Figure 59: Voicemail Group ....................................................................................................................... 139
Figure 60: Ring Group ............................................................................................................................... 141
Figure 61: Ring Group Configuration ........................................................................................................ 142
Figure 62: Paging/Intercom Group ............................................................................................................ 143
Figure 63: Page/Intercom Group Settings ................................................................................................ 144
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.
Thank you for purchasing Grandstream UCM6510 IP PBX appliance. The UCM6510 is an innovative IP
PBX appliance for E1/T1/J1 networks that brings enterprise-grade unified communications and security
protection to enterprises, small-to-medium businesses (SMBs), retail environments and residential
settings in an easy-to-manage fashion. Powered by an advanced hardware platform and revolutionary
software functionalities, the UCM6510 offers a breakthrough turnkey solution for converged voice, video,
data, fax, security surveillance, and mobility applications out of the box without any extra license fees or
recurring costs.
Caution:
Changes or modifications to this product not expressly approved by Grandstream, or operation of this
product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
Warning:
Please do not use a different power adapter with the UCM6510 as it may cause damage to the products
and void the manufacturer warranty.
This document is subject to change without notice. The latest electronic version of this user manual is
available for download here:
http://www.grandstream.com/support
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for
any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.
and dedicated high performance multi-core DSP array for advanced voice processing
1 Integrated 1 T1/E1/J1 interface, 2 PSTN trunk FXO ports, 2 analog telephone/Fax FXS ports with
lifeline capability in case of power outage, and up to 50 SIP trunk accounts
Hardware DSP based 128ms-tail-length carrier-grade line echo cancellation (LEC), hardware based
caller ID/call progress tone and smart automated impedance matching for various countries
Gigabit network port(s) with integrated PoE, USB, SD card; integrated NAT router with advanced QoS
support
Strong defense against malicious attacks (Fail2ban, Whitelist, Blacklist, alerts, etc.)
Data communication via T1/E1/J1 and data-voice combined communication via T1/E1/J1 with SS7 or
PRI
Supports up to 2000 SIP endpoint registrations, up to 200 concurrent calls (up to 100 SRTP
encrypted concurrent calls), and up to 64 conference attendees
Flexible dial plan, call routing, site peering, call recording (manual and automatic per SIP call and SIP
trunk), central control panel for endpoints, integrated NTP server, and integrated LDAP contact
directory
Automated detection and provisioning of IP phones, video phones, ATAs, gateways, SIP cameras,
and other endpoints for easy deployment
Strongest-possible security protection using SRTP, TLS, and HTTPS with hardware encryption
accelerator
Redundant power supply, advanced support for Hot Standby Clustering and High Availability to
minimize system down time (pending)
Automatic export of previous day’s data; periodically cleans up user data
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TECHNICAL SPECIFICATIONS
Interfaces
Analog Telephone FXS Ports
2 RJ11 ports (both with lifetime capability in case of power outage)
PSTN Line FXO Ports
2 RJ11 ports (both with lifeline capability in case of power outage)
T1/E1/J1 Interface
1 RJ45 port
Network Interfaces
Dual Gigabit ports (switched or routed) with PoE;
A 3rd Gigabit port for Hot-Standby Clustering
128x32 dot matrix graphic LCD with DOWN and OK buttons
Reset Switch
Yes, long press for factory reset and short press for reboot
Voice/Video Capabilities
Voice-over-Packet
Capabilities
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
TFTP/HTTP/HTTPS, auto-discovery & auto-provisioning of Grandstream
IP endpoints via ZeroConfig (DHCP Option 66/multicast SIP
SUBSCRIBE/mDNS), eventlist between local and remote trunks
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.
EQUIPMENT PACKAGING
Table 2: UCM6510 Equipment Packaging
CONNECT YOUR UCM6510
CONNECT THE UCM6510
Figure 1: UCM6510 Front View
Figure 2: UCM6510 Back View
Follow the steps below to connect the UCM6510 for initial setup:
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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. (Connect the second
power adapter into the DC 12V power jack 2 for failover purpose in case the first one is down).
3. Wait for the UCM6510 to boot up. The LCD in the front will show its hardware information when the
bootup process is done.
4. Once the UCM6510 is successfully connected to the network, the LED indicator for the WAN port in
the front will be in solid green and the LCD shows up the IP address.
Depending on how the UCM6510 is used, users can follow the steps below for optional setup:
1. PSTN Line Connection: connect PSTN lines from the wall jack to the UCM6510 LINE ports (FXO
ports).
2. Analog Line Connection: connect analog lines (phone and fax) to the PHONE ports (FXS ports).
3. T1/E1 Line Connection: connect one end of the T1/E1 cable provided from the service provider into
the T1/E1 port of the UCM6510; connect the other end into the T1/E1 wall jack.
SAFETY COMPLIANCES
The UCM6510 series IP PBX complies with FCC/CE and various safety standards. The UCM6510 power
adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM6510
package only. The manufacturer’s warranty does not cover damages to the device caused by
unsupported power adapters.
WARRANTY
If the UCM6510 series IP PBX was purchased from a reseller, please contact the company where the
device was purchased for replacement, repair or refund. If the device was purchased directly from
Grandstream Networks, contact our Technical Support Team for a RMA (Return Materials Authorization)
number before the product is returned. Grandstream Networks reserves the right to remedy warranty
policy without prior notification.
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.
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.
This section provides step-by-step instructions on how to use the LCD menu, LED indicators and web
GUI of the UCM6510. Once the basic settings are done, users could start making calls from UCM6510
extension registered on a SIP phone as described at the end of this section.
USE THE LCD MENU
Default LCD Display
By default, when the device is powered up, the LCD will show device model (e.g., UCM6510),
hardware version (e.g., V1.2A) and IP address. Press "Down" button and the system time will be
displayed (e.g., 2014-05-15 14:20).
Menu Access
Press "OK" button to start browsing menu options. Please see menu options in [Table 3: LCD Menu
Options].
Menu Navigation
Press the "Down" arrow key to browser different menu options. Press the "OK" button to select an
entry.
Exit
If "Back" option is available in the menu, select it to go back to the previous menu. For "Device Info"
"Network Info" and "Web Info" which do not have "Back" option, simply press the "OK" button to go
back to the previous menu. Additionally, the LCD will display default idle screen after staying in menu
option for 15 seconds.
LCD Backlight
The LCD backlight will be on upon key pressing. The backlight will go off after the LCD stays in idle
for 30 seconds.
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The following table shows the LCD menu options.
View Events
Critical Events
Other Events
Device Info
Hardware: Hardware version number
Software: Software version number
P/N: Part number
WAN MAC: WAN side MAC address
LAN MAC: LAN side MAC address
Uptime: System up time since the last reboot
Network Info
WAN Mode: DHCP, Static IP, or PPPoE
WAN IP: IP address
WAN Subnet Mask
LAN IP: IP address
LAN Subnet Mask
Network Menu
WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE
Static Routes Reset: Click to reset the static route setting
Factory Menu
Reboot
Factory Reset
LCD Test Patterns
Press "OK" to start. Then press "Down" button to test different LCD
patterns. When done, press "OK" button to exit.
Fan Mode
Select "Auto" or "On".
LED Test Patterns
Select "All On" "All Off" or "Blinking" and check LED status for USB, SD,
T1/E1, Phone 1/Phone 2, Line 1/Line 2 ports. After the LED test, select
"Back" in the menu and the device will show the LED actual status again.
RTC Test Patterns
Select "2022-02-22 22:22" or "2011-01-11 11:11" to start the RTC (RealTime Clock) test pattern. Check the system time from LCD idle screen by
pressing "DOWN" button, or from web GUI->System Status->General
page. After the test, reboot the device manually and the device will display
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
LED Indicator
LED Status
Power 1/Power 2
PoE
LAN
WAN
USB
SD
Phone 1 /Phone 2 (FXS)
Line 1/Line 2 FXO
Solid: Connected Fast Blinking: Data Transferring Slow Blinking: Trying to connect
OFF: Not Connected
T1/E1
Solid: Connected and working Fast Blinking (0.5s on/0.5s off): Connected
The UCM6510 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML
pages allow users to configure the device through a Web browser such as Microsoft IE (version 8+),
Mozilla Firefox, Google Chrome and etc.
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. The default
administrator's username and password is "admin" and "admin". It is highly recommended to change
the default password after login for the first time.
By default, the UCM6510 has "Redirect From Port 80" enabled. Therefore, if users type in the UCM6510
IP address in the web browser, the web page will be automatically redirected to the page using HTTPS
and port 8089.
For example, if the LCD shows 192.168.40.167, please enter 192.168.40.167 in your web browser and
the web page will be redirected to:
https://192.168.40.167:8089
The option "Redirect From Port 80" can be configured under the UCM6510 web GUI->Settings->HTTP Server.
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.
Users can select the displayed language in web GUI login page, or at the upper right of the web GUI after
logging in.
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.
Power up the UCM6510 and your SIP end point phone. Connect both devices to the network. Then follow
the steps below to make your first call.
1. Log in the UCM6510 web GUI, go to PBX->Basic/Call Routes->Extensions.
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. Once successfully logged in to the voicemail, you will be prompted with the Voice Mail Main menu.
6. You are successfully connected to the PBX system now.
Select "Route", "Switch" or "Dual" mode on the network interface of UCM6510.
The default setting is "Route".
Route
WAN port interface will be used for uplink connection. LAN port interface will
be used to serve as router.
Switch
WAN port interface will be used for uplink connection. LAN port interface will
be used as bridge for PC connection.
Dual
Both ports can be used for uplink connection. Users will need assign LAN 1 or
LAN 2 as the default interface in option "Default Interface" and configure
"Gateway IP" for this interface if static IP is used for the interface.
Preferred DNS Server
Enter the preferred DNS server address.
WAN (when "Method" is set to "Route")
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask
Enter the subnet mask address for static IP settings. The default setting is
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 log in the web
GUI and go to Settings->Network Settings to configure the network parameters for the device. Select
each tab in web GUI->Settings->Network Settings page to configure LAN/WAN settings, 802.1X and
Port Forwarding.
BASIC SETTINGS
Please refer to the following tables for basic network configuration parameters on the UCM6510.
Table 5: UCM6510 Network Settings->Basic Settings
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Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
DNS Server 1
Enter the DNS server 1 address for static IP settings. The default setting is
0.0.0.0.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
User Name
Enter the user name to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS
802.1Q/VLAN Tag
Assign the VLAN tag of the layer 2 QoS packets for WAN port. The default value
is 0.
Layer 2 QoS 802.1p
Priority Value
Assign the priority value of the layer 2 QoS packets for WAN port. The default
value is 0.
LAN (when Method is set to "Route")
IP Address
Enter the IP address assigned to LAN port. The default setting is 192.168.2.1.
Subnet Mask
Enter the subnet mask. The default setting is 255.255.255.0.
DHCP Server Enable
Enable or disable DHCP server capability. The default setting is "Yes".
DNS Server 1
Enter DNS server address 1. The default setting is 8.8.8.8.
DNS Server 2
Enter DNS server address 2. The default setting is 208.67.222.222.
Allow IP Address From
Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.
Allow IP Address To
Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.
Default IP Lease Time
Enter the IP lease time (in seconds). The default setting is 43200.
LAN (when Method is set to "Switch")
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask
Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
DNS Server 1
Enter the DNS server 1 address for static IP settings. The default setting is
0.0.0.0.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
User Name
Enter the user name to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS
802.1Q/VLAN Tag
Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is
0.
Layer 2 QoS 802.1p
Priority Value
Assign the priority value of the layer 2 QoS packets for LAN port. The default
value is 0.
LAN 1 / LAN 2 (when Method is set to "Dual")
Default Interface
If "Dual" is selected as "Method", users will need assign the default interface to be
LAN 1 (mapped to UCM6510 WAN port) or LAN 2 (mapped to UCM6510 LAN
port) and then configure network settings for LAN 1 and LAN 2. The default
interface is LAN 2.
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask
Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
Gateway IP
Enter the gateway IP address for static IP settings when the port is assigned as
default interface. The default setting is 0.0.0.0.
DNS Server 1
Enter the DNS server 1 address for static IP settings. The default setting is
0.0.0.0.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
User Name
Enter the user name to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS
802.1Q/VLAN Tag
Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is
0.
Layer 2 QoS 802.1p
Priority Value
Assign the priority value of the layer 2 QoS packets for LAN port. The default
value is 0.
Both WAN port and LAN port are used for uplink connection. WAN port will be mapped to LAN 1
interface; LAN port will be mapped to LAN 2 interface. Users will need assign LAN 1 or LAN 2 as the
default interface in option "Default Interface" and configure "Gateway IP" if static IP is used for this
interface.
IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication
mechanism to device before the device is allowed to access Internet or other LAN resources. The
UCM6510 supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure
show UCM6510 uses 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet.
The following table shows the configuration parameters for 802.1X on UCM6510. Identity and MD5
password are required for authentication, which should be provided by the network administrator obtained
from the RADIUS server. If “EAP-TLS” or“EAP-PEAPv0/MSCHAPv2” is used as the 802.1X mode, users
will also need upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated
from the RADIUS server.
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Table 6: UCM6510 Network Settings->802.1X
802.1X Mode
Select 802.1X mode. The default setting is "Disable". The supported 802.1X
mode are:
EAP-MD5
EAP-TLS
EAP-PEAPv0/MSCHAPv2
Identity
Enter 802.1X mode identity information.
MD5 Password
Enter 802.1X mode MD5 password information.
802.1X CA Certificate
Select 802.1X certificate from local PC and then upload.
802.1X Client
Certificate
Select 802.1X client certificate from local PC and then upload.
Destination
Configure the destination IP address or the destination IP subnet for the
UCM6510 to reach using the static route.
A static route is a pre-determined path that the network traffic travels to reach a specific host or network.
On the UCM6510, the static route function allows the device to use manually configured routes, rather
than dynamically assigned routes or default gateway configured in the UCM6510 web GUI->Network Settings->Basic Settings to forward traffic. It can be used to define a route when no other routes are
available or necessary, or used in complementary with existing routing on the UCM6510 as a failover
backup, and etc.
Click on to create a new static route. The configuration parameters are listed
in the table below.
Once added, users can select to edit the static route.
Select to delete the static route.
Static routes configuration can be reset from LCD menu->Network Menu.
Table 7: UCM6510 Network Settings->Static Routes
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IP address - 192.168.66.4
IP subnet - 192.168.66.0
Netmask
Configure the subnet mask for the above destination address. If left blank, the
default value is 255.255.255.255.
Example:
255.255.255.0
Gateway
Configure the gateway address so that the UCM6510 can reach the destination
via this gateway. Gateway address is optional.
Example:
192.168.40.5
Interface
Specify the network interface "LAN" or "WAN" on the UCM6150 to reach the
destination using the static route.
The following diagram shows a sample application of static route usage on UCM6510.
Figure 11: UCM6510 Static Route Sample
The network topology of the above diagram is as below:
Network 192.168.69.0 has IP phones registered to UCM6510 LAN 1 address
Network 192.168.40.0 has IP phones registered to UCM6510 LAN 2 address
Network 192.168.66.0 has IP phones registered to UCM6510 via VPN
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Network 192.168.40.0 has VPN connection established with network 192.168.66.0
WAN Port
Specify the WAN port number. Up to 8 ports can be configured.
LAN IP
Specify the LAN IP address.
LAN Port
Specify the LAN port number.
Protocol Type
Select protocol type "UDP Only", "TCP Only" or "TCP/UDP" for the forwarding in
the selected port. The default setting is "UDP Only".
In this network, by default the IP phones in network 192.168.69.0 are unable to call IP phones in network
192.168.66.0 when registered on different interfaces on the UCM6510. Therefore, we need configure a
static route on the UCM6510 so that the phones in isolated networks can make calls between each other.
Figure 12: UCM6510 Static Route Configuration
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.
network 192.168.2.x. The UCM6510 is used as a router, with gateway address 192.168.2.1
There is a GXP2160 connected under the LAN interface network of the UCM6510. It obtains IP
address 192.168.2.100 from UCM6510 DHCP pool
On the UCM6510 web UI->Settings->Network Settings->Port Forwarding, configure a port
forwarding entry as the figure shows below.
WAN Port: This is the port opened up on the WAN side for access purpose.
LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM6510.
Protocol Type: We select TCP here for web UI access using HTTP.
Figure 13: UCM6510 Port Forwarding Configuration
This will allow users to access the GXP2160 web UI from public side, by typing in address
“96.31.248.8:8088”.
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Figure 14: GXP2160 Web Access Using UCM6510 Port Forwarding
The UCM6510 provides users firewall configurations to prevent certain malicious attack to the UCM6510
system. Users could configure to allow, restrict or reject specific traffic through the device for security and
bandwidth purpose. The UCM6510 also provides Fail2ban feature for authentication errors in SIP
REGISTER, INVITE and SUBSCRIBE.
To configure firewall settings in UCM6510, go to web GUI->Settings->Firewall page.
STATIC DEFENSE
Under web GUI->Settings->Firewall->Static Defense page, users will see the following information:
Current service information with port, process and type.
Typical firewall settings.
Custom firewall settings.
The following table shows a sample current service status running on the UCM6510.
Table 9: UCM6510 Firewall->Static Defense->Current Service
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22
Dropbear
tcp/IPv4
80
Lighthttpd
tcp/IPv4
8089
Lighthttpd
tcp/IPv4
69
Opentftpd
udp/IPv4
9090
Asterisk
udp/IPv4
6060
zero_config
udp/IPv4
5060
Asterisk
udp/IPv4
4569
Asterisk
udp/IPv4
5353
zero_config
udp/IPv4
37435
Syslogd
udp/IPv4
Ping Defense
Enable
If enabled, ICMP response will not be allowed for Ping request. The default
setting is disabled. To enable or disable it, click on the check box for the LAN
or WAN interface.
SYN-Flood Defense
Enable
Enable to prevent SYN Flood denial-of-service attack to the device. The
default setting is disabled. To enable or disable it, click on the check box for the
LAN or WAN interface.
Ping-of-Death
Defense Enable
Enable to prevent Ping-of-Death attack to the device. The default setting is
disabled. To enable or disable it, click on the check box for the LAN or WAN
interface.
For typical firewall settings, users could configure the following options on the UCM6510.
Table 10: Typical Firewall Settings
Under "Custom Firewall Settings", users could create new rules to accept, reject or drop certain traffic
going through the UCM6510. To create new rule, click on "Create New Rule" button and a new window
will pop up for users to specify rule options.
The following figure shows a firewall rule example that will deny SSH access for the UCM6510 from WAN
side.
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Rule Name
Specify the Firewall rule name to identify the firewall rule.
Action
Select the action for the Firewall to perform.
ACCEPT
REJECT
DROP
Type
Select the traffic type.
IN
If selected, users will need specify the network interface "LAN", "WAN" or
"Both" for the incoming traffic.
If selected, users will need specify Source (IP and port), Destination (IP
and port) and Protocol (TCP, UDP or Both) for the service. Please note if
the source or the destination field is left blank, it will be used as
"Anywhere".
The new rule will be listed at the bottom of the page with sequence number, rule name, action, protocol,
Dynamic Defense
Enable
Enable dynamic defense. The default setting is disabled.
Periodical Time
Interval
Configure the dynamic defense periodic time interval (in minutes). If the
number of TCP connections from a host exceeds the “Connection Threshold”
within this period, this host will be added into Blacklist. The valid value is
between 1 and 59 when dynamic defense is turned on. The default setting is
59.
Blacklist Update
Interval
Configure the blacklist update time interval (in seconds). The default setting is
120. This defines how long the IP will be blocked once added into the
UCM6510 blacklist. For example, if it’s set to 300 seconds, the blocked IP
address will only be able to establish TCP connection with the UCM6510 again
after 300 seconds.
Connection
Threshold
Configure the connection threshold. Once the number of connections from the
same host reaches the threshold during “Periodical Time Interval”, it will be
added into the blacklist. The default setting is 100.
Dynamic Defense
Whitelist
Configure the dynamic defense whitelist. This is a list of IPs that will not be
blocked by the UCM6510.
type, source, destination and operation. Users can click on to edit the rule, or click on to delete the
rule. Save the change and reboot the device for the configuration to take effect.
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 via TCP connection
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.
Table 12: UCM6510 Firewall Dynamic Defense
The following figure shows a configuration example like this:
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If a host at IP address 192.168.40.7 initiates more than 20 TCP connections to the UCM6510 within 1
Global Settings
Enable Fail2Ban
Enable Fail2Ban. The default setting is disabled. Please make sure both "Enable
Fail2Ban" and "Asterisk Service" are turned on in order to use Fail2Ban for SIP
authentication on the UCM6510.
Banned Duration
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.
Max Retry Duration
Within this duration (in seconds), if a host exceeds the max times of retry as
defined in "MaxRetry", the host will be banned. The default setting is 5.
This host 192.168.40.7 will be blocked by the UCM6510 for 300 seconds.
Since IP address 192.168.40.5 is in whitelist, if the host at IP address 192.168.40.5 initiates more
than 20 TCP connections to the UCM6510 within 1 minute, it will not be added into UCM6510
blacklist. It can still establish TCP connection with the UCM6510.
Figure 16: Configure Dynamic Defense
FAIL2BAN
Fail2Ban feature on the UCM6510 provides intrusion detection and prevention for authentication errors in
SIP REGISTER, INVITE and SUBSCRIBE. Once the entry is detected within "Max Retry Duration", the
UCM6510 will take action to forbid the host for certain period as defined in "Banned Duration". This
feature helps prevent SIP brute force attacks to the PBX system.
Table 13: Fail2Ban Settings
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MaxRetry
Configure the number of authentication failures during "Max Retry Duration"
before the host is banned. The default setting is 10.
Fail2Ban Whitelist
Configure IP address, CIDR mask or DNS host in the whiltelist. Fail2Ban will not
ban the host with matching address in this list. Up to 5 addresses can be added
into the list.
Local Settings
Asterisk Service
Enable Asterisk service for Fail2Ban. The default setting is disabled. Please make
sure both "Enable Fail2Ban" and "Asterisk Service" are turned on in order to use
Fail2Ban for SIP authentication on the UCM6510.
Port
Configure the listening port number for the service. Currently only 5060 (for UDP)
is supported.
MaxRetry
Configure the number of authentication failures during "Max Retry Duration"
before the host is banned. The default setting is 10. Please make sure this option
is properly configured as it will override the "MaxRetry" value under "Global
Settings".
After logging in the web GUI for the first time, it is highly recommended for users to change the default
password "admin" to a more complicated password for security purpose. Follow the steps below to
change the web GUI access password.
1. Go to web GUI->Settings->Change Password page.
2. Enter the old password first.
3. Enter the new password and retype the new password to confirm. The new password has to be at
least 4 characters. The maximum length of the password is 16 characters.
4. Click on "Save" and the user will be automatically logged out.
5. Once the web page comes back to the login page again, enter the username "admin" and the new
password to login.
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.
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Users could add new phonebook with a different Phonebook DN for other external contacts. For
All the phonebooks in the UCM6510 LDAP server have the same Base DN "dc=pbx,dc=com".
If users have the Grandstream phone provisioned by the UCM6510, the LDAP directory has been set up
on the phone and can be used right away for users to access all phonebooks.
Additionally, users could manually configure the LDAP client settings to manipulate the built-in LDAP
server on the UCM6510. If the UCM6510 has multiple LDAP phonebooks created, in the LDAP client
configuration, users could use "dc=pbx,dc=com" as Base DN to have access to all phonebooks on the
UCM6510 LDAP server, or use a specific phonebook DN, for example "ou=people,dc=pbx,dc=com", to
access to phonebook with Phonebook DN "ou=people,dc=pbx,dc=com " only.
To access LDAP Server settings, go to web GUI->Settings->LDAP Server.
LDAP SERVER CONFIGURATIONS
The following figure shows the default LDAP server configurations on the UCM6510.
Figure 17: LDAP Server Configurations
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.
Users could use the default phonebook, edit the default phonebook as well as add new phonebook on the
LDAP server. The first phonebook with default phonebook dn "ou=pbx,dc=pbx,dc=com" displayed on the
LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly. The
contacts information will need to be modified via web GUI->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.
Configure the "Phonebook Prefix" first. The "Phonebook DN" will be automatically filled in. For example, if
configuring "Phonebook Prefix" as "people", the "Phonebook DN" will be filled with
"ou=people,dc=pbx,dc=com".
Once added, users can select to edit the phonebook attributes and contact list (see figure below), or
select to delete the phonebook.
Figure 21: Edit LDAP Phonebook
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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Assuming the server base dn is "dc=pbx,dc=com", configure the LDAP clients as follows (case
Base DN: dc=pbx,dc=com
Login DN: Please leave this field empty
Password: Please leave this field empty
Anonymous: Please enable this option
Filter: (|(CallerIDName=%)(AccountNumber=%))
Port: 389
To configure Grandstream IP phones as the LDAP client, please refer to the following example:
Server Address: The IP address or domain name of the UCM6510
Base DN: dc=pbx,dc=com
User Name: Please leave this field empty
Password: Please leave this field empty
LDAP Name Attribute: CallerIDName Email Department FirstName LastName
LDAP Number Attribute: AccountNumber MobileNumber HomeNumber Fax
LDAP Number Filter: (AccountNumber=%)
LDAP Name Filter: (CallerIDName=%)
LDAP Display Name: AccountNumber CallerIDName
LDAP Version: If existed, please select LDAP Version 3
Port: 389
The following figure shows the configuration information on a Grandstream GXP2200 to successfully use
the LDAP server as configured in [Figure 17: LDAP Server Configurations].
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). Users could also change the access protocol and port as preferred under
web GUI->Settings->HTTP Server.
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Redirect From Port 80
Enable or disable redirect from port 80. On the PBX, the default access
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
Select HTTP or HTTPS as the protocol to access the HTTP server. The
default setting is "HTTPS". This also defines whether to use HTTP or
HTTPS to download the config file in zero config as the UCM6510 is
served as HTTP/HTTPS server that has the device config files for zero
config.
Port
Specify port number to access the HTTP server. The default port number
is 8089.
TLS Enable
Enable or disable TLS during transferring/submitting your Email to other
SMTP server. The default setting is "Yes".
Type
MTA: Mail Transfer Agent. The Email will be sent from the configured
domain. When MTA is selected, there is no need to set up SMTP
server for it or no user login is required. However, the Emails sent
from MTA might be considered as spam by the target SMTP server.
Client: Submit Emails to the SMTP server. A SMTP server is required
and users need login with correct credentials.
Domain
Specify the domain name to be used in the Email when using type "MTA".
Server
Specify the SMTP server when using type "Client". For example, if using
Gmail as the SMTP server, you can configure it as smtp.gmail.com:465.
Username
Username is required when using type "Client". Normally it's the Email
address.
Once the change is saved, the web page will be redirected to the login page using the new URL. Enter
the username and password to login again.
EMAIL SETTINGS
The Email application on the UCM6510 can be used to send out alert event Emails, Fax (Fax-To-Email),
Voicemail (Voicemail-To-Email) and etc. The configuration parameters can be accessed via web GUI>Settings->Email Settings.
Table 15: Email Settings
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Password
Password to log in for the above Username (Email address) is required
when using type "Client".
Display Name
Specify the display name in the FROM header in the Email.
Sender
Specify the sender's Email address.
For example, pbx@example.mycompany.com.
The following figure shows a sample Email settings on the UCM6510, assuming the Email is using
smtp.gmail.com as the SMTP server and the port number is 465.
Figure 23: UCM6510 Email Settings
Once the configuration is finished, click on "Save" first. Then click on "Test" button to make sure the Email
setting is working.
The following figure shows the new dialog prompted to test the Email setting. Fill in a valid Email address
to send a test Email to verify the Email settings on the UCM6510.
Figure 24: UCM6510 Email Settings: Send Test Email
AUTO TIME UPDATING
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.
Note:
The configurations under Web GUI->Settings->Time Settings->Time Auto Updating page require
reboot to take effect. Please consider configuring auto time updating related changes when setting up the
UCM6510 for the first time to avoid service interrupt after installation and deployment in production.
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Table 16: Auto Time Updating
Remote NTP Server
Specify the URL or IP address of the NTP server for the UCM6510 to
synchronize the date and time. The default NTP server is
ntp.ipvideotalk.com.
Enable DHCP Option 2
If set to "Yes", the UCM6510 is allowed to get provisioned for Time Zone
from DHCP Option 2 in the local server automatically. The default setting
is "Yes".
Enable DHCP Option 42
If set to "Yes", the UCM6510 is allowed to get provisioned for NTP Server
from DHCP Option 42 in the local server automatically. This will override
the manually configured NTP Server. The default setting is "Yes".
Time Zone
Select the proper time zone option so the UCM6510 can display correct
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.
Self-Defined Time Zone
If "Self-Defined Time Zone" is selected in "Time Zone" option, users will
need define their own time zone following the format below.
The syntax is: std offset dst [offset], start [/time], end [/time]
Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0
MTZ+6MDT+5
This indicates a time zone with 6 hours offset and 1 hour ahead for DST,
which is U.S central time. If it is positive (+), the local time zone is west of
the Prime Meridian (A.K.A: International or Greenwich Meridian); If it is
negative (-), the local time zone is east.
M4.1.0,M11.1.0
The 1st number indicates Month: 1, 2, 3..., 12 (for Jan, Feb...Dec.).
The 2nd number indicates the nth iteration of the weekday: (1st Sunday,
3rd Tuesday…). Normally 1, 2, 3, 4 are used. If 5 is used, it means the
last iteration of the weekday.
The 3rd number indicates weekday: 0, 1, 2…6 (for Sun, Mon, Tues...
Sat).
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.
To manually set the time on the UCM6510, go to Web GUI->Settings->Time Settings->Set Time
Manually. The format is YYYY-MM-DD HH:MI:SS.
Figure 25: Set Time Manually
Note:
Manually setup time will take effect immediately after saving and applying change in the web UI. If users
would like to reboot the UCM6510 and keep the manually setup time setting, please make sure "Remote
NTP Server", "Enable DHCP Option 2" and "Enable DHCP Option 42" options under Web GUI->Settings>Time Settings->Time Auto Updating page are unchecked or set to empty. Otherwise, time auto
updating settings in this page will take effect after reboot.
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.
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. This allows users to finish the installation with
ease and start using the SIP devices in a managed way.
To provision a phone, three steps are involved, i.e., discovery, assignment and provisioning. The
UCM6510 creates XML config file to the detected/assigned Grandstream device and accomplishes the
following configurations on the device after the provisioning:
A UCM6510 extension will be assigned and registered on the phone.
SIP-related network settings such as "NAT traversal" and "Use Random Port" are configured on the
phone.
Call feature settings such as "Public Mode", "Voicemail User ID", "Dial Plan" and "Auto Answer".
LDAP client configurations will be set up automatically on the phone to use the default LDAP directory
generated in the UCM6510 LDAP server.
This section explains how zero config works on the UCM6510. The settings for this feature can be
accessed via web GUI->PBX->Basic/Call Routes->Zero Config.
AUTO PROVISIONING
By default, the Zero Config feature is enabled on the UCM6510 for auto provisioning. Three methods of
auto provisioning are used.
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.
DHCP OPTION 66
This method should be used only when the UCM6510 is set to "Route" mode under web GUI>Settings->Network Settings->Basic Settings: Method. When the phone restarts (by default DHCP
Option 66 is turned on), it will send out a DHCP DISCOVER request. The UCM6510 receives it and
returns DHCP OFFER with the config server path URL in the Option 66, for example,
https://192.168.2.1:8089/zccgi/. The phone will then use the path to download the config file
generated in the UCM6510.
mDNS
When the phone boots up, it sends out mDNS query to get the TFTP server address. The UCM6510
will respond with its own address. The phone will then send TFTP request to download the XML
config file from the UCM6510.
To start the auto provisioning process, under web GUI->PBX->Basic/Call Routes->Zero Config, click on
"Auto Provision Settings" and fill in the auto provision information.
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Enable Zero Config
Enable or disable the zero config feature on the PBX. The default setting
is enabled.
Automatically Assign Extension
If enabled, when the device is discovered, the PBX will automatically
assign an extension within the range defined in "Zero Config Extension
Segment" to the device. The default setting is disabled.
Zero Config Extension
Segment
Click on the link "Zero Config Extension Segment" to specify the
extension range to be assigned if "Automatically Assign Extension" is
enabled. The default range is 5000-6299. Zero Config Extension
Segment range can be defined in web UI->PBX->Internal Options>General page->Extension Preference section: "Auto Provision
Extensions".
Enable Pick Extension
If enabled, the extension list will be sent out to the device after receiving
the device's request. This feature is for the GXP series phones that
support selecting extension to be provisioned via phone's LCD. The
default setting is disabled.
Pick Extension Segment
Click on the link "Pick Extension Segment" to specify the extension list to
be sent to the device. The default range is 4000 to 4999. Pick Extension
Segment range can be defined in web UI->PBX->Internal Options-
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. Three methods are supported to scan the devices.
PING
ARP
SIP Message (NOTIFY)
Click on "Auto Discover", fill in the "Scan Method" and "Scan IP". The IP address segment will be
automatically filled in based on the network mask detected on the UCM6510. If users need scan the
entire network segment, enter 255 (for example, 192.168.40.255) instead of a specific IP address. Then
click on "Save" to start discovering the devices within the same network. To successfully discover the
devices, "Zero Config" needs to be enabled on the UCM6510 web GUI->PBX->Basic/Call Routes->Zero
Config->Auto Provisioning Settings.
Figure 28: Auto Discover
The following figure shows a list of discovered phones. The MAC address, IP Address, Extension (if
assigned), Version, Vendor, Model, Connection Status, Create Config, Options (Edit/Delete/Update) are
displayed in the list.
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 30: Assign Extension To Device
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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 icon for the entry in the zero config device
list.
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. Click on "Save" to add the device to the provision list.
Figure 31: Create New Device
PROVISIONING
After the successful discovery and assignment configuration on the UCM6510, the device will start
downloading the config file and take the new configuration with the extension registered.
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EXTENSIONS
General
Extension
The extension number associated with the user. This is the SIP UserID
for registration.
CallerID Number
Configure the CallerID Number that would be applied for outgoing calls
from this user.
Note:
The ability to manipulate your outbound Caller ID may be limited by your
VoIP provider.
Permission
Assign permission level to the user. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level. The default setting is "Internal".
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege in order to make outbound calls using this rule. If the
outbound rule privilege is disabled, this option will not take effect.
SIP/IAX Password
Configure the password for the user. A random secure password will be
automatically generated when the extension is created. It is
recommended to use this password or other strong password for security
purpose.
Enable Voicemail
Enable voicemail for the user so that the call will be forwarded to the
user’s voicemail if there is no answer or the call is rejected. The default
setting is "Yes".
Voicemail Password
Configure voicemail password (digits only) for the user to access the
voicemail box. A random numeric password is automatically generated
when the extension is created. It is recommended to use the random
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 18: SIP Extension Configuration Parameters
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generated password for security purpose.
Call Forward Unconditional
Configure the Call Forward Unconditional target number so that the
incoming call to this extension will be always forwarded to the target
number. If not configured, the Call Forward Unconditional feature is
deactivated. The default setting is deactivated.
Call Forward No Answer
Configure the Call Forward No Answer target number so that the
incoming call to this extension will be forwarded to the target number if
the call is not answered until the ringing times out. If not configured, the
Call Forward No Answer feature is deactivated. The default setting is
deactivated.
Call Forward Busy
Configure the Call Forward Busy target number so that the incoming call
to this extension will be forwarded to the target number if the call is
rejected or the extension is in talking/busy status. If not configured, the
Call Forward Busy feature is deactivated. The default setting is
deactivated.
Ring Timeout
Configure the number of seconds to ring the user before the call is
forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the
UCM6510, which can be configured in the global ring timeout setting
under web GUI->Internal Options: General Preference. The valid range
is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default
setting is disabled. The recording files will be saved in external storage if
plugged in and can be accessed under web GUI->CDR->Recording Files.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
Support Hot-Desking Mode
This mode can be used for devices that support hot-desking feature. For
example, the GXP21xx series phones support hot-desking feature by
turning on “Public Mode”. On the device, users can log in and log out
using the SIP UserID and password. If enabled on the UCM6510, the SIP
Password for the extension will accept only alphabet characters and
digits; AuthID will also be changed to the same as Extension.
Configure the first name of the user. The first name can contain
characters, letters, digits and _.
Last Name
Configure the last name of the user. The last name can contain
characters, letters, digits and _.
Email Address
Fill in the Email address for the user. Voicemail will be sent to this Email
address.
Language
Select the voice prompt language to be used for this extension. The
default setting is "Default" which is the selected voice prompt language
under web GUI->PBX->Internal Options->Language. The dropdown list
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.
SIP Settings
NAT
Use NAT when the UCM6510 is on a public IP communicating with
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.
Can Reinvite
By default, the UCM6510 will route the media steams from SIP endpoints
through itself. If enabled, the PBX will attempt to negotiate with the
endpoints to route the media stream directly. It is not always possible for
the UCM6510 to negotiate endpoint-to-endpoint media routing. The
default setting is "No".
DTMF Mode
Select DTMF mode for the user to send DTMF. The default setting is
"RFC2833". If "Info" is selected, SIP INFO message will be used. If
"Inband" is selected, 64-kbit PCMU and PCMA are required. When "Auto"
is selected, RFC2833 will be used if offered, otherwise "Inband" will be
used.
Insecure
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
required.
No: Normal IP-based peers matching and authentication of incoming
INVITE.
The default setting is "Port".
Enable Keep-alive
If enabled, empty SDP packet will be sent to the SIP server periodically to
keep the NAT port open. The default setting is "Yes".
Configure the Keep-alive interval (in seconds) to check if the host is up.
The default setting is 60 seconds.
Auth ID
Configure the authentication ID for the user. If not configured, the
extension number will be used for authentication.
Other Settings
SRTP
Enable SRTP for the call. The default setting is disabled.
Fax Detection
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
default Email address in Fax setting page under UCM6510 web GUI>PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
Strategy
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
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".
Codec Preference
Select audio and video codec for the extension. The available codecs are:
PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, ILBC,
ADPCM, H.264, H.263 and H.263p. In the selected codec list, users can
click on UP or DOWN arrow to adjust the order for the codec priority.
To manually create new IAX user, go to web GUI->PBX->Basic/Call Routes->Extensions. Click on
"Create New User"->"Create New IAX Extension" and a new dialog window will show for users to fill in
the extension information. The configuration parameters are as follows.
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Table 19: IAX Extension Configuration Parameters
General
Extension
The extension number associated with the user.
CallerID Number
Configure the CallerID Number that would be applied for outgoing calls
from this user.
Note:
The ability to manipulate your outbound Caller ID may be limited by your
VoIP provider.
Permission
Assign permission level to the user. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level. The default setting is "Internal".
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege in order to make outbound calls using this rule. If the
outbound rule privilege is disabled, this option will not take effect.
SIP/IAX Password
Configure the password for the user. A random secure password will be
automatically generated when the extension is created. It is
recommended to use this password or other strong password for security
purpose.
Enable Voicemail
Enable voicemail for the user. The default setting is "Yes".
Voicemail Password
Configure voicemail password (digits only) for the user to access the
voicemail box. A random numeric password is automatically generated. It
is recommended to use the random generated password for security
purpose.
Call Forward Unconditional
Configure the Call Forward Unconditional target number so that the
incoming call to this extension will be always forwarded to the target
number. If not configured, the Call Forward Unconditional feature is
deactivated. The default setting is deactivated.
Call Forward No Answer
Configure the Call Forward No Answer target number so that the
incoming call to this extension will be forwarded to the target number if
the call is not answered until the ringing times out. If not configured, the
Call Forward No Answer feature is deactivated. The default setting is
deactivated.
Call Forward Busy
Configure the Call Forward Busy target number so that the incoming call
to this extension will be forwarded to the target number if the call is
rejected or the extension is in talking/busy status. If not configured, the
Call Forward Busy feature is deactivated. The default setting is
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
under web GUI->Internal Options: General Preference. The valid range
is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default
setting is disabled. The recording files will be saved in external storage if
plugged in and can be accessed under web GUI->CDR->Recording Files.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
User Settings
First Name
Configure the first name of the user. The first name can contain
characters, letters, digits and _.
Last Name
Configure the last name of the user. The last name can contain
characters, letters, digits and _.
Email Address
Fill in the Email address for the user. Voicemail will be sent to this Email
address.
Language
Select the voice prompt language to be used for this extension. The
default setting is "Default" which is the selected voice prompt language
under web GUI->PBX->Internal Options->Language. The dropdown list
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.
IAX Settings
Max Number of Calls
Configure the maximum number of calls allowed for each remote IP
address.
Require Call Token
Configure to enable/disable requiring call token. If set to "Auto", it might
lock out users who depend on backward compatibility when peer
authentication credentials are shared between physical endpoints. The
default setting is "Yes".
Enable SRTP for the call. The default setting is disabled.
Fax Detection
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
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.
Strategy
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
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".
Codec Preference
Select audio and video codec for the extension. The available codecs are:
PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, ILBC,
ADPCM, H.264, H.263 and H.263p. In the selected codec list, users can
click on UP or DOWN arrow to adjust the order for the codec priority.
To manually create new FXS user, go to web GUI->PBX->Basic/Call Routes->Extensions. Click on
"Create New User"->"Create New FXS Extension" and a new dialog window will show for users to fill in
the extension information. The configuration parameters are as follows.
Table 20: FXS Extension Configuration Parameters
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Analog Station
Select the FXS port to be assigned for this extension.
CallerID Number
Configure the CallerID Number that would be applied for outgoing calls
from this user.
Note:
The ability to manipulate your outbound Caller ID may be limited by your
VoIP provider.
Permission
Assign permission level to the user. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level. The default setting is "Internal".
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege in order to make outbound calls using this rule. If the
outbound rule privilege is disabled, this option will not take effect.
Enable Voicemail
Enable voicemail for the user so that the call will be forwarded to the
user’s voicemail if there is no answer or the call is rejected. The default
setting is "Yes".
Voicemail Password
Configure voicemail password (digits only) for the user to access the
voicemail box. A random numeric password is automatically generated
when the extension is created. It is recommended to use the random
generated password for security purpose.
Call Forward Unconditional
Configure the Call Forward Unconditional target number so that the
incoming call to this extension will be always forwarded to the target
number. If not configured, the Call Forward Unconditional feature is
deactivated. The default setting is deactivated.
Call Forward No Answer
Configure the Call Forward No Answer target number so that the
incoming call to this extension will be forwarded to the target number if
the call is not answered until the ringing times out. If not configured, the
Call Forward No Answer feature is deactivated. The default setting is
deactivated.
Call Forward Busy
Configure the Call Forward Busy target number so that the incoming call
to this extension will be forwarded to the target number if the call is
rejected or the extension is in talking/busy status. If not configured, the
Call Forward Busy feature is deactivated. The default setting is
deactivated.
Ring Timeout
Configure the number of seconds to ring the user before the call is
forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the
UCM6510, which can be configured in the global ring timeout setting
under web GUI->Internal Options: General Preference. The valid range
is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default
setting is disabled. The recording files will be saved in external storage if
plugged in and can be accessed under web GUI->CDR->Recording Files.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
User Settings
First Name
Configure the first name of the user. The first name can contain
characters, letters, digits and _.
Last Name
Configure the last name of the user. The last name can contain
characters, letters, digits and _.
Email Address
Fill in the Email address for the user. Voicemail will be sent to this Email
address.
Language
Select the voice prompt language to be used for this extension. The
default setting is "Default" which is the selected voice prompt language
under web GUI->PBX->Internal Options->Language. The dropdown list
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
Configure to enable/disable call waiting feature for the FXS extension.
When enabled, the FXS extension currently in an active call allows a new
call to come in and can hear call waiting tone on the new incoming call.
The default setting is "No".
User # as SEND
If configured, the # key can be used as SEND key after dialing the
number on the analog phone. The default setting is "Yes".
RX Gain
Configure the RX gain for the receiving channel of analog FXS port. The
valid range is -30dB to +6dB. The default setting is 0.
TX Gain
Configure the TX gain for the transmitting channel of analog FXS port.
The valid range is -30dB to +6dB. The default setting is 0.
MIN RX Flash
Configure the minimum period of time (in milliseconds) that the hook-flash
must remain unpressed for the PBX to consider the event as a valid flash
event. The valid range is 30ms to 1000ms. The default setting is 200ms.
MAX RX Flash
Configure the maximum period of time (in milliseconds) that the hookflash must remain unpressed for the PBX to consider the event as a valid
flash event. The minimum period of time is 256ms and it can't be
modified. The default setting is 1250ms.
Enable Polarity Reversal
If enabled, a polarity reversal will be marked as received when an
outgoing call is answered by the remote party. For some countries, a
polarity reversal is used for signaling the disconnection of a phone line
and the call will be considered as hangup on a polarity reversal. The
default setting is "Yes".
Echo Cancellation
Specify "ON", "OFF" or a value (the power of 2) from 32 to 1024 as the
number of taps of cancellation.
Note:
When configuring the number of taps, the number 256 is not translated
into 256ms of echo cancellation. Instead, 256 taps means 256/8 = 32 ms.
The default setting is "ON", which is 128 taps.
3-Way Calling
Configure to enable/disable 3-way calling feature on the user. The default
setting is enabled. For example, when enabled, if the FXS extension has
established a call with User A -> Press “Flash” to open a new line -> FXS
extension calls User B -> Press flash again, it will establish 3-way
conference call with User A and User B.
Send CallerID After
Configure the number of rings before sending CID. The default setting is
1.
Other Settings
Fax Detection
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
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.
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".
Configure the starting extension number of the batch of extensions to be
added.
Create Number
Specify the number of extensions to be added. The default setting is 5.
Permission
Assign permission level to the user. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level. The default setting is "Internal".
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege in order to make outbound calls using this rule. If the
outbound rule privilege is disabled, this option will not take effect.
Enable Voicemail
Enable Voicemail for the user. The default setting is "Yes".
SIP/IAX Password
Configure the SIP/IAX password for the users. Three options are
available to create password for the batch of extensions.
User Random Password.
A random secure password will be automatically generated. It is
recommended to use this password for security purpose.
Enter a password to be used on all the extensions in the batch.
Voicemail Password
Configure Voicemail password (digits only) for the users.
User Random Password.
A random password in digits will be automatically generated. It is
recommended to use this password for security purpose.
Enter a password to be used on all the extensions in the batch.
Ring Timeout
Configure the number of seconds to ring the user before the call is
forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the
UCM6510, which can be configured in the global ring timeout setting
Under web GUI->PBX->Basic/Call Routes->Extensions, click on "Batch Add Extensions"->"Batch Add
SIP Extensions".
Table 21: Batch Add SIP Extension Parameters
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under web GUI->Internal Options: General Preference. The valid range
is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default
setting is disabled. The recording files will be saved in external storage if
plugged in and can be accessed under web GUI->CDR->Recording Files.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
SIP Settings
NAT
Use NAT when the PBX is on a public IP communicating with devices
hidden behind NAT (e.g., broadband router). If there is one-way audio
issue, usually it's related to NAT configuration or Firewall's support of SIP
and RTP ports. The default setting is enabled.
Can Reinvite
By default, the PBX will route the media steams from SIP endpoints
through itself. If enabled, the PBX will attempt to negotiate with the
endpoints to route the media stream directly. It is not always possible for
the PBX to negotiate endpoint-to-endpoint media routing. The default
setting is "No".
DTMF Mode
Select DTMF mode for the user to send DTMF. The default setting is
"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.
Insecure
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
required.
No: Normal IP-based peers matching and authentication of incoming
INVITE.
The default setting is "Port".
Enable Keep-alive
If enabled, empty SDP packet will be sent to the SIP server periodically to
keep the NAT port open. The default setting is "Yes".
Configure the number of seconds for the host to be up for Keep-alive. The
default setting is 60 seconds.
Other Settings
SRTP
Enable SRTP for the call. The default setting is "No".
Fax Detection
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
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.
Strategy
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
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".
Codec Preference
Select audio and video codec for the extension. The available codecs are:
PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, ILBC,
ADPCM, LPC10, H.264, H.263 and H.263p. In the selected codec list,
users can click on UP or DOWN arrow to adjust the order for the codec
priority.
Under web GUI->PBX->Basic/Call Routes->Extensions, click on "Batch Add Extensions"->"Batch Add
IAX Extensions".
Table 22: Batch Add IAX Extension Parameters
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General
Start Extension
Configure the starting extension number of the batch of extensions to be
added.
Create Number
Specify the number of extensions to be added. The default setting is 5.
Permission
Assign permission level to the user. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level. The default setting is "Internal".
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege in order to make outbound calls using this rule. If the
outbound rule privilege is disabled, this option will not take effect.
Enable Voicemail
Enable Voicemail for the user. The default setting is "Yes".
SIP/IAX Password
Configure the SIP/IAX password for the users. Three options are
available to create password for the batch of extensions.
User Random Password.
A random secure password will be automatically generated. It is
recommended to use this password for security purpose.
Enter a password to be used on all the extensions in the batch.
Voicemail Password
Configure Voicemail password (digits only) for the users.
User Random Password.
A random password in digits will be automatically generated. It is
recommended to use this password for security purpose.
Enter a password to be used on all the extensions in the batch.
Ring Timeout
Configure the number of seconds to ring the user before the call is
forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the
UCM6510, which can be configured in the global ring timeout setting
under web GUI->Internal Options: General Preference. The valid range
is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default
setting is disabled. The recording files will be saved in external storage if
plugged in and can be accessed under web GUI->CDR->Recording Files.
Skip Voicemail Password
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
IAX Settings
Max Number of Calls
Configure the maximum number of calls allowed for each remote IP
address.
Require Call Token
Configure to enable/disable requiring call token. If set to "Auto", it might
lock out users who depend on backward compatibility when peer
authentication credentials are shared between physical endpoints. The
default setting is "Yes".
Other Settings
SRTP
Enable SRTP for the call. The default setting is "No".
Fax Detection
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
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.
Strategy
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
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".
Codec Preference
Select audio and video codec for the extension. The available codecs are:
PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, ILBC,
ADPCM, LPC10, H.264, H.263 and H.263p. In the selected codec list,
users can click on UP or DOWN arrow to adjust the order for the codec
priority.
All the UCM6510 extensions are listed under web GUI->PBX->Basic/Call Routes->Extensions, with
status, Extension, CallerID Name, Technology (SIP, IAX and FXS), IP and Port. Each extension has a
checkbox for users to "Modify Selected Extensions" or "Delete Selected Extensions". Also, options
"Edit", "Reboot" and "Delete" are available per extension.
Status
Users can see the following icon for each extension to indicate the SIP status.
Green: Free
Blue: Ringing
Yellow: In Use
Grey: Unavailable
Edit single extension
Click on to start editing the extension parameters.
Reboot the user
Click on to send NOTIFY reboot event to the device which has an UCM6510 extension already
registered. To successfully reboot the user, "Zero Config" needs to be enabled on the UCM6510 web
The extensions configured on the UCM6510 can be exported to csv format file with selected technology
"SIP", "IAX" or "FXS". Click on "Export Extensions" button and select technology in the prompt.
Figure 32: Export Extensions
The exported csv file can also serve as a template for users to fill in desired extension information to be
imported to the UCM6510.
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 33: Export Extensions
4. Select the option in "On Duplicate Extension" to define how the duplicate extension(s) in the imported
csv file should be treated by the PBX.
Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension
information as previously configured without change.
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Delete and Recreate: The current extension previously configured will be deleted and the
duplicate extension in the csv file will be loaded to the PBX.
Update Information: The current extension previously configured in the PBX will be kept.
However, if the duplicate extension in the csv file has different configuration for any options, it will
override the configuration for those options in the extension.
5. Click on to select csv file from local directory in the PC for uploading.
6. Click on "Save" to import the csv file.
7. Click on "Apply Changes" to apply the imported file on the UCM6510.
EMAIL TO USER
Once the extensions are created with Email address, the PBX administrator can click on button “Email To
User” to send the registration and configuration information for this account to the user who is going to
register and use this extension. Please make sure Email setting under web UI->Settings->Email
Settings is properly configured and tested on the UCM6510 before using “Email To User”.
When click on “Email To User” button, the following message will be prompted in the web page. Click on
OK to confirm sending the account information to all users’ Email addresses.
Figure 34: Email To User: Prompt Information
The user will receive Email including account registration information and LDAP configuration. A QR code
is also generated for the text information.
Specify a unique label to identify the trunk when listed in outbound routes,
inbound routes and etc.
Advanced Options
Enable Polarity Reversal
If enabled, a polarity reversal will be marked as received when an
outgoing call is answered by the remote party. For some countries, a
polarity reversal is used for signaling the disconnection of a phone line
and the call will be considered as "hangup" on a polarity reversal. The
default setting is "No".
Polarity on Answer Delay
When FXO port answers the call, FXS may send a Polarity Reversal. If
this interval is shorter than the value of "Polarity on Answer Delay", the
Polarity Reversal will be ignored. Otherwise, the FXO will onhook to
disconnect the call. The default setting is 600ms.
Current Disconnect Threshold
(ms)
This is the periodic time (in ms) that the UCM6510 will use to check on a
voltage drop in the line. The default setting is 200. The valid range is 50
to 3000.
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.
The analog trunk options are listed in the table below.
Table 23: Analog Trunk Configuration Parameters
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Ring Timeout
Configure the ring timeout (in ms). Trunk (FXO) devices must have a
timeout to determine if there was a hangup before the line is answered.
This value can be used to configure how long it takes before the
UCM6510 considers a non-ringing line with hangup activity. The default
setting is 8000.
RX Gain
Configure the RX gain for the receiving channel of analog FXO port. The
valid range is from -13.5 (dB) to + 12.0 (dB). The default setting is 0.
TX Gain
Configure the TX gain for the transmitting channel of analog FXO port.
The valid range is from -13.5 (dB) to + 12.0 (dB). The default setting is 0.
Use CallerID
Configure to enable CallerID detection. The default setting is "Yes".
Fax Detection
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. The default setting is "No".
Note:
If enabled, Fax Pass-through cannot be used.
Caller ID Scheme
Select the Caller ID scheme for this trunk. If you are not sure which
scheme to choose, please select “Auto Detect”. The default setting is
"Bellcore/Telcordia".
Auto Record
Enable automatic recording for the calls using this trunk. The default
setting is disabled. The recording files are saved in external storage
device if plugged in and can be accessed under web GUI->CDR>Recording Files.
Tone Settings
Busy Detection
Busy Detection is used to detect far end hangup or for detecting busy
signal. The default setting is "Yes".
Busy Tone Count
If "Busy Detection" is enabled, users can specify the number of busy
tones to be played before hanging up. The default setting is 2. Better
results might be achieved if set to 4, 6 or even 8. Please note that the
higher the number is, the more time is needed to hangup the channel.
However, this might lower the probability to get random hangup.
Congestion Detection
Congestion detection is used to detect far end congestion signal. The
default setting is "Yes".
Congestion Count
If "Congestion Detection" is enabled, users can specify the number of
congestion tones to wait for. The default setting is 2.
Tone Country
Select the country for tone settings. If "Custom" is selected, users could
manually configure the values for Busy Tone and Congestion Tone. The
default setting is "United States of America (USA)".
Syntax:
f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];
Frequencies are in Hz and cadence on and off are in ms.
Frequencies Range: [0, 4000)
Busy Level Range: (-300, 0)
Cadence Range: [0, 16383].
Select Tone Country "Custom" to manually configure Busy Tone value.
Default value:
f1=480@-50,f2=620@-50,c=500/500
Congestion Tone
Syntax:
f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];
Frequencies are in Hz and cadence on and off are in ms.
Frequencies Range: [0, 4000)
Busy Level Range: (-300, 0)
Cadence Range: [0, 16383].
Select Tone Country "Custom" to manually configure Busy Tone value.
Default value:
f1=480@-50,f2=620@-50,c=250/250
PSTN Detection
Click on "Detect" to detect the busy tone, Polarity Reversal and Current
Disconnect by PSTN. Before the detecting, please make sure there are
more than one channel configured and working properly. If the detection
has busy tone, the "Tone Country" option will be set as "Custom".
The UCM6510 provides PSTN detection function to help users detect the busy tone, Polarity Reversal
and Current Disconnect by making a call from the PSTN line to another destination. The detecting call will
be answered and up for about 1 minute. Once done, the detecting result will show and can be used for
the UCM6510 settings.
1. Go to UCM6510 web GUI->PBX->Basic/Call Routes->Analog Trunks page.
2. Click to edit the analog trunk created for the FXO port.
3. In the dialog window to edit the analog trunk, go to "Tone Settings" section and click on for
“PSTN Detection”.
If there are two FXO ports connected to PSTN lines, use the following settings for auto-detection.
Detect Model: Auto Detect.
Source Channel: The source channel to be detected.
Destination Channel: The channel to help detecting. For example, the second FXO port.
Destination Number: The number to be dialed for detecting. This number must be the actual
PSTN number for the FXO port used as the destination channel.
Detect Model: Semi-auto Detect.
Source Channel: The source channel to be detected.
Destination Number: The number to be dialed for detecting. This number could be a cell phone
number or other PSTN number that can be reached from the source channel PSTN number.
5. Click "Detect" to start detecting. The source channel will initiate a call to the destination number. For
"Auto Detect", the call will be automatically answered. For "Semi-auto Detect", the UCM6510 web
GUI will display prompt to notify the user to answer or hang up the call to finish the detecting process.
6. Once done, the detected result will show. Users could save the detecting result as the current
UCM6510 settings.
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Detect Model
Select "Auto Detect" or "Semi-auto Detect" for PSTN detection.
Auto Detect
Please make sure two or more channels are connected to the
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.
Semi-auto Detect
Semi-auto detection requires answering or hanging up the call
manually. Please make sure one channel is connected to the
UCM6510 and in idle status before starting the detection. During the
detection, source channel will be used as caller and send the call to
the configured Destination Number. Users will then need follow the
prompts in web GUI to help finish the detection.
The default setting is "Auto Detect".
Source Channel
Select the channel to be detected.
Destination Channel
Select the channel to help detect when "Auto Detect" is used.
Destination Number
Configure the number to be called to help the detection.
The PSTN detection process will keep the call up for about 1 minute.
If "Semi-auto Detect' is used, please pick up the call only after informed from the web GUI
prompt.
Once the detection is successful, the detected parameters "Busy Tone", "Polarity Reversal" and
"Current Disconnect by PSTN" will be filled into the corresponding fields in the analog trunk
configuration.
The analog hardware (FXS port and FXO port) on the UCM6510 can be configured under web GUI-
>PBX->Ports Config->Analog Hardware. Click on to edit signaling preference for FXS port or
configure ACIM settings for FXO port.
Select "Loop Start" or "Kewl Start" for each FXS port. And then click on "Update" to save the change.
Figure 41: 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.
Figure 42: FXO Ports ACIM Settings
Note:
ACIM setting is very important for the FXO/PSTN line to work properly on the UCM6510. If the users
experience echo, caller ID or disconnecting issue, please make sure to run the ACIM detection to find out
the correct value for impedance setting.
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Table 25: Analog Hardware Configuration Parameters
Tone Region
Select country to set the default tones for dial tone, busy tone, ring tone
and etc to be sent from the FXS port. The default setting is "United States
of America (USA)".
Advanced Settings
FXO Opermode
Select country to set the On Hook Speed, Ringer Impedance, Ringer
Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum
Operational Loop Current, and AC Impedance as predefined for your
country's analog line characteristics. The default setting is "United States
of America (USA)".
FXS Opermode
Select country to set the On Hook Speed, Ringer Impedance, Ringer
Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum
Operational Loop Current, and AC Impedance as predefined for your
country's analog line characteristics. The default setting is "United States
of America (USA)".
FXS TISS Override
Configure to enable or disable override Two-Wire Impedance Synthesis
(TISS). The default setting is No.
If enabled, users can select the impedance value for Two-Wire
Impedance Synthesis (TISS) override. The default setting is 600Ω.
PCMA Override
Select the codec to be used for analog lines. North American users
should choose PCMU. All other countries, unless already known, should
be assumed to be PCMA. The default setting is PCMU.
Note:
This option requires system reboot to take effect.
Boost Ringer
Configure whether normal ringing voltage (40V) or maximum ringing
voltage (89V) for analog phones attached to the FXS port is required. The
default setting is "Normal".
Fast Ringer
Configure to increase the ringing speed to 25HZ. This option can be used
with "Low Power" option. The default setting is "Normal".
Low Power
Configure the peak voltage up to 50V during "Fast Ringer" operation. This
option is used with "Fast Ringer". The default setting is "Normal".
Ring Detect
If set to "Full Wave", false ring detection will be prevented for lines where
Caller ID is sent before the first ring and proceeded by a polarity reversal,
as in UK. The default setting is "Standard".
FXS MWI Mode
Configure the type of Message Waiting Indicator on FXS lines. The
default setting is "FSK".
The UCM6510 supports E1/T1/J1 which are physical connection technology used in digital network. T1 is
the North American standard, J1 is used in Japan, whereas E1 is the European standard.
UCM6510 supports three signaling protocols: PRI, MFC/R2 and SS7. PRI provides a varying number of
channels depending on the standards in the country of implementation (E1, T1 or J1); MFC/R2 is a
signaling protocol heavily used over E1 trunks; SS7 uses out-of-band signaling, which travels on a
separate, dedicated channel rather than within the same channel as the telephone call, providing more
efficiency and higher security level when the telephone calls are set up.
To set up digital trunk on the UCM6510:
1. Go to web UI->PBX->Ports Config->Digital Hardware to configure port type and channels.
2. Go to web UI->PBX->Basic/Call Routes->Digital Trunks to add and edit digit trunk.
3. Go to web UI->PBX->Basic/Call Routes->Outbound Routes and Inbound Routes to configure
outbound and inbound rule for the digital trunk.
DIGITAL HARDWARE CONFIGURATION
Go to web GUI->PBX->Ports Config->Digital Hardware page and configure the following:
Figure 43: Digital Hardware Configuration
Step 1: Click on to edit digital ports. Please see configuration parameters in the tables below.
Step 2: Click on to edit group. This assigns channels to be used for the digital port. For E1, 30 B
channels can be assigned to the default group; for T1/J1, 23 B channels can be assigned to the
default group.
Step 3: If fewer than 30 B channels for E1 or 23 B channels for T1/J1 are assigned in default group,
users can click on to add more groups. This is not necessary in most cases and only default
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group is needed.
Basic Settings
Clock
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is
used as the master, the PBX system clock will synchronize to it.
Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.
Slave: The equipment at the far end of the E1/T1/J1 link is the
preferred source of the master clock.
LBO
The line build-out (LBO) is the distance between the operators and the
PBX. Please use the default value 0dB unless the distance is long.
RX Gain
Configure the RX gain for the receiving channel of digital port. The valid
range is from -24dB to +12dB.
TX Gain
Configure the TX Gain for the transmitting channel of digital port. The
valid range is -24dB to +12dB.
Codec
Select alaw or ulaw. If set to default, alaw will be used for E1.
Play Local RBT
This configured whether to play the ringback tone from local UCM6510 or
not. If enabled, the local UCM6510 will play ringback tone to the caller.
Currently, the group configuration in digit trunks settings is to manage outbound routes only. It doesn't
control inbound routes. Therefore, if the users have configured multiple groups for the digital trunk, please
make sure the inbound routes for those groups have the same inbound rule configured. Otherwise,
inbound call using the digital trunk might not work properly.
The UCM6510 currently supports E1, T1 and J1 digital hardware type. When different signaling is
selected for E1, T1 or J1, the settings in basic options and advanced options will be different. The
following tables list all the settings to configure digital ports when selecting each signaling.
Table 26: Digital Hardware Configuration Parameters: E1 - PRI_NET/PRI_CPE
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Otherwise, the caller will listen to the tone from peer device. The default
setting is disabled.
Advanced Settings
Switch Type
Select switch type.
EuroISDN: EuroISDN (common in Europe)
NI2: National ISDN type 2 (common in the US)
DMS100: Nortel DMS100
4ESS: AT&T 4ESS
5ESS: Lucent 5ESS
NI1: old national ISDN type 1
Q.SIG
Coding
Select "HDB3" or "AMI".
CRC
Select whether to use CRC4 or not.
PRI Dial Plan
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".
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.
PRI Local Dial Plan
This setting is used to specify the type of the caller number. The service
provider will usually verify this.
International Prefix
National Prefix
Local Prefix
Private Prefix
Unknown Prefix
Configure the prefix in PRI Local Dial Plan and PRI Dial Plan for each
type.
PRI Indication
Select the PRI Indication.
outofband: Use RELEASE, DISCONNECT or other messages with
CAUSE to indicate call progress (e.g., cause: unassigned number or
user busy).
inband: use in-band tones to play busy or congestion signal to the
other side. This is the default setting.
Reset Interval
The interval that restarts idle channels.
PRI Exclusive
This setting is used to set up the ChannelID in SETUP message. If
enabled, only the specified B channel can be used. Otherwise, select one
of the channels in B channel. If you need override the existing channels
selection routine and force all PRI channels to be marked as exclusively
selected, please enable it.
Facility Enable
If selected, transmission of facility-based ISDN supplementary services
(such as caller name from CPE over facility) will be enabled.
NSF
Some switches (AT&T especially) require network specific facility.
Currently the supported values are "none", "sdn", "megacom",
"tollfreemegacom", "accunet".
Basic Settings
Clock
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is
used as the master, the PBX system clock will synchronize to it.
Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.
Slave: The equipment at the far end of the E1/T1 link is the preferred
source of the master clock.
SS7 Variant
Select ITU, ANSI or CHINA.
Originating Point Code
Originating point code is used to identify the node originating the
message, always provided by the operator/ISP.
ITU Format: decimal number.
ANSI & CHINA Format: decimal number or XXX-XXX-XXX.
Destination Point Code
Destination point code is the address to send the message to, always be
provided by the operator/ISP.
ITU Format: decimal number.
ANSI & CHINA Format: decimal number or XXX-XXX-XXX.
Network Indicator
Network Indicator (NI) should match in nodes, otherwise it might cause
issues. Users can select "National", "National Spare", "International", or
"International Spare". Usually "National" or "International" is used.
LBO
The line build-out (LBO) is the distance between the operators and the
PBX. Please use the default value 0dB unless the distance is long.
RX Gain
Configure the RX gain for the receiving channel of digital port. The valid
range is from -24dB to +12dB.
TX Gain
Configure the TX Gain for the transmitting channel of digital port. The
valid range is -24dB to +12dB.
Table 27: Digital Hardware Configuration Parameters: E1 - SS7
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Codec
Select alaw or ulaw. If set to default, alaw will be used for E1.
Advanced Settings
Coding
Select "HDB3" or "AMI".
CRC
Select whether to use CRC4 or not.
Called Nature of Address
Indicator
Indicates the type of the called number. The receiving switch may use this
indicator during translations to apply the number’s proper dial plan. Users
can select "Unknown", "Subscriber", "National", "International" or
"Dynamic".
Calling Nature of Address
Indicator
Indicates the type of the calling number. The receiving switch may use
this indicator during translations to apply the number’s proper dial plan.
Users can select "Unknown", "Subscriber", "National", "International" or
"Dynamic".
International Prefix
National Prefix
Local Prefix
Private Prefix
Unknown Prefix
Configure the prefix in PRI Local Dial Plan and PRI Dial Plan for each
type.
Basic Settings
Clock
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is
used as the master, the PBX system clock will synchronize to it.
Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.
Slave: The equipment at the far end of the E1/T1 link is the preferred
source of the master clock.
Variant
MFC/R2 multinational adaption. UCM6510 supports MFC/R2 standards
by ITU and MFC/R2 standards in different countries or regions including
Argentina, Brazil, China, Czech Republic, Colombia, Ecuador, Indonesia,
Mexico, the Philippines and Venezuela.
Get ANI First
If enabled, the callee side will request the caller to send caller number
first and then called number.
Note:
Options "Get ANI First" and "Skip Category" cannot be enabled at the
Table 28: Digital Hardware Configuration Parameters: E1 - MFC/R2
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same time.
Category
Select the category of the caller. UCM6510 supports four categories:
National Subscriber, National Priority Subscriber, International Subscriber
and International Priority Subscriber.
LBO
The line build-out (LBO) is the distance between the operators and the
PBX. Please use the default value 0dB unless the distance is long.
RX Gain
Configure the RX gain for the receiving channel of digital port. The valid
range is from -24dB to +12dB.
TX Gain
Configure the TX Gain for the transmitting channel of digital port. The
valid range is -24dB to +12dB.
Play Local RBT
This configured whether to play the ringback tone from local UCM6510 or
not. If enabled, the local UCM6510 will play ringback tone to the caller.
Otherwise, the caller will listen to the tone from peer device. The default
setting is disabled.
Advanced Settings
Coding
Select "HDB3" or "AMI".
CRC
Select whether to use CRC4 or not.
MF Back Timeout(ms)
MFC/R2 value in milliseconds for MF timeout. Values smaller than 500ms
are not recommended. -1 represents default value.
Metering Pulse Timeout(ms)
MFC/R2 value in milliseconds for the metering pulse timeout. Metering
pulse is sent by some telcos for some R2 variants during a call
presumably for billing purposes to indicate costs. Should not last more
than 500ms, -1 represents default value, and for Argentina the default
value is 400ms, for others is 0ms.
Alllow Collect Calls
Brazil has a special calling party category for collect calls (llamadas por
cobrar) instead of using the operator (as in Mexico). The R2 spec in Brazil
says a special GB tone should be used to reject collect calls.
By default, this is disabled, which means collect calls will be blocked.
Double Answer
Some PBXs require a double-answer process to block collect calls. If
users have problem blocking collect calls using Group B signals, please
try enabling this option.
Accept On Offer
By default it’s enabled. In most of cases, this option should be enabled.
Skip Category
If enabled, the callee side will request the caller to send caller category
before sending caller number.
Note:
"Get ANI First" and "Skip Category" cannot be enabled at the same time.
Whether or not report to the other end "accept call with charge". This
setting has no effect with most telecos. The default setting is enabled
(recommended).
Basic Settings
Clock
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is
used as the master, the PBX system clock will synchronize to it.
Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.
Slave: The equipment at the far end of the E1/T1/J1 link is the
preferred source of the master clock.
LBO
The line build-out (LBO) is the distance between the operators and the
PBX. Please use the default value 0dB unless the distance is long.
RX Gain
Configure the RX gain for the receiving channel of digital port. The valid
range is from -24dB to +12dB.
TX Gain
Configure the TX Gain for the transmitting channel of digital port. The
valid range is -24dB to +12dB.
Codec
Select alaw or ulaw. If set to default, ulaw will be used for T1/J1.
Play Local RBT
This configured whether to play the ringback tone from local UCM6510 or
not. If enabled, the local UCM6510 will play ringback tone to the caller.
Otherwise, the caller will listen to the tone from peer device. The default
setting is disabled.
Advanced Settings
Switch Type
Select switch type.
EuroISDN: EuroISDN (common in Europe)
NI2: National ISDN type 2 (common in the US)
DMS100: Nortel DMS100
4ESS: AT&T 4ESS
5ESS: Lucent 5ESS
NI1: old national ISDN type 1
Q.SIG
Coding
Select "B8ZS" or "AMI".
PRI Dial Plan
This setting is used to specify the type of the callee number. The service
Table 29: Digital Hardware Configuration Parameters: T1/J1 - PRI_NET/PRI_CPE
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provider will usually verify this. The default setting is "unknown". In some
very unusual circumstances, you may need set to "Dynamic" or
"Redundant".
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.
PRI Local Dial Plan
This setting is used to specify the type of the caller number. The service
provider will usually verify this.
International Prefix
National Prefix
Local Prefix
Private Prefix
Unknown Prefix
Configure the prefix in PRI Local Dial Plan and PRI Dial Plan for each
type.
PRI Indication
Select the PRI Indication.
outofband: Use RELEASE, DISCONNECT or other messages with
CAUSE to indicate call progress (e.g., cause: unassigned number or
user busy).
inband: use in-band tones to play busy or congestion signal to the
other side. This is the default setting.
Reset Interval
The interval that restarts idle channels.
PRI Exclusive
This setting is used to set up the ChannelID in SETUP message. If
enabled, only the specified B channel can be used. Otherwise, select one
of the channels in B channel. If you need override the existing channels
selection routine and force all PRI channels to be marked as exclusively
selected, please enable it.
Facility Enable
If selected, transmission of facility-based ISDN supplementary services
(such as caller name from CPE over facility) will be enabled.
NSF
Some switches (AT&T especially) require network specific facility.
Currently the supported values are "none", "sdn", "megacom",
"tollfreemegacom", "accunet".
Basic Settings
Clock
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is
used as the master, the PBX system clock will synchronize to it.