|
Virtual Queue Outbound Prefix | System will add this prefix to dialed numbers when calling back users. |
Enable Virtual Queue Timeout | If enabled, agents will have a set amount of time to answer a virtual queue callback. |
Write Timeout | Configure the virtual queue callback timeout period in seconds. |
If enabled, callers’ positions in queue will be announced based on the frequency set in Position Announcement Interval. Note: Queue position will now be announced to the caller upon entering the queue. | |
Position Announcement Interval | Configure the frequency of the caller position announcement. |
Enable the announcement of the estimated hold time to the caller periodically. Note: Hold time will not be announced if less than one minute. | |
Select the extension to act as chairman of the queue (monitoring).
| |
Enables quick static agent login/logout via GXP21xx softkeys. Supports GXP21XX endpoint on firmware 1.0.9.18 and higher. Notes:
| |
Autofill | Toggles the way UCM handles and distributes call traffic. Please see the section below for more details. |
In previous UCM firmware, the call queue has a serial type behavior in that the queue will make all waiting callers wait in the queue even if there is more than one available member ready to take calls until the head caller is connected with the member they were trying to get to.
The next waiting caller in line then becomes the head caller, and they are then connected with the next available member and all available members and waiting callers waits while this happens.
Starting from 1.0.14.x, the waiting callers are connecting with available members in a parallel fashion until there are no more available members or no more waiting callers.
For example, in a call queue with linear method, if there are two available agents, when two callers call in the queue at the same time, UCM will assign the two callers to each of the two available agents at the same time, rather than assigning the second caller to second available agent after the first agent answers the call from the first caller.
Queue Statistics
Along with the mentioned call center features, UCM offers detailed call queue statistics, allowing system administrators to make informed decisions regarding call traffic distribution and handling.
Select the time interval along with the queue(s) and agent(s) to get detailed statistics.
User can download statistics on CSV format by clicking on the download button
, also the statistics can be cleared using “Clear Statistics” button
.
The statistics can be automatically sent to a specific email address on a preconfigured Period, this can be done by clicking on “Automatic Download Settings” button
, and user will be directed to below page where he can configure the download period (Day/Week/Month) and the Email where the statistics will be sent (Email settings should be configured correctly):
Significantly more information is now available UCM’s queue statistics page. In addition to the information presented in previous firmware, users can now view a call log that displays calls to all agents and queues, a dynamic agent login/logout record, and a pause log. Statistics reports for these new pages can be obtained by pressing the Download button in the top left corner of the Call Queue Statistics page. The reports are in .CSV format and will be packaged into a single tar.gz file upon download.
Agent Details
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 that was rung for the call.
- Queue – the queue that the call went to.
- Caller ID Number – the CID of the caller
- Abandoned – indicates whether the call was picked up or not by that specific agent. If the call rang several agents simultaneously, and this specific agent did not pick up the call, the call will be considered abandoned even if a different agent in the same queue picked it up.
- Wait Time – the amount of time that the call was waiting in queue after dialing in.
- Talk Time – the duration of the call after it was picked up by agent.
Login Record
Login Record is a report that shows the timestamps of dynamic agent logins and logouts and calculates the amount of time the dynamic agents were logged in. Dynamic agents are extensions that log in and out either via agent login/logout codes (configured in Global Queue Settings page) or by using the GXP21xx call queue softkey. A new record will be created only when an agent logs out. The following information is available:
- Agent – the extension that logged in and out.
- Queue – the queue that the extension logged in and out of.
- Login Time – the time that the extension logged into the queue.
- Logout Time – the time that the extension logged out of the queue.
- Login Duration – the total length of time that the extension was logged in.
Pause Log
Pause Log is a report that shows the times of agent pauses and unpauses and calculates the amount of time that agents are paused. If an agent is part of several queues, an entry will be created for each queue. An entry will only be created after an agent unpauses. The following information is available:
- Agent – the extension that paused and unpaused
- Queue – the queue that the agent is in.
- Pause Time – the time that the agent paused.
- Resume Time – the time that the agent unpaused.
- Pause Duration – the total length of time the agent was paused for.
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.

Table 84: Global Queue Settings
Dynamic Agent Login Settings | |
Agent Login Code Suffix |
Configure the code to dial after the queue extension to log into the queue (i.e. queue extension + suffix).
|
Agent Logout Code Suffix |
Configure the code to dial after the queue extension to log out of the queue (i.e. queue extension + suffix).
|
Call Back Current Number | Press the feature key configured to set your current number as callback number. |
Custom Callback Number | Press the feature key configured to set a custom callback number. |
Continue Waiting | Press the feature key configured to continue waiting. |
Switchboard
Switchboard is a tool that allows system administrators and users to monitor and manage queues. This can be accessed from the Call Features->Call Queue page.
Clicking on one of the queues shown above will bring up the following window:

The table below gives a brief description for the main menus:
Table 85: Switchboard Parameters
There are three different privilege levels for Call Queue management from the switchboard: Super Admin, Queue Chairman, and Queue Agent.
- Super Admin – Default admin of the UCM. Call queue privileges include being able to view and edit all queue agents, monitor and execute actions for incoming and ongoing calls for each extension in Switchboard, and generate Call Queue reports to track performance.
- Queue Chairman – User appointed by Super Admin to monitor and manage an assigned queue extension via Switchboard. The Queue Chairman can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on “Value-added Features” in the side menu and click on “Call Queue”. In the image below, User 1012 is the Queue Chairman appointed to manage Queue Extension 6500 and can see all the agents of the queue in the Switchboard.
Note: Super Admin can assign 3 Chairmen - Queue Agent – User appointed by Super Admin to be a member of a queue extension. A queue agent can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on “Value-added Features” in the side menu and click on “Call Queue”. However, a queue agent can view and manage only his own calls and statistics, but not other agents’ in the queue extension. In the image below, User 1000 is a queue agent and can see only his own information in the Switchboard.
Pickup Groups
The UCM6200 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🡪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🡪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 playlist and upload music files. The “default” Music On Hold playlist already has 5 audio files defined for users to use.
- Click on “Create New MOH Class” to add a new Music On Hold playlist.
-
Click on
to configure the MOH class sort method to be “Name” or “Random” for the sound files.
-
Click on
next to the selected Music On Hold playlist to delete this Music On Hold playlist.
-
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
next to the sound file to disable it from the selected Music On Hold Playlist.
-
Click on
next to the sound file to enable it from the selected Music On Hold Playlist.
-
Select the sound files and click on
to delete all selected music on hold files.
Note: the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire MoH.
The UCM can play selected MOH files to extensions by initiating calls to them from the Manage Music on Hold page
Steps to play the music on hold file:
-
Click on the
button for the Music on Hold file.
-
In the prompted window, select the extension to playback and click
.

- The selected extension will ring.
- 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:
-
Click on
.
- A prompt of confirmation will pop up, as shown below.

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

- Answer the call and start to record your new music on hold.
- Hangup the call and refresh Music On Hold page then you can listen to the new recorded file.
Notes:
Once the MOH file is deleted, there are two ways to recover the music files.
- Users could download the MOH file from this link:
https://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-moh-opsound-wav-2.03.tar.gz
After downloading and unzip the pack, users could then upload the music files to UCM.
- Factory reset could also recover the MOH file on the UCM.
Fax Server
The UCM6200 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 GUI🡪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.

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 UCM6200. Two modes “Only” and “More” are supported.
Under this mode, the UCM6200 allows only single user to send fax at a time.
Under this mode, the UCM6200 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. |
If enabled this this will give users the option to send a special header in SIP fax messages. | |
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:
|
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 Value-Added Features🡪Fax Sending to allow manual resending. | |
Configures the maximum attempts number to resend the fax. |
Receiving Fax
Example Configuration to Receive Fax from PSTN Line
The following instructions describe how to use the UCM6200 to receive fax from PSTN line on the Fax machine connected to the UCM6200 FXS port.
- Connect Fax machine to the UCM6200 FXS port.
- Connect PSTN line to the UCM6200 FXO port.
- Go to Web GUI🡪Extension/Trunk page.
- Create or edit the analog trunk for Fax as below.
Fax Mode: Make sure “Fax Mode” option is set to “None”.
- Go to UCM6200 Web GUI🡪Extension/Trunk🡪Extensions page.
- 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.
- Go to Web GUI🡪Extension/Trunk🡪Inbound Routes page.
- 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 UCM6200.
- Connect PSTN line to the UCM6200 FXO port.
- Go to UCM6200 Web GUI🡪Call Features🡪Fax/T.38 page. Create a new Fax extension.
