User's Manual

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Grandstream Networks, Inc.

UCM6200 Series IP PBX

User Manual

Summary of content (324 pages)

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Grandstream Networks, Inc.
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UCM6200 Series IP PBX User Manual Table of Content GNU GPL INFORMATION ........................................................................ 17 CHANGE LOG ......................................................................................... 18 FIRMWARE VERSION 1.0.0.1 ............................................................................................................ 18 WELCOME ...............................................................................................
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OPERATION LOG ........................................................................................................................ 43 CHANGE PASSWORD ....................................................................................................................... 45 CHANGE BINDNG EMAIL ........................................................................................................... 46 NETWORK SETTINGS .......................................................................................
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CREATE NEW DEVICE.............................................................................................................. 105 MANAGE DEVICES ................................................................................................................... 106 SAMPLE APPLICATION .................................................................................................................... 113 EXTENSIONS.........................................................................................
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CONFERENCE BRIDGE CONFIGURATIONS .......................................................................... 181 JOIN A CONFERENCE CALL .................................................................................................... 183 INVITE OTHER PARTIES TO JOIN CONFERENCE ................................................................. 183 DURING THE CONFERENCE ................................................................................................... 184 RECORD CONFERENCE ..................
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CONFIGURE PICKUP FEATURE CODE.......................................................................................... 219 MUSIC ON HOLD ................................................................................... 221 FAX/T.38 ................................................................................................. 223 CONFIGURE FAX/T.38 .....................................................................................................................
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INTERNAL OPTIONS/GENERAL ..................................................................................................... 263 INTERNAL OPTIONS/JITTER BUFFER ........................................................................................... 265 INTERNAL OPTIONS/RTP SETTINGS ............................................................................................ 266 INTERNAL OPTIONS/PAYLOAD ......................................................................................................
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ALERT LOG ................................................................................................................................ 294 ALERT CONTACT ...................................................................................................................... 295 CDR ................................................................................................................................................... 295 CDR IMPROVEMENT ............................................................
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Table of Tables UCM6200 Series IP PBX User Manual Table 1: Technical Specifications................................................................................................................. 21 Table 2: UCM6202/UCM6204 Equipment Packaging ................................................................................. 25 Table 3: LCD Menu Options ........................................................................................................................
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Table 39: IAX Extension Configuration Parameters->Features ................................................................ 127 Table 40: IAX Extension Configuration Parameters->Specific Time ......................................................... 129 Table 41: FXS Extension Configuration Parameters->Basic Settings ...................................................... 130 Table 42: FXS Extension Configuration Parameters->Media ...................................................................
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Table 80: IAX Settings/General ................................................................................................................. 269 Table 81: IAX Settings/Registration .......................................................................................................... 269 Table 82: IAX Settings/Static Defense ...................................................................................................... 270 Table 83: SIP Settings/General ........................................
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Table of Figures UCM6200 Series IP PBX User Manual Figure 1: UCM6202 Front View................................................................................................................... 25 Figure 2: UCM6202 Back View ................................................................................................................... 26 Figure 3: UCM6204 Front View...................................................................................................................
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Figure 39: Add LDAP Phonebook ............................................................................................................... 68 Figure 40: Edit LDAP Phonebook ............................................................................................................... 68 Figure 41: Import Phonebook...................................................................................................................... 69 Figure 42: Phonebook CSV File Format ...................................
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Figure 80: Zero Config Sample - Device Preview 1 .................................................................................. 115 Figure 81: Zero Config Sample - Device Preview 2 .................................................................................. 116 Figure 82: Zero Config Sample - Device Preview 3 .................................................................................. 117 Figure 83: Create New Device ...........................................................................
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Figure 121: Ring Group Configuration ...................................................................................................... 207 Figure 122: Sync LDAP Server option ...................................................................................................... 208 Figure 123: Manually Sync LDAP Server ................................................................................................. 208 Figure 124: Ring Group Remote Extension .............................................
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Figure 162: Status->PBX Status ............................................................................................................... 283 Figure 163: Trunk Status ........................................................................................................................... 283 Figure 164: Extension Status .................................................................................................................... 284 Figure 165: Queue Status .....................................
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Figure 203: Troubleshooting Analog Trunks ............................................................................................. 318 Figure 204: Service Check ........................................................................................................................ 318 Figure 205: Network Status ....................................................................................................................... 319 Figure 206: SSH Access ...............................................
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GNU GPL INFORMATION UCM6200 firmware contains third-party software licensed under the GNU General Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license. Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu-general-public-license/gnu-gpl-information-download Firmware Version 1.0.0.
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CHANGE LOG This section documents significant changes from previous versions of the UCM6200 user manuals. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here. FIRMWARE VERSION 1.0.0.1  This is the initial version. Firmware Version 1.0.0.
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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.
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Firmware Version 1.0.0.
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PRODUCT OVERVIEW TECHNICAL SPECIFICATIONS Table 1: Technical Specifications Interfaces Analog Telephone FXS Ports PSTN Line FXO Ports Network Interfaces 2 ports (both with lifeline capability in case of power outage)  UCM6202: 2 ports  UCM6204: 4 ports  UCM6208: 8 ports  UCM6202/6204: Dual Gigabit RJ45 ports with integrated PoE Plus (IEEE 802.3at-2009)  UCM6208: Single Gigabit RJ45 port with integrated PoE Plus (IEEE 802.
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Security Media SRTP, TLS, HTTPS, SSH Physical Universal Power Supply Dimensions Environmental Mounting Weight  Output: 12VDC, 1.5A  Input: 100-240VAC, 50-60Hz  UCM6202/6204: 226mm (L) x 155mm (W) x 34.5mm (H)  UCM6208: 440mm (L) x 185mm (W) x 44mm (H)  Operating: 32 - 104 F / 0 - 40 C, 10-90% (non-condensing)  Storage: 14 - 140 F / -10 - 60 C  UCM6202/6204: Wall mount and Desktop  UCM6208: Rack mount and Desktop  UCM6202: Unit weight 0.51kg, Package weight 0.
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Call Features Compliance Call park, call forward, call transfer, DND, ring/hunt group, paging/intercom and etc  FCC: Part 15 (CFR 47) Class B, Part 68  CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, TBR21, RoHS  A-TICK: AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS 60950, AS/ACIF S002 and ITU-T K.
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Firmware Version 1.0.0.
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INSTALLATION Before deploying and configuring the UCM6200 series, the device needs to be properly powered up and connected to network. This section describes detailed information on installation, connection and warranty policy of the UCM6200 series.
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Figure 2: UCM6202 Back View To set up the UCM6202, follow the steps below: 1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6202. 2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub. 3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6202. Insert the main plug of the power adapter into a surge-protected power outlet. 4. Wait for the UCM6202 to boot up.
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Figure 4: UCM6204 Back View To set up the UCM6204, follow the steps below: 1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6204. 2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub. 3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6204. Insert the main plug of the power adapter into a surge-protected power outlet. 4. Wait for the UCM6204 to boot up.
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6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax) to the FXS ports. Figure 5: UCM6208 Front View Figure 6: UCM6208 Back View SAFETY COMPLIANCES The UCM6200 series IP PBX complies with FCC/CE and various safety standards. The UCM6200 power adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM6200 package only.
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Warning: Use the power adapter provided with the UCM6200 series IP PBX. Do not use a different power adapter as this may damage the device. This type of damage is not covered under warranty. -------------------------------------------------------------------------------------------------------------------------------------------- Firmware Version 1.0.0.
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Firmware Version 1.0.0.
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GETTING STARTED The UCM6200 series provides LCD interface, LED indication and web GUI configuration interface.  The LCD displays hardware, software and network information. Users could also navigate in the LCD menu for device information and basic network configuration.  The LED indication at the front of the device provides interface connection and activity status.  The web GUI gives users access to all the configurations and options for UCM6200 series setup.
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Table 3: LCD Menu Options View Events Device Info  Critical Events  Other Events  Hardware: Hardware version number  Software: Software version number  P/N: Part number  WAN MAC: WAN side MAC address (UCM6202/UCM6204 only)  LAN MAC: LAN side MAC address  Uptime: System up time For UCM6208: Network Info  LAN Mode: DHCP, Static IP, or PPPoE  LAN IP: IP address  LAN Subnet Mask For UCM6202/UCM6204:  WAN Mode: DHCP, Static IP, or PPPoE  WAN IP: IP address  WAN Subne
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Select "All On" "All Off" or "Blinking" and check LED status.  RTC Test Patterns Select "2022-02-22 22:22" or "2011-01-11 11:11" to start the RTC (Real-Time Clock) test pattern. Then check the system time from LCD idle screen by pressing "DOWN" button, or from web GUI->System Status->General page. Reboot the device manually after the RTC test is done.  Hardware Testing Select "Test SVIP" to perform SVIP test on the device.
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Table 5: UCM6208 LED INDICATORS LED NETWORK LED Status Solid: Connected OFF: Not Connected ACT USB Solid: Connected SD Flashing: Data Transferring Phone (FXS) OFF: Not Connected Line (FXO) USE THE WEB GUI ACCESS WEB GUI The UCM6200 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow users to configure the device through a Web browser such as Microsoft IE, Mozilla Firefox, Google Chrome and etc. Figure 7: UCM6204 Web GUI Login Page Firmware Version 1.0.0.
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To access the Web GUI: 1. Connect the computer to the same network as the UCM6200. 2. Ensure the device is properly powered up and shows its IP address on the LCD. 3. Open a Web browser on the computer and enter the web GUI URL in the following format: http(s)://IP-Address:Port where the IP-Address is the IP address displayed on the UCM6200 LCD. By default, the protocol is HTTPS and the Port number is 8089. For example, if the LCD shows 192.168.40.
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SETUP WIZARD When the user logs in the UCM6200 web UI for the first time, a setup wizard will guide the user to set up basic configuration. Configurations in setup wizard includes: Time zone, Change password, Network settings, Extensions, Trunk and routes. Figure 8: UCM6200 Setup Wizard During the wizard, the user can quit the setup wizard at any time to start over with manual configuration.
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WEB GUI CONFIGURATIONS There are four main sections in the Web GUI for users to view the PBX status, configure and manage the PBX.  Status: Displays PBX status, System Status, System Events and CDR.  PBX: To configure extensions, trunks, call routes, zero config for auto provisioning, call features, internal options, IAX settings and SIP settings.
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SAVE AND APPLY CHANGES Click on "Save" button after configuring the web GUI options in one page. After saving all the changes, make sure click on "Apply Changes" button on the upper right of the web page to submit all the changes. If the change requires reboot to take effect, a prompted message will pop up for you to reboot the device. MAKE YOUR FIRST CALL Power up the UCM6200 and your SIP end point phone. Connect both devices to the network. Then follow the steps below to make your first call. 1.
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SYSTEM SETTINGS This section explains configurations for system-wide parameters on the UCM6200. System settings are under “Settings” tag on UCM6200 web GUI. System settings include User Management, Network Settings, Firewall, Change Password, LDAP server, HTTP server, Email settings, Time Settings, NTP Server, Recordings Storage and Login Timeout settings. USER MANAGEMENT User management is on web GUI->Settings->User Management page.
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- This is the highest privilege. Super Admin can access all pages on UCM6200 web GUI, change configuration for all options and execute all the operations. - Super Admin can create, edit and delete one or more users with “Admin” privilege - Super Admin can edit and delete one or more users with “Consumer” privilege - Super Admin can view operation logs generated by all users.
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Figure 11: Create New User Table 6: User Management->Create New User User Name User Password Privilege Configure a username to identify the user which will be required in web UI login. Letters, digits and underscore are allowed in the user name. Configure a password for this user which will be required in web UI login. Letters, digits and underscore are allowed. This is the role of the web UI user. Currently only “Admin” is supported when Super Admin creates a new user.
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USER PORTAL The user could log in web UI user portal using the extension number and password. When there is an extension created in the UCM6200, the corresponding user account for the extension is automatically created. The user portal allows limited access including user information, extension configuration and CDR information of the extension. The login username is the extension number and the password is configured by Super Admin.
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Figure 15: User Portal Layout For the configuration parameter information in each page, please refer to [Table 6: User Management->Create New User] for options in User Portal->Basic Information->User Information page; please refer to [EXTENSIONS] for options in User Portal->Basic Information->Extension page; please refer to [CDR] for User Portal->Basic Information->CDR page. CONCURRENT MULTI-USER LOGIN When there are multiple web UI users created, concurrent multi-user login is supported on the UCM6200.
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Figure 17: Operation Logs The operation log can be sorted and filtered for easy access. Click on the header of each column to sort. For example, clicking on "Date" will sort the logs according to operation date and time. Clicking on "Date" again will reverse the order. Table 7: Operation Log Column Header Date The date and time when the operation is executed. User Name The username of the user who performed the operation. IP Address The IP address from which the operation is made.
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Figure 18: Operation Logs Filter The above figure shows an example that operations made by user “support” on device with IP 192.168.40.173 from 2014-11-01 00:00 to 2014-11-06 15:38 are filtered out and displayed. To delete operation logs, users can perform filtering first and then click on delete the filtered result of operation logs. Or users can click on to to delete all operation logs at once.
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Figure 19 : Change Password Enter Old Password Enter the Old Password for UCM6200 Enter New Password Enter the New Password for UCM6200 Retype New Password Email Address Retype the New Password for UCM6200 Configure the Email address for UCM6200. In case login credential is lost, Email address is used to retrieve login credential CHANGE BINDNG EMAIL UCM6200 allows user to configure binding email in case login password is lost. UCM6200 login credential will be sent to the designated email address.
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Table 8: Change Binding Email option Enter the password of the account Email Address Enter the current login user credential for UCM6200 Email Address is used to retrieve password when password is lost NETWORK SETTINGS After successfully connecting the UCM6200 to the network for the first time, users could login the Web GUI and go to Settings->Network Settings to configure the network parameters for the device.
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Preferred DNS Server Enter the preferred DNS server address. If Preferred DNS is used, UCM will try to use it as 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 Gateway IP DNS Server 1 Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.
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User Name Enter the user name to connect via PPPoE. Password Enter the password to connect via PPPoE. Layer 2 QoS Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 802.1Q/VLAN Tag 0. Layer 2 QoS 802.1p Assign the priority value of the layer 2 QoS packets for LAN port. The default Priority Value value is 0.
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DNS Server 1 Enter the DNS server 1 address for static IP settings. DNS Server 2 Enter the DNS server 2 address for static IP settings. User Name Enter the user name to connect via PPPoE. Password Enter the password to connect via PPPoE. Layer 2 QoS Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 802.1Q/VLAN Tag 0. Layer 2 QoS 802.1p Assign the priority value of the layer 2 QoS packets for LAN port. The default Priority Value value is 0.
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 Method: Switch WAN port interface is used for uplink connection; LAN port interface is used as bridge for PC connection. Figure 22: UCM6200 Network Interface Method: Switch  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. Firmware Version 1.0.0.
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Figure 23: UCM6200 Network Interface Method: Dual 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 is allowed to 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 uses 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet. Figure 24: UCM6200 Using 802.
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Figure 25: UCM6200 Using 802.1X EAP-MD5 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 upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated from the RADIUS server. Table 11: UCM6200 Network Settings->802.
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 Click on to create a new static route. The configuration parameters are listed in the table below.  Once added, users can select  Select to edit the static route. to delete the static route. Table 12: UCM6200 Network Settings->Static Routes Configure the destination IP address or the destination IP subnet for the UCM6200 to reach using the static route. Destination Example: IP address - 192.168.66.4 IP subnet - 192.168.66.0 Configure the subnet mask for the above destination address.
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Figure 26: UCM6204 Static Route Sample 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.
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PORT FORWORDING The UCM network interface supports router function which provides users the ability to do port forwarding. If the UCM6202/UCM6204 LAN mode is set to "Route" under web GUI->Settings->Network Settings->Basic Settings page, port forwarding is available for configuration. The port forwarding configuration is under web GUI->Settings->Network Settings->Port Forwarding page. Please see related settings in the table below.
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 There is a GXP2160 connected under the LAN interface network of the UCM6202/UCM6204. It obtains IP address 192.168.2.100 from UCM6200 DHCP pool  On the UCM6202/UCM6204 web UI->Settings->Network Settings->Port Forwarding, configure a port forwarding entry as the figure shows below. WAN Port: This is the port opened up on the WAN side for access purpose. LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM6202/UCM6204.
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Figure 29: GXP2160 Web Access Using UCM6202 Port Forwarding DDNS SETTINGS DDNS setting allows user to access UCM6200 via domain name instead of IP address. The UCM supports DDNS service from the following DDNS provider:  dydns.org  noip.com  freedns.afraid.org  zoneedit.com  oray.net Here is an example of using noip.com for DDNS. 1. Register domain in DDNS service provider. Please note the UCM6200 needs to have public IP access. Firmware Version 1.0.0.
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Figure 30: Register Domain Name on noip.com 2. On web UI->Settings->Network Settings->DDNS Settings, enable DDNS service and configure username, password and host name. Figure 31: UCM6200 DDNS Setting 3. Now you can use domain name instead of IP address to connect to the UCM6200 web UI. Firmware Version 1.0.0.
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Figure 32: Using Domain Name to Connect to UCM6200 FIREWALL The UCM6200 provides users firewall configurations to prevent certain malicious attack to the UCM6200 system. Users could configure to allow, restrict or reject specific traffic through the device for security and bandwidth purpose. The UCM6200 also provides Fail2ban feature for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE. To configure firewall settings in the UCM6200, go to Web UI->Settings->Firewall page.
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Table 14: UCM6200 Firewall->Static Defense->Current Service Port Process Type Protocol or Service 7777 Asterisk tcp/IPv4 SIP 389 Slapd tcp/IPv4 LDAP 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
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Figure 33: Create New Firewall Rule Table 16: Firewall Rule Settings Rule Name Specify the Firewall rule name to identify the firewall rule. Select the action for the Firewall to perform. Action  ACCEPT  REJECT  DROP Select the traffic type.  IN If selected, users will need specify the network interface "LAN" or "WAN" Type (for UCM6202/UCM6204) for the incoming traffic.  OUT Select the service type.
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 Click on to edit the rule  Click on to delete the rule DYNAMIC DEFENSE Dynamic defense is supported on the UCM6200 series. It can blacklist hosts dynamically when the LAN mode is set to "Route" under web GUI->Settings->Network Settings->Basic Settings page. If enabled, the traffic coming into the UCM6200 can be monitored, which helps prevent massive connection attempts or brute force attacks to the device.
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 Since IP address 192.168.40.5 is in whitelist, if the host at IP address 192.168.40.5 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. Figure 34: Configure Dynamic Defense FAIL2BAN Fail2Ban feature on the UCM6200 provides intrusion detection and prevention for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE.
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ban the host with matching address in this list. Up to 5 addresses can be added into the list. Local Settings Enable Asterisk service for Fail2Ban. The default setting is disabled. Please make sure both "Enable Fail2Ban" and "Asterisk Service" are turned on in order to use Asterisk Service Fail2Ban for SIP authentication on the UCM6200. Configure the listening port number for the service. Currently only 5060 (for UDP) Protocol is supported.
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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 other 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.
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Figure 37: Default LDAP Phonebook Attributes 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.
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 Add new phonebook A new sibling phonebook of the default PBX phonebook can be added by clicking on "Add" under "LDAP Phonebook" section. Figure 39: Add LDAP Phonebook 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".
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 Import phonebook from your computer to LDAP server Click on “Import Phonebook” and a dialog will prompt as shown in the figure below. Figure 41: Import Phonebook The file to be imported must be a CSV file with UTF-8 encoding. Users can open the CSV file with Notepad and save it with UTF-8 encoding. Here is how a sample file looks like. Please note “Account Number” and “Phonebook DN” fields are required.
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UCM6200. Figure 43: LDAP Phonebook After Import As the default LDAP phonebook with DN “ou=pbx,dc=pbx,dc=com” cannot be edited or deleted in LDAP phonebook section, users cannot import contacts with Phonebook DN field “pbx” if existed in the CSV file.  Export phonebook to your computer from UCM6200 LDAP server Select the checkbox for the LDAP phonebook and then click on “Export Selected Phonebook” to export the selected phonebook.
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Base DN: dc=pbx,dc=com User Name: cn= “LDAP server login name”, dc=pbx, dc=com [matching LDAP server format] Password: “LDAP server login password” Filter: (|(CallerIDName=%)(AccountNumber=%)) Port: 389 The following figure gives a sample configurations for UCM6200 acting as a LDAP client.
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Figure 46: GXP2200 LDAP Phonebook Configuration HTTP SERVER The UCM6200 embedded web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow the users to configure the PBX through a Web browser such as Microsoft IE, Mozilla Firefox and Google Chrome. By default, the PBX can be accessed via HTTPS using Port 8089 (e.g., https://192.168.40.50:8089). Users could also change the access protocol and port as preferred under Web GUI->Settings->HTTP Server. Firmware Version 1.0.0.
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Table 19: HTTP Server Settings Enable or disable redirect from port 80. On the PBX, the default access Redirect From Port 80 protocol is HTTPS and the default port number is 8089. When this option is enabled, the access using HTTP with Port 80 will be redirected to HTTPS with Port 8089. The default setting is "Enable". Select HTTP or HTTPS. The default setting is "HTTPS". This is also the Protocol Type protocol used for zero config when the end point device downloads the config file from the UCM6200.
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The following figure shows a sample Email settings on the UCM6200, assuming the Email is using smtp.gmail.com as the SMTP server. Figure 47: UCM6200 Email Settings Once the configuration is finished, click on "Test". In the prompt, fill in a valid Email address to send a test Email to verify the Email settings on the UCM6200. TIME SETTINGS AUTO TIME UPDATING The current system time on the UCM6200 is displayed on the upper right of the web page.
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Table 21: Time Auto Updating Specify the URL or IP address of the NTP server for the UCM6200 to Remote NTP Server synchronize the date and time. The default NTP server is ntp.ipvideotalk.com. If set to "Yes", the UCM6200 is allowed to get provisioned for Time Zone Enable DHCP Option 2 from DHCP Option 2 in the local server automatically. The default setting is "Yes".
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SET TIME MANUALLY To manually set the time on the UCM6200, go to Web UI->Settings->Time Settings->Set Time Manually. The format is YYYY-MM-DD HH:MI:SS. Figure 48: Set Time Manually -------------------------------------------------------------------------------------------------------------------------------------------Note: Manually setup time will take effect immediately after saving and applying change in the web UI.
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Figure 49: Create New Office Time Table 22: Create New Office Time Start Time Configure the start time for office hour. End Time Configure the end time for office hour Week Select the work days in one week. Show Advanced Options Check this options to show advanced options. Once selected, please specify "Month" and "Day" below. Month Select the months for office time. Day Select the work days in one month. Select "Start Time", "End Time" and the day for the "Week" for the office time.
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Figure 50: Settings->Time Settings->Office Time  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 On the UCM6200, the system administrator can define "holiday", which can be used to configure time condition for extension call forwarding schedule and inbound rule schedule. To configure holiday, go to Web UI->Settings->Time Settings->Holiday.
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Table 23: Create New Holiday Name Specify the holiday name to identify this holiday. Holiday Memo Create a note for the holiday. Month Select the month for the holiday. Day Select the day for the holiday. Show Advanced Options Check this option to show advanced options. If selected, please specify the days as holiday in one week below. Select the days as holiday in one week. Week Enter holiday "Name" and "Holiday Memo" for the new holiday. Then select "Month" and "Day".
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NTP SERVER The UCM6200 can be used as a NTP server for the NTP clients to synchronize their time with. To configure the UCM6200 as the NTP server, set "Enable NTP server" to "Yes" under web GUI->Settings->NTP Server. On the client side, point the NTP server address to the UCM6200 IP address or host name to use the UCM6200 as the NTP server.
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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. Figure 54: Recordings Storage Prompt Information Click on “OK” to continue. The users will be prompted a new dialog to select the categories for the files to be copied over.
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LOGIN TIMEOUT SETTINGS After the user logs in the UCM6200 web UI, the user will be automatically logged out after certain timeout. This timeout value can be specified under UCM100 web GUI->Settings->Login Timeout Settings page. The “User Login Timeout” value is in minute and the default setting is 10 minutes. If the user doesn’t make any operation on web UI within the timeout, the user will be logged out automatically.
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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-> Settings-> Google Service Settings-> Google Service Settings. Figure 57: Google Service Settings->OAuth2.0 Authentication If you already have OAuth2.
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Figure 58: Google Service->New Project 2. Enable Calendar API from API Library. 3. Click “Credentials” on the left drop down menu to create new OAuth2.0 login credentials. Figure 59: Google Service->Create New Credential 4. Use the newly created login credential to fill in “OAuth2.0 Client ID” and “OAuth2.0 Client Secret”. 5. Click “Get Authentication Code” to obtain authentication code from Google Service. Firmware Version 1.0.0.
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Figure 60: Google Service->OAuth2.0 Login 6. Now UCM6200 is connected with Google Service. Firmware Version 1.0.0.
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Firmware Version 1.0.0.
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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.
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Figure 61: Zero Config Configuration Architecture for End Point Device 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.
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Figure 62: UCM6200 Zero Config  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.
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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->PBX->Zero Config->Zero Config Settings, fill in the auto provision information. Figure 63: Auto Provision Settings Table 24: Auto Provision Settings Enable Zero Config Enable or disable the zero config feature on the PBX.
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Segment 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 UI->PBX->Internal Options->General page->Extension Preference section: "Auto Provision Extensions". If enabled, the extension list will be sent out to the device after receiving Enable Pick Extension the device's request.
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Figure 64: Auto Discover 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) are displayed in the list. Figure 65: Discovered Devices 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.
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 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.
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Table 26: Global Policy Parameters->Phone Settings 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. When enabled, “Do Not Disturb”, “Call Forward” and other call features Enable Call Features can be used via the local feature code on the phone. Otherwise, the ITSP feature code will be used.
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 Frequencies are in Hz and cadence on and off are in 10ms).  “on” is the period (in ms) of ringing while “off” is the period of silence. Up to three cadences are supported.  Please refer to user manual of the SIP devices to be provisioned for more details Select “Default Mode” or “Toggle Headset/Speaker” for the Headset key. HEADSET Key Mode Please refer to user manual of the SIP devices to be provisioned for more details.
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LDAP search result. Example:  telephoneNumber  telephoneNumber Mobile Configure the entry information to be shown on phone's LCD. Up to 3 Display Name fields can be displayed. Example:  Max Hits %cn %sn %telephoneNumber Specify the maximum number of results to be returned by the LDAP server. Valid range is 1 to 3000. The default value is 50. Specify the interval (in seconds) for the server to process the request and Search Timeout client waits for server to return. Valid range is 0 to 180.
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Table 28: Global Policy Parameters->Maintenance Upgrade and Provision Firmware source via ZeroConfig provisioning could a URL for external server address, local UCM directory or USB media if plugged in to the UCM6200. Select a source to get the firmware file:  URL If select to use URL to upgrade, complete the configuration for the following four parameters: “Upgrade Via”, “Server Path”, “File Prefix” and “File Postfix”.
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 By week Once selected, specify the day of the week to check HTTP/TFTP server for firmware upgrades or configuration files changes.  By day Once selected, specify the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes.  By minute Once selected, specify the interval X that the SIP end device will request for new firmware every X minutes. Firmware Upgrade Rule Specify how firmware upgrading and provisioning request to be sent.
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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.
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Configure the location where wallpapers are stored.  File 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. GLOBAL TEMPLATES Global Templates can be accessed in web GUI->PBX->Zero Config->Global Templates. Users can create multiple global templates with different sets of configurations and save the templates.
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Figure 67: Edit 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 option from the template. On the right side of each option, users can click on value to the default value. to remove this to reset the option Click on “Save” to save this global template.  The created global templates will show in the web UI->PBX->Zero Config->Global Templates page.
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 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 PBX->Zero Config->Model Templates.
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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.
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 Click on Save when done. The model template will be displayed on web UI->PBX->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.
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Figure 70: Upload Model Template Manually DEVICE CONFIGURATION On web GUI, page PBX->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.
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Figure 71: Create New Device MANAGE DEVICES The device manually created or discovered from Auto Discover will be listed in the web UI->PBX->Zero Config->Zero Config page. Users can see the devices with their MAC address, IP address, vendor, model and etc. Figure 72: Manage Devices  Click on  Click on to access the web UI of the phone. to edit the device configuration. Firmware Version 1.0.0.
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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. Figure 73: Edit Device A preview of the “Advanced” settings is shown in the above figure.
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(2) 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.
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Figure 74: Edit Customize Device Settings 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 following figure shows setting P value “P1362” to “en”, which means the display language on the LCD is set to English.
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Figure 75: Add P Value in Customize Device Settings  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. Firmware Version 1.0.0.
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Figure 76: Modify Selected Devices - Same Model If on selected devices are of different models, the configuration dialog is like the following figure. Click to view more devices of other models. Users are only allowed to make modifications in Global Templates and Global Policy level. Firmware Version 1.0.0.
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Figure 77: Modify Selected Devices - Different Models -------------------------------------------------------------------------------------------------------------------------------------------Note: Performing batch operation will override all the existing device configuration on the page.
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Figure 78: Device List in Zero Config In this web page, users can also click on “Reset All Extensions” to reset the extensions of all the devices. SAMPLE APPLICATION Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1 GXV3275 phone used by customer support supervisor. 3 of the 8 customer support members speak Spanish and the rest speak English. We could deploy the following configurations to provisioning the office phones for the customer support team. 1.
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Figure 79: Zero Config Sample - Global Policy 3. Go to web GUI->PBX->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. 4. Go to web GUI->PBX->Zero Config->Model Templates, create another model template “Spanish Support Template” for GXP2140. Add option “Language” and set it to “Español”. 5.
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7. For each of the 5 phones used by English speaking customer support, in “Basic” settings select an available extension for account 1 and click on “Save”. Then click on “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. Figure 80: Zero Config Sample - Device Preview 1 8.
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Figure 81: Zero Config Sample - Device Preview 2 4 “Model Template”. The preview of the device settings is Select “Spanish Support Template” in ○ 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. 9. For the GXV3275 used by the customer support supervisor, select an available extension for account 1 on “Basic” settings and click on “Save”.
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Figure 82: Zero Config Sample - Device Preview 3 10. Click on “Apply Changes” to apply saved changes. 11. On the web UI->PBX->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 an 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.
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Firmware Version 1.0.0.
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EXTENSIONS CREATE NEW USER CREATE NEW SIP EXTENSION To manually create new SIP user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on "Create New User"->"Create New SIP Extension" and a new dialog window will show for users to fill in the extension information. Figure 83: Create New Device SIP extension options are divided into four categories:  Basic Settings  Media  Features  Specific Time Firmware Version 1.0.0.
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Click on the tag to view or edit options belonging to that category. The configuration parameters are as follows. Table 33: SIP Extension Configuration Parameters->Basic Settings General Extension The extension number associated with the user. Configure the CallerID Number that would be applied for outbound calls from this user. CallerID Number Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider. Assign permission level to the user.
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characters, letters, digits and _. Email Address Fill in the Email address for the user. Voicemail will be sent to this Email address. Configure the password for user portal access. A random numeric User Password password is automatically generated. It is recommended to use the randomly generated password for security purpose. Select the voice prompt language to be used for this extension.
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Configure the Keep-alive interval (in seconds) to check if the host is up. Keep-alive Frequency The default setting is 60 seconds. Enable T.38 UDPTL Enable or disable T.38 UDPTL support. SRTP Enable SRTP for the call. The default setting is disabled. Select Fax mode. The default setting is “None”.  None: Disable Fax.  Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address Fax Mode configured for this extension.
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time.  Office Time and Holiday could be configured on page Settings->Time Settings->Office Time/Holiday page. Configure the Call Forward No Answer target number. If not configured, Call Forward No Answer the Call Forward No Answer feature is deactivated. The default setting is deactivated. Select time condition for Call Forward No Answer. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.
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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. Configure the maximum number of monitor structures which may be CC Max Monitors 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.
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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 WebRTC. Specify which Music On Hold class to suggest to the bridged channel Music On Hold when putting them on hold. The maximum duration of call-blocking.
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Enable Voicemail Enable voicemail for the user. The default setting is "Yes". Configure voicemail password (digits only) for the user to access the Voicemail Password voicemail box. A random numeric password is automatically generated. It is recommended to use the random generated password for security purpose. Skip Voicemail Password Verification When user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access.
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Enable SRTP for the call. The default setting is disabled. SRTP Select Fax Mode. The default setting is “None”.  None: Disable Fax. This is the default setting.  Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address Fax Mode configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under web UI->PBX->Internal Options->Fax/T.
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Settings->Office Time/Holiday page. Configure the Call Forward No Answer target number. If not configured, Call Forward No Answer the Call Forward No Answer feature is deactivated. The default setting is deactivated. Select time condition for Call Forward No Answer. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.
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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->Internal Options->IVR 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.
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Table 41: 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. Configure the CallerID Number that would be applied for outbound calls from this user. CallerID Number Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider. Assign permission level to the user.
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Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt language under web GUI->PBX->Internal Options->Language. The dropdown list Language 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 UI->PBX->Internal Options->Language.
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For FXS extension, there are three options available in Fax Mode. The default setting is “None”.  None: Disable Fax.  Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the Fax Mode user, the Fax will be sent to the default Email address configured in Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
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time.  Office Time and Holiday could be configured on page Settings->Time Settings->Office Time/Holiday page. Call Forward Busy Configure the Call Forward Busy target number. If not configured, the Call Forward Busy feature is deactivated. The default setting is deactivated. Select time condition for Call Forward Busy. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.
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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->Internal Options->IVR Prompt: General Preference. The Ring Timeout valid range is between 5 seconds and 600 seconds.
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Table 45: Batch Add SIP Extension Parameters General Start Extension Create Number Configure the starting extension number of the batch of extensions to be added. Specify the number of extensions to be added. The default setting is 5. Assign permission level to the user. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to Permission the highest level. The default setting is "Internal".
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option is disabled. Music On Hold Select which Music On Hold class to suggest to extensions when putting them on hold. If enabled, the batch added extensions will be added to LDAP Phonebook Enable LDAP PBX list; if disabled, the batch added extensions will be skipped when creating LDAP Phonebook. Enable WebRTC Support Call Duration Limit If enabled, extensions will be able to login to user portal and use Web RTC features. Configure the maximum duration of call-blocking.
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 None: Disable Fax.  Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under web UI->PBX->Internal Options->Fax/T.38. This option controls how the extension can be used on devices within different types of network.
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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. Enable Voicemail Enable Voicemail for the user. The default setting is “Yes”. Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions.  User Random Password. A random secure password will be automatically generated. It is SIP/IAX Password recommended to use this password for security purpose.
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address. Configure to enable/disable requiring call token. If set to "Auto", it might Require Call Token lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints. The default setting is "Yes". Other Settings Enable SRTP for the call. The default setting is "No". SRTP Select Fax Mode for this user. The default setting is “None”.  None: Disable Fax.
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Figure 84: Manage Extensions  Status Users can see the following icon for each extension to indicate the SIP status.  Green: Free 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. Reboot the user Click on to send NOTIFY reboot event to the device which has an UCM6200 extension already registered.
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EXPORT EXTENSIONS The extensions configured on the UCM6200 can be exported to csv format file with selected technology "SIP", "IAX" or "FXS". Click on "Export Extensions" button and select technology in the prompt below. Figure 85: Export Extensions The exported csv file can be serve as a template for users to fill in desired extension information to be imported to the UCM6200.
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information as previously configured without change.  Delete and Recreate: The current extension previously configured will be deleted and the duplicate extension in the csv file will be loaded to the PBX.  Update Information: The current extension previously configured in the PBX will be kept. However, if the duplicate extension in the csv file has different configuration for any options, it will override the configuration for those options in the extension. 5.
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Figure 88: Account Registration Information and QR Code Figure 89: LDAP Client Information and QR Code Firmware Version 1.0.0.
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MULTIPLE REGISTRATIONS PER EXTENSION UCM6200 supports multiple registrations per extension so that users can use the same extension on devices in different locations. Figure 90: Multiple Registrations per Extension This feature can be enabled by configuring option “Concurrent Registrations” under web UI->PBX->Basic/Call Routes->Edit Extension. The default value is set to 1 for security purpose. Figure 91: Extension - Concurrent Registration Firmware Version 1.0.0.
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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. Figure 92: SMS Message Support Firmware Version 1.0.0.
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Firmware Version 1.0.0.
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TRUNKS ANALOG TRUNKS Go to Web GUI->PBX->Basic/Call Routes->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 47: Analog Trunk Configuration Parameters Select the channel for the analog trunk.
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default setting is “No”. When FXO port answers the call, FXS may send a Polarity Reversal. If Polarity on Answer Delay 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.
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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's idle). The default setting is “Ascend” mode. Tone Settings Busy Detection Busy Detection is used to detect far end hangup or for detecting busy signal. The default setting is "Yes".
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Click on "Detect" to detect the busy tone, Polarity Reversal and Current PSTN Detection Disconnect by PSTN. Before the detecting, please make sure there are 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.
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Figure 94: UCM6200 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.
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 If there is only one FXO port connected to PSTN line, use the following settings for auto-detection. Figure 96: UCM6200 PSTN Detection: Semi-Auto Detect Detect Model: Semi-auto Detect. Source Channel: The source channel to be detected. Destination Number: The number to be dialed for detecting. This number could be a cell phone number or other PSTN number that can be reached from the source channel PSTN number. 5. Click "Detect" to start detecting.
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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.
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For VoIP trunk example, please refer to the document in the following link: http://www.grandstream.com/sites/default/files/Resources/ucm_to_ucm_peer_guide.pdf The VoIP trunk options are listed in the table below. Table 49: Create New SIP Trunk Select the VoIP trunk type. Type Provider Name Host Name  Peer SIP Trunk  Register SIP Trunk Configure a unique label to identify this trunk when listed in outbound rules, inbound rules and etc.
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 If no CallerID is configured for the extension, the CallerID configured for the trunk will be used.  If the above two are missing, the "Global Outbound CID" defined in Web GUI->PBX->Internal Options->General will be used. Need Registration Username Password Auth ID 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.
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with devices behind NAT. If there is one-way audio issue, usually it’s related to NAT configuration or SIP/RTP port configuration on the firewall. If selected, the trunk will be disabled. Disable This Trunk Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider. If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone".
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“Send PPI Header” and “Send PAI Header” cannot be enabled at the same time. Only one of the two headers is allowed to be contained in the SIP INVITE message. If enabled, the SIP INVITE message sent to the trunk will contain PAI (P-Asserted-Identity) header. The default setting is “No”. Send PAI Header Note: “Send PPI Header” and “Send PAI Header” cannot be enabled at the same time. Only one of the two headers is allowed to be contained in the SIP INVITE message.
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configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under web UI->PBX->Internal Options->Fax/T.38. 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.
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Turn on this option when the PBX is using public IP and communicating NAT awith devices behind NAT. If there is one-way audio issue, usually it’s related to NAT configuration or SIP/RTP port configuration on the firewall. If selected, the trunk will be disabled. Disable This Trunk Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider. If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone".
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 Info: Send DTMF using SIP INFO message.  Inband: Send DTMF using inband audio. This requires 64 bit codec, i.e., PCMU and PCMA.  Auto: Send DTMF using RFC2833 if offered. Otherwise, inband will be used. Enable Qualify 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". When "Enable Qualify" option is set to "Yes", configure the timeout (in ms) Qualify Timeout for the Qualify SIP message.
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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. Configure the maximum number of CCSS agents which may be allocated CC Max Agents 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. Configure the maximum number of monitor structures which may be CC Max Monitors created for this device.
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to set the CallerID with this option and this option will be ignored. When making outgoing calls, the following rules are used to determine which CallerID will be used if they exist:  The CallerID configured for the extension will be looked up first.  If no CallerID configured for the extension, the CallerID configured for the trunk will be used.  If the above two are missing, the "Global Outbound CID" defined in Web GUI->PBX->Internal Options->General will be used.
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Host Name Keep Trunk CID Disable This Trunk Configure the IP address or URL for the VoIP provider’s server of the trunk. If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No". If selected, the trunk will be disabled. Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls.
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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.
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6. 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. Figure 98: Edit DOD Firmware Version 1.0.0.
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Firmware Version 1.0.0.
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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 similar to BLF but SLA is used to monitor external line i.e., analog trunk on the UCM6200.
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Configure the time (in seconds) to ring the station before the call is Ring Timeout considered unanswered. No timeout is set by default. If set to 0, there will be no timeout. Configure the time (in seconds) for delay before ringing the station when Ring Delay a call first coming in on the shared line. No delay is set by default. If set to 0, there will be no delay. This option defines the competence of the hold action for one particular Hold Access trunk.
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Figure 100: Enable SLA Mode for Analog Trunk Click on “Save”. The analog trunk will be listed with trunk mode “SLA”. Figure 101: Analog Trunk with SLA Mode Enabled 2. On the UCM6200, go to web UI->Basic/Call Routes->SLA Station page, click on “Create New SLA Station”. 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.
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and the value must be set to “extension_trunkname”, which is 1002_fxo1 in this case. 4. 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. Figure 103: SLA Example - MPK Configuration Now the SLA station is ready to use. The following functions can be achieved by this configuration.
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CALL ROUTES 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 a 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->PBX->Basic/Call Routes->Outbound Routes to add and edit outbound rules.
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Maximum Call Duration Configure the maximum duration of the call (in seconds). The default setting is 0, which means no limit. Configure the warning time for the call using this outbound route. If set to Warning Time x seconds, the warning tone will be played to the caller when x seconds are left to end the call. Configure the warning repeat interval for the call using this outbound Warning Repeat Interval route.
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Example: [12345-9] - Any digit from 1 to 9. Send This Call Through Trunk Select the trunk for this outbound rule. Use Trunk 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: Strip 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.
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 Click on to delete the inbound route. INBOUND RULE CONFIGURATIONS Table 57: Inbound Rule Configuration Parameters Trunks Select the trunk to configure the inbound rule.  All patterns are prefixed with the "_".  Special characters: X: Any Digit from 0-9. Z: Any Digit from 1-9. N: Any Digit from 2-9. ".": Wildcard. Match one or more characters. "!": Wildcard. Match zero or more characters immediately. DID Pattern Example: [12345-9] - Any digit from 1 to 9.
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 External Number  By DID 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 Prepend Configure the number of digits to be stripped from the beginning of the DID.
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Figure 104: Inbound Route feature: Prepend The following example demonstrates the process, 1. If Trunk provides a DID pattern of 18005251163. 2. If Strip is set to 8, UCM6200 will strip the first 8 digits. 3. If Prepend is set to 2, UCM6200 will then prepend a 2 to the stripped number, now the number become 2163. 4. UCM6200 will now forward the incoming call to extension 2163.
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Figure 105: Inbound Route - Multiple Mode When Multiple Mode is enabled for the inbound route, the user can configure a “Default Destination” and a “Mode 1” destination for this route. By default, the call coming into this inbound route 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”.
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FAX WITH TWO MEDIA The UCM6200 supports Fax re-invite with multiple codec negotiation. If a Fax re-invite contains both T.38 and PCMA/PCMU codec, UCM6200 will choose T.38 codec over PCMA/PCMU. 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.
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Figure 107: Blacklist csv File ----------------------------------------------------------------------------------------- --------------------------------------------------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->PBX->Internal Options->Feature Codes. Firmware Version 1.0.0.
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Firmware Version 1.0.0.
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CONFERENCE BRIDGE The UCM6200 supports conference bridge allowing multiple bridges used at the same time:  UCM6202/6204 supports up to 3 conference bridges allowing up to 25 simultaneous PSTN or IP participants.  The UCM6208 supports up to 6 conference bridges allowing up to 32 simultaneous PSTN or IP participants. conference bridge configurations can be accessed under Web GUI->PBX->Call Features->Conference.
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Note:  If "Public Mode" is enabled, the password is not required to join the conference bridge thus this field is invalid.  Enable Caller Menu The password has to be at least 4 characters. If enabled, conference participant could press the * key to access the conference bridge menu. The default setting is "No". If enabled, the calls in this conference bridge will be recorded Record Conference automatically in a .wav format file.
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Select the music on hold class to be played in conference call. Music On Music On Hold Hold class can be set up under web UI->PBX->Internal Options->Music On Hold. Skip Authentication When If enabled, the invitation from Web GUI for a conference bridge with Inviting User via Trunk from password will skip the authentication for the invited users. The default Web GUI setting is "No". JOIN A CONFERENCE CALL Users could dial the conference bridge extension to join the conference.
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A conference participant can invite other parties to the conference by dialing from the phone during the conference call. Please make sure option "Enable User Invite" is turned on for the conference bridge 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.
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Table 59: Conference Caller IVR Menu Conference Administrator IVR Menu 1 Mute/unmute yourself. 2 Lock/unlock the conference bridge. 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. More options. 8  1: List all users currently in the conference call.  2: Kick all non-Administrator participants from the conference call.
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To record the conference call, when the conference bridge is in idle, enable "Record Conference" from the conference bridge 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 recording or click on to delete the recording.
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CONFERENCE SCHEDULE CONFERENCE SCHEUDLE CONFIGURATION Conference Schedule can be found under UCM6200 web UI->PBX->Call Features->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.
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Select the administrator of scheduled conference from selected extensions. Conference Administrator Note: “Public Mode” must be disabled from Conference Room Options tab. Local Extension Select available extensions from the list to attend scheduled conference. Select available extensions from the remote peer PBX. Remote Extension Note: “LDAP Sync” must be enabled on the UCM6200 in order to view remote extensions here. Add extensions that are not in the list (both local and remote list).
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 Press ‘3’ to drop all current multi-conference bridges Note: Conference Administrator is always allowed to access this menu. If this option is enabled, when a participant joins the conference room, participant’s name will be announced to all members in the conference Announce Callers room. Note: Option “Quiet Mode” and option “Announce Caller” cannot be enabled at the same time. If this option is enabled, no authentication is required for entering the conference room.
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Figure 110: Conference Schedule 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, UCM6200 will send INVITE to the extensions that have been selected for conference.
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IVR CONFIGURE IVR IVR configurations can be accessed under the UCM6200 Web GUI->PBX->Call Features->IVR. Users could create, edit, view and delete an IVR.  Click on "Create New IVR" to add a new IVR.  Click on to edit the IVR configuration.  Click on to delete the IVR. Table 61: 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.
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allowed to go through. Select an audio file to play as the welcome prompt for the IVR. Click on Welcome Prompt "Prompt" to add additional audio file under web GUI->Internal Options->IVR Prompt. Configure the timeout between digit entries. After the user enters a digit, Digit Timeout 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.
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Press 9  Fax Press *  Custom Prompt Timeout  Hangup Invalid  DISA  Dial By Name  External Number  Callback 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->Internal Options->Custom Prompt page directly.
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Figure 112: Record New Custom 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.
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LANGUAGE SETTINGS FOR VOICE PROMPT The UCM6200 supports multiple languages in web GUI as well as system voice prompt. Currently, there are 16 languages supported in system voice prompt: English (United States), Arabic, Chinese, Dutch, English (United Kingdom), French, German, Greek, Hebrew, Italian, Polish, Portuguese, Russian, Spanish, Swedish and Turkish. English (United States) and Chinese voice prompts are built in with the UCM6200 already.
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Figure 115: Voice Prompt Package List Click on to download the language to the UCM6200. The installation will be automatically started once the downloading is finished. Figure 116: New Voice Prompt Language Added 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. Firmware Version 1.0.0.
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CUSTOMIZE SPECIFIC PROMPT On the UCM6200, if the user needs to replace some specific customized prompt, the user can upload a single specific customized prompt from web UI->PBX->Internal Options->Language instead of the entire language pack. Figure 117: Upload Single Voice Prompt for Entire Language Pack Firmware Version 1.0.0.
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Firmware Version 1.0.0.
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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->PBX->Call Features->Voicemail. Table 62: Voicemail Settings Max Greeting Configure the maximum number of seconds for the voicemail greeting. The default setting is 60 seconds. If enabled, the caller can press 0 to exit the voicemail application and Dial ‘0’ For Operator connect to the configured operator’s extension.
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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". If enabled, a brief introduction (received time, received from, and etc) of Play Envelope each message will be played when accessed from the voicemail application. The default setting is "Yes".
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1 - Send a reply 3- 2 - Call the person who sent this message Advanced 3 - Hear the message envelop options 4 - Leave a message * - Return to the main menu 1 - Accept this recording 1 - Record your unavailable message 2 - Listen to it 3 - Re-record your message 1 - Accept this recording 2 - Record your busy message 2 - Listen to it 3 - Re-record your message 1 - Accept this recording 0 - Mailbox options 3 - Record your name 2 - Listen to it 3 - Re-record your message 1 - Accept this recording
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 ${VM_DUR}: The duration of the voicemail message  ${VM_MAILBOX}: The recipient's extension  ${VM_CALLERID}: The caller ID of the person who has left the message  ${VM_MSGNUM}: The number of messages in the mailbox  ${VM_DATE}: The date and time when the message is left Figure 118: Voicemail Email Settings Click on "Load Default Settings" button to view the default template as an example.
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Figure 119: Voicemail Group Table 65: Voicemail Group Settings Extension Name Voicemail Password Email Address Enter the Voicemail Group Extension. The voicemail messages left to this extension will be forwarded to all the voicemail group members. Configure the Name to identify the voicemail group. Letters, digits, _ and are allowed. Configure the voicemail password for the users to check voicemail messages. Configure the Email address for the voicemail group extension.
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Firmware Version 1.0.0.
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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->PBX->Call Features->Ring Group. Figure 120: Ring Group  Click on “Create New Ring Group” 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.
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This option is to set a custom prompt for a ring group to announce to Custom Prompt caller. Click on ‘Prompt’, it will direct to the page PBX->Internal Options->Custom Prompt, where users could record new prompt or upload prompt files. Configure the number of seconds to ring each member. If set to 0, it will keep ringing. The default setting is 30 seconds. Ring Timeout on Each Member Note: The actual ring timeout might be overridden by users if the phone has ring timeout settings as well.
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Figure 121: Ring Group Configuration REMOTE EXTENSION IN RING GROUP Remote extensions from the peer trunk of a remote UCM6200 can be included in the ring group with local extension. An example of Ring Group with peer extensions is presented in the following: 1. Creating SIP Peer Trunk between both UCM6200_A and UCM6200_B. SIP Trunk can be found under web UI-> PBX-> Basic/Call Routes-> VoIP Trunks. Also, please configure their Inbound/Outbound routes accordingly. 2.
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Figure 122: Sync LDAP Server option 3. In case if LDAP server doesn’t sync automatically, user can manually sync LDAP server. Under VoIP Trunks page, click sync button shown in the following figure to manually sync LDAP contacts from peer UCM6200. Figure 123: Manually Sync LDAP Server Firmware Version 1.0.0.
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4. Under Ring Groups setting page, click . Ring Groups can be found under web UI-> PBX-> Call Features-> Ring Groups. 5. If LDAP server is synced correctly, Available LDAP Numbers box will display available remote extensions that can be included in the current ring group. Please also make sure the extensions in the peer UCM6200 can be included into that UCM6200’s LDAP contact. Figure 124: Ring Group Remote Extension Firmware Version 1.0.0.
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Firmware Version 1.0.0.
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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 sections describes the configuration of paging/intercom group under Web GUI->PBX->Call Features->Paging/Intercom.
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 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. Figure 126: Page/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.
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CALL QUEUE The UCM6200 supports call queue by using static agents or dynamic agents. Call Queue system can accept more calls than the available agents. Incoming calls will be held until next representative is available in the system. This section describes the configuration of call queue under Web GUI->PBX->Call Features->Call Queue. CONFIGURE CALL QUEUE Call queue settings can be accessed via Web GUI->PBX->Call Features->Call Queue.
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 Least Recent Ring the agent who has been called the least recently.  Fewest Calls Ring the agent with the fewest completed calls.  Random Ring a random agent.  Round Robin Ring the agents in Round Robin scheduling with memory. The default setting is "Ring All". Select the Music On Hold class for the call queue. Music On Hold Note: Music On Hold classes can be managed from Web GUI-> PBX->Internal Options->Music On Hold.
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after the last call on the agent is completed. If set to 0, there will be no delay between calls to the queue. The default setting is 15 seconds. Configure the maximum number of calls to be queued at once. This number does not include calls that have been connected with agents. It Max Queue Length only includes calls not connected yet. The default setting is 0, which means unlimited.
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Figure 128: Agent Login Settings For example, if the call queue extension is 6500, Agent Login Extension Postfix is * and Agent Logout Extension Postfix is **, users could dial 6500* to login to the call queue as dynamic agent and dial 6500** to logout from the call queue. Dynamic agent doesn't 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->PBX->Internal Options->Feature Codes.
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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.
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USING EXTENSION GROUPS Here is an example where the extension group can be used. Go to Web GUI->PBX->Basic/Call Routes->Outbound Routes and select "Enable Filter on Source Caller ID". Both single extensions and extension groups will show up for users to select. Figure 130: Select Extension Group in Outbound Route Firmware Version 1.0.0.
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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->PBX->Call Features->Pickup Groups.  Click on "Create New Pickup Group" to create a new pickup group.  Click on to edit the pickup group. Select extensions from the list on the left side to the right side.
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Figure 132: Edit Pickup Feature Code Firmware Version 1.0.0.
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MUSIC ON HOLD Music On Hold settings can be accessed via Web GUI->PBX->Internal Options->Music On Hold. In this page, users could configure music on hold class and upload music files. The "default" Music On Hold class already has 5 audio files defined for users to use. Figure 133: Music On Hold Default Class  Click on "Create New MOH Class" to add a new Music On Hold class.  Click on to configure the MOH class sort method to be "Alpha" or "Random" for the sound files.
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 Click on to select music file from local PC and click on to start uploading. The music file uploaded has to be 8 KHz Mono format with size smaller than 5M.  Click on  Select the sound files and click on next to the sound file to delete it from the selected Music On Hold Class. to delete all selected music on hold files.
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FAX/T.38 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->PBX->Internal Options->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.
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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 be not be received from Email. Fill in the "Subject:" and "Message:" content, to be used in the Email when sending the Fax to the users.
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Figure 134: Configure Analog Trunk without Fax Detection 5. Go to UCM6200 web GUI->PBX->Basic/Call Routes->Extensions page. 6. Create or edit the extension for FXS port.  Analog Station: Select FXS port to be assigned to the extension. By default, it's set to "None".  Once selected, analog related settings for this extension will show up in "Analog Settings" section. Figure 135: Configure Extension for Fax Machine: FXS Extension Firmware Version 1.0.0.
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Figure 136: Configure Extension for Fax Machine: Analog Settings 7. Go to web GUI->PBX->Basic/Call Routes->Inbound Routes page. 8. Create an inbound route to use the Fax analog trunk. Select the created extension for Fax machine in step 4 as the default destination. Figure 137: Configure Inbound Rule for Fax Firmware Version 1.0.0.
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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. SAMPLE CONFIGURATION FOR FAX-TO-EMAIL The following instructions describe a sample configuration on how to use Fax-to-Email feature on the UCM6200. 1. Connect PSTN line to the UCM6200 FXO port. 2. Go to UCM6200 web GUI->Internal Options->Fax/T.38 page. Create a new Fax extension. Figure 138: Create Fax Extension 3.
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Figure 139: Inbound Route to Fax Extension 5. Once successfully configured, the incoming Fax from external Fax machine to the PSTN line number will be converted to PDF file and sent to the Email address Faxtest@ucm6200mycompany.com as attachment. Firmware Version 1.0.0.
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ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS) The UCM6200 supports Asterisk Manager Interface (AMI) with restricted access. AMI allows a client program to connect to an Asterisk instance commands or read events over a TCP/IP stream. It’s particularly useful when the system admin tries to track the state of a telephony client inside Asterisk. User could configure AMI parameters on UCM6200 web GUI->PBX->Internal Options->AMI.
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Firmware Version 1.0.0.
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BUSY CAMP-ON The UCM6200 supports busy camp-on/call completion feature that allows the PBX to camp on a called party and inform the caller as soon as the called party becomes available given the previous attempted call has failed. The configuration and instructions on how to use busy camp-on/call completion feature can be found in the following guide: http://www.grandstream.com/sites/default/files/Resources/ucm6200_busy_camp_on_guide.pdf Firmware Version 1.0.0.
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Firmware Version 1.0.0.
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FOLLOW ME Follow Me is a feature on the UCM6200 that allows users to direct calls to other phone numbers and have them ring all at once or one after the other. Calls can be directed to users’ home phone, office phone, mobile and etc. The calls will get to the user no matter where they are. Follow Me option can be found under web GUI-> PBX-> Call Features->Follow Me. To configure follow me:  Click on "Create New Follow Me" and then select an extension to be configured with Follow Me.
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 Click on “Add Follow Me Number” to add local extensions or external numbers to be called after ringing the extension selected in the first step.  Once created, it will be displayed on the follow me web page list. Click on configuration. Click on to edit the Follow Me to delete the Follow Me. The following table shows the Follow Me configuration parameters. Table 70: Follow Me Settings Enable Configure to enable or disable Follow Me for this user.
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 Click on “Follow Me Options” to enable or disable the options listed in the following table. Table 71: Follow Me Options Playback Incoming Status If enabled, the PBX will playback the incoming status message before Message starting the Follow Me steps. Record the Caller’s Name If enabled, the PBX will record the caller’s name from the phone so it can be announced to the callee in each step.
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Firmware Version 1.0.0.
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ONE-KEY DIAL The UCM6200 supports One-Key Dial that allows users to call a certain destination by pressing one digit 0 to 9 on the keypad. This creates a system-wide speed dial access for all the extensions on the UCM6200. To enable One-Key Dial, on the UCM6200 web GUI, go to page PBX->Call Features->One-Key Dial. Figure 142: Configure One-Key Dial User should first decide a digit used for One-Key Dial and check the option “Enable Destination” for the digit.
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Figure 143: One-Key Dial Destinations Firmware Version 1.0.0.
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DISA In many situations the user will find the need to access his own IP PBX resources but he is not physically near one of his extensions. However, he does have access to his own cell phone. In this case we can use what is commonly known as DISA (Direct Inward System Access). Under this scenario the user will be able to call from the outside, whether it’s using his cell phone, pay phone, regular PSTN, etc.
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Configure the permission level for DISA. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal". If the user tries to dial Permission outbound calls after dialing into the DISA, the UCM6200 will compared the DISA's permission level with the outbound route's privilege level.
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CALLBACK FEATURE Callback is mainly designed for users who often use their mobile phones to make long distance or international calls which may have high service charges. The callback feature provides an economic solution for reduce the cost from this. The callback feature works as follows: 1. Configure a new callback on the UCM6200. 2. On the UCM6200, configure destination of the inbound route for analog trunk to callback. 3. Save and apply the settings. 4.
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Delay Before Callback Configure the number of seconds to be delayed before calling back the user. Configure the destination which the callback will direct the caller to. Two destinations are available: Destination  IVR  DISA The caller can then enter the desired number to dial out via UCM6200 trunk. Firmware Version 1.0.0.
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BLF AND EVENT LIST BLF The UCM6200 supports BLF monitoring for extensions, ring group, call queue, conference room and parking lot. For example, on the user's phone, configure the parking lot number 701 as the BLF monitored number. When there is a parked call on 701, the LED for this BLF key will light up in red, meaning a call is parked against this parking lot. Pressing this BLF key can pick up the call from this parking lot.
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manually enter the remote extensions under "Special Extensions" field. Manually enter the remote extensions in the peer/register trunk to be Special Extensions monitored in the event list. Valid format: 5000,5001,9000 Figure 145: Create New Event List Remote extension monitoring works on the UCM6200 via event list BLF, among Peer SIP trunks or Register SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the UCM6200 first before using remote BLF feature.
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----------------------------------------------------------------------------------------------------------------------------- --------------Note:  To configure LDAP sync, please go to UCM6200 web GUI->PBX->Basic/Call Routes->VoIP Trunk. You will see "Sync LDAP Enable" option. Once enabled, please configure password information for the remote peer UCM6200 to connect to the local UCM6200. Additional information such as port number, LDAP outbound rule, LDAP Dialed Prefix will also be required.
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Firmware Version 1.0.0.
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DIAL BY NAME Dial By Name is a feature on the PBX that allows caller to search a person by first or last name via his/her phone's keypad. The administrator can define the Dial By Name directory including the desired extensions in the directory and the searching type by "first name" or "last name". After dialing in, the PBX IVR/Auto Attendant will guide the caller to spell the digits to find the person in the Dial By Name directory.
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1. Group Name Enter the Group Name. This is to identify the Dial By Name group. The Dial By Name group can be used as the destination for inbound route and key pressing event for IVR. The group name defined here will show up in the destination list when configuring IVR and inbound route. If Dial By Name is set as a key pressing event for IVR, user could use ‘*’ to exit from Dial By Name, then re-enter IVR and start a new event. The following example shows how to use this option.
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Figure 148: Dial By Name Group In Inbound Rule 2. Extension Configure the direct dial extension for the Dial By Name group. 3. Available Extensions/Selected Extensions Select available extensions from the left side to the right side as the directory for the Dial By Name group. Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group.
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Figure 149: Configure Extension First Name and Last Name 4. Query Type Specify the query type. This defines how the caller will need to enter to search the directory. By First Name: enter the first 3 digits of the first name to search the directory. By Last Name: enter the first 3 digits of the last name to search the directory. By Full Name: enter the first 3 digits of the first name or last name to search the directory. 5. Select Type Specify the select type on the searching result.
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ACTIVE CALLS AND MONITOR The active calls on the UCM6200 are displayed in web UI->Status->Active Calls page. Users can monitor the status, hang up the call as well as barge in the active calls in real time manner. ACTIVE CALLS STATUS To view the status of active calls, navigate to web GUI->Status->Active Calls. The following figure shows extension 1000 is calling 1001. 1001 is ringing.
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In active call web page, click on to refresh the active call status. HANG UP ACTIVE CALLS To hang up an active call, click on icon in the active call dialog. Users can also click on to hang up all active calls. CALL MONITOR During an active call, click on icon and the monitor dialog will pop up. Figure 152: Configure to Monitor an Active Call In the “Monitor” dialog, configure the following to monitor an active call: 1.
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4. Enable or disable “Require Confirmation” option. If enabled, the confirmation of the invited monitor’s extension is required before the active call can be monitored. This option can be used to avoid adding participant who has auto-answer configured or call forwarded to voicemail. 5. Click on “Add”. An INVITE will be sent to the monitor’s extension. The monitor can answer the call and start monitoring. If “Require Confirmation” is enabled, the user will be asked to confirm to monitor the call.
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Firmware Version 1.0.0.
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CALL FEATURES The UCM6200 supports call recording, transfer, call forward, call park and other call features via feature code. This section lists all the feature codes in the UCM6200 and describes how to use the call features. FEATURE CODES Table 75: UCM6200 Feature Codes Feature Maps  Default code: #1.  Enter the code during active call. After hearing "Transfer", you will hear dial tone. Enter the number to transfer to. Then the user will be disconnected and transfer is completed.
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 Default code: #72.  Enter the code during active call to park the call.  Options Call Park Disable Allow Caller: Enable the feature code on caller side only. Allow Callee: Enable the feature code on callee side only. Allow Both: Enable the feature code on both caller and callee.  Default code: *3.  Enter the code followed by # or SEND to start recording the audio call and the UCM6200 will mix the streams natively on the fly as the call is in progress.
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Feature Code Digits Timeout  Default Setting: 1000.  Configure the maximum interval (in milliseconds) between the digits input to activate the feature code. Call Park  Default Extension: 700.  During an active call, initiate blind transfer and then enter this code to park the call. Parked Lots  Default Extension: 701-720.  These are the extensions where the calls will be parked, i.e., parking lots that the parked calls can be retrieved.
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Blacklist Add  Default Code: *40.  To add a number to blacklist for inbound route, dial *40 and follow the voice prompt to enter the number. Blacklist Remove  Default Code: *41.  To remove a number from current blacklist for inbound route, dial *41 and follow the voice prompt to remove the number. Call Pickup on Ringing  Default Code: **.  To pick up a call for any extension xxxx, enter the code followed by the extension number xxxx. Pickup Extension  Default Code: *8.
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This is the feature code to join in on the call to assist both parties. Barge Spy The default setting is *56. If enabled, user can switch between different inbound route modes Enable Inbound Multiple Mode with feature code. By default, this option is disabled. This feature code is used to switch inbound route mode to default Inbound Default Mode mode. The default setting is *61. This feature code is used to switch inbound route mode to mode 1. Inbound Mode 1 The default setting is *62.
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The above recorded call's recording files are also listed under the UCM6200 web GUI->CDR->Recording Files. Figure 154: Download Recording File from Recording Files Page CALL PARK The UCM6200 provides call park and call pickup features via feature code. PARK A CALL There are two feature codes that can be used to park the call.  Feature Maps->Call Park (Default code #72) During an active call, press #72 and the call will be parked.
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Assume a call is on-going between extension A and extension B, user could dial the feature code from extension C to listen on their call (*54 by default), whisper to one side (*55 by default), or barge into the call (*56 by default). Then the user will be asked to enter the number to call, which should be either side of the active call, extension A or B in this example.
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Firmware Version 1.0.0.
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INTERNAL OPTIONS This section describes internal options that haven't been mentioned in previous sections yet. The settings in this section can be applied globally to the UCM6200, including general configurations, jitter buffer, RTP settings, ports config and STUN monitor. The options can be accessed via Web GUI->PBX->Internal Options-> General.
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digits, with a minimum length of 4 digits. Repetitive digits pattern (such as 0000, 1111, 1234, 2345, and etc), or common digits pattern (such as 111222, 321321 and etc) are not allowed to be configured as password. 2. Password for extension registration, web GUI admin login, LDAP and LDAP sync requires alphanumeric characters containing at least two categories of the following, with a minimum length of 4 characters.
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 Auto Provision Extensions: 5000-6299 This sets the range for "Zero Config Extension Segment" which is the extensions can be assigned on the UCM6200 to provision the end device.
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INTERNAL OPTIONS/RTP SETTINGS Table 78: Internal Options/RTP Settings RTP Start Configure the RTP port starting number. The default setting is 10000. RTP End Configure the RTP port ending address. The default setting is 20000. Configure to enable or disable strict RTP protection. If enabled, RTP Strict RTP packets that do not come from the source of the RTP stream will be dropped. The default setting is "Disable". RTP Checksums Configure to enable or disable RTP Checksums on RTP traffic.
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G.721 Compatible G.726 Configure to enable/disable G.721 compatible. The default setting is Yes. Configure the payload type for G.726 if "G.721 Compatible" is disabled. The default setting is 111. iLBC Configure the payload type for iLBC. The default setting is 97. H.264 Configure the payload type for H.264. The default setting is 99. H.263P Configure the payload type for H.263+. The default setting is 100 103. VP8 Configure the payload type for VP8. The default settings is 108.
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Firmware Version 1.0.0.
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IAX SETTINGS The UCM6200 IAX global settings can be accessed via Web GUI->PBX->IAX Settings. IAX SETTINGS/GENERAL Table 80: IAX Settings/General Bind Port Bind Address IAX1 Compatibility Configure the port number that the IAX2 will be allowed to listen to. The default setting is 4569. Configure the address that the IAX2 will be forced to bind to. The default setting is 0.0.0.0, which means all addresses. Select to configure IAX1 compatibility. The default setting is "No".
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If set to "yes", the connection will be terminated if ACK for the NEW Auto Kill message is not received within 2000ms. Users could also specify number (in milliseconds) in addition to "yes" and "no". The default setting is "yes". Authentication Debugging If enabled, authentication traffic in debugging will not show. The default setting is "No". Configure codec negotiation priority. The default setting is "Reqonly".  Caller Consider the callers preferred order ahead of the host's.
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SIP SETTINGS The UCM6200 SIP global settings can be accessed via Web GUI->PBX->SIP Settings. SIP SETTINGS/GENERAL Table 83: SIP Settings/General Realm For Digest Authentication Bind UDP Port Bind IP Address Configure the host name or domain name for the UCM6200. Realms MUST be globally unique according to RFC3261. The default setting is Grandstream. Configure the UDP port used for SIP. The default setting is 5060. Configure the IP address to bind to. The default setting is 0.0.0.
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Support SIP Video Select to enable video support in SIP calls. The default setting is "Yes". If enabled, when rejecting an incoming INVITE or REGISTER request, the UCM6200 will always reject with "401 Unauthorized" instead of notifying Reject Non-Matching INVITE the requester whether there is a matching user or peer for the request. This reduces the ability of an attacker to scan for valid SIP usernames. The default setting is "No".
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Note: The IP address must match the common name (hostname) in the certificate. Please do not bind a TLS socket to multiple IP addresses. For details on how to construct a certificate for SIP, please refer to the following document: http://tools.ietf.org/html/draft-ietf-sip-domain-certs TLS Client Protocol TLS Do Not Verify Select the TLS protocol for outbound client connections. The default setting is TLSv1. If enabled, the TLS server's certificate won't be verified when acting as a client.
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External TCP Port External TLS Port Configure the externally mapped TCP port when the UCM6200 is behind a static NAT or PAT. Configures the externally mapped TLS port when UCM6200 is behind a static NAT or PAT. Specify a list of network addresses that are considered inside of the NAT network. Multiple entries are allowed. If not configured, the external IP Local Network Address address will not be set correctly. A sample configuration could be as follows: 192.168.0.
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should be larger than RTP Timeout. The default setting is no timeout. Trust Remote Party ID Send Remote Party ID Configure whether the Remote-Party-ID should be trusted. The default setting is "No". Configure whether the Remote-Party-ID should be sent or not. The default setting is "No". Configure whether the UCM6200 should generate inband ringing or not. The default setting is "Never".  Yes: The UCM6200 will send 180 Ringing followed by 183 Session Progress and in-band audio.
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Firmware Version 1.0.0.
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PORTS CONFIG The analog hardware (FXS port and FXO port) on the UCM6200 will be listed in this page. Click on to edit signaling preference for FXS port or configure ACIM settings for FXO port. Select "Loop Start" or "Kewl Start" for each FXS port. And then click on "Update" to save the change. Figure 155: FXS Ports Signaling Preference For FXO port, users could manually enter the ACIM settings by selecting the value from dropdown list for each port.
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Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum Operational Loop Current, and AC Impedance as predefined for your country's analog line characteristics. The default setting is "United States of America (USA)". Select country to set the On Hook Speed, Ringer Impedance, Ringer Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum FXS Opermode Operational Loop Current, and AC Impedance as predefined for your country's analog line characteristics.
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VALUE-ADDED FEATURES FAX SENDING The UCM6200 supports sending Fax via web UI access. This feature can be found on web UI->PBX->Value-added Features->Fax Sending page. In order to send fax, pre-setup for analog trunk and outbound route is required. Please refer to [ANALOG TRUNKS], [VOIP TRUNKS] and [OUTBOUND ROUTES] sections for configuring analog trunk and outbound route.
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Figure 158: Announcements Center ANNOUNCEMENTS CENTER SETTINGS Table 90: Announcements Center Settings Name Configure a name for the newly created Announcements Center to identify this announcement center. Enter a code number for the custom prompt. This code will be used in combination with the group number. For example, if the code is 55, and group number is 666. The user can dial 55666 to send prompt 55 to all Code members in group 666.
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Configure the group number. The group number is used in combination with the code. For example, if group number is 666, and code is 55. The user can dial 55666 to send prompt 55 to all members in group 666. Number Note: The combination number must not conflict with any number in the system such as extension number or conference number. Announcements Center feature can be found under web UI->PBX->Value-added Features-> Announcements Center. The following example demonstrates the usage of this feature. 1.
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a new Prompt, please click “Prompt” link and follow the instructions in that page. Figure 160: Announcements Center Code Configuration Code and Group number are used together to direct specified message to the target group. All extensions in the group will receive the message. For example, we can send code 55 to group 666 by dialing 55666 from any extension registered to the UCM6200.
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STATUS AND REPORTING PBX STATUS The UCM6200 monitors the status for Trunks, Extensions, Queues, Conference Rooms, Interfaces and Parking lot. It presents administrators the real time status in different sections under web GUI->Status->PBX Status. Figure 162: Status->PBX Status TRUNKS Users could see all the configured trunk status in this section. Figure 163: Trunk Status Table 92: Trunk Status Display trunk status.
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Unmonitored: QUALIFY feature is not turned on to be monitored. Reachable: The hostname can be reached.  SIP Register trunk status: Registered Unrecognized Trunk Trunks Display trunk name Display trunk Type: Type  Analog  SIP  IAX Username Display username for this trunk. Port/Hostname/IP Display Port for analog trunk, or Hostname/IP for VoIP (SIP/IAX) trunk.
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Table 93: Extension Status Display extension number (including feature code). The color indicator has the following definitions. Status  Green: Free  Blue: Ringing  Yellow: In Use  Grey: Unavailable Extension Display the extension number. Name/Label First name and last name of the extension. Display message status for the extension. Example: 2/4/1 Message Description: There are 2 urgent messages, 4 messages in total and 1 message that has been already read. Displays extension type.
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Figure 165: Queue Status The current call status (caller ID, duration), agent status, service level, calls summary (completed/abandoned) are shown for the call queue. The agent status is defined as below. Table 94: Agent Status The agent is available/idle. The agent is ringing. The agent is talking/busy. The agent has been logged out.
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Figure 166: Conference Room Status Other operations are also available in conference room status section:  Click on "Conference Rooms", the web page will redirect to conference room configuration page which can also be accessed via web GUI->PBX->Call Features->Conference.  Click on  Click on [ + ] to expand the conference room details.  Click on [ - ] to refresh the conference room status. to hide the conference room details.
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LAN/WAN connected. LAN/WAN not configured. LAN/WAN disconnected. FXS/FXO connected. FXS/FXO waiting. FXS/FXO busy. FXS/FXO not configured. FXS/FXO disconnected. Other operations are also available in interface status section:  Click on "Interfaces Status", the web page will redirect to ports configuration page which can also be accessed via web GUI->PBX->Internal Options->Ports Config.  Click on  Click on [ + ] to expand the interface details.  Click on [ - ] to hide the interface details.
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reaches 0, the caller who parks the call will be called back. Other operations are also available in parking lot status section:  Click on "Parking Lot", the web page will redirect to feature codes page which can also be accessed via web GUI->PBX->Internal Options->Feature Codes.  Click on  Click on [ + ] to expand the parking lot details.  Click on [ - ] to hide the parking details. to refresh the parking lot status.
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Recovery Recovery version. NETWORK Under Web GUI->Status->System Status->Network, users could check the network information for the UCM6200. Please see details in the following table. Table 98: System Status->Network Status -> System Status -> Network MAC Address Global unique ID of device, in HEX format. The MAC address can be found on the label coming with original box and on the label located on the bottom of the device. IP Address IP address. Gateway Default gateway address.
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Figure 169: System Status->Storage Usage RESOURCE USAGE When configuring and managing the UCM6200, users could access resource usage information to estimate the current usage and allocate the resources accordingly. Under web UI->Status->System Status->Resource Usage, the current CPU usage and Memory usage are shown in the pie chart. Figure 170: System Status->Resource Usage Firmware Version 1.0.0.
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SYSTEM EVENTS The UCM6200 can monitor important system events, log the alerts and send Email notifications to the system administrator. ALERT EVENTS LIST The system alert events list can be found under Web GUI->Status->System Events->Alert Events List.
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 Detect Cycle: The UCM6200 will perform the internal disk usage detection based on this cycle. Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.  Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM6200 system will send the alert. 2. Memory Usage Figure 172: System Events->Alert Events Lists: Memory Usage  Detect Cycle: The UCM6200 will perform the memory usage detection based on this cycle.
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 Detect Cycle: The UCM will detect the event at each cycle based on the specified time. Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle. Click on the switch to turn on/off the alert and Email notification for the event. Users could also select the checkbox for each event and then click on button "Alert On", "Alert Off", "Email Notification On", "Email Notification Off" to control the alert and Email notification configuration.
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Figure 176: Filter for Alert Log ALERT CONTACT Users could add administrator's Email address under Web GUI->Status->System Events->Alert Contact to send the alert notification to. Up to 10 Email addresses can be added. CDR A Call Detail Record (CDR) is a data record produced by telephone exchange activities or other telecommunications equipment documenting the details of a phone call that passed through the PBX. The CDR is composed of the following data fields on the UCM6200.  Start Time.
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Figure 177: CDR Filter Table 99: CDR Filter Criteria Inbound calls Inbound calls are calls originated from a non-internal source (like a VoIP trunk) and sent to an internal extension. Outbound calls Outbound calls are calls sent to a non-internal source (like a VoIP trunk) from an internal extension. Internal calls Internal calls are calls from one internal extension to another extension, which are not sent over a trunk.
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be filtered out. Callee Number Enter the callee number to filter the CDR report. CDR with the matching callee number will be filtered out. The call report will display as the following figure shows. Figure 178: Call Report Users could perform the following operations on the call report.  Sort Click on the header of the column to sort by this category. For example, clicking on "Start Time" will sort the report according to start time. Clicking on "Start Time" again will reverse the order.
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 Automatic Download CDR Records User could configure the UCM6200 to automatically download the CDR records and send the records to an Email address. Click on “Automatic Download Settings”, and configure the parameters in the dialog below. Figure 180: Automatic Download Settings To receive CDR record automatically from Email, check “Enable” and select a time period “By Day” “By Week” or “By Month” for the automatic download period. Make sure you have entered an Email address to receive the CDR records.
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Figure 182: Detailed CDR Information DOWNLOADED CDR FILE The downloaded CDR (.csv file) has different format from the web UI CDR. Here are some descriptions.  Call From, Call To "Call From": the caller ID. "Call To": the callee ID. If "Call From" shows empty, "Call To" shows "s" (see highlight part in the picture below) and the "Source Channel" contains "DAHDI", this means the call is from FXO/PSTN line.
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Figure 184: Downloaded CDR File Sample - Source Channel and Dest Channel 1 DAHDI means it is an analog call, FXO or FXS. For UCM6202, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports. For UCM6204, DAHDI/(1-4) are FXO ports, and DAHDI(5-6) are FXS ports. For UCM6208, DAHDI/(1-8) are FXO ports, and DAHDI(9-10) are FXS ports. Sample 2: Figure 185: Downloaded CDR File Sample - Source Channel and Dest Channel 2 "SIP" means it's a SIP call.
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STATISTICS CDR Statistics is an additional feature on the UCM6200 which provides users a visual overview of the call report across the time frame. Users can filter with different criteria to generate the statistics chart. Figure 187: CDR Statistics Table 100: CDR Statistics Filter Criteria Trunk Type Call Type Time Range Select one of the following trunk type.  All  SIP Calls  PSTN Calls Select one or more in the following checkboxes.
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Figure 188: CDR->Recording Files  Click on “Delete Selected Recording Files” to delete the recording files.  Click on “Delete All Recording Files” to delete all recording files.  Click on to download the recording file in .wav format.  Click on to delete the recording file.  To sort the recording file, click on the title "Caller", "Callee" or "Call Time" for the corresponding column. Click on the title again can switch the sorting mode between ascending order or descending order.
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Username Configure the Username for API Authentication. Password Configure the Password for API Authentication. Permitted Specify a list of IP addresses permitted by API. This creates an AIP-specific access control list. Multiple entries are allowed. For example, "192.168.40.3/255.255.255.255" denies access from all IP addresses except 192.168.40.3. The default setting is blank, meaning all IPs will be denied. Users must set permitted IP address before connecting to the API.
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Firmware Version 1.0.0.
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UPGRADING AND MAINTENANCE UPGRADING The UCM6200 can be upgraded to a new firmware version remotely or locally. This section describes how to upgrade your UCM6200 via network or local upload. UPGRADING VIA NETWORK The UCM6200 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for the TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP, HTTP or HTTPS; the server name can be FQDN or IP address. Examples of valid URLs: firmware.grandstream.
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Table 102: Network Upgrade Configuration Upgrade Via Allow users to choose the firmware upgrade method: TFTP, HTTP or HTTPS. Firmware Server Path Define the server path for the firmware server. Firmware File Prefix If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the UCM6200. Firmware File Suffix If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the UCM6200.
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Figure 191: Upgrading Firmware Files  Wait until the upgrading process is successful and a window will be popped up in the Web GUI. Figure 192: Reboot UCM6200  Click on "OK" to reboot the UCM6200 and check the firmware version after it boots up. ----------------------------------------------------------------------------------------------------------------------------- --------------Note: Please do not interrupt or power cycle the UCM6200 during upgrading process.
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NO LOCAL FIRMWARE SERVERS Service providers should maintain their own firmware upgrade servers. For users who do not have TFTP/HTTP/HTTPS server, some free windows version TFTP servers are available for download from http://www.solarwinds.com/products/freetools/free_tftp_server.aspx http://tftpd32.jounin.net Please check our website at http://www.grandstream.com/support/firmware for latest firmware. Instructions for local firmware upgrade via TFTP: 1.
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Figure 193: Create New Backup 1. Choose the type(s) of files to be included in the backup. 2. Choose where to store the backup file: USB Disk, SD Card or Local. 3. Name the backup file. 4. Click on "Backup" to start backup. Once the backup is done, the list of the backups will be displayed with date and time in the web page. Users can download , restore , or delete it from the UCM6200 internal storage or the external device. Click on to upload backup file from the local device to UCM6200.
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Figure 194: Backup / Restore option allows UCM to perform automatically backup on the user specified time. Regular backup file can only be stored in USB / SD card / SFTP server. User is allowed to set backup time from 0-23 and how frequent the backup will be performed. Figure 195: Local Backup Firmware Version 1.0.0.
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DATA SYNC Besides local backup, users could backup the voice records/voice mails/CDR/FAX in a daily basis to a remote server via SFTP protocol automatically under Web GUI->Maintenance->Backup->Data Sync. Figure 196: Data Sync Table 103: Data Sync Configuration Enable Data Sync Enable the auto data sync function. The default setting is "No". Account Enter the Account name on the SFTP backup server. Password Enter the Password associate with the Account on the SFTP backup server.
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manually synchronize all data by clicking on instead of waiting for the backup time interval to come. RESTORE CONFIGURATION FROM BACKUP FILE To restore the configuration on the UCM6200 from a backup file, users could go to Web GUI->Maintenance->Backup->Local Backup.  A list of previous configuration backups is displayed on the web page. Users could click on of the desired backup file and it will be restored to the UCM6200.
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CLEANER Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under Web GUI->Maintenance->Cleaner. Figure 198: Cleaner Table 104: Cleaner Configuration Enable CDR Cleaner Enable the CDR Cleaner function. CDR Clean Time Enter 0-23 to specify the hour of the day to clean up CDR. Clean Interval Enter 1-30 to specify the day of the month to clean up CDR. Enable VR Cleaner Enter the Voice Records Cleaner function.
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VR Clean Time Enter 0-23 to specify the hour of the day to clean up Voice Records. Clean Interval Enter 1-30 to specify the day of the month to clean up Voice Records. All the cleaner logs will be listed on the bottom of the page. ----------------------------------------------------------------------------------------------------------------------------- --------------Note: Cleaner will delete data based on Recording Storage selection.
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SYSLOG On the UCM6200, users could dump the syslog information to a remote server under Web GUI->Maintenance->Syslog. Enter the syslog server hostname or IP address and select the module/level for the syslog information. The default syslog level for all modules is "error", which is recommended in your UCM6200 settings because it can be helpful to locate the issues when errors happen.
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Figure 200: Ethernet Capture The output result is in .pcap format. Therefore, users could specify the capture filter as used in general network traffic capture tool (host, src, dst, net, protocol, port, port range) before starting capturing the trace. IP PING Enter the target host in host name or IP address. Then press "Start" button. The output result will dynamically display in the window below. Figure 201: PING Firmware Version 1.0.0.
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TRACEROUTE Enter the target host in host name or IP address. Then press "Start" button. The output result will dynamically display in the window below. Figure 202: Traceroute ANALOG RECORD TRACE Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM6200 user has caller ID issue for incoming call from Analog trunk. Users can access analog record trance under web GUI->Maintenance->Troubleshooting ->Analog Record Trace. Here is the step to capture trace: 1.
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Figure 203: Troubleshooting Analog Trunks After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream Technical support in the following link for further assistance if the issue is not resolved. http://www.grandstream.com/index.php/support SERVICE CHECK Enable Service Check to periodically check UCM6200. Check Cycle is configurable in seconds and the default setting is 60 sec. Check Times is the maximum number of failed checks before restart the UCM6200.
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Figure 205: Network Status REMOTE ACCESS SSH ACCESS SSH switch now is available via web UI and LCD. User can enable or disable SSH access directly from web UI or LCD screen. For web SSH access, please log in UCM6200 web interface and go to Maintenance->Remote Access->SSH Access. By default, SSH access is disabled for security concerns. It is highly recommended to only enable SSH access for debugging purpose. Firmware Version 1.0.0.
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Figure 206: SSH Access Firmware Version 1.0.0.
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EXPERIENCING THE UCM6200 SERIES IP PBX Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products. We encourage you to browse our product related documentation, FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream Certified Partner or Reseller, please contact them directly for immediate support.
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FCC Caution: This device complies with part15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation. This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules.
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Regulatory Information U.S. FCC Part 68 Statement This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA. The unit bears a label on the back which contains among other information a product identifier in the format US: GNIIS00BUCM6202. If requested, this number must be provided to the telephone company. This equipment uses the following standard jack types for network connection: RJ11C. This equipment contains an FCC compliant modular jack.