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UCM6510 IP PBX User Manual

Grandstream Networks, Inc.

UCM6510 IP PBX

User Manual

Summary of content (193 pages)

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Grandstream Networks, Inc.
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UCM6510 IP PBX User Manual Index CHANGE LOG ......................................................................................... 12 FIRMWARE VERSION 1.0.0.5 ............................................................................................................ 12 WELCOME ............................................................................................... 13 PRODUCT OVERVIEW ............................................................................ 14 FEATURE HIGHTLIGHTS .............
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LDAP SERVER .................................................................................................................................... 35 LDAP SERVER CONFIGURATIONS ........................................................................................... 35 LDAP PHONEBOOK .................................................................................................................... 36 LDAP CLIENT CONFIGURATIONS ................................................................................
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VOIP TRUNKS ......................................................................................... 82 VOIP TRUNK CONFIGURATION ........................................................................................................ 82 DIRECT OUTWARD DIALING (DOD) VIA VOIP TRUNKS ................................................................. 89 CALL ROUTES ........................................................................................ 91 OUTBOUND ROUTES .......................................
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EXTENSION GROUPS........................................................................... 123 CONFIGURE EXTENSION GROUPS ............................................................................................... 123 USE EXTENSION GROUPS ............................................................................................................. 123 PICKUP GROUPS .................................................................................. 125 CONFIGURE PICKUP GROUPS ............................
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SIP SETTINGS ....................................................................................... 151 SIP SETTINGS/GENERAL ................................................................................................................ 151 SIP SETTINGS/MISC ........................................................................................................................ 152 SIP SETTINGS/SESSION TIMER ...................................................................................................
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CLEANER .......................................................................................................................................... 186 RESET AND REBOOT ...................................................................................................................... 187 SYSLOG ............................................................................................................................................ 188 TROUBLESHOOTING .....................................................
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Table of Tables UCM6510 IP PBX User Manual Table 1: Technical Specifications................................................................................................................. 14 Table 2: UCM6510 Equipment Packaging .................................................................................................. 17 Table 3: LCD Menu Options ........................................................................................................................
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Table 39: Call Queue Configuration Parameters ...................................................................................... 120 Table 40: FAX/T.38 Settings ...................................................................................................................... 127 Table 41: DISA Settings ............................................................................................................................ 132 Table 42: Event List Settings .........................................
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Table of Figures UCM6510 IP PBX User Manual Figure 1: UCM6510 Front View................................................................................................................... 17 Figure 2: UCM6510 Back View ................................................................................................................... 17 Figure 3: UCM6510 web GUI Login Page ..................................................................................................
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Figure 39: Upload IVR Prompt .................................................................................................................. 107 Figure 40: Language Settings For Voice Prompt ...................................................................................... 109 Figure 41: Voice Prompt Package List ...................................................................................................... 110 Figure 42: New Voice Prompt Language Added ........................................
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Figure 80: System Events->Alert Events Lists: System Update ............................................................... 167 Figure 81: System Events->Alert Events Lists: System Crash ................................................................. 168 Figure 82: System Events->Alert Log ....................................................................................................... 168 Figure 83: CDR Filter .........................................................................................
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CHANGE LOG This section documents significant changes from previous versions of the UCM6510 user manual. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here. FIRMWARE VERSION 1.0.0.5  This is the initial version. Firmware Version 1.0.0.
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WELCOME Thank you for purchasing Grandstream UCM6510 IP PBX appliance. UCM6500 is an innovative IP PBX appliance designed to bring enterprise-grade unified communications and security protection features to small-to-medium businesses (SMBs) in an easy-to-manage fashion.
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PRODUCT OVERVIEW FEATURE HIGHTLIGHTS  1 GHz 4-core Cortex A9 application processor, large memory (1GB DDR3 RAM, 32GB NAND Flash), and dedicated high performance multi-core DSP array for advanced voice processing.  Integrated 1 T1/E1/J1 (J1 is TBD) interface, 2 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability in case of power outage, and up to 50 SIP trunk accounts.  Gigabit network port(s) with integrated PoE, USB, SD card; integrated NAT router with advanced QoS support.
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LED Indicators Power 1/Power 2, PoE, USB, SD, T1/E1/J1(J1 is TBD), FXS 1/FXS 2, FXO 1/FXO 2, LAN, WAN, Heartbeat LCD Display 128x32 graphic LCD with DOWN and OK button Reset Switch Yes Voice/Video Capabilities Voice-over-Packet Capabilities Voice and Fax Codecs LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection and auto-switch to G.711 G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.
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Mounting Rack mount and Desktop Additional Features English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech for web GUI; Multi-language Support Customizable IVR/extension to support English, Chinese, British English, German, Spanish, Greeks, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew and Arabic Caller ID Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 - BT, NTT Japan Polarity Reversal/ Wink Yes, with en
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INSTALLATION Before deploying and configuring the UCM6510 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 UCM6510 series.
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Follow the steps below to connect the UCM6510 for initial setup: 1. Connect one end of an RJ-45 Ethernet cable (cable type: straight through) into the WAN port of the UCM6510; connect the other end into the uplink port of an Ethernet switch/hub. 2. Connect the 12V DC power adapter into the DC 12V power jack 1 on the back of the UCM6510. Insert the main plug of the power adapter into a surge-protected power outlet.
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Warning: Use the power adapter provided with the UCM6510 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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GETTING STARTED The UCM6510 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 UCM6510 setup.
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Table 3: LCD Menu Options View Events Device Info Network Info Network Menu  Critical Events  Other Events  Hardware: Hardware version number  Software: Software version number  P/N: Part number  WAN MAC: WAN side MAC address (UCM6510 only)  LAN MAC: LAN side MAC address  Uptime: System up time since the last reboot.
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 Hardware Testing Select "Test SVIP" to perform SVIP test on the device. This is mainly for factory testing purpose which verifies the hardware connection inside the device. The diagnostic result displays on the LCD after the test is done. Web Info  Protocol: Web access protocol. HTTP or HTTPS. By default it's HTTPS  Port: Web access port number. By default it's 8089 USE THE LED INDICATORS The UCM6510 has LED indicators in the front to display connection status.
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Figure 3: UCM6510 web GUI Login Page To access the web GUI: 1. Connect the computer to the same network as the UCM6510. 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 IP address in the address bar. The web login page will display as shown in [Figure 3: UCM6510 web GUI Login Page]. 4. Enter the administrator’s login and password to access the web configuration menu.
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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, SIP settings, as well as ports configuration for digital trunks.
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Figure 4: UCM6510 web GUI Language 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. Figure 5: UCM6510 web GUI: Apply Changes MAKE YOUR FIRST CALL Power up the UCM6510 and your SIP end point phone.
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2. Click on "Create New SIP Extension" to create a new extension. You will need User ID, Password and Voicemail Password information to register and use the extension later. 3. Register the extension on your phone with the SIP User ID, SIP server and SIP Password information. The SIP server address is the UCM6510 IP address. 4. When your phone is registered with the extension, dial *97 to access the voicemail box. Enter the Voicemail Password once you hear "Password" voice prompt. 5.
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SYSTEM SETTINGS This section explains configurations for system-wide parameters on the UCM6510. Those parameters include Network Settings, Firewall, Change Password, LDAP server, HTTP server, Email settings, Time Settings and NTP Server settings. NETWORK SETTINGS After successfully connecting the UCM6510 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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255.255.0.0. IP Address DNS Server 1 Enter the IP address for static IP settings. The default setting is 192.168.0.160. Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0. DNS Server 2 Enter the DNS server 2 address for static IP settings. 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 WAN port. The default value 802.1Q/VLAN Tag is 0.
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LAN 1 / LAN 2 (when Method is set to "Dual") If "Dual" is selected as "Method", users will need assign the default interface to Default Interface be LAN 1 (mapped to UCM6510 WAN port) or LAN 2 (mapped to UCM6510 LAN port) and then configure network settings for LAN 1/LAN 2. The default interface is LAN 2. IP Method Gateway IP IP Address Subnet Mask DNS Server 1 Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
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PORT FORWORDING The UCM6510 network interface supports router functions which provides users the ability to do port forwarding. If the UCM6510 is set to "Route" under web GUI->Settings->Network Settings->Basic Settings: Method, 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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Table 8: UCM6510 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 UCM6510 zero_config service 5060 Asterisk udp/IPv4 SIP 4569 Asterisk udp/IPv4 SIP 5353 zero_config udp/IPv4 UCM6510 zero_config service 37435 Syslogd
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Figure 6: Create New Firewall Rule Table 10: 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.  Type IN If selected, users will need specify the network interface "LAN", "WAN" or "Both" for the incoming traffic.  OUT Select the service type.
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DYNAMIC DEFENSE Dynamic defense can blacklist hosts dynamically when the UCM6510 is set to "Route" under web GUI->Settings->Network Settings->Basic Settings: Method. If enabled, the traffic coming into the UCM6510 can be monitored, which helps prevent massive connection attempts or brute force attacks to the device. The blacklist can be created and updated by the UCM6510 firewall, which will then be displayed in the web page. Please refer to the following table for dynamic defense options on the UCM6510.
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Table 12: Fail2Ban Settings Global Settings Enable Fail2Ban. The default setting is disabled. Please make sure both "Enable Enable Fail2Ban Fail2Ban" and "Asterisk Service" are turned on in order to use Fail2Ban for SIP authentication on the UCM6510. Banned Duration Max Retry Duration MaxRetry Configure the duration (in seconds) for the detected host to be banned. The default setting is 300. If set to -1, the host will be always banned.
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LDAP SERVER The UCM6510 has an embedded LDAP server for users to manage corporate phonebook in a centralized manner.  By default, the LDAP server has generated the first phonebook with PBX DN "ou=pbx,dc=pbx,dc=com" based on the UCM6510 user extensions already.  Users could add new phonebook with a different Phonebook DN for other external contacts. For example, "ou=people,dc=pbx,dc=com".  All the phonebooks in the UCM6510 LDAP server have the same Base DN "dc=pbx,dc=com".
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The UCM6510 LDAP server supports anonymous access (read-only) by default. Therefore the LDAP client doesn't have to configure username and password to access the phonebook directory. The "Root DN" and "Root Password" here are for LDAP management and configuration where users will need provide for authentication purpose before modifying the LDAP information. The default phonebook list in this LDAP server can be viewed and edited by clicking on for the first phonebook under LDAP Phonebook.
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information will need to be modified via web GUI->PBX->Basic/Call Routes->Extensions first. The default LDAP phonebook will then be updated automatically. A new sibling phonebook of the default PBX phonebook can be added by clicking on "Add" under "LDAP Phonebook" section. Figure 10: Add LDAP Phonebook Configure the "Phonebook Prefix" first. The "Phonebook DN" will be automatically filled in.
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LDAP CLIENT CONFIGURATIONS The configuration on LDAP client is similar when you use other LDAP servers. Here we provide an example on how to configure the LDAP client on the SIP end points to use the default PBX phonebook.
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Figure 12: GXP2200 LDAP Phonebook Configuration HTTP SERVER The UCM6510 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 directly by typing IP address in the PC's web browser (e.g., 192.168.40.50). It will then be automatically redirected to HTTPS using Port 8089 (e.g., https://192.168.40.50:8089).
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Table 13: 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". Protocol Type Port Select HTTP or HTTPS. The default setting is "HTTPS". Specify port number to access the HTTP server. The default port number is 8089.
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when using type "Client". Display Name Sender Specify the display name in the FROM header in the Email. Specify the sender's Email address. For example, pbx@example.mycompany.com. The following figure shows a sample Email settings on the UCM6510, assuming the Email is using smtp.gmail.com as the SMTP server. Figure 13: UCM6510 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 UCM6510.
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Figure 14: UCM6510 Email Settings: Send Test Email TIME SETTINGS The current system time on the UCM6510 is displayed on the upper right of the web page. It can also be found under web GUI->Status->System Status->General. To configure the UCM6510 to update time automatically, go to web GUI->Settings->Time Settings-> Auto Time Updating. Table 15: Auto Time Updating Specify the URL or IP address of the NTP server for the UCM6510 to Remote NTP Server synchronize the date and time.
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time accordingly. If "Self-Defined Tome Zone" is selected, please specify the time zone parameters in "Self-Defined Time Zone" field as described in below option. If "Self-Defined Time Zone" is selected in "Time Zone" option, users will need define their own time zone following the format below. The syntax is: std offset dst [offset], start [/time], end [/time] Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0 MTZ+6MDT+5 This indicates a time zone with 6 hours offset and 1 hour ahead for DST, which is U.
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NTP SERVER The UCM6510 can be used as a NTP server for the NTP clients to synchronize their time with. To configure the UCM6510 as the NTP server, set "Enable NTP server" to "Yes" under web GUI->Settings->Time Settings->NTP Server. On the client side, point the NTP server address to the UCM6510 IP address or host name to use the UCM6510 as the NTP server. 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 UCM6510 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 16: UCM6510 Zero Config  SIP SUBSCRIBE When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The UCM6510 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 UCM6510 and reboot again to take the new configuration.
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Figure 17: Auto Provision Settings Table 16: Auto Provision Settings Enable Zero Config Enable or disable the zero config feature on the PBX. The default setting is disabled. If enabled, when the device is discovered, the PBX will automatically Automatically Assign Extension assign an extension within the range defined in "Zero Config Extension Segment" to the device. The default setting is disabled.
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Please make sure an extension is manually assigned to the phone or "Automatically Assign Extension" is enabled during provisioning. After the configuration on the UCM6510 web GUI, click on "Save" and "Apply Changes". Once the phone boots up and picks up the config file from the UCM6510, it will take the configuration right away. MANUAL PROVISIONING DISCOVERY Users could manually discover the device by specifying the IP address or scanning the entire LAN network.
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Figure 19: Discovered Devices ASSIGNMENT In the discovered list, click on to open the edit dialog to assign an extension or multiple extensions to this device. Hot-Desking can also be enabled from this edit page. Figure 20: Assign Extension To Device After saving the edit dialog, the XML config file will be generated in the UCM6510. Reboot the phone or trigger the phone to download the config file by clicking on list. Firmware Version 1.0.0.
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CREATE NEW DEVICE Users could also directly create a new device and assign the extension before the device is discovered by the UCM6510. Once the device is plugged in, it can then be discovered and provisioned by the UCM6510. Click on "Create New Device" and the following dialog will show. Enabled Hot-Desking (optional), fill in the MAC address (required), IP address (optional), Version (optional), Model (optional) and the extension (required) to assign to the device.
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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. The configuration parameters are as follows. Table 17: SIP Extension Configuration Parameters General Extension The extension number associated with the user.
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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. Configure the Call Forward Busy target number. If not configured, the Call Forward Busy Call Forward Busy feature is deactivated. The default setting is deactivated. 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).
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GUI->PBX->Internal Options->Language. SIP Settings Use NAT when the UCM6510 is on a public IP communicating with NAT devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it's related to NAT configuration or Firewall's support of SIP and RTP ports. The default setting is enabled. By default, the UCM6510 will route the media steams from SIP endpoints through itself.
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Note: If enabled, Fax Pass-through cannot be used. This option controls how the extension can be used on devices within different types of network.  Allow All Device in any network can register this extension.  Local Subnet Only Only the user in specific subnet can register this extension. Up to Strategy three subnet addresses can be specified.  A Specific IP Address Only the device on the specific IP address can register this extension. The default setting is "Allow All".
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"Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal". Note: Users need to have the same level as or higher level than an outbound rule's privilege in order to make outbound calls using this rule. Configure the password for the user. A random secure password will be SIP/IAX Password automatically generated. It is recommended to use this password for security purpose. Enable Voicemail Enable voicemail for the user.
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Verification skipped. If enabled, this would allow one-button voicemail access. By default this option is disabled. User Settings First Name Last Name Email Address Configure the first name of the user. The first name can contain characters, letters, digits and _. Configure the last name of the user. The last name can contain characters, letters, digits and _. Fill in the Email address for the user. Voicemail will be sent to this Email address.
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Only the user in specific subnet can register this extension. Up to three subnet addresses can be specified.  A Specific IP Address Only the device on the specific IP address can register this extension. The default setting is "Allow All". Skip Trunk Auth If enabled, users will not need enter the "PIN Set" required by the outbound rule to make outbound calls. The default setting is "No". Select audio and video codec for the extension. The available codecs Codec Preference are: PCMU, PCMA, GSM, AAL2-G.
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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. Configure the Call Forward Unconditional target number. If not Call Forward Unconditional configured, the Call Forward Unconditional feature is deactivated. The default setting is deactivated. Configure the Call Forward No Answer target number.
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default setting is "Default" which is the selected voice prompt language under web GUI->PBX->Internal Options->Language. The dropdown list shows all the current available voice prompt languages on the UCM6510. To add more languages in the list, please download voice prompt package by selecting "Check Prompt List" under web GUI->PBX->Internal Options->Language. Analog Settings Call Waiting User # as SEND RX Gain TX Gain Configure to enable/disable call waiting feature. The default setting is "No".
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Send CallerID After Configure the number of rings before sending CID. The default setting is 1. Other Settings Enable to detect Fax signal from the user/trunk during the call and send the received Fax to the Email address configured for this extension. If no Email address can be found for the user, send the received Fax to the Fax Detection default Email address in Fax setting page under web GUI->PBX->Internal Options->Fax/T.38. Note: If enabled, Fax Pass-through cannot be used.
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 User Random Password. A random secure password will be automatically generated. It is recommended to use this password for security purpose.  Use Extension as Password.  Enter a password to be used on all the extensions in the batch. Configure Voicemail password (digits only) for the users.  User Random Password. A random password in digits will be automatically generated. It is Voicemail Password recommended to use this password for security purpose.  Use Extension as Password.
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"RFC2833". If "Info" is selected, SIP INFO message will be used. If "Inband" is selected, 64-kbit codec PCMU and PCMA are required. When "Auto" is selected, RFC2833 will be used if offered, otherwise "Inband" will be used.  Port: Allow peers matching by IP address without matching port number.  Very: Allow peers matching by IP address without matching port number. Also, authentication of incoming INVITE messages is not Insecure required.
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Skip Trunk Auth If enabled, users will not need enter the "PIN Set" required by the outbound rule to make outbound calls. The default setting is "No". Select audio and video codec for the extension. The available codecs Codec Preference are: PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, ILBC, ADPCM, LPC10, H.264, H.263 and H.263p. BATCH ADD IAX EXTENSIONS Under web GUI->PBX->Basic/Call Routes->Extensions, click on "Batch Add Extensions"->"Batch Add IAX Extensions".
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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 UCM6510, which can be configured in the global ring timeout setting Ring Timeout under web GUI->Internal Options->IVR Prompt: General Preference. The valid range is between 5 seconds and 600 seconds.
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 Local Subnet Only Only the user in specific subnet can register this extension. Up to three subnet addresses can be specified.  A Specific IP Address. Only the device on the specific IP address can register this extension. The default setting is "Allow All". If enabled, users will not need enter the "PIN Set" required by the Skip Trunk Auth outbound rule to make outbound calls. The default setting is "No". Select audio and video codec for the extension.
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Click on to send NOTIFY reboot event to the device which has an UCM6510 extension already registered. To successfully reboot the user, "Zero Config" needs to be enabled on the UCM6510 web GUI->PBX->Basic/Call Routes->Zero Config->Auto Provisioning Settings.  Delete single extension Click on to delete the extension. Or select the checkbox of the extension and then click on "Delete Selected Extensions".  Modify selected extensions Select the checkbox for the extension(s).
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IMPORT EXTENSIONS The capability to import extensions to the UCM6510 provides users flexibility to batch add extensions with similar or different configurations quickly. 1. Export extension csv file from the UCM6510 by clicking on "Export Extensions" button. 2. Fill up the extension information you would like in the exported csv template. 3. Click on "Import Extensions" button. The following dialog will be prompted. Figure 23: Export Extensions 4.
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ANALOG TRUNKS To set up analog trunk on the UCM6510:  Go to web GUI->PBX->Basic/Call Routes->Analog Trunks to add and edit analog trunks.  Go to web GUI->PBX->Ports Config->Analog Hardware to configure analog hardware settings. ANALOG TRUNKS CONFIGURATION 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.
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Configure the ring timeout (in ms). Trunk (FXO) devices must have a timeout to determine if there was a hangup before the line is answered. Ring Timeout This value can be used to configure how long it takes before the UCM6510 considers a non-ringing line with hangup activity. The default setting is 8000. RX Gain TX Gain Use CallerID Configure the RX gain for the receiving channel of analog FXO port. The valid range is from -13.5 (dB) to + 12.0 (dB). The default setting is 0.
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f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]]; Frequencies are in Hz and cadence on and off are in ms. Frequencies Range: [0, 4000) Busy Level Range: (-300, 0) Cadence Range: [0, 16383]. Select Tone Country "Custom" to manually configure Busy Tone value. Default value: f1=480@-50,f2=620@-50,c=500/500 Syntax: f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]]; Frequencies are in Hz and cadence on and off are in ms.
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Figure 24: UCM6510 FXO Tone Settings 4. Click on "Detect" to start PSTN detection. Figure 25: UCM6510 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.
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Figure 26: UCM6510 PSTN Detection: Auto Detect  If there is only one FXO port connected to PSTN line, use the following settings for auto-detection. Figure 27: UCM6510 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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Table 23: PSTN Detection For Analog Trunk Select "Auto Detect" or "Semi-auto Detect" for PSTN detection.  Auto Detect Please make sure two or more channels are connected to the UCM6510 and in idle status before starting the detection. During the detection, one channel will be used as caller (Source Channel) and another channel will be used as callee (Destination Channel). The UCM6510 will control the call to be established and hang up between caller and callee to finish the detection.
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Select "Loop Start" or "Kewl Start" for each FXS port. And then click on "Update" to save the change. Figure 28: FXS Ports Signaling Preference For FXO port, users could manually enter the ACIM settings by selecting the value from dropdown list for each port. Or users could click on "Detect" for the UCM6510 to automatically detect the ACIM value. The detecting value will be automatically filled into the settings.
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Select country to set the On Hook Speed, Ringer Impedance, Ringer Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum FXO Opermode Operational Loop Current, and AC Impedance as predefined for your country's analog line characteristics. The default setting is "United States of America (USA)".
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DIGITAL TRUNKS The UCM6510 supports E1/T1 which are physical connection technology used in digital network. T1 is the North American format whereas E1 is the European format with different transmission speed. Currently PRI signaling is supported for the E1/T1 interface on the UCM6510. To set up digital trunk on the UCM6510:  Go to web GUI->PBX->Ports Config->Digital Hardware to configure port type and channels.  Go to web GUI->PBX->Basic/Call Routes->Digital Trunks to add and edit digit trunks.
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The following dialog shows the digital port configuration parameters. Click on "Show Advanced Options" to view more options. Table 25: Ports Config/Digital Hardware: Edit Digital Ports Span Type Select the digital channel mode "E1", "T1" or "J1" (J1 is TBD). All T1/E1 spans generate a clock signal on their transmit side. The parameter determines whether the clock signal from the far end of the T1/E1 is used as the master source of clock timing.
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is used by default. This setting is used to specify the type of the callee number. The service provider will usually verify this. The default setting is "unknown". In some very unusual circumstances, you may need set to "dynamic" or "redundant". PRI Dial Plan Note: When one type is selected, you might not be able to dial another class of numbers. For example, if "national" is configured, you won't be able to dial local or international numbers.
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DIGITAL TRUNK CONFIGURATION After configuring digital hardware, go to web GUI->PBX->Basic/Call Routes->Digital Trunks.  Click on "Create New Digital Trunk" to add a new digital trunk.  Click on  Click on  Click on to configure detailed parameters for the digital trunk. to configure Direct Outward Dialing (DOD) for the digital Trunk. to delete the digital trunk. The digital trunk parameters are listed in the table below.
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Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web Fax Detection GUI->PBX->Internal Options->Fax/T.38. Note: If enabled, Fax Pass-through cannot be used. DIRECT OUTWARD DIALING (DOD) VIA DIGITAL TRUNKS Please refer to section [DIRECT OUTWARD DIALING (DOD) VIA VOIP TRUNKS].
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 Click on "Start" to start capturing trace. The output result shows "Capturing...".  Once the test is done, click on "Stop" to stop the trace.  Click on "Download" to download the trace.  To delete the trace, click on "Delete". Firmware Version 1.0.0.
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VOIP TRUNKS VOIP TRUNK CONFIGURATION VoIP trunks can be configured in UCM6510 under web GUI->PBX->Basic/Call Routes->VoIP Trunks. Once created, the VoIP trunks will be listed with Provider Name, Type, Hostname/IP, Username and Options to edit/detect the trunk.  Click on "Create New SIP Trunk" or "Create New IAX Trunk" to add a new VoIP trunk.  Click on  Click on to configure Direct Outward Dialing (DOD) for the SIP Trunk.  Click on to start LDAP Sync.
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Trunk" type. Enable automatic recording for the calls using this trunk (for SIP trunk Auto Record only). The default setting is disabled. The recording files are saved in external storage if plugged in and can be accessed under web GUI->CDR->Recording Files. Peer SIP Trunk Configuration Parameters Provider Name Host Name Configure the provider name for the VoIP trunk. This is a unique label to identify the trunk when listed in outbound rules, inbound rules and etc.
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Select audio and video codec for the VoIP trunk. The available codecs Codec Preference are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, ILBC, ADPCM, H.264, H.263, H.263p. Enable automatic recording for the calls using this trunk. The default Auto Record setting is disabled. The recording files are saved in external storage if plugged in and can be accessed under web GUI->CDR->Recording Files.
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automatically modify the remote contacts by adding this prefix. Register SIP Trunk Configuration Parameters Provider Name Host Name Configure the provider name for the VoIP trunk. This is a unique label to identify the trunk when listed in outbound rules, inbound rules and etc. Configure the IP address or URL for the VoIP provider server of the trunk. Configure the SIP transport protocol to be used in this trunk. The default setting is "All - UDP Primary".
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Outbound Proxy Support Outbound Proxy Select to enable outbound proxy in this trunk. The default setting is "No". When outbound proxy support is enabled, enter the IP address or URL of the outbound proxy. Enable automatic recording for the calls using this trunk. The default Auto Record setting is disabled. The recording files can be accessed under web GUI->CDR->Recording Files.
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"Register IAX Trunk" type is selected. Password Enter the password to register to the trunk from the provider when "Register IAX Trunk" type is selected. Peer IAX Trunk Configuration Parameters Provider Name Host Name Keep Trunk CID Configure the provider name for the VoIP trunk. This is a unique label to identify the trunk when listed in outbound rules, inbound rules and etc. Configure the IP address or URL for the VoIP provider server of the trunk.
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GUI->PBX->Internal Options->Fax/T.38. Note: If enabled, Fax Pass-through cannot be used. Register IAX Trunk Configuration Parameters Provider Name Host Name Keep Trunk CID Configure the provider name for the VoIP trunk. This is a unique label to identify the trunk when listed in outbound rules, inbound rules and etc. Configure the IP address or URL for the VoIP provider server of the trunk. When enabled, it can avoid overridden by extension's CID if the extension has CID configured.
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timeout, the device is considered offline. The default setting is 1000ms. When "Enable Qualify" option is set to "Yes", configure the interval (in Qualify Frequency seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds. Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web GUI->PBX->Internal Options->Fax/T.38.
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Figure 32: DOD extension selection 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 33: Edit DOD Firmware Version 1.0.0.
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CALL ROUTES OUTBOUND ROUTES In the UCM6510, 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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to use this rule.  National: Users with National or International level are allowed to use this rule.  International: The highest level required. Only users with international level can use this rule. The default setting is "International". Please be aware of the potential security risks when using "Internal" level, which means all users can use this outbound rule to dial out from the trunk. When enabled, users could specify extensions allowed to use this outbound route.
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Specify the digits to be prepended before the call is placed via the trunk. Prepend Those digits will be prepended after the dialing number is stripped. Use Failover Trunk Failover trunks can be used to make sure that a call goes through an alternate route, when the primary trunk is busy or down. If "Use Failover Trunk" is enabled and "Failover trunk" is defined, the calls that cannot be placed via the regular trunk may have a secondary trunk to go through.
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INBOUND RULE CONFIGURATIONS Table 30: 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.  The pattern can be composed of two parts, divided by a ‘/’ character.
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 Fax  DISA  IVR  By DID When "By DID" is used, the UCM6510 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 Dial By Name Specify the number of digits to strip from the beginning of the DID.
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 Voicemail  Conference Room  Call Queue  Ring Group  Paging/Intercom  Voicemail Group  Fax  DISA  IVR  By DID When "By DID" is used, the UCM6510 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.
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 Select the checkbox for "Blacklist Enable" to turn on Blacklist feature for all inbound routes. Blacklist is disabled by default.  Enter a number in "Add Blacklist Number" field and then click  To remove a number from the Blacklist, select the number in "Blacklist list" and click on to add to the list. .
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CONFERENCE BRIDGE The UCM6510 supports conference bridge allowing 32 participants with up to 5 bridges at the same time. The conference bridge configurations can be accessed under web GUI->PBX->Call Features->Conference. In this page, users could create, edit, view, invite, manage the participants and delete conference bridges. The conference bridge status and conference call recordings (if recording is enabled) will be displayed in this web page as well.
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Enable Caller Menu 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. All the recording files will be displayed and can be downloaded in the conference web page. The default setting is "No". If enabled, if there are users joining or leaving the conference, voice prompt or notification tone won't be played.
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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. If password is required, enter the password to join the conference as a normal user, or enter the admin password to join the conference as administrator.
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0: If 0 is entered to invite other party, once the invited party picks up the invitation call, a permission will be asked to "accept" or "reject" the invitation before joining the conference. 1: If 1 is entered to invite other party, no permission will be required from the invited party. Note: Conference administrator can always invite other parties from the phone during the call by entering 0 or 1. To join a conference bridge as administrator, enter the admin password when joining the conference.
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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.  3: Mute/Unmute all non-Administrator participants from the conference call.  4: Record the conference call.  8: Exit the caller menu and return to the conference.
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The recording files will be listed as below once available. Users could click on recording or click on to download the to delete the recording. Figure 36: Conference Recording Firmware Version 1.0.0.
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IVR CONFIGURE IVR IVR configurations can be accessed under the UCM6510 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 33: IVR Configuration Parameters 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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detected within the timeout, a timeout prompt will be played. The default setting is 10 seconds. Response Timeout Prompt Invalid Prompt Response Timeout Repeat Loops Select the prompt message to be played when timeout occurs. Select the prompt message to be played when an invalid extension is pressed. Configure the number of times to repeat the prompt if no DTMF input is detected. When the loop ends, it will go to the timeout destination if configured, or hang up. The default setting is 3.
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CREATE IVR 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 IVR Prompt page. Or users could go to web GUI->PBX->Internal Options->IVR Prompt page directly. Figure 37: Click On Prompt To Create IVR Prompt Once the IVR prompt file is successfully added to the UCM6510, it will be added into the prompt list options for users to select in different IVR scenarios.
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 Click the "Record" button. A request will be sent to the UCM6510. The UCM6510 will then call the extension for recording the IVR prompt from the phone.  Pick up the call from the extension and start the recording following the voice prompt.  The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play or delete the recording.
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LANGUAGE SETTINGS FOR VOICE PROMPT The UCM6510 supports multiple languages in web GUI as well as system voice prompt. The following languages are currently supported in system voice prompt: English (United States) Arabic Chinese Dutch English (United Kingdom) French German Greek Hebrew Italian Polish Portuguese Russian Spanish Swedish Turkish English (United States) and Chinese voice prompts are built in with the UCM6510 already.
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Figure 40: Language Settings For Voice Prompt A new dialog window of voice prompt package list will be displayed. Users can see the version number (latest version available V.S. current installed version), package size and options to upgrade or download the language. Firmware Version 1.0.0.
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Figure 41: Voice Prompt Package List Click on to download the language to the UCM6510. The installation will be automatically started once the downloading is finished. Firmware Version 1.0.0.
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Figure 42: New Voice Prompt Language Added A new language option will be displayed after successfully installed. Users then could select it to apply in the UCM6510 system voice prompt or delete it from the UCM6510. CUSTOMIZE AND UPLOAD VOICE PROMPT PACKAGE The UCM6510 provides interface from web GUI for users to customize their own voice prompts. Users could directly upload the package from web GUI.
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VOICEMAIL CONFIGURE VOICEMAIL If the voicemail is enabled for UCM6510 extensions, the configurations of the voicemail can be globally set up and managed under web GUI->PBX->Call Features->Voicemail. Table 34: 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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announced at the beginning of the voicemail message. The default setting is "No". Announce Message Duration If enabled, the message duration will be announced at the beginning of the voicemail message. The default setting is "No". 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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Table 35: Voicemail Email Settings Attach Recordings to E-Mail If enabled, voicemails will be sent to user's Email address. The default setting is "Yes". Fill in the "Subject:" and "Message:" content, to be used in the Email when sending to the user. The template variables are: Template For Voicemail Emails  \t: TAB  ${VM_NAME}: Recipient's first name and last name  ${VM_DUR}: The duration of the voicemail message  ${VM_MAILBOX}: The recipient's extension  ${VM_CALLERID}: The caller ID of
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Table 36: Voicemail Group Settings Voicemail Group Extension Name Enter the Voicemail Group Extension. The voicemail messages left to this extension will be forwarded to all the voicemail group members. Configure the Name to identify the voicemail group. Letters, digits, _ and - are allowed. Select available mailboxes from the left list and add them to the right list. Voicemail Group Mailboxes The extensions need to have voicemail enabled to be listed in available mailboxes list. Firmware Version 1.0.
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RING GROUP The UCM6510 supports ring group feature with different ring strategies applied to the ring group members. This section describes the ring group configuration on the UCM6510. CONFIGURE RING GROUP Ring group settings can be accessed via web GUI->PBX->Call Features->Ring Group. Figure 45: 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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Note: The actual ring timeout might be overridden by users if the phone has ring timeout settings as well. If enabled, users could select extension, voicemail, ring group, IVR, call Enable Destination queue, voicemail group as the destination if the call to the ring group has no answer. Secret and Email address are required if voicemail is selected as the destination. Configure the password to access the ring group extension's voicemail. Secret Note: The password has to be at least 4 characters.
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PAGING AND INTERCOM GROUP The UCM6510 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. CONFIGURE PAGING/INTERCOM GROUP  Click on "Create New Paging/Intercom Group" to add paging/intercom group. Figure 47: Paging/Intercom Group Table 38: Paging/Intercom Group Configuration Parameters Name Configure paging/intercom group name.
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 Click on  Click on "Paging/Intercom Group Settings" to edit Alert-Info Header. This header will be included in the to delete the paging/intercom group. SIP INVITE message sent to the callee in paging/intercom call. Figure 48: Page/Intercom Group Settings  The UCM6510 has pre-configured paging/intercom feature code. By default, the Paging Prefix is *81 and the Intercom Prefix is *80. To edit page/intercom feature code, click on "Feature Codes" in the "Paging/Intercom Group Settings" dialog.
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CALL QUEUE The UCM6510 supports call queue by using static agents or dynamic agents. This sections 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.  Click on "Create New Queue" to add call queue. Figure 49: Call Queue  Click on to edit the call queue. The call queue configuration parameters are listed in the table below.
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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. Configure whether the callers will be disconnected from the queue or not if the queue has no agent anymore. The default setting is "Strict".  Yes Callers will be disconnected from the queue if all agents are paused or invalid.
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If enabled, the UCM6510 will report (to the agent) the duration of time of Report Hold Time the call before the caller is connected to the agent. The default setting is "No". If enabled, users will be disconnected after the configured number of seconds. The default setting is "No". Wait Time Note: It is recommended to configure "Wait Time" longer than the "Wrapup Time". Select the available users to be the static agents in the call queue.
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EXTENSION GROUPS The UCM6510 extension group feature allows users to assign extensions to different groups to better manage the configurations on the PBX. 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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Figure 52: Select Extension Group in Outbound Route Firmware Version 1.0.0.
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PICKUP GROUPS The UCM6510 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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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 54: 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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FAX/T.38 The UCM6510 supports T.30/T.38 Fax and Fax Pass-through. It can also 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. CONFIGURE FAX/T.38  Click on "Create New Fax Extension". In the popped up window, fill the extension, name and Email address to send the received Fax to.  Click on "Fax Settings" to configure the Fax parameters. Table 40: FAX/T.
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 Click on to edit the Fax extension.  Click on to delete the Fax extension. SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE The following instructions describes how to use the UCM6510 to receive Fax from PSTN line on the Fax machine connected to the UCM6510 FXS port. 1. Connect Fax machine to the UCM6510 FXS port. 2. Connect PSTN line to the UCM6510 FXO port. 3. Go to web GUI->PBX->Analog Trunks page. 4. Create or edit the analog trunk for Fax as below.
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Figure 56: Configure Extension For Fax Machine 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 57: Configure Inbound Rule For Fax 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. Firmware Version 1.0.0.
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SAMPLE CONFIGURATION FOR FAX-TO-EMAIL The following instructions describes a sample configuration on how to use Fax-to-Email feature on the UCM6510. 1. Connect PSTN line to the UCM6510 FXO port. 2. Go to UCM6510 web GUI->Internal Options->Fax/T.38 page. Create a new Fax extension. Figure 58: Create Fax Extension 3. Go to UCM6510 web GUI->Basic/Call Routes->Analog Trunks page. Create a new analog trunk. Please make sure "Fax Detection" is set to "Yes". Figure 59: Enable Fax Detection In Analog Trunk 4.
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Figure 60: 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@ucm6510mycompany.com as attachment. Firmware Version 1.0.0.
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DISA The UCM6510 supports DISA to be used in IVR or inbound route. Before using it, create new DISA under web GUI->Call Features->DISA.  Click on "Create New IVR" to add a new DISA.  Click on to edit the DISA configuration.  Click on to delete the DISA. Figure 61: Create New DISA Table 41: DISA Settings Name Configure DISA name to identify the DISA. Configure the password (digit only) required for the user to enter before using DISA to dial out.
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route's privilege level, the call will be allowed to go through. Configure the maximum amount of time the UCM6510 will wait before Response Timeout hanging up if the user dials an incomplete or invalid number. The default setting is 10 seconds. Digit Timeout Configure the maximum amount of time permitted between digits when the user is typing the extension. The default setting is 5 seconds.
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BLF AND EVENT LIST BLF The UCM6510 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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Figure 62: Create New Event List Table 42: Event List Settings Configure the name of this event list (for example, office_event_list). URI Please note the URI name cannot be the same as the extension name on the UCM6510. The valid characters are letters, digits, _ and -. Local Extensions Select the available extensions listed on the local UCM6510 to be monitored in the event list.
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UCM6510 first before using remote BLF feature. Please note the SIP end points need support event list BLF in order to monitor remote extensions. When an event list is created on the UCM6510 and remote extensions are added to the list, the UCM6510 will send out SIP SUBSCIRBE to the remote UCM6510 to obtain the remote extension status. When the SIP end points registers and subscribes to the local UCM6510 event list, it can obtain the remote extension status from this event list.
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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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Figure 64: Dial By Name Group In IVR Key Pressing Events Figure 65: Dial By Name Group In IVR Key Pressing Events Firmware Version 1.0.0.
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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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CALL FEATURES The UCM6510 supports call recording, transfer, call forward, call park and other call features via feature code. This section lists all the feature codes in the UCM6510 and describes how to use the call features. FEATURE CODES Table 43: UCM6510 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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callee.  Default code: #72.  Enter the code during active call to park the call.  Options Disable Call Park 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 UCM6510 will mix the streams natively on the fly as the call is in progress.
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 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. Parking Timeout (s)  Default setting: 300.  This is the timeout allowed for a call to be parked.
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 Default Code: *8.  This code is for the pickup group which can be assigned for each extension on the extension configuration page. Pickup Extension  If there is an incoming call to an extension, the other extensions within the same pickup group can dial *8 directly to pick up the call.  Default Code: *  This code is for the user to directly dial or transfer to an extension's voicemail.
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The above recorded call's recording files are also listed under the UCM6510 web GUI->CDR->Recording Files. CALL PARK The UCM6510 provides call park and call pickup features via feature code. PARK A CALL There are two feature codes that can be used to park the call.  Feature Maps->Call Park (Default code #72) During an active call, press #72 and the call will be parked. Parking lot number (default range 701 to 720) will be announced after parking the call.
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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 UCM6510, including general configurations, jitter buffer, RTP settings, hardware config and STUN monitor. The options can be accessed via web GUI->PBX->Internal Options.
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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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Configure the time (in ms) to buffer. This is the jitter buffer size used in Jitter Buffer Size "Fixed" jitter buffer, or used as the initial time for "adaptive" jitter buffer. The default setting is 100. Configure the maximum time (in ms) to buffer for "Adaptive" jitter buffer Max Jitter Buffer implementation, or used as the jitter buffer size for "Fixed" jitter buffer implementation. The default setting is 200. Configure the jitter buffer implementation on the sending side of a SIP channel.
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IAX SETTINGS The UCM6510 IAX global settings can be accessed via web GUI->PBX->IAX Settings. IAX SETTINGS/GENERAL Table 48: 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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setting is 100. 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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Enter the IP address or a range of IP addresses to be considered for call number limits. IP or IP Range For example: 11.11.11.11 11.11.11.11/22.22.22.22. Firmware Version 1.0.0.
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SIP SETTINGS The UCM6510 SIP global settings can be accessed via web GUI->PBX->SIP Settings. SIP SETTINGS/GENERAL Table 51: SIP Settings/General Realm For Digest Authentication Bind UDP Port Bind IP Address Configure the host name or domain name for the UCM6510. 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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which can direct the call to a specific context if desired. By default, all domains are accepted and sent to the default context or the context associated with the user/peer placing the call. Register to non-local domains will be automatically denied if a domain list is configured. Up to 10 domains can be added. From Domain Auto Domain Allow External Domains Configure the domain in the "From:" header of the SIP message. It may be required by some providers for authentication.
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 Originate Always request and run session timer.  Accept Run session timer only when requested by other UA.  Refuse Do not run session timer. Session Expire Min SE Session Refresher Configure the maximum session refresh interval (in seconds). The default setting is 1800. Configure the minimum session refresh interval (in seconds). The default setting is 90. Select the session refresher to be UAC or UAS. The default setting is UAC.
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This is the CA certificate if the TLS server being connected to requires self-signed certificate, including server's public key. This file will be TLS Self-Signed CA renames as "TLS.ca" automatically. Note: The size of the uploaded ca file must be under 2MB. This is the Certificate file (*.pem format only) used for TLS connections. It contains private key for client and signed certificate for the server. This TLS Cert file will be renamed as "TLS.pem" automatically.
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A sample configuration could be as follows: 192.168.0.0/16 SIP SETTINGS/TOS Table 56: SIP Settings/ToS ToS For SIP ToS For RTP Audio ToS For RTP Video Configure the Type of Service for SIP packets. The default setting is None. Configure the Type of Service for RTP audio packets. The default setting is None. Configure the Type of Service for RTP video packets. The default setting is None.
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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 UCM6510 should generate inband ringing or not. The default setting is "Never".  Yes: The UCM6510 will send 180 Ringing followed by 183 Session Progress and in-band audio.
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STATUS AND REPORTING PBX STATUS The UCM6510 monitors the status for Trunks, Extensions, Queues, Conference Rooms, Interfaces, Digital Channels and Parking lot. It presents administrators the real time status in different sections under web GUI->Status->PBX Status. Figure 68: Status->PBX Status TRUNKS Users could see all the configured trunk status in this section. Figure 69: Trunk Status Firmware Version 1.0.0.
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Table 57: Trunk Status Display trunk status.  Analog trunk/Digital trunk status: Available Busy Unavailable Unknown Error  Status SIP Peer trunk status: Unreachable: The hostname cannot be reached. 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  E1/T1  SIP  IAX Username Display username for this trunk.
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Figure 70: Extension Status Table 58: 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 Display name (callerID name) or label for the extension. Display message status for the extension.
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 Click on one of the tabs to display the corresponding extensions accordingly.  Click on [ + ] to expand the status detail table.  Click on [ - ] to hide the status detail table. QUEUES Users could see all the configured call queue status in this section. The following figure shows the call queue 6500 being in used. Figure 71: Queue Status The current call status (caller ID, duration), agent status, service level, calls summary (completed/abandoned) are shown for the call queue.
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CONFERENCE ROOMS Users could see all the conference room status in this section. It shows all the configured conference rooms, current users, call duration for each user and conference call. Figure 72: 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.
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LAN/WAN/Heartbeat disconnected. FXS/FXO/Digital connected. FXS/FXO/Digital waiting. FXS/FXO/Digital busy. FXS/FXO/Digital not configured. FXS/FXO/Digital disconnected. Other operations are also available in interface status section:  Click on  Click on [ + ] to expand the interface details.  Click on [ - ] to hide the interface details. to refresh the interface status. PARKING LOT The UCM6510 supports call park using feature code.
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 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. SYSTEM STATUS The UCM6510 system status can be accessed via web GUI->Status->System Status, which displays the following system information.
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NETWORK Under web GUI->Status->System Status->Network, users could check the network information for the UCM6510. Please see details in the following table. Table 63: 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. Subnet Mask Subnet mask address.
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Figure 74: System Status->Storage Usage RESOURCE USAGE When configuring and managing the UCM6510, users could access resource usage information to estimate the current usage and allocate the resources accordingly. Under web GUI->Status->System Status->Resource Usage, the current CPU usage and Memory usage are shown in the pie chart. Figure 75: System Status->Resource Usage Firmware Version 1.0.0.
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SYSTEM EVENTS The UCM6510 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. Click on to configure the parameters for each event. 1. Disk Usage Figure 76: System Events->Alert Events Lists: Disk Usage  Detect Cycle: The UCM6510 will perform the internal disk usage detection based on this cycle.
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3. Memory Usage Figure 78: System Events->Alert Events Lists: Memory Usage  Detect Cycle: The UCM6510 will perform the memory 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 UCM6510 system will send the alert. 4.
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6. System Crash Figure 81: System Events->Alert Events Lists: System Crash  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.
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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 UCM6510.  Start Time. Format: 2013-03-27 16:47:03.  Call From. Format: "John Doe"<6012>.  Call To. Format: 6005.  Call Time. Format: 0:00:10.  Talk Time. Format: 0:00:10  Status. Format: NO ANSWER, BUSY, ANSWERED, or FAILED.  Options.
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any internal extension. Caller Number Enter the caller number to be filtered in the CDR report. Caller Name Enter the caller name to be filtered in the CDR report. From Date Specify "From" date and time to be filtered for the CDR report. Click on the field and the calendar will show for users to select the exact date and time. To Date Specify "To" date and time to be filtered for the CDR report. Click on the field and the calendar will show for users to select the exact date and time.
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Click on on to play the recording file; click on to download the recording file in .wav format; click to delete the recording file (the call record entry will not be deleted). Figure 85: Call Report Entry With Audio Recording File 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.
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 Source Channel, Dest Channel Sample 1: Figure 87: Downloaded CDR File Sample - Source Channel and Dest Channel 1 DAHDI means it is an analog call, FXO or FXS. For UCM6510, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports. For UCM6510, DAHDI/(1-4) are FXO ports, and DAHDI(5-6) are FXS ports. For UCM6510, DAHDI/(1-8) are FXO ports, and DAHDI(9-10) are FXS ports. For UCM6510, DAHDI/(1-16) are FXO ports, and DAHDI/(17-18) are FXS ports.
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abnormal cases. Playback: play some prompts to you, such as 183 response or run into an IVR. ReadExten: collect numbers from user. It may occur when you input PIN codes or run into DISA STATISTICS CDR Statistics is an additional feature on the UCM6510 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.
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RECORDING FILES The recording files recorded by "Auto Record" per extension/per trunk, or via feature code "Audio Mix Record" are listed here. Users could click on recording file in .wav format; or click on to play the recording file; click on to download the 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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The format of the HTTPS request for the CDR API is as below. https://[UCM IP]:[Port]/cdrapi?[option1]=[value]&[option2]=[value]&... By default, the port number for the API is 8443. The options included in the request URI control the record matching and output format. For CDR matching parameters, all non-empty parameters must have a match to return a record. Parameters can appear in the URI in any order. Multiple values given for caller or callee will be concatenated.
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Date and/or time of day in any of startTime the following formats: Filters based on the start (call start time) value. Calls YYYY-MM-DDTHH:MM which start within this period (inclusive of boundaries) YYYY-MM-DDTHH:MM:SS YYYY-MM-DDTHH:MM:SS.SSS will match, regardless of the call answer or end time. An empty value for either field will be interpreted as range with no minimum or maximum respectively.
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Query 2: Request all records of calls placed on extension 5300 or in the range 6300-6399 to extensions starting with 5, with results in XML format. https://192.168.254.200:8088/cdrapi?format=XML&caller=5300,6300-6399&callee=5@ -ORhttps://192.168.254.200:8088/cdrapi?cdrapi?format=XML&caller=5300&caller=6300-6399&callee=5@ Query 3: Request all records of calls placed on extensions containing substring "53" prior to January 23, 2013 00:00:00 UTC to extensions 5300-5309, with results in CSV format.
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Note:  Disallowed characters in the caller, callee, startTime, or endTime strings, and non-digit characters in the values of numRecords, offset, minDur, or maxDur, will result in no records returned - the appropriate container/header for the output format will be the only output. If the format parameter is in error, the CSV header will be used. Error messages will appear in the Asterisk log (along with errors stemming from failed database connections, etc.).
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XML: 6253005301from-internal"pn01" <5300>SIP/5300-00000000SIP/5301-00000001DialSIP/5301,60,2013-12-03 11:46:402013-12-03 11:46:432013-12-03 11:46:4996ANSWEREDDOCUMENTATION
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UPGRADING AND MAINTENANCE UPGRADING The UCM6510 can be upgraded to a new firmware version remotely or locally. This section describes how to upgrade your UCM6510 via network or local upload. UPGRADING VIA NETWORK The UCM6510 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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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 UCM6510. Firmware File Suffix If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the UCM6510. HTTP/HTTPS User Name The user name for the HTTP/HTTPS server. HTTP/HTTPS Password The password for the HTTP/HTTPS server.
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Figure 93: Upgrading Firmware Files  Wait until the upgrading process is successful and a window will be popped up in the web GUI. Figure 94: Reboot UCM6510  Click on "OK" to reboot the UCM6510 and check the firmware version after it boots up. Firmware Version 1.0.0.
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Note: Please do not interrupt or power cycle the UCM6510 during upgrading process. NO LOCAL FIRMWARE SERVERS For users that would like to use remote upgrading without a local TFTP server, Grandstream offers a NAT-friendly HTTP server. This enables users to download the latest software upgrades for their devices via this server. Please refer to the webpage: http://www.grandstream.com/support/firmware. Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade.
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LOCAL BACKUP Users could backup the UCM6510 configurations for restore purpose under web GUI->Maintenance->Backup->Local Backup. Before creating new backup file, select the backup option first.  If the Config-File is selected only, the backup file will be saved in the flash of the UCM6510.  If Voice-File, Voicemail-File, Voice-Records, CDR or VFAX is selected, external storage devices (USB Flash drive or SD Card) will be required because the backup file might be too large.
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Figure 96: Data Sync Table 69: Data Sync Configuration Enable Backup Enable the auto backup function. The default setting is "No". Account Enter the Account name on the SFTP backup server. Password Enter the Password associate with the Account on the SFTP backup server. Server Address Enter the SFTP server address. Backup Time Enter 0-23 to specify the backup hour of the day. Before saving the configuration, users could click on "Test Connection".
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be displayed in the list of previous configuration backups for restore purpose. Click on to restore from the backup file. Figure 97: Restore UCM6510 From Backup File Note:  The uploaded backup file must be a tar file with no special characters like *,!,#,@,&,$,%,^,(,),/,\,space in the file name.  The uploaded back file size must be under 10MB. CLEANER Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under web GUI->Maintenance->Cleaner.
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Figure 98: Cleaner Table 70: 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. VR Clean Threshold Specify the Voice Records threshold from 0 to 99 by using local storage status in percentage.
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Figure 99: Reset and Reboot SYSLOG On the UCM6510, 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 UCM6510 settings because it can be helpful to locate the issues when errors happen.
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ETHERNET CAPTURE The captured trace can be downloaded for analysis. Also the instructions or result will be displayed in the web GUI output result. Figure 100: 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.
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Figure 101: PING 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 102: Traceroute PRI SIGNALING TRACE Please see section [DIGITAL TRUNK TROUBLESHOOTING]. Firmware Version 1.0.0.
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EXPERIENCING THE UCM6510 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 Compliance Statement: This device complies with part 15 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. Important: Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment.