UCM6200 Series - User Manual

  • Updated on September 22, 2022

Welcome

Thank you for purchasing Grandstream UCM6200 series IP PBX appliance. The UCM6200 series IP PBX appliance is designed to bring enterprise-grade voice, video, data, and mobility features to small-to-medium businesses (SMBs) in an easy-to-manage fashion. This IP PBX series allows businesses to unify multiple communication technologies, such comprehensive voice, video calling, video conferencing, video surveillance, data tools and facility access management onto one common network that that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprise-grade features without any licensing fees, costs-per-feature or recurring fees.

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 adaptor with the UCM6200 as it may cause damage to the product and void the manufacturer warranty.

<p>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

</p>

<p>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.</p>

Product Overview

Technical Specifications

The following table resumes all the technical specifications including the protocols / standards supported, voice codecs, telephony features, languages and upgrade/provisioning settings for UCM6200 series.

Table 1: Technical Specifications

Interfaces

Analog Telephone FXS Ports

2x RJ11 ports with lifeline support

Each port supports 2 REN

PSTN Line FXO Ports

  • UCM6202: 2 ports
  • UCM6204: 4 ports
  • UCM6208: 8 ports

Network Interfaces

UCM6202/6204/6208:

1x Gigabit WAN port

1x Gigabit LAN port with PoE Plus (IEEE 802.3at-2009)

NAT Router

Yes

Peripheral Ports

USB 2.0, SD

LED Indicators

Power/Ready, Network, PSTN Line, USB, SD

LCD Display

128×32 graphic LCD with DOWN and OK button

Reset Switch

Yes

Voice/Video Capabilities

Voice-over-Packet

Capabilities

128ms tail-length carrier-grade Line Echo Cancellation with NLP Packetized Voice Protocol Unit, dynamic jitter buffer, modem detection, and auto-switch to G.711.

Voice and Fax Codecs

G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC (30ms only), GSM, AAL2-G.726-32, RTX, ADPCM; T.38

Video Codecs

H.264, H, H.263, H.263+, VP8

QoS

Layer 3 QoS, Layer 2 QoS

Signaling and Control

DTMF Methods

Inband, RFC4733, and SIP INFO

Provisioning Protocol and

Plug-and-Play

TFTP/HTTP/HTTPS, auto-discovery and auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66/multicast SIP SUBSCRIBE/mDNS), Eventlist between local and remote trunk

Network Protocols

TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS, LDAP/LDAPS

Disconnect Methods

Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect, Busy Tone

Security

Media

SRTP, TLS1.2, HTTPS, SSH

Advanced Defense

Fail2ban, alert events, whitelist, blacklist and strong password requirement.

Physical

Universal Power Supply

  • Output: 12VDC, 1.5A
  • Input: 100-240VAC, 50-60Hz

Dimensions

  • UCM6202/6204: 226mm (L) x 155mm (W) x 34.5mm (H)
  • UCM6208: 440mm (L) x 185mm (W) x 44mm (H)

Environmental

  • Operating: 32 – 104oF / 0 – 40oC, 10-90% (non-condensing)
  • Storage: 14 – 140oF / -10 – 60oC

Mounting

  • UCM6202/6204: Wall mount and Desktop
  • UCM6208: Rack mount and Desktop

Weight

  • UCM6202: Unit weight 0.51kg, Package weight 0.94kg
  • UCM6204: Unit weight 0.51kg, Package weight 0.94kg
  • UCM6208: Unit weight 2.23kg, Package weight 3.09kg

Additional Features

Multi-language Support

English/Simplified Chinese/Traditional Chinese/Spanish/French/

Portuguese/German/Russian/Italian/Polish/Czech for Web GUI; Customizable IVR/voice prompts for English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic;

Customizable language pack to support any other languages

Caller ID

Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT

Polarity Reversal/ Wink

Yes, with enable/disable option upon call establishment and termination

Call Center

Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability busy level, in-queue announcement

Customizable Auto Attendant

Up to 5 layers of IVR (Interactive Voice Response)

Maximum Call Capacity

  • UCM6202:

Concurrent audio calls up to 50, concurrent WebRTC calls up to 25.

  • UCM6204:

Concurrent audio calls up to 75, concurrent WebRTC calls up to 35.

  • UCM6208:

Concurrent audio calls up to 100, concurrent WebRTC calls up to 50.

Or up to 66% performance if calls are SRTP encrypted

SIP Devices

  • UCM6202/6204 up to 500 registered SIP endpoints.
  • UCM6208 up to 800 registered SIP endpoints.

Conference Rooms

  • UCM6202/6204: Up to 3 password-protected conference rooms allowing up to 25 simultaneous PSTN or IP participants
  • UCM6208: Up to 6 password-protected conference rooms allowing up to 32 simultaneous PSTN or IP participants

Call Features

Call park, call forward, call transfer, DND, ring/hunt group, paging/intercom and etc.

Compliance

  • FCC: Part 15 (CFR 47) Class B, Part 68
  • CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, TBR21, RoHS
  • A-TICK: AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS 60950, AS/ACIF S002 and ITU-T K.21 (Basic Level)
  • UL 60950 (power adapter)
UCM6200 FXS ports lifeline functionality:

The UCM6200 FXS interfaces are metallic through to the FXO interfaces. If there is power outage, FXS1 port will fail over to FXO 1 port, FXS 2 port will fail over to FXO 2 port. The user can still access the PSTN connected with the FXO interfaces from FXS interfaces.

Installation

Before deploying and configuring the UCM6200 series, the device needs to be properly powered up and connected to a network. This section describes detailed information on installation, connection and warranty policy of the UCM6200 series.

Equipment Packaging

Table 2: UCM6200 Equipment Packaging

Main Case

1x

Power Adaptor

1x

Ethernet Cable

1x

Quick Installation Guide

1x

GPL License

1x

Connect Your UCM6200

Connect the UCM6202

Figure 1: UCM6202 Front View

Figure 2: UCM6202 Back View

To set up the UCM6202, follow the steps below:

  1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6202.
  2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
  3. Connect the 12V DC power adapter into the power jacks located on the back of the UCM6202. It is highly recommended to connect the other end of the plug to a surge protected power outlet.
  4. Wait for the UCM6202 to boot up. The LCD in the front will show the device hardware information when the boot process is done.
  5. Once the UCM6202 is successfully connected to network, the LED indicator for WAN in the front will be solid green and the LCD will display the IP address.
  6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax) to the FXS ports.

The ground screw needs to be connected.

Connect the UCM6204

Figure 3: UCM6204 Front View

Figure 4: UCM6204 Back View

To set up the UCM6204, follow the steps below:

  1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6204.
  2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
  3. Connect the 12V DC power adapter into the power jacks located on the back of the UCM6204. It is highly recommended to connect the other end of the plug to a surge protected power outlet.
  4. Wait for the UCM6204 to boot up. The LCD in the front will show the device hardware information when the boot process is done.
  5. Once the UCM6204 is successfully connected to network, the LED indicator for WAN in the front will be solid green and the LCD will display the IP address.
  6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax) to the FXS ports.

The ground screw needs to be connected.

Connect the UCM6208

To set up the UCM6208, follow the steps below:

  1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6208.
  2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
  3. Connect the 12V DC power adapter into the power jacks located on the back of the UCM6208. It is highly recommended to connect the other end of the plug to a surge protected power outlet.
  4. Wait for the UCM6208 to boot up. The LCD in the front will show the device hardware information when the boot process is done.
  5. Once the UCM6208 is successfully connected to network, the LED indicator for NETWORK in the front will be solid green and the LCD will display the IP address.
  6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax) to the FXS ports.

The ground screw needs to be connected.

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Figure 5: UCM6208 Front View

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Figure 6: UCM6208 Back View

Safety Compliances

The UCM6200 series IP PBX complies with FCC/CE and various safety standards. The UCM6200 power adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM6200 package only. The manufacturer’s warranty does not cover damages to the device caused by unsupported power adapters.

Warranty

If the UCM6200 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, contact our Technical Support Team for an RMA (Return Materials Authorization) number before the product is returned. Grandstream reserves the right to remedy warranty policy without prior notification.

Warning:

Use the power adapter provided with the UCM6200 series IP PBX. Do not use a different power adapter as this may damage the device. This type of damage is not covered under warranty.

Getting Started

To get started with the UCM6200 setup process, use the following available interfaces: LCD display, LED indicators, and web portal.

  • The LCD display shows hardware, software, and network information and can be navigated via the DOWN and OK buttons next to the display. From here, users can configure basic network settings, run diagnostic tests, and factory reset.
  • The LED indicators at the front of the device provides interface connection and activity status.
  • The web portal (may also be referred to as web UI in this guide) is the primary method of configuring the UCM.

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

Use the LCD Menu

  • Idle Screen

Once the device has booted up completely, the LCD will show the UCM model, hardware version (e.g., V1.4A), and IP address. Upon menu key press timeout (15 seconds), the screen will default back to this information. Pressing the DOWN button will show the system time.

  • Menu

Pressing the OK button will show the main menu. All available menu options are found in [Table 3: LCD Menu Options].

  • Menu Navigation

Pressing the DOWN button will scroll through the menu options. Press the OK button to select an option.

  • Exit

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

  • LCD Backlight

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

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

Table 3: 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 uptime

Network Info

  • WAN Mode: DHCP, Static IP, or PPPoE
  • WAN IP: IP address
  • WAN Subnet Mask
  • LAN IP: IP address
  • LAN Subnet Mask

Network Menu

  • WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE
  • Static Route Reset: Select this to reset static route settings.

Factory Menu

  • Reboot
  • Factory Reset
  • LCD Test Patterns

Press DOWN and OK buttons to scroll through and select different LCD patterns to test. Once a test is done, press the OK button to return to the previous menu.

  • Fan Mode

Select Auto or On.

  • LED Test Patterns

All On, All Off, and Blinking are the available options. Selecting Back in the
menu will revert the LED indicators back to their actual status.

  • RTC Test Patterns

Select either 2022-02-22 22:22 or 2011-01-11 11:11 to start the RTC (Real-Time Clock) test pattern. Check the system time from either the LCD idle screen or in the web portal System Status->System Information->General page. To revert back to the correct time, manually reboot the device.

  • Hardware Testing

Select Test SVIP to verify hardware connections within the device. The result will display on the LCD when the test is complete.

Web Info

  • Protocol: Web access protocol (HTTP/ HTTPS). HTTPS is used by default.
  • Port: Web access port number, which is 8089 by default.

SSH Switch

  • Enable SSH
  • Disable SSH

SSH access is disabled by default

Use the LED Indicators

The UCM6200 has LED indicators in the front to display connection status. The following table shows the status definitions.

Table 4: UCM6202/UCM6204 LED Indicators

LED Indicator

LED Status

LAN

WAN

USB

SD

FXS (Phone/Fax)

FXO (Telco Line)

Solid: Connected

Flashing: Data Transferring

OFF: Not Connected

Table 5: UCM6208 LED Indicators

LED

LED Status

NETWORK

Solid: Connected

OFF: Not Connected

ACT

USB

SD

Phone (FXS)

Line (FXO)

Solid: Connected

Flashing: Data Transferring

OFF: Not Connected

Using the Web UI

Accessing the Web UI

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

Figure 7: UCM6202 Web GUI Login Page

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

Units manufactured starting January 2017 have a unique random password printed on the sticker located on the back of the unit. It is highly recommended to change the default password after logging in for the first time. Older units have default password “admin”. t

By default, the UCM6200 has Redirect From Port 80 enabled. As such, if users type in the UCM6200 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, and 192.168.40.167 is entered into the web browser, the web page will be redirected to: https://192.168.40.167:8089

The option Redirect From Port 80 can be found in the System Settings🡪HTTP Server page.

Setup Wizard

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

Figure 8: UCM6200 Setup Wizard

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

Main Settings

There are 8 main sections in the web portal to manage various features of the UCM.

  • System Status: Displays the dashboard, system information, current active calls, and network status.
  • Extensions/Trunks: Manages extensions, trunks, and routing rules.
  • Call Features: Manages various features of the UCM such as the IVR and voicemail.
  • PBX Settings: Manages the settings related to PBX functionality such as SIP settings and interface settings.
  • System Settings: Manages the settings related to the UCM system itself such as network and security settings.
  • CDR: Contains the call detail records, statistics, and audio recordings of calls processed by the UCM.
  • Value-Added Features: Manages the settings of features unrelated to core PBX functionality such as Zero Config provisioning and CRM/PMS integrations.
  • Maintenance: Manages settings and logs related to system management and maintenance such as
    user management, activity logs, backup settings, upgrade settings and troubleshooting tools.

Web GUI Languages

Currently the UCM6200 series Web GUI supports English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, Russian, Italian, Polish, German, Turkish and Czech.

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

Figure 9: UCM6200 Web GUI Language

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

Figure 10: Web GUI Search Bar

Saving and Applying Changes

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

Setting Up an Extension

Power on the UCM6200 and your SIP endpoint. Connect both devices to the same network and follow the steps below to set up an extension.

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

System Settings

This section will explain the available system-wide parameters and configuration options on the UCM62xx. This includes settings for the following items: HTTP server, network, OpenVPN, DDNS, LDAP server and email server.

HTTP Server

The UCM6200’s embedded web server responds to HTTP/HTTPS GET/POST requests and allows users to configure the UCM via web browsers such as Microsoft IE, Mozilla Firefox, and Google Chrome. By default, users can access the UCM by just typing its IP address into a browser address bar. The browser will automatically be redirected to HTTPS using port 8089. For example, typing in “192.168.40.50” into the address bar will redirect the browser to “https://192.168.40.50:8089”. This behavior can be changed in the System Settings->HTTP Server page.

Table 6: HTTP Server Settings

Redirect From Port 80

Toggles automatic redirection to UCM’s web portal from port 80. If disabled, users will need to manually add the UCM’s configured HTTP/HTTPS port to the server address when accessing the UCM web portal via browser. Default is “Enabled”.

Protocol Type

Select either HTTP or HTTPS as the protocol to access the UCM’s HTTP server. This will also determine what is used when endpoints download config files from the UCM via Zero Config. Default is “HTTPS”.

Port

Specifies the port number used to access the UCM HTTP server. Default is “8089”.

Enable IP Whitelist

If enabled, only the server addresses in whitelist will be able to access the UCM’s web portal. It is highly recommended to add the IP address currently used to access the UCM web page before enabling this option. Default is “Disabled”.

Permitted IP(s)

List of addresses that can access the UCM web portal.

Ex: 192.168.6.233 / 255.255.255.255

Certificate Options

Selects the method of acquiring SSL certificates for the UCM web server. Two methods are currently available:

– Upload Certificate: Upload the appropriate files from one’s own PC.

– Request Certificate: Enter the domain for which to request a certificate for from Let’s Encrypt.

TLS Private Key

Uploads the private key for the HTTP server.

Note: Key file must be under 2MB in file size and in *.pem format. File name will automatically be changed to “private.pem”.

TLS Cert

Uploads the certificate for the HTTP server.

Note: Certificate must be under 2MB in file size and in *.pem format. This will be used for TLS connections and contains private key for the client and signed certificate for the server.

Domain

Enter the domain to request the certificate for and click on to request the certificate.

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

Network Settings

After successfully connecting the UCM6200 to the network for the first time, users could login the Web GUI and go to System Settings🡪Network Settings to configure the network parameters for the device.

  • UCM6200 supports Route/Switch/Dual mode functions.

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

Basic Settings

Please refer to the following tables for basic network configuration parameters on UCM6202, UCM6204 and UCM6208, respectively.

Table 7: UCM6200 Network Settings🡪Basic Settings

Method

Select “Route”, “Switch” or “Dual” mode on the network interface of UCM6200. The default setting is “Route”.

  • Route

WAN port will be used for uplink connection. LAN port will function similarly to a regular router port.

  • Switch

WAN port will be used for uplink connection. LAN port will be used as a bridge for connections.

  • Dual

Both WAN and LAN ports will be used for uplink connections labeled as LAN2 and LAN1, respectively. The port selected as the Default Interface will need to have a gateway IP address configured if it is using a static IP.

MTU

Specifies the maximum transmission unit value. Default is 1500.

IPv4 Address

Preferred DNS Server

If configured, this will be used as the Primary DNS server.

WAN (when “Method” is set to “Route”)

IP Method

Select DHCP, Static IP, or PPPoE. The default setting is DHCP.

IP Address

Enter the IP address for static IP settings. The default setting is 192.168.0.160.

Subnet Mask

Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.

Gateway IP

Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.

DNS Server 1

Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.

DNS Server 2

Enter the DNS server 2 address for static IP settings.

Username

Enter the username to connect via PPPoE.

Password

Enter the password to connect via PPPoE.

Layer 2 QoS 802.1Q/VLAN Tag

Assign the VLAN tag of the layer 2 QoS packets for WAN port. The default value is 0.

Layer 2 QoS 802.1p Priority Value

Assign the priority value of the layer 2 QoS packets for WAN port. The default value is 0.

LAN (when Method is set to “Route”)

IP Address

Enter the IP address assigned to LAN port. The default setting is 192.168.2.1.

Subnet Mask

Enter the subnet mask. The default setting is 255.255.255.0.

DHCP Server Enable

Enable or disable DHCP server capability. The default setting is “Yes”.

DNS Server 1

Enter DNS server address 1. The default setting is 8.8.8.8.

DNS Server 2

Enter DNS server address 2. The default setting is 208.67.222.222.

Allow IP Address From

Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.

Allow IP Address To

Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.

Default IP Lease Time

Enter the IP lease time (in seconds). The default setting is 43200.

LAN (when Method is set to “Switch”)

IP Method

Select DHCP, Static IP, or PPPoE. The default setting is DHCP.

IP Address

Enter the IP address for static IP settings. The default setting is 192.168.0.160.

Subnet Mask

Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.

Gateway IP

Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.

DNS Server 1

Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.

DNS Server 2

Enter the DNS server 2 address for static IP settings.

Username

Enter the username to connect via PPPoE.

Password

Enter the password to connect via PPPoE.

Layer 2 QoS 802.1Q/VLAN Tag

Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 0.

Layer 2 QoS 802.1p Priority Value

Assign the priority value of the layer 2 QoS packets for LAN port. The default value is 0.

LAN 1 / LAN 2 (when Method is set to “Dual”)

Default Interface

If “Dual” is selected as the Method, users must select either LAN 1 (WAN port) or LAN 2 (LAN port) to be used as the default interface. Default setting is LAN 2.

IP Method

Select DHCP, Static IP, or PPPoE. The default setting is DHCP.

IP Address

Enter the IP address for static IP settings.

The default setting is 192.168.0.160.

Subnet Mask

Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.

Gateway IP

Enter the gateway IP address for static IP settings when the port is assigned as default interface. The default setting is 0.0.0.0.

DNS Server 1

Enter the DNS server 1 address for static IP settings.

The default setting is 0.0.0.0.

DNS Server 2

Enter the DNS server 2 address for static IP settings.

Username

Enter the username to connect via PPPoE.

Password

Enter the password to connect via PPPoE.

Layer 2 QoS 802.1Q/VLAN Tag

Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 0.

Layer 2 QoS 802.1p Priority Value

Assign the priority value of the layer 2 QoS packets for LAN port. The default value is 0.

IPv6 Address

WAN (when “Method” is set to “Route”)

IP Method

Select Auto or Static. The default setting is Auto

IP Address

Enter the IP address for static IP settings.

IP Prefixlen

Enter the Prefix length for static settings. Default is 64

DNS Server 1

Enter the DNS server 1 address for static settings.

DNS Server 2

Enter the DNS server 2 address for static settings.

LAN (when Method is set to “Route”)

DHCP Server

Select Disable, Auto or DHCPv6.

Disable: Disables the DHCPv6 server.

Auto: Stateless address auto configuration using NDP protocol.

DHCPv6: Stateful address auto configuration using DHCPv6 protocol.

DHCP Prefix

Enter DHCP prefix. (Default is 2001:db8:2:2::)

DHCP prefixlen

Enter the Prefix length for static settings. Default is 64

DNS Server 1

Enter the DNS server 1 address for static settings. Default is (2001:4860:4860::8888 )

DNS Server 2

Enter the DNS server 2 address for static settings. Default is (2001:4860:4860::8844 )

Allow IP Address From

Configure starting IP address assigned by the DHCP prefix and DHCP prefixlen.

Allow IP Address To

Configure the ending IP address assigned by the DHCP Prefix and DHCP prefixlen.

Default IP Lease Time

Configure the lease time (in second) of the IP address.

LAN (when Method is set to “Switch”)

IP Method

Select Auto or Static. The default setting is Auto

IP Address

Enter the IP address for static IP settings.

IP Prefixlen

Enter the Prefix length for static settings. Default is 64

DNS Server 1

Enter the DNS server 1 address for static settings.

DNS Server 2

Enter the DNS server 2 address for static settings.

LAN 1 / LAN 2 (when Method is set to “Dual”)

Default Interface

Users must select either LAN 1 (WAN port) or LAN 2 (LAN port) to be used as the default interface. Default setting is LAN 2.

IP Method

Select Auto or Static. The default setting is Auto

IP Address

Enter the IP address for static IP settings.

IP Prefixlen

Enter the Prefix length for static settings. Default is 64

DNS Server 1

Enter the DNS server 1 address for static settings.

DNS Server 2

Enter the DNS server 2 address for static settings.

  • Method: Route

When the UCM6200 has, method set to Route in network settings, WAN port interface is used for uplink connection and LAN port interface is used as a router. Please see the sample diagram below.

Figure 11: UCM6202 Network Interface Method: Route

  • Method: Switch

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

Timeline

Description automatically generated

Figure 12: UCM6202 Network Interface Method: Switch

  • Method: Dual

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

Timeline

Description automatically generated

Figure 13: UCM6202 Network Interface Method: Dual

DHCP Client List

This page lists all the detected devices on the LAN and the IP addresses that were provided to them. Additionally, users can manually link a MAC address to an IP address.

When devices receive IP addresses from the UCM’s DHCP server, they will be listed in the System Settings->Network Settings->DHCP Client List page as shown below.

Figure 14: DHCP Client List

To manually add and bind a MAC address to an IP address, click on . The following menu will then be displayed.

Figure 15: Add MAC Address Bind

Enter the device’s MAC address and the IP address to bind it to. This IP address must be in the UCM’s DHCP range.

To bind multiple existing MAC addresses that are in the list to their respective IP addresses, check the boxes next to each MAC address and click on the button. A confirmation message will appear on the screen. Click to bind the addresses.

Figure 16: Batch Add MAC Address Bind

The Bind Status for the selected MAC addresses should now be changed from “Unbound” to “Bound”.

802.1X

IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication mechanism to device before the device can access Internet or other LAN resources. The UCM6200 supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure show UCM6200 use 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet.

Diagram

Description automatically generated

Figure 17: UCM6200 Using 802.1X as Client

Figure 18: UCM6200 Using 802.1X EAP-MD5

The following table shows the configuration parameters for 802.1X on UCM6200. Identity and MD5 password are required for authentication, which should be provided by the network administrator obtained from the RADIUS server. If “EAP-TLS” or “EAP-PEAPv0/MSCHAPv2” is used as the 802.1X mode, users will also need to upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated from the RADIUS server.

Table 8: UCM6200 Network Settings🡪802.1X

802.1X Mode

Select 802.1X mode. The default setting is “Disable”. The supported 802.1X modes are:

  • 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 Certificate

Select 802.1X certificate from local PC and then upload.

802.1X Client Certificate

Select 802.1X client certificate from local PC and then upload.

Static Routes

The UCM6200 provides users static routing capability that allows the device to use manually configured routes, instead of dynamically assigned routes or the default gateway. Static routes can be used to define a route when no others are available or can serve as complementary routes alongside existing routes as failover routes, with existing routing on the UCM6200 as a failover backup, etc.

  • Click on to create a new IPv4 static route or click on to create a new IPv6 static route. The configuration parameters are listed in the table below.
  • Once added, users can select to edit the static route.
  • Select to delete the static route.

Table 9: UCM6200 Network Settings🡪Static Routes

Destination

Configure the destination IPv4 address or the destination IPv6 subnet for the UCM6200 to reach using the static route.

Example:

IPv4 address – 192.168.66.4

IPv6 subnet – 2001:740:D::1/64

Netmask

Configure the subnet mask for the above destination address. If left blank, the default value is 255.255.255.255.

Example:

255.255.255.0

Gateway

Configure the IPv4 or IPv6 gateway address so that the UCM6200 can reach the destination via this gateway. Gateway address is optional.

Example:

192.168.40.5 or 2001:740:D::1

Interface

Specify the network interface on the UCM6200 to reach the destination using the static route.

LAN interface is eth0; WAN interface is eth1.

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

The following diagram shows a sample application of static route usage on UCM6204.

Untitled

Figure 19: UCM6204 Static Route Sample

The network topology of the above diagram is as below:

  • Network 192.168.69.0 has IP phones registered to UCM6204 LAN 1 address
  • Network 192.168.40.0 has IP phones registered to UCM6204 LAN 2 address
  • Network 192.168.66.0 has IP phones registered to UCM6204 via VPN
  • Network 192.168.40.0 has VPN connection established with network 192.168.66.0

In this network, by default the IP phones in network 192.168.69.0 are unable to call IP phones in network 192.168.66.0 when registered on different interfaces on the UCM6204. Therefore, we need configure a static route on the UCM6204 so that the phones in isolated networks can make calls between each other.

Figure 20: UCM6204 Static Route Configuration

Port Forwarding

The UCM network interface supports router functionality that allows users to configure port forwarding. If the UCM is set to “Route” in System Settings->Network Settings->Basic Settings->Method, the Port Forwarding tab will be available.

Table 10: UCM6200 Network Settings🡪Port Forwarding

WAN Port

Specify the WAN port number or a range of WAN ports. Unlimited number of ports can be configured.

Note:

When set to a range, both the WAN port and LAN port must be configured with the same range (e.g., WAN port: 1000-1005, LAN port: 1000-1005). Access from the WAN port will be forwarded to the LAN port with the same port number.

LAN IP

Specify the LAN IP address.

LAN Port

Specify the LAN port number or a range of LAN ports.

Please see the note for WAN Port.

Protocol Type

Select protocol type “UDP Only”, “TCP Only” or “TCP/UDP” for the forwarding in the selected port. The default setting is “UDP Only”.

Here is an example of an environment and how to configure port forwarding to an endpoint’s web portal.

  • Configure the UCM as a router by selecting “Route” in System Settings->Network Settings->Basic Settings->Method.
  • WAN port is connected to an uplink switch with a public IP address configured (e.g., 1.1.1.1).
  • LAN port provides a DHCP pool for devices on the LAN network (gateway address is 192.168.2.1 by default).
  • Connect a GXP2160 to the UCM’s LAN network. It should obtain an IP address from UCM’s DHCP pool.
  • While still on the Network Settings page, navigate to the Port Forwarding tab.
  • Click on button to start setting up port forwarding.

Figure 21: Create New Port Forwarding

WAN Port: Enter the port that will be opened on the WAN side to allow access.

LAN IP: Enter the GXP2160’s IP address.

LAN Port: Enter the port that will be opened on the GXP2160 to allow access.

Protocol Type: Select the protocol to use for this port forwarding.

Figure 22: UCM6200 Port Forwarding Configuration

This will allow users to access the GXP2160 web portal from an external network.

Figure 23: GXP2160 Web Access using UCM6202 Port Forwarding

OpenVPN®

The UCM can be configured as an OpenVPN client.

Table 11: UCM6200 System Settings🡪Network Settings🡪OpenVPN®

Enable

Enable / Disable the OpenVPN® feature.

Configuration Method

Select OpenVPN® configuration method.

Manual Configuration: Allows to configure OpenVPN settings manually.

Upload Configuration File: Allows to upload. ovpn and .conf files to the UCM and to automatically configure OpenVPN settings.

OpenVPN® Server Address

Configures the hostname/IP and port of the server. For example: 192.168.1.2:22

OpenVPN® Server Protocol

Specify the protocol user, user should use the same settings as used on the server

OpenVPN® Device mode

Use the same setting as used on the server.
Dev TUN: Create a routed IP tunnel.
Dev TAP: Create an Ethernet tunnel.

OpenVPN® Use Compression

Compress tunnel packets using the LZO algorithm on the VPN link. Do not enable this unless it is also enabled in the server config file.

Allow Weak SSL Ciphers

Enable/Disable allowing Weak SSL Ciphers.

OpenVPN® Encryption Algorithm

Specify the cryptographic cipher. Users should make sure to use the same setting that they are using on the OpenVPN server.

OpenVPN® CA Cert

Upload as SSL/TLS root certificate. This file will be renamed as ‘ca.crt’ automatically.

OpenVPN® Client Cert 

Upload a client certificate. This file will be renamed as ‘cliend.crt’ automatically.

OpenVPN® Client Key 

Upload a client private key. This file will be renamed as ‘client.key’ automatically.

Figure 24: Open VPN® feature on the UCM6200

DDNS Settings

Configuring UCM DDNS settings will allow the UCM to be accessed via domain name instead of IP address. UCM supports the following DDNS services:

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

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

  1. The UCM must be publicly accessible in order to work with the service provider.

Figure 25: Register Domain Name on noip.com

  1. Navigate to System Settings->Network Settings->DDNS Settings, check the Enable DDNS option, select your service provider, and configure all the displayed fields.

Figure 26: UCM6200 DDNS Setting

  1. You can now use the configured domain to access the UCM web portal.

Figure 27: Using Domain Name to Connect to UCM6200

Security Settings

The UCM offers several methods of protection against malicious attacks and unauthorized access such as firewall rules, connection thresholds, and Fail2ban.

To get started on configuring security settings, navigate to the System Settings🡪Security Settings page.

Static Defense

On the Static Defense page, users can configure firewall rules and view the ports used by various UCM services and processes.

The following table shows a few examples of the information available on the Static Defense page.

Table 12: UCM6200 Firewall🡪Static Defense🡪Current Service

Port

Process

Type

Protocol or Service

7777

Asterisk

TCP/IPv4

SIP

389

Slapd

TCP/IPv4

LDAP

2000

Asterisk

TCP/IPv4

SCCP

22

Dropbear

TCP/IPv4

SSH

80

Lighthttpd

TCP/IPv4

HTTP

8089

Lighthttpd

TCP/IPv4

HTTPS

69

Opentftpd

UDP/IPv4

TFTP

9090

Asterisk

UDP/IPv4

SIP

6060

zero_config

UDP/IPv4

UCM6200 zero_config service

5060

Asterisk

UDP/IPv4

SIP

4569

Asterisk

UDP/IPv4

SIP

5353

zero_config

UDP/IPv4

UCM6200 zero_config service

37435

Syslogd

UDP/IPv4

Syslog

For general firewall defense mechanisms, UCM supports Ping Defense, SYN-Flood Defense, and Ping-of-Death Defense. These can be configured separately for the WAN interface and LAN interface.

Table 13: Typical Firewall Settings

Ping Defense Enable

If enabled, the UCM will not send ICMP messages in response to ping requests.

SYN-Flood Defense Enable

If enabled, the UCM will limit the amount of SYN packets accepted from one source to 10 packets per second, preventing the UCM web portal from becoming inaccessible. Excess packets will be discarded.

Ping-of-Death Defense Enable

If enabled, the UCM will not be affected by ping of death attacks.

In the Custom Firewall Settings section, users can create rules to accept, reject, or drop specific traffic going through the UCM. To create a new rule, click on the https://lh3.googleusercontent.com/-4vbaLaV0BV8/X3tcT1l2yhI/AAAAAAAACk4/gNmOjjA-BZoNv834SrX2zak1asC3M9ZDgCK8BGAsYHg/s0/11.PNG button.

The following menu will appear. Here is an example firewall rule created to reject SSH access to the UCM from the WAN interface.

Figure 28: Create New Firewall Rule

Table 14: Firewall Rule Settings

Rule Name

Enter a name for the firewall rule.

Action

Select the action for the Firewall to perform.

  • ACCEPT
  • REJECT
  • DROP

Type

Select the traffic type.

  • IN

If selected, the Interface field will appear. Users must specify the interface that the inbound rule will be applied to.

  • OUT

Service

Select the connection type the rule will apply to.

  • FTP
  • SSH
  • Telnet
  • TFTP
  • HTTP
  • LDAP
  • Custom

If selected, users must configure the Source IP and Port,Destination IP Address and Port, and the Protocol fields that appear. If the source and destination are left blank, the “Anywhere” values will be used.

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

  • Click on to edit the rule.
  • Click on to delete the rule.

Dynamic Defense

On the Dynamic Defense page, users can configure the UCM to monitor incoming TCP connections and prevent excessive traffic from hosts. The UCM must have “Route” configured in the System Settings->Network Settings->Basic Settings page. The blacklist on this page is automatically updated.

The following options are available:

Table 15: UCM6200 Firewall Dynamic Defense

Dynamic Defense Enable

Toggle dynamic defense on and off. This is disabled by default.

Blacklist Update Interval

Configure the blacklist update time interval (in seconds). The default setting is 120.

For example, if set to “300”, blocked IP addresses will not be able to establish TCP connections with the UCM until after 300 seconds have passed.

Connection Threshold

Configure the connection threshold. Once a host exceeds this threshold, it will be added to the blacklist. Default setting is 100.

Dynamic Defense Whitelist

Allowed IPs and ports range, multiple IP addresses and port range.

For example:

192.168.5.100-

192.168.5.200 1500:2000

The following figure shows a configuration example like this:

  • If a host at IP address 192.168.5.7 initiates more than 20 TCP connections to the UCM6200 within 1 minute, it will be added into UCM6200 blacklist.
  • This host 192.168.5.7 will be blocked by the UCM6200 for 500 seconds.
  • Since IP range 192.168.5.100-192.168.5.200 is in whitelist, if a host initiates more than 20 TCP connections to the UCM6200 within 1 minute, it will not be added into UCM6200 blacklist. It can still establish TCP connection with the UCM6200.

Figure 29: Configure Dynamic Defense

Fail2ban

Fail2Ban feature on the UCM6200 provides intrusion detection and prevention for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE and prevents SIP brute force attacks on the PBX system.

Once an IP address exceeds the allowed number of login or SIP authentication attempts within the configured “Max Retry Duration” period, all SIP and HTTP requests from that IP address will be dropped, forbidding web access and blocking further authentication attempts.

Figure 30: Fail2ban Settings

Table 16: Fail2Ban Settings

Global Settings

Enable Fail2Ban

Enable Fail2Ban. The default setting is disabled. Please make sure both “Enable Fail2Ban” and “Asterisk Service” are turned on to use Fail2Ban for SIP authentication on the UCM6200.

Banned Duration

Configure the duration (in seconds) for the detected host to be banned. The default setting is 600. If set to 0, the host will be always banned.

Max Retry Duration

If a host exceeds the maximum allowed number attempts configured for Max Retry within the configured Max Retry Duration window, the host will be banned. The default setting is 600 seconds.

MaxRetry

Configures the maximum number of allowed authentication failures within the configured Max Retry Duration window. The default setting is 5.

Fail2Ban Whitelist

Configures the IP addresses, CIDR masks, and DNS hosts in the Fail2Ban whitelist. Whitelisted entries will not be banned by Fail2Ban even after exceeding the allowed number of authentication failures. Up to 20 addresses can be added.

Local Settings

Asterisk Service

Enable Asterisk service for Fail2Ban. The default setting is disabled. Please make sure both “Enable Fail2Ban” and “Asterisk Service” are turned on to use Fail2Ban for SIP authentication on the UCM6200.

Listening Port Number

Configure the listening port number for the service. By default, port 5060 will be used for UDP and TCP, and port 5061 will be used for TLS.

MaxRetry

Configures the maximum number of authentication failures before the host is banned. The default setting is 10. Please note that this will override the Global Settings->MaxRetry setting.

Login Attack Defense

Enables defense against excessive login attacks to the UCM’s web GUI.

The default setting is disabled.

Listening Port Number

This is the Web GUI listening port number which is configured under System Settings🡪HTTP Server🡪Port. The default is 8089.

MaxRetry

Configures the maximum allowed number of failed login attempts from an IP address before it is added to the Fail2Ban blacklist.

Blacklist

Blacklist

Users will be able to view the IPs that have been blocked by UCM.

TLS Security

TLS security to specify minimum and maximum TLS versions supported by the UCM.

Please log in UCM62xx web interface and go to System Settings🡪Security Settings🡪TLS Security.

By default, minimum TLS version is set to TLS1.1, and maximum TLS version is set to TLS1.2.

Supported versions are 1.0, 1.1 and 1.2

Figure 31: TLS Security

SSH Access

SSH access can be toggled from the UCM’s webUI and physical LCD screen. The webUI option can be found under System Settings->Security Settings->SSH Access. SSH access is disabled by default and should only be turned on for troubleshooting and debugging.

Figure 32: SSH Access

LDAP Server

The UCM6200 has an embedded LDAP/LDAPS server for users to manage corporate phonebooks in a centralized manner.

  • The LDAP server automatically generates an initial phonebook with PBX DN “ou=pbx,dc=pbx,dc=com” based on the UCM6200’s existing extensions.
  • Users could add new phonebook with a different Phonebook DN for other external contacts. For example, “ou=people,dc=pbx,dc=com”.
  • All the phonebooks in the UCM6200 LDAP server have the same Base DN “dc=pbx,dc=com”.
Term Explanation:

cn= Common Name

ou= Organization Unit

dc= Domain Component

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

If users have the Grandstream phone provisioned by the UCM6200, the LDAP directory will be set up on the phone and can be used right away for users to access all phonebooks.

Additionally, users could manually configure the LDAP client settings to manipulate the built-in LDAP server on the UCM6200. If the UCM6200 has multiple LDAP phonebooks created, in the LDAP client configuration, users could use “dc=pbx,dc=com” as Base DN to have access to all phonebooks on the UCM6200 LDAP server, or use a specific phonebook DN, for example “ou=people,dc=pbx,dc=com”, to access to phonebook with Phonebook DN “ou=people,dc=pbx,dc=com ” only.

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

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

LDAP Server Configurations

The following figure shows the default LDAP server configurations on the UCM6200.

Figure 33: LDAP Server Configurations

The UCM6200 LDAP server supports anonymous access (read-only) by default. Therefore, the LDAP client does not have to configure username and password to access the phonebook directory. The “Root DN” and “Root Password” (limited with 64 and 32 characters respectively) here are for LDAP management and configuration where users will need provide for authentication purpose before modifying the LDAP information.

The default phonebook list in this LDAP server can be viewed and edited by clicking on for the first phonebook under LDAP Phonebook.

Figure 34: Default LDAP Phonebook DN

Figure 35: Default LDAP Phonebook Attributes

LDAP Phonebook

Users could use the default phonebook, edit the default phonebook, add new phonebook, import phonebook on the LDAP server as well as export phonebook from the LDAP server. The first phonebook with default phonebook dn “ou=pbx,dc=pbx,dc=com” displayed on the LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly. The contacts information will need to be modified via Web GUI🡪Extension/Trunk🡪Extensions first. The default LDAP phonebook will then be updated automatically.

Figure 36: LDAP Server🡪LDAP Phonebook

  • Add new phonebook

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

Figure 37: Add LDAP Phonebook

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

Once added, users can select to edit the phonebook attributes and contact list (see figure below) or select to delete the phonebook.

Figure 38: Edit LDAP Phonebook

  • Import phonebook from your computer to LDAP server

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

Figure 39: Import Phonebook

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

Here is an example of CSV file. Please note “Account Number” and “Phonebook DN” fields are required. Users can export a phonebook file from the UCM’s LDAP phonebook section and use it as a template.

Figure 40: Phonebook CSV File Format

The Phonebook DN field is the same “Phonebook Prefix” entry as when the user clicks on “Add” to create a new phonebook. Therefore, if the user enters “phonebook” in “Phonebook DN” field in the CSV file, the actual phonebook DN “ou=phonebook,dc=pbx,dc=com” will be automatically created by the UCM6200 once the CSV file is imported.

In the CSV file, users can specify different phonebook DN fields for different contacts. If the phonebook DN already exists on the UCM6200 LDAP Phonebook, the contacts in the CSV file will be added into the existing phonebook. If the phonebook DN does not exist on the UCM6200 LDAP Phonebook, a new phonebook with this phonebook DN will be created.

The sample phonebook CSV file in above picture will result in the following LDAP phonebook in the UCM6200.

Figure 41: LDAP Phonebook After Import

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

  • Export phonebook to your computer from UCM6200 LDAP server

Select the checkbox for the LDAP phonebook and then click on “Export Selected Phonebook” to export the selected phonebook. The exported phonebook can be used as a record or a sample CSV, VFC or XML file for the users to add more contacts in it and import to the UCM6200 again.

Figure 42: Export Selected LDAP Phonebook

LDAP Client Configurations

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

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

  • LDAP Server: Enter a name for the remote LDAP server
  • Server Address: Enter the IP address or domain name for remote LDAP server.
  • Base DN: dc=pbx,dc=com
  • Username: Enter username if authentication is required. This field cannot exceed 64 characters and can contain space.
  • Password: Enter password if authentication is required.
  • Filter: Enter the filter. Ex: (|(CallerIDName=%)(AccountNumber=%))
  • Port: Enter the port number. Ex:389
  • LDAP Name Attributes: Enter the name attributes for remote server
  • LDAP Number Attributes: Enter the number attributes for remote server
  • Client type: Protocol of LDAP or LDAPS.
  • LDAP Client CA cert: Upload LDAP client CA certificate, The following file types are supported: .crt .der and .pem.
  • LDAP Client Private Key: Upload LDAP client private key.

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

Figure 43: LDAP Client Configurations

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

  • Server Address: The IP address or domain name of the UCM6200
  • Base DN: dc=pbx,dc=com
  • Username: Please leave this field empty
  • Password: Please leave this field empty
  • LDAP Name Attribute: CallerIDName Email Department FirstName LastName
  • LDAP Number Attribute: AccountNumber MobileNumber HomeNumber Fax
  • LDAP Number Filter: (AccountNumber=%)
  • LDAP Name Filter: (CallerIDName=%)
  • LDAP Display Name: AccountNumber CallerIDName
  • LDAP Version: If existed, please select LDAP Version 3
  • Port: 389

The following figure shows the configuration information on a Grandstream GXP2170 to successfully use the LDAP server as configured in [Figure 33: LDAP Server Configurations].

https://lh3.googleusercontent.com/-HX48dNb6acI/WSwyQUNq4fI/AAAAAAAABHs/qm3Vm2qXvmQWWi3Sqm7kNZ2MtRvXNTMtgCL0B/h759/2017-05-29.png

Figure 44: GXP2170 LDAP Phonebook Configuration

Time settings

Automatic Date and Time

The current system time can be found under System Status->Dashboard and System Information.

To configure the UCM to automatically update time, navigate to System Settings->Time Settings->Automatic Date and Time.

Note:

The configurations under Web GUI🡪Settings🡪Time Settings🡪Automatic Date and Time page require reboot to take effect. Please consider configuring auto time updating related changes when setting up the UCM6200 for the first time to avoid service interrupt after installation and deployment in production.

Table 17: Time Auto Updating

Remote NTP Server

Specify the server address of the NTP server for the UCM6510 to sync date and time with. The default NTP server is pool.ntp.org.

Enable DHCP Option 2

If set to “Yes”, the UCM’s time zone can be provisioned via DHCP Option 2 from a local server automatically.

Enable DHCP Option 42

If set to “Yes”, the UCM’s NTP server can be provisioned via DHCP Option 42 from a local server automatically. This will override the manually configured NTP server. The default setting is “Yes”.

Time Zone

Select the time zone for the UCM to use.

If “Self-Defined Tome Zone” is selected, please specify the time zone parameters in “Self-Defined Time Zone” field as described in below option.

Self-Defined Time Zone

If “Self-Defined Time Zone” is selected in “Time Zone” option, users will need define their own time zone following the format below.

The syntax is: std offset dst [offset], start [/time], end [/time]

Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0

MTZ+6MDT+5

This indicates a time zone with 6 hours offset and 1 hour ahead for DST, which is U.S central time. If it is positive (+), the local time zone is west of the Prime Meridian (A.K.A: International or Greenwich Meridian); If it is negative (-), the local time zone is east.

M4.1.0,M11.1.0

The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec).

The 2nd number indicates the nth iteration of the weekday: (1st Sunday,

3rd Tuesday…). Normally 1, 2, 3, 4 are used. If 5 is used, it means the last iteration of the weekday.

The 3rd number indicates weekday: 0,1,2,..,6 ( for Sun, Mon, Tues, … ,Sat).

Therefore, this example is the DST which starts from the First Sunday of

April to the 1st Sunday of November.

Set Time Manually

To manually set the time on the UCM6200, go to Web GUI🡪System Settings🡪Time Settings🡪Set Date and Time. The format is YYYY-MM-DD HH:MM:SS.

Figure 45: Set Time Manually

Note:

Manually setup time will take effect immediately after saving and applying change in the Web GUI. If users would like to reboot the UCM6200 and keep the manually setup time setting, please make sure “Remote NTP Server”, “Enable DHCP Option 2” and “Enable DHCP Option 42” options under Web GUI🡪Settings🡪Time Settings🡪Auto Time Updating page are unchecked or set to empty. Otherwise, time auto updating settings in this page will take effect after reboot.

NTP Server

The UCM can act as an NTP server for clients to sync their system times with. To configure this, navigate to System Settings->Time Settings->NTP Server and set Enable NTP Server to “Yes”. On the client side, use the UCM’s IP address or hostname as the NTP server address.

Office Time

The system administrator can define office hours which can be used as conditions for call forwarding and inbound routing. To configure this, navigate to System Settings->Time Settings->Office Time and click on the Add button to create office hours.

Figure 44: Create New Office Time

Start Time

Configure the start time for office hour.

End Time

Configure the end time for office hour

Week

Select the workdays in one week.

Show Advanced Options

Check this option to show advanced options. Once selected, please specify “Month” and “Day” below.

Month

Select the months for office time.

Day

Select the workdays in one month.

Table 18: Create New Office Time

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

Figure 47: Settings🡪Time Settings🡪Office Time

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

Holiday

System administrators can define holidays which can be used as conditions for call forwarding and inbound routing. To configure this, navigate to System Settings->Time Settings->Holiday and click on the Add button to create a new holiday.

Figure 48: Create New Holiday

Table 19: Create New Holiday

Name

Specify the holiday name to identify this holiday.

Holiday Memo

Create a note for the holiday.

Year

Select the year of the Holiday.

Note: Users can now configure holidays for the next 4 years.

Month

Select the month for the holiday.

Day

Select the day for the holiday.

Show Advanced Options

Check this option to show advanced options. If selected, please specify the days as holiday in one week below.

Week

Select the days as holiday in one week.

Enter holiday “Name” and “Holiday Memo” for the new holiday. Then select “Month” and “Day”. The system administrator can also define days in one week as advanced options. Once done, click on “Save” and then “Apply Change” for the holiday to take effect. The holiday will be listed in the web page as the figure shows below.

Figure 49: Settings🡪Time Settings🡪Holiday

  • Click on to edit the holiday.
  • Click on to delete the holiday.
  • Click on “Delete Selected Holidays” to delete multiple selected holidays at once.

Note:

For more details on how to use office time and holiday, please refer to the link below:

https://documentation.grandstream.com/knowledge-base/managing-office-time-holidays/

Email Settings

Email settings

The UCM’s email module can send out alert event emails, fax (Fax-to-Email), voicemail (Voicemail-to-Email), etc. Email settings can be configured in System Settings->Email Settings.

Table 20: Email Settings

TLS Enable

Enable or disable TLS during transferring/submitting your Email to another SMTP server. The default setting is “Yes”.

Type

Select Email type.

  • MTA: Mail Transfer Agent. UCM will act as a virtual mail server, and emails will be sent from the configured domain. There is no need to configure SMTP server settings, and no credentials will be required. However, emails sent using MTA may be considered as spam by destination SMPT servers.
  • Client: UCM will send emails to the configured SMTP server to forward to the final destination. Credentials for the SMTP server will be required.

Domain

Specify the domain name to be used in the Email when using type “MTA”.

Server

Specify the SMTP server when using type “Client”.

Username

Username is required when using type “Client”. This is typically the email address.

Password

Password for the entered Username.

Enable Email-to-Fax

Toggles the Email-to-Fax feature.

If enabled, the UCM will monitor the configured email inbox (using provided [Username] and [Password]) for emails with the subject “SendFaxMail To XXX” or “XXX”.

Subject example: SendFaxMail To 7200 Or 7200.

The UCM will extract the attachments of detected emails and send it to the XXX extension by fax. The attachment must be in PDF/TIF/TIFF format.

Note: This field will appear when using Type “Client”.

Email-to-Fax Blacklist/Whitelist

Enables the Email to fax Blacklist/Whitelist functionality.

Email-to-Fax Subject

Specify the Email subject format for fax sending, the subject can be either “SendFaxMail To XXX” or “XXX” with XXX the fax number.

Internal Blacklist/Whitelist

Specify the Email address blacklist/whitelist for local extensions. This feature prevents faxing from unauthorized email addresses. The internal list includes only contacts with local extensions.

External Blacklist/Whitelist

Specify the Email address blacklist/whitelist for non-local contacts. This feature prevents faxing from unauthorized email addresses. The external list is for non-local contacts.

Note: Multiple addresses can be separated with semicolon (;) i.e. ” XXX;YYY “.

POP/POP3 Server Address

Configure the POP/POP3 server address for the configured username
Example: pop.gmail.com

POP/POP3 Server Port

Configure the POP/POP3 server port for the configured username
Example: 995

Display Name

Specify the display name in the FROM header in the Email.

Sender

Specify the sender’s Email address.

For example: pbx@example.mycompany.com.

The following figure shows example email settings, where 192.168.6.202 is the SMTP server.

Figure 50: UCM6200 Email Settings

Once configuration is complete, click on the Save button first and then the Test button to verify that the settings work.

Email Templates

The UCM provides various email templates for different email notifications. Email templates can be accessed from System Settings->Email Settings->Email Templates.

Figure 51: Email Templates

Press on to upload pictures to be used on email templates.

Press to reset all email templates to default ones.

To configure the email template, click the button under Options column, and edit the template as desired.

Figure 52: Conference Schedule Template

  • Users have the ability to preview mail sample by clicking on .
  • Click on in order to restore the default email template.
  • Finally, users can click on to upload a custom picture to the email template to display their own logo in the sent mails for example

Email Send Log

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

Figure 53: Email Send log

Table 21: Email Log

Field

Description

Start Time

Enter the start time for filter

End Time

Enter the end time for filter

Receivers

Enter the email recipient, while searching for multiple recipients, please separate then with comma and no spaces.

Send Result

Enter the status of the send result to filter with

Return Code

Enter the email code to filter with

Email Send Module

Select the email module to filter with from the drop-down list, which contains:

  • All Modules
  • Extension
  • Voicemail
  • Conference Schedule
  • Emergency calls
  • Video conference Schedule
  • User Password
  • Alert Events
  • CDR
  • Fax
  • Fax sending
  • Call Queue Statistics
  • Conference Reports
  • PMS
  • Test
  • Missed calls

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

Figure 54: Email Logs

Recordings Storage

The UCM supports both automatic and manual call recordings. Recordings can be saved to UCM’s local storage, external storage (SD/USB disks), and NAS.

Figure 55: PBX Settings🡪Recordings Storage

  • If Enable Auto Change is selected, the recording storage location will automatically change to NAS, USB Disk, or SD card if they are available. If all storage location types are available, the priority will be NAS->USB Disk->SD Card->Local storage.
  • If “Local” is selected, the recordings will be stored in UCM62xx internal storage.
  • If “USB Disk” or “SD Card” is selected, the recordings will be stored in the corresponding plugged in external storage device. Please note the options “USB Disk” and “SD Card” will be displayed only if they are plugged into the UCM62xx.

Once “USB Disk” or “SD Card” is selected, click on “OK”. The user will be prompted to confirm to copy the local files to the external storage device.

Figure 56: Recordings Storage Prompt Information

Click on “OK” to continue. The users will be prompted a new dialog to select the categories for the files to be copied over.

Figure 57: Recording Storage Category

On the UCM62xx, recording files are generated and exist in 3 categories: normal call recording files, conference recording files, and call queue recording files. Therefore, users have the following options when select the categories to copy the files to the external device:

  • Recording Files: Copy the normal recording files to the external device.
  • Conference: Copy the conference recording files to the external device.
  • Queue: Copy the call queue recording files to the external device.
  • All: Copy all recording files to the external device.

Provisioning

Overview

Grandstream SIP Devices can be configured via Web interface as well as via configuration file through TFTP/HTTP/HTTPS download. All Grandstream SIP devices support a proprietary binary format configuration file and XML format configuration file. The UCM6200 provides a Plug and Play mechanism to auto-provision the Grandstream SIP devices in a zero-configuration manner by generating XML config file and having the phone to download it within LAN area. This allows users to finish the installation with ease and properly manage SIP devices.

To provision a phone, three steps are involved, i.e., discovery, configuration and provisioning. This section explains how Zero Config works on the UCM6200. The settings for this feature can be accessed via Web GUI🡪Value-added Features🡪Zero Config.

Configuration Architecture for End Point Device

Started from firmware version 1.0.7.10, the end point device configuration in zero config is divided into the following three layers with priority from the lowest to the highest:

  • Global

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

  • Model

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

  • Device

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

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

Figure 58: Zero Config Configuration Architecture for End Point Device

The configuration options in model layer and device layer have all the option in global layers already, i.e., the options in global layer is a subset of the options in model layer and device layer. If an option is set in all three layers with different values, the highest layer value will override the value in lower layer. For example, if the user selects English for Language setting in Global Policy and Spanish for Language setting in Default Model Template, the language setting on the device to be provisioned will use Spanish as model layer has higher priority than global layer. To sum up, configurations in higher layer will always override the configurations for the same options/fields in the lower layer when presented at the same time.

After understanding the zero-config configuration architecture, users could configure the available options for end point devices to be provisioned by the UCM6200 by going through the three layers. This configuration architecture allows users to set up and manage the Grandstream end point devices in the same LAN area in a centralized way.

Auto Provisioning Settings

By default, the Zero Config feature is enabled on the UCM6200 for auto provisioning. Three methods of auto provisioning are used.

Zero config

Figure 59: UCM6200 Zero Config

  • SIP SUBSCRIBE

When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The UCM6200 discovers it and then sends a NOTIFY with the XML config file URL in the message body. The phone will then use the path to download the config file generated in the UCM6200 and take the new configuration.

  • DHCP OPTION 66

Route mode need to be set to use this feature. When the phone restarts (by default DHCP Option 66 is turned on), it will send out a DHCP DISCOVER request. The UCM6200 receives it and returns DHCP OFFER with the config server path URL in Option 66, for example, https://192.168.2.1:8089/zccgi/. The phone will then use the path to download the config file generated in the UCM6200.

  • mDNS

When the phone boots up, it sends out mDNS query to get the TFTP server address. The UCM6200 will respond with its own address. The phone will then send TFTP request to download the XML config file from the UCM6200.

To start the auto provisioning process, under Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config Settings, fill in the auto provision information.

Figure 60: Auto Provision Settings

Table 22: Auto Provision Settings

Enable Zero Config

Enable or disable the zero-config feature on the PBX. The default setting is enabled.

Enable Automatic Configuration Assignment

By default, this is disabled. If disabled, when SIP device boots up, the UCM6200 will not send the SIP device the URL to download the config file and therefore the SIP device will not be automatically provisioned by the UCM6200.

Note:

When disabled, SIP devices can still be provisioned by manually sending NOTIFY from the UCM6200 which will include the XML config file URL for the SIP device to download.

Automatically Assign Extension

If enabled, when the device is discovered, the PBX will automatically assign an extension within the range defined in “Zero Config Extension Segment” to the device. The default setting is disabled.

Zero Config Extension Segment

Click on the link “Zero Config Extension Segment” to specify the extension range to be assigned if “Automatically Assign Extension” is enabled. The default range is 5000-6299. Zero Config Extension Segment range can be defined in Web GUI🡪PBX Settings🡪General Settings🡪General page🡪Extension Preference section: “Auto Provision Extensions”.

Enable Pick Extension

If enabled, the extension list will be sent out to the device after receiving the device’s request. This feature is for the GXP series phones that support selecting extension to be provisioned via phone’s LCD. The default setting is disabled.

Pick Extension Segment

Click on the link “Pick Extension Segment” to specify the extension list to be sent to the device. The default range is 4000 to 4999. Pick Extension Segment range can be defined in Web GUI🡪PBX Settings🡪General Settings🡪General page🡪Extension Preference section: “Pick Extensions”.

Pick Extension Period (hour)

Specify the number of minutes to allow the phones being provisioned to pick extensions.

Subnet Whitelist

This feature allows the UCM to provision devices in different subnets other than UCM network.

Enter subnets IP addresses to allow devices within these subnets to be provisioned. The syntax is <IP>/<CIDR>.

Examples:

10.0.0.1/8

192.168.6.0/24

Note: Only private IP ranges (10.0.0.0 | 172.16.0.0 | 192.168.0.0) are supported.

Please make sure an extension is manually assigned to the phone or “Automatically Assign Extension” is enabled during provisioning. After the configuration on the UCM6200 Web GUI, click on “Save” and “Apply Changes”. Once the phone boots up and picks up the config file from the UCM6200, it will immediately apply the configuration.

Discovery

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

  • PING
  • ARP
  • SIP Message (NOTIFY)

Click on “Auto Discover” under Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config, fill in the “Scan Method” and “Scan IP”. The IP address segment will be automatically filled in based on the network mask detected on the UCM6200. If users need scan the entire network segment, enter 255 (for example, 192.168.40.255) instead of a specific IP address. Then click on “Save” to start discovering the devices within the same network. To successfully discover the devices, “Zero Config” needs to be enabled on the UCM6200 Web GUI🡪Value-added Features🡪Zero Config🡪Auto Provisioning Settings.

Figure 61: 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 /Reboot /Access Device WebGUI) are displayed in the list.


Auto Discover can also search for devices located on other subnets, in condition for the subnet to be added under Zero Config Settings> Subnet Whitelist. The method allowed to auto discover other subnets then the UCM’s is SIP-Message like shown below.

Figure 62: Auto Discover other subnets

Figure 63: Discovered Devices

Uploading Devices List

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

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

Figure 64: Device list – CSV file sample

Managing discovered devices

  • Sorting: Press or to sort per MAC Address, IP Address, Version, Vendor, Model or Create Config columns from lower to higher or higher to lower, respectively.
  • Filter: Select a filter to display corresponding results.
  • All: Display all discovered devices.
  • Scan Results: Display only manually discovered devices. [Discovery]
  • IP Address: Enter device IP and press Search button.
  • MAC Address: Enter device MAC and press Search button.
  • Model: Enter a model name and press Search button. Example: GXP2130.

Figure 65: Managing Discovered Devices

From the main menu of zero config, users can perform the following operations:

  • Click on in order to access to the discovery menu as shown on [Discovery] section.
  • Click on to add a new device to zero config database using its MAC address.
  • Click on to delete selected devices from the zero-config database.
  • Click on to modify selected devices.
  • Click on to batch update a list of devices, the UCM on this case will send SIP NOTIFY message to all selected devices in order to update them at once.
  • Click on to reboot selected devices (the selected devices, should have been provisioned with extensions since the phone will authenticate the server which is trying to send it reboot command).
  • Click on to clear all devices configurations.
  • Click on to upload CSV file containing list of devices.
  • Click on A close up of a logo

Description generated with very high confidence to copy configuration from one device to another. This can be useful for easily replace devices and note that this feature works only between devices of same model.

All these operations will be detailed on the next sections.

Global configuration

Global policy

Global configuration will apply to all the connected Grandstream SIP end point devices in the same LAN with the UCM6200 no matter what the Grandstream device model it is. It is divided into two levels:

  • Web GUI🡪Value-added Features🡪Zero Config🡪Global Policy
  • Web GUI🡪Value-added Features🡪Zero Config🡪Global Templates.
  • Global Templates configuration has higher priority to Global Policy configuration.

Global Policy can be accessed in Web GUI🡪Value-added Features🡪Zero Config🡪Global Policy page. On the top of the configuration table, users can select category in the “Options” dropdown list to quickly navigate to the category. The categories are:

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

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

Figure 66: Global Policy Categories

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

Table 23: Global Policy Parameters – Localization

Language settings

Language

Select the LCD display language on the SIP end device.

Date and Time

Date Format

Configure the date display format on the SIP end device’s LCD.

Time Format

Configure the time display in 12-hour or 24-hour format on the SIP end device’s LCD.

NTP Server

Configure the URL or IP address of the NTP server. The SIP end device may obtain the date and time from the server.

Time Zone

Configure the time zone used on the SIP end device.

Table 24: Global Policy Parameters – Phone Settings

Default Call Settings

Dial Plan

Configure the default dial plan rule. For syntax and examples, please refer to user manual of the SIP devices to be provisioned for more details.

Enable Call Features

When enabled, “Do Not Disturb”, “Call Forward” and other call features can be used via the local feature code on the phone. Otherwise, the ITSP feature code will be used.

Use # as Dial Key

If set to “Yes”, pressing the number key “#” will immediately dial out the input digits.

Auto Answer by Call-info

If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep, based on the SIP Call-Info header sent from the server/proxy.

The default setting is enabled.

NAT Traversal

Configure if NAT traversal mechanism is activated.

User Random Port

If set to “Yes”, this parameter will force random generation of both the local SIP and RTP ports.

General Settings

Call Progress Tones

Configure call progress tones including ring tone, dial tone, second dial tone, message waiting tone, ring back tone, call waiting tone, busy tone and reorder tone using the following syntax:

f1=val, f2=val[, c=on1/ off1[- on2/ off2[- on3/ off3]]];

  • Frequencies are in Hz and cadence on and off are in 10ms).
  • “on” is the period (in ms) of ringing while “off” is the period of silence. Up to three cadences are supported.
  • Please refer to user manual of the SIP devices to be provisioned for more details

HEADSET Key Mode

Select “Default Mode” or “Toggle Headset/Speaker” for the Headset key. Please refer to user manual of the SIP devices to be provisioned for more details.

Table 25: Global Policy Parameters – Contact List

LDAP Phonebook

Source

Select “Manual” or “PBX” as the LDAP configuration source.

  • If “Manual” is selected, the LDAP configuration below will be applied to the SIP end device.
  • If “PBX” is selected, the LDAP configuration built-in from UCM6200 Web GUI🡪System Settings🡪LDAP Server will be applied.

Address

Configure the IP address or DNS name of the LDAP server.

Port

Configure the LDAP server port. The default value is 389.

Base DN

This is the location in the directory where the search is requested to begin. Example:

  • dc=grandstream, dc=com
  • ou=Boston, dc=grandstream, dc=com

Username

Configure the bind “Username” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

Password

Configure the bind “Password” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

Number Filter

Configure the filter used for number lookups. Please refer to user manual for more details.

Name Filter

Configure the filter used for name lookups. Please refer to user manual for more details.

Version

Select the protocol version for the phone to send the bind requests. The default value is 3.

Name Attribute

Specify the “name” attributes of each record which are returned in the LDAP search result.

Example:

gn

cn sn description

Number Attribute

Specify the “number” attributes of each record which are returned in the LDAP search result.

Example:

telephoneNumber

telephoneNumber Mobile

Display Name

Configure the entry information to be shown on phone’s LCD. Up to 3 fields can be displayed. 

Example:

%cn %sn %telephoneNumber

Max Hits

Specify the maximum number of results to be returned by the LDAP server. Valid range is 1 to 3000. The default value is 50.

Search Timeout

Specify the interval (in seconds) for the server to process the request and client waits for server to return. Valid range is 0 to 180. Default value is 30.

Sort Results

Specify whether the searching result is sorted or not. Default setting is No.

Incoming Calls

Configure to enable LDAP number searching when receiving calls. The default setting is No.

Outgoing Calls

Configure to enable LDAP number searching when making calls. The default setting is No.

Lookup Display Name

Configures the display name when LDAP looks up the name for incoming call or outgoing call.

It must be a subset of the LDAP Name Attributes.

XML Phonebook

Phonebook XML Server

Select the source of the phonebook XML server.

  • Disable

Disable phonebook XML downloading.

  • Manual

Once selected, users need specify downloading protocol HTTP, HTTPS or TFTP and the server path to download the phonebook XML file. The server path could be IP address or URL, with up to 256 characters.

  • Local UCM Server

Once selected, click on the Server Path field to upload the phonebook XML file. Please note after uploading the phonebook XML file to the server, the original file name will be used as the directory name and the file will be renamed as phonebook.xml under that directory.

Phonebook Download Interval

Configure the phonebook download interval (in Minute). If set to 0, automatic download will be disabled. Valid range is 5 to 720.

Remove manually-edited entries on download

If set to “Yes”, when XML phonebook is downloaded, manually added entries will be removed.

Table 26: Global Policy Parameters – Maintenance

Upgrade and Provision

Firmware Source

UCM can provision the firmware source to devices. The following options are available:

Select a source to get the firmware file:

  • URL

If selected, complete the configuration for the following four parameters: “Upgrade Via”, “Server Path”, “File Prefix” and “File Postfix”.

  • Local UCM Server

Firmware can be uploaded to the UCM6200 internal storage for firmware upgrade. If selected, click on “Manage Storage” icon next to “Directory” option, upload firmware file and select directory for the end device to retrieve the firmware file.

  • Local USB Media

If selected, the USB storage device needs to be plugged into the UCM6200 and the firmware file must be put under a folder named “ZC_firmware” in the USB storage root directory.

  • Local SD Card Media

If selected, an SD card needs to be plugged into the UCM6200 and the firmware file must be put under a folder named “ZC_firmware” in the USB storage root directory.

Upgrade via

When URL is selected as firmware source, configure upgrade via TFTP, HTTP or HTTPS.

Server Path

When URL is selected as firmware source, configure the firmware upgrading server path.

File Prefix

When URL is selected as firmware source, configure the firmware file prefix. If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone, if URL is selected as firmware source.

File Postfix

When URL is selected as firmware source, configure the firmware file postfix. If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone.

Allow DHCP Option 43/66

If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can be redirected.

Automatic Upgrade

If enabled, the end point device will automatically upgrade if a new firmware is detected. Users can select automatic upgrading by day, by week or by minute.

  • By week

Once selected, specify the day of the week to check HTTP/TFTP server for firmware upgrades or configuration files changes.

  • By day

Once selected, specify the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes.

  • By minute

Once selected, specify the interval X that the SIP end device will request for new firmware every X minutes.

Firmware Upgrade Rule

Specifies how firmware upgrade and provision requests will be sent.

Web Access

Admin Password

Configure the administrator password for admin level login.

End-User Password

Configure the end-user password for the end user level login.

Web Access Mode

Select HTTP or HTTPS as the web access protocol.

Web Server Port

Configure the port for web access. The valid range is 1 to 65535.

Security

Disable Telnet/SSH

Enable Telnet/SSH access for the SIP end device. If the SIP end device supports Telnet access, this option controls the Telnet access of the device; if the SIP end device supports SSH access, this option controls the SSH access of the device.

Syslog

Syslog Server

Configure the URL/IP address for the syslog server.

Syslog Level

Select the level of logging for syslog.

Send SIP Log

Configures whether SIP messages will be included in the syslog.

Table 27: Global Policy Parameters – Network Settings

Basic Settings

IP Address

Configures how the SIP endpoint will obtain IP addresses. The following options are available:

  • DHCP

Once selected, users can specify the Host Name (option 12) of the SIP end device as DHCP client, and Vendor Class ID (option 60) used by the client and server to exchange vendor class ID information.

  • PPPoE

Once selected, users need specify the Account ID, Password and Service Name for PPPoE.

Advanced Setting

Layer 3 QoS

Define the Layer 3 QoS parameter. This value is used for IP Precedence, Diff-Serv or MPLS. Valid range is 0-63.

Layer 2 QoS Tag

Assign the VLAN Tag of the Layer 2 QoS packets. Valid range is 0 -4095.

Layer 2 QoS Priority Value

Assign the priority value of the Layer 2 QoS packets. Valid range is 0-7.

STUN Server

Configure the IP address or Domain name of the STUN server. Only non-symmetric NAT routers work with STUN.

Keep Alive Interval

Specify how often the phone will send a blank UDP packet to the SIP server in order to keep the “ping hole” on the NAT router to open. Valid range is 10-160.

Table 28: Global Policy Parameters – Customization

Wallpaper

Screen Resolution 1024 x 600

Check this option if the SIP endpoint uses a 1024×600 resolution wallpaper.

  • Source

Configure the location where wallpapers are stored.

  • File

If “URL” is selected as source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as source, click to upload wallpaper file to the UCM6200.

Screen Resolution 800 x 400

Check this option if the SIP endpoint uses a 800×400 resolution wallpaper.

  • Source

Configure the location where wallpapers are stored.

  • File

If “URL” is selected as source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as source, click to upload wallpaper file to the UCM6200.

Screen Resolution 480 x 272

Check this option if the SIP endpoint uses a 480×272 resolution wallpaper.

  • Source

Configure the location where wallpapers are stored.

  • File

If “URL” is selected as source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as source, click to upload wallpaper file to the UCM6200.

Screen Resolution 320 x 240

Check this option if the SIP endpoint uses a 320×240 resolution wallpaper.

  • Source

Configure the location where wallpapers are stored.

  • File

If “URL” is selected as source, specify the URL of the wallpaper file. If “Local UCM Server” is selected as source, click to upload wallpaper file to the UCM6200.

Table 29: Global Policy Parameters – Communication Settings

Email Settings

SMTP Settings

Check this option to configure the email settings that will be sent to the provisioned phones:

  • Server

IP address of the SMTP server

  • Port

SMTP server port

  • From E-Mail address

Email address

  • Sender Username

Username of the sender

  • Password Recovery Email

Email where recovered password will be sent

  • Alarm receive Email 1

Email address where alarms notifications will be sent

  • Alarm receive Email 2

Email address where alarms notifications will be sent

  • Enable SSL

Enable SSL protocol for SMTP

FTP

FTP

Check this option to configure the FTP settings that will be sent to the provisioned phones:

  • Storage Server Type

Either FTP or Central Storage

  • Server

FTP server address

  • Port

FTP port to be used

  • Username

FTP username

  • Path

FTP Directory path

Global Templates

Global Templates can be accessed in Web GUI🡪Value-added Features🡪Zero Config🡪Global Templates. Users can create multiple global templates with different sets of configurations and save the templates. Later on, when the user configures the device in Edit Device dialog🡪Advanced Settings, the user can select to use one of the global templates for the device. Please refer to section [Manage Devices] for more details on using the global templates.

When creating global template, users can select the categories and the parameters under each category to be used in the template. The global policy and the selected global template will both take effect when generating the config file. However, the selected global template has higher priority to the global policy when it comes to the same setting option/field. If the same option/field has different value configured in the global policy and the selected global template, the value for this option/field in the selected global template will override the value in global policy.

Click on “Create New Template” to add a global template. Users will see the following configurations.

Table 30: Create New Template

Template Name

Create a name to identify this global template.

Description

Provide a description for the global template. This is optional.

Active

Check this option to enable the global template.

  • Click on to edit the global template.

The window for editing global template is shown in the following figure. In the “Options” field, after entering the option name key word, the options containing the key word will be listed. Users could then select the options to be modified and click on “Add Option” to add it into the global template.

Figure 67: Edit Global Template

The added options will show in the list. Users can then enter or select value for each option to be used in the global template. On the left side of each added option, users can click on to remove this option from the template. On the right side of each option, users can click on to reset the option value to the default value.

Click on “Save” to save this global template.

  • The created global templates will show in the Web GUI🡪Value-added Features🡪Zero Config🡪Global Templates page. Users can click on to delete the global template or click on “Delete Selected Templates” to delete multiple selected templates at once.
  • Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected templates.

Model configuration

Model templates

Model layer configuration allows users to apply model-specific configurations to different devices. Users could create/edit/delete a model template by accessing Web GUI, page Value-added Features🡪Zero Config🡪Model Templates. If multiple model templates are created and enabled, when the user configures the device in Edit Device dialog🡪Advanced Settings, the user can select to use one of the model templates for the device. Please refer to section [Manage Devices] for more details on using the model template.

For each created model template, users can assign it as default model template. If assigned as default model template, the values in this model template will be applied to all the devices of this model. There is always only one default model template that can be assigned at one time on the UCM6200.

The selected model template and the default model template will both take effect when generating the config file for the device. However, the model template has higher priority to default model template when it comes to the same setting option/field. If the same option/field has different value configured in the default model template and the selected model template, the value for this option/field in the selected model template will override the value in default model template.

  • Click on “Create New Template” to add a model template.

Table 31: Create New Model Template

Model

Select a model to apply this template. The supported Grandstream models are listed in the dropdown list for selection.

Template Name

Create a name for the model template.

Description

Enter a description for the model template. This is optional.

Default Model Template

Select to assign this model template as the default model template. The value of the option in default model template will be overwritten if other selected model template has a different value for the same option.

Active

Check this option to enable the model template.

  • Click on to edit the model template.

The editing window for model template is shown in the following figure. In the “Options” field, enter the option name key word, the option that contains the key word will be listed. User could then select the option and click on “Add Option” to add it into the model template.

Once added, the option will be shown in the list below. On the left side of each option, users can click on to remove this option from the model template. On the right side of each option, users can click on to reset the option to the default value.

User could also click on “Add New Field” to add a P value number and the value to the configuration. The following figure shows setting P value “P1362” to “en”, which means the display language on the LCD is set to English. For P value information of different models, please refer to configuration template here https://content.grandstream.com/hubfs/Grandstream_Feb_2021/Zip%20File/config-template.zip

Note: Some devices use Alias and not P-values.

Figure 68: Edit Model Template

  • Click on Save when done. The model template will be displayed on Web GUI🡪Value-added Features🡪Zero Config🡪Model Templates page.
  • Click on to delete the model template or click on “Delete Selected Templates” to delete multiple selected templates at once.
  • Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected model templates.

Model Update

UCM6200 zero config feature supports provisioning all models of Grandstream SIP end devices including OEM device models.

OEM Models

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

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

Figure 69: OEM Models

Model Template Package List

Templates for most of the Grandstream models are built in with the UCM6200 already. Templates for GS Wave and Grandstream surveillance products require users to download and install under Web GUI🡪Value-added Features🡪Zero Config🡪Model Update first before they are available in the UCM6200 for selection. After downloading and installing the model template to the UCM6200, it will show in the dropdown list for “Model” selection when editing the model template.

  • Click on to download the template.
  • Click on to upgrade the model template. Users will see this icon available if the device model has template updated in the UCM6200.

Figure 70: Template Management

Upload Model Template Package

In case the UCM6200 is placed in the private network and Internet access is restricted, users will not be able to get packages by downloading and installing from the remote server. Model template packages can be manually uploaded to the UCM.. Please contact Grandstream customer support if the model package is needed for manual uploading.

Figure 71: Upload Model Template Manually

Device Configuration

On Web GUI, page Value-added Features🡪Zero Config🡪Zero Config, users could create new device, delete existing device(s), make special configuration for a single device, or send NOTIFY to existing device(s).

Create New Device

Besides configuring the device after the device is discovered, users could also directly create a new device and configure basic settings before the device is discovered by the UCM6200. Once the device is plugged in, it can then be discovered and provisioned. This gives the system administrator adequate time to set up each device beforehand.

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

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

Figure 72: Create New Device

Manage Devices

The device manually created or discovered from Auto Discover will be listed in the Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config page. Users can see the devices with their MAC address, IP address, vendor, model etc.

Figure 73: Manage Devices

  • Click on to access the Web GUI of the phone.
  • Click on to edit the device configuration.

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

Figure 74: Edit Device

A preview of the “Advanced” settings is shown in the above figure. There are five levels configurations as described in (1) (2) (3) (4) (5) below, with priority from the lowest to the highest. The configurations in all levels will take effect for the device. If there are same options existing in different level configurations with different value configured, the higher-level configuration will override the lower level configuration.

  1. Global Policy

This is the lowest level configuration. The global policy configured in Web GUI🡪Value-added Features🡪Zero Config🡪Global Policy will be applied here. Clicking on “Modify Global Policy” to redirect to page Value-added Features🡪Zero Config🡪Global Policy.

  1. Global Templates

Select a global template to be used for the device and click on to add. Multiple global templates can be selected, and users can arrange the priority by adjusting orders via and . All the selected global templates will take effect. If the same option exists on multiple selected global templates, the value in the template with higher priority will override the one in the template with lower priority. Click on to remove the global template from the selected list.

  1. Default Model Template

Default Model Template will be applied to the devices of this model. Default model template can be configured in model template under Web GUI🡪Value-added Features🡪Zero Config🡪Model Templates page. Please see default model template option in [Table 31: Create New Model Template].

  1. Model Templates

Select a model template to be used for the device and click on to add. Multiple global templates can be selected, and users can arrange the priority by adjusting orders via and . All the selected model templates will take effect. If the same option exists on multiple selected model templates, the value in the template with higher priority will override the one in the template with lower priority. Click on to remove the model template from the selected list.

  1. Customize Device Settings

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

Figure 75: Edit Customize Device Settings

Scroll down in the dialog to view and edit the device-specific options. If the users would like to add more options which are not in the pre-defined list, click on “Add New Field” to add a P value number and the value to the configuration. The above figure shows setting P value “P1362” to “en”, which means the display language on the LCD is set to English. The warning information on right tells that the option matching the P value number exists and clicking on it will lead to the matching option. For P value information of different models, please refer to configuration template here http://www.grandstream.com/sites/default/files/Resources/config-template.zip.

  • Select multiple devices that need to be modified and then click on to batch modify devices.

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

Figure 76: Modify Selected Devices – Same Model

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

Figure 77: Modify Selected Devices – Different Models

Note:

Performing batch operation will override all the existing device configuration on the page.

After the above configurations, save the changes and go back to Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config page. Users could then click on to send NOTIFY to the SIP end point device and trigger the provisioning process. The device will start downloading the generated configuration file from the URL contained in the NOTIFY message.

Figure 78: Device List in Zero Config

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

Example Application

Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1 GXV3370 phone used by customer support supervisor. 3 of the 8 customer support members speak Spanish and the rest speak English. We could deploy the following configurations to provisioning the office phones for the customer support team.

  1. Go to Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config Settings, select “Enable Zero Config”.
  2. Go to Web GUI🡪Value-added Features🡪Zero Config🡪Global Policy, configure Date Format, Time Format and Firmware Source as follows.

Figure 79: Zero Config Sample – Global Policy

  1. Go to Web GUI🡪Value-added Features🡪Zero Config🡪Model Templates, create a new model template “English Support Template” for GXP2140. Add option “Language” and set it to “English”. Then select the option “Default Model Template” to make it the default model template.
  2. Go to Web GUI🡪Value-added Features🡪Zero Config🡪Model Templates, create another model template “Spanish Support Template” for GXP2140. Add option “Language” and set it to “Español”.
  3. After 9 devices are powered up and connected to the LAN network, use “Auto Discover” function or “Create New Device” function to add the devices to the device list on Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config.
  4. On Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config page, users could identify the devices by their MAC addresses or IP addresses displayed on the list. Click on to edit the device settings.
  5. For each of the 5 phones used by English speaking customer support, in “Basic” settings select an available extension for account 1 and click on “Save”. Then click on “Advanced” settings tab to bring up the following dialog. Users will see the English support template is applied since this is the default model template. A preview of the device settings will be listed on the right side.

Figure 80: Zero Config Sample – Device Preview 1

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

Figure 81: Zero Config Sample – Device Preview 2

Select “Spanish Support Template” in “Model Template”. The preview of the device settings is displayed on the right side and we can see the language is set to “Español” since Model Template has the higher priority for the option “Language”, which overrides the value configured in default model template.

  1. For the GXV3370 used by the customer support supervisor, select an available extension for account 1 on “Basic” settings and click on “Save”. Users can see the preview of the device configuration in “Advanced” settings. There is no model template configured for GXV3370.

Figure 82: Zero Config Sample – Device Preview 3

  1. Click on “Apply Changes” to apply saved changes.
  2. On the Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config page, click on to send NOTIFY to trigger the device to download config file from UCM6200.

Now all the 9 phones in the network will be provisioned with a unique extension registered on the UCM6200. 3 of the phones will be provisioned to display Spanish on LCD and the other 5 will be provisioned to display English on LCD. The GXV3370 used by the supervisor will be provisioned to use the default language on LCD display since it is not specified in the global policy.

Extensions

Create New User

Create New SIP Extension

To create a new SIP extension, navigate to Extension/Trunk->Extensions and click on the Add button. The following window will appear:

Figure 83: Create New Device

Extension options are divided into four categories:

  • Basic Settings
  • Media
  • Features
  • Specific Time
  • Follow me

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

Table 32: SIP Extension Configuration Parameters – Basic Settings

General

Extension

The extension number associated with the user.

Note: This filed supports (+) sign.

CallerID Number

Configure the CallerID Number that would be applied for outbound calls from this user.

Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider.

Permission

Assign permission level to the user. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level. The default setting is “Internal”.

Note: Users need to have the same level as or higher level than an outbound rule’s privilege to make outbound calls using this rule.

SIP/IAX Password

Configure the password for the user. A random secure password will be automatically generated. It is recommended to use this password for security purpose.

Auth ID

Configure the authentication ID for the user. If not configured, the extension number will be used for authentication.

Note: This filed supports (+) sign.

Voicemail

Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”.

  • Disable Voicemail: Disable Voicemail.
  • Enable Local Voicemail: Enable voicemail for the user.
  • Enable Remote Voicemail: Forward the notify message from remote voicemail system for the user, and the local voicemail will be disabled. Note: Remote voicemail feature is used only for Infomatec (Brazil).

Voicemail Password

Configure the password to access the extension’s voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported.

Skip Voicemail Password Verification

If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default.

Send Voicemail to Email

If enabled, the voicemail will be attached to the email notification that is sent when a voicemail is received. Note: When set to “Default”, the global settings in Call Features->Voicemail->Voicemail Email Settings will be used.

Keep Voicemail after Emailing

Configure whether or not to retain the voicemail in local storage after sending the voicemail attachment in the email notification.

Enable Keep-alive

If enabled, empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is “Yes”.

Keep-alive Frequency

Configure the Keep-alive interval (in seconds) to check if the host is up. The default setting is 60 seconds.

Disable This Extension

If selected, this extension will be disabled on the UCM6200.

Note: The disabled extension still exists on the PBX but cannot be used on the end device.

User Settings

First Name

Configure the user’s first name. This field supports alphanumeric characters, underscores (_), and periods.

Last Name

Configures the user’s last name. This field supports alphanumeric characters, underscores (_), and periods.

Email Address

Configure the user’s email address. Email notifications will be sent to this address.

User Password

Configure the password for user portal access. A random password is automatically generated by default and is highly recommended for security.

Language

Select the voice prompt language that will be used for this extension. By default, the selected voice prompt language under PBX Settings->Voice Prompt->Language Settings will be used. To add more supported languages, please download the voice prompt language packages in the same page.

Concurrent Registrations

The maximum allowed number of endpoint registrations to this extension. Default value is 1.

Mobile Phone Number

Configure a phone number for the extension. Users can dial the Direct Dial Mobile Phone Prefix feature code (*88 by default) + extension number to directly dial this number. Example: Dial *881000 to call the mobile number associated with extension 1000.

Table 33: SIP Extension Configuration Parameters – Media

SIP Settings

NAT

Use NAT when the UCM6200 is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it is related to NAT configuration or Firewall’s support of SIP and RTP ports. The default setting is enabled.

Enable Direct Media

By default, the UCM6200 will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to negotiate with the endpoints to route the media stream directly. It is not always possible for the UCM6200 to negotiate endpoint-to-endpoint media routing. The default setting is “No”.

DTMF Mode

Select DTMF mode for the user to send DTMF. The default setting is “RFC4733”. If “Info” is selected, SIP INFO message will be used. If “Inband” is selected, 64-kbit PCMU and PCMA are required. When “Auto” is selected, RFC4733 will be used if offered, otherwise “Inband” will be used.

TEL URI

If the phone has an assigned PSTN telephone number, this field should be set to “User=Phone”. “User=Phone” parameter will be attached to the Request-Line and “TO” header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel” will be used instead of “SIP” in the SIP request.

Alert-Info

Configure the Alert-Info, when UCM6200 receives an INVITE request, the Alert-Info header field specifies an alternative ring tone to the UAS.

Enable T.38 UDPTL

Enable or disable T.38 UDPTL support.

SRTP

Enable SRTP for the call. The default setting is disabled.

Fax Mode

Select Fax mode. The default setting is “None”.

  • None: Disable Fax.
  • Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38.

Fax to Email

  • Yes – Allow Fax to Email for this extension. Faxes will be sent to the user’s email address configured in the extension’s Basic Settings.
  • No – Do not send any faxes to the user’s email address configured in the extension’s Basic Settings.

ACL Policy

Access Control List manages the IP addresses that can register to this extension.

  • Allow All: Any IP address can register to this extension.
  • Local Network Address: Only IP addresses in the configured network segments can register to this extension.

Local Network Address

Specifies allowed IP address or networks from where the extension can be registered. Up to 10 entries are allowed.

Format: “xxx.xxx.xxx.xxx”, “xxx.xxx.xxx.xxx/32”, “[::]” or “[::]/128”.

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, OPUS, iLBC, ADPCM, H.264, H.263, H.263p and RTX.

Table 34: SIP Extension Configuration Parameters – Features

Call Transfer

Presence Status

Select which presence status to set for the extension and configure call forward conditions for each status. Six possible options are possible: “Available”, “Away”, “Chat”, “Custom”, “DND” and “Unavailable”.

Call Forward Unconditional

Enable and configure the Call Forward Unconditional target number. Available options for target number are:

  • None”: Call forward deactivated.

  • Extension”: Select an extension from dropdown list as CFU target.

  • Custom Number”: Enter a customer number as target. For example: *97.

  • Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension.

  • “Ring Group”: Select a ring group from dropdown list as CFU target.

  • “Queues”: Select a queue from dropdown list as CFU target.

  • “Voicemail Group”: Select a voicemail group from dropdown list as CFU target.

The default setting is “None”.

CFU Time Condition

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes: 

  • Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific Time can be configured in the Specific Time tab.

  • Office Time and Holiday can be configured in System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward No Answer

Configure the Call Forward No Answer target number. Available options for target number are:

  • None”: Call forward deactivated.

  • Extension”: Select an extension from dropdown list as CFN target.

  • Custom Number”: Enter a customer number as target. For example: *97.

  • Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension.

  • Ring Group”: Select a ring group from dropdown list as CFN target.

  • Queues”: Select a queue from dropdown list as CFN target.

  • Voicemail Group”: Select a voicemail group from dropdown list as CFN target.

CFN Time Condition

Select time condition for Call Forward No Answer. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes: 

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific Time can be configured in the Specific Time tab.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward Busy

Configure the Call Forward Busy target number. Available options for target number are:

  • None”: Call forward deactivated.

  • Extension”: Select an extension from dropdown list as CFB target.

  • Custom Number”: Enter a customer number as target. For example: *97.

  • Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension.

  • Ring Group”: Select a ring group from dropdown list as CFB target.

  • Queues”: Select a queue from dropdown list as CFB target.

  • “Voicemail Group”: Select a voicemail group from dropdown list as CFB target.

CFB Time Condition

Select time condition for Call Forward Busy. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes: 

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Do Not Disturb

If enabled the extension will ignore all incoming calls

DND Time Condition

Select time condition for Do Not Disturb. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes:

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

DND Whitelist

If DND is enabled, all calls to this extension will be rejected except the numbers listed on this list.

Note: The maximum number on the Whitelist is 10.

FWD Whitelist

If call forward is enabled, all calls to this extension will be forwarded except the calls coming from the specified numbers on this list.

Note: The Maximum number on the whitelist is 10.

CC Settings

Enable CC

If enabled, UCM6200 will automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason. By default, it is disabled.

CC Mode

Two modes for Call Completion are supported:

  • Normal: This extension is used as ordinary extension.

  • For Trunk: This extension is registered from a PBX.

The default setting is “Normal”.

Note: The number of CC agents (for “Normal” mode) will now be limited by the extensions’ allowed number of concurrent registrations.

CC Max Agents

Configure the maximum number of CCSS agents which may be allocated for this channel (when CC Mode is set to “For Trunk”). In other words, this number serves as the maximum number of CC requests this channel is allowed to make. Min. value is 1.

CC Max Monitors

Configure the maximum number of monitor structures which may be created for this device (when CC Mode is set to “For Trunk”). This number tells how many callers may request CC services for a specific device at one time. The minimum value is 1.

Ring Simultaneously

Ring Simultaneously

Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.

External Number

Set the external number to be rang simultaneously. ‘-’ is the connection character which will be ignored.

This field accepts only letters, numbers, and special characters + = * #.

Time Condition for Ring Simultaneously

Ring the external number simultaneously along with the extension based on this time condition.

Use callee DOD on FWD or Ring Simultaneously

Use the DOD number when calls are being diverted/forwarded to external destinations or when ring simultaneous is configured.

Monitor privilege control

Allow call-barging

Add members from “Available Extensions” to “Selected Extensions” so that the selected extensions can spy on the used extension using feature code.

Seamless transfer privilege control

Allowed to seamless transfer

Any extensions on the UCM can perform seamless transfer. When using Pickup Incall feature, only extensions available on the “Selected Extensions” list can perform seamless transfer to the edited extension.

Other Settings

Ring Timeout

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM6200, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪General Settings: General Preference. The valid range is between 5 seconds and 600 seconds.

Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Auto Record

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.

Skip Trunk Auth

  • If set to ‘Yes’, users can skip entering the password when making outbound calls.

  • If set to ‘By Time’, users can skip entering the password when making outbound calls during the selected time condition.

  • If set to ‘No’, users will be asked to enter the password when making outbound calls.

Time Condition for Skip Trunk Auth

If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering password when making outbound calls.

Dial Trunk Password

Configure personal password when making outbound calls via trunk.

Support Hot-Desking Mode

If enabled, SIP Password will accept only alphabet characters and digits. Auth ID will be changed to the same as Extension.

Enable LDAP

If enabled, the extension will be added to LDAP Phonebook PBX list.

Enable WebRTC Support

Enable registration and call from Web RTC.

Music On Hold

Specify which Music On Hold playlist to suggest to the bridged channel when putting them on hold.

Enable Seamless Transfer

Enable the seamless transfer for this extension.

Permission

Set the permission for this extension when using the seamless transfer

Call Duration Limit

The maximum duration of call-blocking.

Maximum Call Duration

The maximum call duration (in seconds). The default value 0 means no limit.

Custom Call-info for Auto Answer

If enabled, when a call is sent to this extension from UCM, the SIP INVITE message will contain a Call-info header indicating auto answer.

Enable Call Waiting

If disabled, UCM will not ring the extension if it is already in a call and will give the caller a busy response.

Email Missed Calls

When enabled, the UCM sends a log of missed calls to the extension's email address.

Table 34: SIP Extension Configuration Parameters – Features

Specific Time

Time Condition

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

Table 35: SIP Extension Configuration Parameters – Specific Time

Create New IAX Extension

The UCM6200 supports Inter-Asterisk eXchange (IAX) protocol. IAX is used for transporting VoIP telephony sessions between servers and terminal devices. IAX is like SIP but also has its own characteristic. For more information, please refer to RFC 5465.

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

Table 36: IAX Extension Configuration Parameters – Basic Settings

General

Extension

The extension number associated with the user.

CallerID Number

Configure the CallerID Number that would be applied for outbound calls from this user.

Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider.

Permission

Assign permission level to the user. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level. The default setting is “Internal”.

Note: Users need to have the same level as or higher level than an outbound rule’s privilege to make outbound calls using this rule.

SIP/IAX Password

Configure the password for the user. A random secure password will be automatically generated. It is recommended to use this password for security purpose.

Voicemail

Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”.

  • Disable Voicemail: Disable Voicemail.
  • Enable Local Voicemail: Enable voicemail for the user.

Voicemail Password

Configure the password to access the extension’s voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported.

Skip Voicemail Password Verification

If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default.

Disable This Extension

If selected, this extension will be disabled on the UCM6200.

Note:

The disabled extension still exists on the PBX but cannot be used on the end device.

User Settings

First Name

Configure the user’s first name. This field supports alphanumeric characters, underscores (_), and periods.

Last Name

Configures the user’s last name. This field supports alphanumeric characters, underscores (_), and periods.

Email Address

Configure the user’s email address. Email notifications will be sent to this address.

User Password

Configure the password for user portal access. A random password is automatically generated by default and is highly recommended for security.

Language

Select the voice prompt language that will be used for this extension. By default, the selected voice prompt language under PBX Settings->Voice Prompt->Language Settings will be used. To add more supported languages, please download the voice prompt language packages in the same page.

Table 37: IAX Extension Configuration Parameters – Media

IAX Settings

Max Number of Calls

Configure the maximum number of calls allowed for each remote IP address.

Require Call Token

Configure to enable/disable requiring call token. If set to “Auto”, it might lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints. The default setting is “Yes”.

SRTP

Enable SRTP for the call. The default setting is disabled.

Fax Mode

Select Fax Mode. The default setting is “None”.

  • None: Disable Fax. This is the default setting.
  • Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38.

ACL Policy

Access Control List manages the IP addresses that can register to this extension.

  • Allow All: Any IP address can register to this extension.

Local Network Address: Only IP addresses in the configured network segments can register to this extension.

Codec Preference

Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p, RTX and VP8.

Table 38: IAX Extension Configuration Parameters – Features

Call Transfer

Call Forward Unconditional

Configure the Call Forward Unconditional target number. If not configured,
the Call Forward Unconditional feature is deactivated. The default setting
is deactivated.

CFU Time Condition

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes: 

  • Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific Time can be configured in the Specific Time tab.

  • Office Time and Holiday can be configured in System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward No Answer

Configure the Call Forward No Answer target number. Available options for target number are:

CFN Time Condition

Select time condition for Call Forward No Answer. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes: 

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific Time can be configured in the Specific Time tab.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward Busy

Configure the Call Forward Busy target number. Available options for target number are:

  • None”: Call forward deactivated.

  • Extension”: Select an extension from dropdown list as CFB target.

  • Custom Number”: Enter a customer number as target. For example: *97.

  • Voicemail”: Select an extension from dropdown list. Incoming calls will be forwarded to voicemail of selected extension.

  • Ring Group”: Select a ring group from dropdown list as CFB target.

  • Queues”: Select a queue from dropdown list as CFB target.

  • “Voicemail Group”: Select a voicemail group from dropdown list as CFB target.

CFB Time Condition

Select time condition for Call Forward Busy. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes: 

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Ring Simultaneously

Ring Simultaneously

Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.

External Number

Set the external number to be rang simultaneously. ‘-’ is the connection character which will be ignored.

This field accepts only letters, numbers, and special characters + = * #.

Time Condition for Ring Simultaneously

Ring the external number simultaneously along with the extension based on this time condition.

Other Settings

Ring Timeout

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM6200, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪General Settings: General Preference. The valid range is between 5 seconds and 600 seconds.

Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Auto Record

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.

Skip Trunk Auth

  • If set to ‘Yes’, users can skip entering the password when making outbound calls.

  • If set to ‘By Time’, users can skip entering the password when making outbound calls during the selected time condition.

  • If set to ‘No’, users will be asked to enter the password when making outbound calls.

Time Condition for Skip Trunk Auth

If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering password when making outbound calls.

Dial Trunk Password

Configure personal password when making outbound calls via trunk.

Enable LDAP

If enabled, the extension will be added to LDAP Phonebook PBX list.

Music On Hold

Specify which Music On Hold playlist to suggest to the bridged channel when putting them on hold.

Call Duration Limit

The maximum duration of call-blocking.

Enable Call Waiting

If disabled, UCM will not ring the extension if it is already in a call and will give the caller a busy response.

Email Missed Calls

When enabled, the UCM sends a log of missed calls to the extension's email address.

Table 38: IAX Extension Configuration Parameters – Features

  1. Table 39: IAX Extension Configuration Parameters – Specific Time

Specific Time

Time Condition

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

Create New FXS Extension

The UCM6200 supports Foreign eXchange Subscriber (FXS) interface. FXS is used when user needs to connect analog phone lines or FAX machines to the UCM6200.

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

Table 40: FXS Extension Configuration Parameters – Basic Settings

General

Extension

The extension number associated with the user.

Analog Station

Select the FXS port to be assigned for this extension.

Caller ID Number

Configure the CallerID Number that would be applied for outbound calls from this user.

Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider.

Permission

Assign permission level to the user. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level. The default setting is “Internal”.

Note:

Users need to have the same level as or higher level than an outbound rule’s privilege to make outbound calls using this rule.

Voicemail

Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”.

  • Disable Voicemail: Disable Voicemail.
  • Enable Local Voicemail: Enable voicemail for the user.

Voicemail Password

Configure the password to access the extension’s voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported.

Skip Voicemail Password Verification

If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default.

Disable This Extension

If selected, this extension will be disabled on the UCM6200.

Note: The disabled extension still exists on the PBX but cannot be used on the end device.

User Settings

First Name

Configure the user’s first name. This field supports alphanumeric characters, underscores (_), and periods.

Last Name

Configures the user’s last name. This field supports alphanumeric characters, underscores (_), and periods.

Email Address

Configure the user’s email address. Email notifications will be sent to this address.

User Password

Configure the password for user portal access. A random password is automatically generated by default and is highly recommended for security.

Language

Select the voice prompt language that will be used for this extension. By default, the selected voice prompt language under PBX Settings->Voice Prompt->Language Settings will be used. To add more supported languages, please download the voice prompt language packages in the same page.

Table 41: FXS Extension Configuration Parameters – Media

Analog Settings

Call Waiting

Configure to enable/disable call waiting feature.

The default setting is “No”.

User ‘#’ as SEND

If configured, the # key can be used as SEND key after dialing the number on the analog phone. The default setting is “Yes”.

RX Gain

Configure the RX gain for the receiving channel of analog FXS port. The valid range is -30dB to +6dB. The default setting is 0.

TX Gain

Configure the TX gain for the transmitting channel of analog FXS port. The valid range is -30dB to +6dB. The default setting is 0.

MIN RX Flash

Configure the minimum period of time (in milliseconds) that the hook-flash must remain unpressed for the PBX to consider the event as a valid flash event. The valid range is 30ms to 1000ms. The default setting is 200ms.

MAX RX Flash

Configure the maximum period of time (in milliseconds) that the hook-flash must remain unpressed for the PBX to consider the event as a valid flash event. The minimum period of time is 256ms and it cannot be modified. The default setting is 1250ms.

Enable Polarity Reversal

If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as Hangup on a polarity reversal. The default setting is “Yes”.

Echo Cancellation

Specify “ON”, “OFF” or a value (the power of 2) from 32 to 1024 as the number of taps of cancellation.

Note: When configuring the number of taps, the number 256 is not translated into 256ms of echo cancellation. Instead, 256 taps mean 256/8 = 32 ms. The default setting is “ON”, which is 128 taps.

3-Way Calling

Configure to enable/disable 3-way calling feature on the user. The default setting is enabled.

Send CallerID After

Configure the number of rings before sending CID. Default setting is 1.

Fax Mode

For FXS extension, there are three options available in Fax Mode. The default setting is “None”.

  • None: Disable Fax.
  • Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38.
  • Fax Gateway: If selected, the UCM6200 can support conversation and processing of Fax data from T.30 to T.38 or T.38 to T.30. only for FXS ports.

Table 42: FXS Extension Configuration Parameters – Features

Call Transfer

Call Forward Unconditional

Configure the Call Forward Unconditional target number. If not configured,
the Call Forward Unconditional feature is deactivated. The default setting is deactivated.

CFU Time Condition

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes: 

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific Time can be configured in the Specific Time tab.

  • Office Time and Holiday can be configured in System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward No Answer

Configure the Call Forward No Answer target number. If not configured,
the Call Forward No Answer feature is deactivated. The default setting is
deactivated.

CFN Time Condition

Select time condition for Call Forward No Answer. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes: 

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific Time can be configured in the Specific Time tab.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

Call Forward Busy

Configure the Call Forward Busy target number. If not configured, the Call
Forward Busy feature is deactivated. The default setting is deactivated.

CFB Time Condition

Select time condition for Call Forward Busy. The available time conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes: 

  • “Specific” has higher priority to “Office Times” if there is a conflict in terms of time period.

  • Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

  • Office Time and Holiday could be configured on page System Settings🡪Time Settings🡪Office Time/Holiday page.

CC Settings

Enable CC

If enabled, UCM6200 will automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason. By default, it is disabled.

Ring Simultaneously

Ring Simultaneously

Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.

External Number

Set the external number to be rang simultaneously. ‘-’ is the connection character which will be ignored.

This field accepts only letters, numbers, and special characters + = * #.

Time Condition for Ring Simultaneously

Ring the external number simultaneously along with the extension based on this time condition.

Hotline

Enable Hotline

If enabled, hotline dialing plan will be activated, a pre-configured number
will be used based on the selected Hotline Type.

Hotline Number

Configure the Hotline Number

Hotline Type

Configure the Hotline Type:

  • Immediate Hotline: When the phone is off-hook, UCM6200 will
    immediately dial the preset number

  • Delay Hotline: When the phone is off-hook, if there is no dialing within
    5 seconds, UCM6200 will dial the preset number.

Other Settings

Ring Timeout

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM6200, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪General Settings: General Preference. The valid range is between 5 seconds and 600 seconds.

Note: 

If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Auto Record

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.

Skip Trunk Auth

  • If set to ‘Yes’, users can skip entering the password when making outbound calls.

  • If set to ‘By Time’, users can skip entering the password when making outbound calls during the selected time condition.

  • If set to ‘No’, users will be asked to enter the password when making outbound calls.

Time Condition for Skip Trunk Auth

If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can skip entering password when making outbound calls.

Dial Trunk Password

Configure personal password when making outbound calls via trunk.

Enable LDAP

If enabled, the extension will be added to LDAP Phonebook PBX list.

Music On Hold

Specify which Music On Hold playlist to suggest to the bridged channel when putting them on hold.

Call Duration Limit

The maximum duration of call-blocking.

Email Missed Calls

When enabled, the UCM sends a log of missed calls to the extension's email address.

Table 43: FXS Extension Configuration Parameters – Specific Time

Specific Time

Time Condition

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

Batch Add Extensions

Batch Add SIP Extensions

To create multiple SIP extensions quickly, users can select “Batch” for the Select Add Method option during the extension creation process.

Table 44: Batch Add SIP Extension Parameters

General

Create Number

Specify the number of extensions to be added. The default setting is 5.

Extension Incrementation

Specify how much to increment each additional extension (e.g., if the value is 2, the extensions will be 1000, 1002, 1004, etc.). Note: Supports up to 3 characters.

Extension

Configure the starting extension number of the batch of extensions to be added.

Permission

Assign permission level to the user. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level. The default setting is “Internal”.

Note: Users need to have the same level as or higher level than an outbound rule’s privilege to make outbound calls from this rule.

Voicemail

Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”.

  • Disable Voicemail: Disable Voicemail.
  • Enable Local Voicemail: Enable voicemail for the user.
  • Enable Remote Voicemail: Forward the notify message from remote voicemail system for the user, and the local voicemail will be disabled. Note: Remote voicemail feature is used only for Infomatec (Brazil).

Enable WebRTC Support

Enable WebRTC support.

SIP/IAX Password

Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions.

  • User Random Password.

A random secure password will be automatically generated. It is recommended to use this password for security purpose.

  • Use Extension as Password.
  • Enter a password to be used on all the extensions in the batch.

Voicemail Password

Configure the password to access the extension’s voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported.

CallerID Number

Configure CallerID Number when adding Batch Extensions.

  • Use Extension as Number

Users can choose to use the extension number as CallerID

  • Use as Number

Users can choose to set a specific number instead of using the extension number.

Ring Timeout

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM6200, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt: General Preference. The valid range is between 5 seconds and 600 seconds.

Note:

If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Auto Record

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.

Skip Voicemail Password Verification

If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default.

Music On Hold

Select which Music On Hold class to suggest to extensions when putting them on hold.

Enable LDAP

If enabled, the batch added extensions will be added to LDAP Phonebook PBX list; if disabled, the batch added extensions will be skipped when creating LDAP Phonebook.

Enable WebRTC Support

If enabled, extensions will be able to login to user portal and use Web RTC features.

Call Duration Limit

Configure the maximum duration of call-blocking.

SIP Settings

NAT

Use NAT when the PBX is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it is related to NAT configuration or Firewall’s support of SIP and RTP ports.

The default setting is enabled.

Enable Direct Media

By default, the PBX will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to negotiate with the endpoints to route the media stream directly. It is not always possible for the PBX to negotiate endpoint-to-endpoint media routing. The default setting is “No”.

DTMF Mode

Select DTMF mode for the user to send DTMF. The default setting is “RFC4733”. If “Info” is selected, SIP INFO message will be used. If “Inband” is selected, 64-kbit codec PCMU and PCMA are required. When “Auto” is selected, RFC4733 will be used if offered, otherwise “Inband” will be used.

Enable Keep-alive

If enabled, empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is “Yes”.

Keep-alive Frequency

Configure the number of seconds for the host to be up for Keep-alive. The default setting is 60 seconds.

TEL URI

If the end device/phone has an assigned PSTN telephone number, this field should be set to “User=Phone”. Then a “User=Phone” parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is disabled.

Concurrent Registrations

The maximum allowed number of endpoint registrations to this extension. Default value is 1.

Other Settings

SRTP

Enable SRTP for the call. The default setting is “No”.

Fax Mode

Select Fax mode for this user. The default setting is “None”.

  • None: Disable Fax.
  • Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38.

ACL Policy

Access Control List manages the IP addresses that can register to this extension.

  • Allow All: Any IP address can register to this extension.

Local Network Address: Only IP addresses in the configured network segments can register to this extension.

Enable T.38 UDPTL

Enable or disable T.38 UDPTL Support.

Skip Trunk Auth

If enable “All”, users do not need to enter password when making an outbound call. If enable “Follow Me”, the user can dial out via follow me without password.

Codec Preference

Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, iLBC, ADPCM, LPC10, H.264, H.263, H.263p and VP8.

Batch Add IAX Extensions

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

Table 45: Batch Add IAX Extension Parameters

General

Start Extension

Configure the starting extension number of the batch of extensions to be added.

Create Number

Specify the number of extensions to be added. The default setting is 5.

Permission

Assign permission level to the user. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level. The default setting is “Internal”.

Note: Users need to have the same level as or higher level than an outbound rule’s privilege in order to make outbound calls from this rule.

Voicemail

Configure Voicemail. There are three valid options and the default option is “Enable Local Voicemail”.

  • Disable Voicemail: Disable Voicemail.
  • Enable Local Voicemail: Enable voicemail for the user.

SIP/IAX Password

Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions.

  • User Random Password.

A random secure password will be automatically generated. It is recommended to use this password for security purpose.

  • Use Extension as Password.
  • Enter a password to be used on all the extensions in the batch.

Voicemail Password

Configure the password to access the extension’s voicemail. A randomly generated password is used by default and is highly recommended for security. Only digits are supported.

Ring Timeout

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM6200, which can be configured in the global ring timeout setting under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt: General Preference. The valid range is between 5 seconds and 600 seconds.

Note:

If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Auto Record

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.

Skip Voicemail Password Verification

If enabled, users can skip password verification when dialing in via the My Voicemail feature code. This option is disabled by default.

Music On Hold

Select which Music On Hold class to suggest to extensions when putting them on hold.

Enable LDAP

If enabled, the batch added extensions will be added to LDAP Phonebook PBX list; if disabled, the batch added extensions will be skipped when creating LDAP Phonebook.

Call Duration Limit

Configure the maximum duration of call-blocking.

IAX Settings

Max Number of Calls

Configure the maximum number of calls allowed for each remote IP address.

Require Call Token

Configure to enable/disable requiring call token. If set to “Auto”, it might lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints.

The default setting is “Yes”.

Other Settings

SRTP

Enable SRTP for the call. The default setting is “No”.

Fax Mode

Select Fax Mode for this user. The default setting is “None”.

  • None: Disable Fax.
  • Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38.

ACL Policy

Access Control List manages the IP addresses that can register to this extension.

  • Allow All: Any IP address can register to this extension.

Local Network Address: Only IP addresses in the configured network segments can register to this extension.

Skip Trunk Auth

If enable “All”, users do not need to enter password when making an outbound call. If enable “Follow Me”, the call can dial out via follow me without password.

Codec Preference

Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, iLBC, ADPCM, LPC10, H.264, H.263, H.263p and VP8.

Batch Extension Resetting Functionality

To reset multiple extensions to default settings, select the extensions on the main Extensions/Trunks->Extensions page and click on the Reset button. Once reset, all settings except for Concurrent Registrations will be reverted to default. User voicemail password will be randomized again, and user voicemail prompts and recordings will be deleted. The settings for the extension user in the User Management page will also be reverted to default except for usernames and custom privileges.

Search and Edit Extension

All the UCM6200 extensions are listed under Web GUI🡪Extension/Trunk🡪Extensions, with status, Extension, CallerID Name, Technology (SIP, IAX and FXS), IP and Port. Each extension has a checkbox for users to “Edit Selected Extensions” or “Delete Selected Extensions”. Also, options “Edit” , “Reboot” , “Delete” and reset are available per extension. User can search an extension by specifying the extension number to find an extension quickly.

Figure 84: Manage Extensions

  • Status

Users can see the following icon for each extension to indicate the SIP status.

Green: Idle

Blue: Ringing

Yellow: In Use

Grey: Unavailable (the extension is not registered or disabled on the PBX)

  • Edit single extension

Click on to start editing the extension parameters.

  • Reset single extension

Click on A close up of a logo

Description generated with high confidence to reset the extension parameters to default (except concurrent registration).

Other settings will be restored to default in Maintenance🡪User Management🡪User Information except username and permissions and delete the user voicemail prompt and voice messages.

  • Reboot the user

Click on to send NOTIFY reboot event to the device which has an UCM6200 extension already registered. To successfully reboot the user, “Zero Config” needs to be enabled on the UCM6200 Web GUI🡪Value-added Features🡪Zero Config🡪Zero Config Settings.

  • Delete selected extensions

Click on the button to delete a single extension. Users can also select multiple extensions and click on the Delete button to delete several extensions at once.

  • Modify selected extensions

Select the checkbox for the extension(s). Then click on “Edit” to edit the extensions in a batch.

Export Extensions

UCM extensions can be exported to a CSV file. Different extension types (SIP, IAX, FXS) cannot be exported to the same CSV file. To export extensions, click on the Export button and select the extension type to export.

Figure 85: Export Extensions

Users can also use the exported CSV file to use as a template to manually edit extension information.

Import Extensions

The capability to import extensions to the UCM6200 provides users flexibility to batch add extensions with similar or different configuration quickly into the PBX system.

  • Export extension csv file from the UCM6200 by clicking on “Export Extensions” button.
  • Fill up the extension information you would like in the exported csv template.
  • Click on “Import Extensions” button. The following dialog will be prompted.

Figure 86: Import Extensions

  • Select the option in “On Duplicate Extension” to define how the duplicate extension(s) in the imported csv file should be treated by the PBX.
  • Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension information as previously configured without change.
  • Delete and Recreate: The current extension previously configured will be deleted and the duplicate extension in the csv file will be loaded to the PBX.
  • Update Information: The current extension previously configured in the PBX will be kept. However, if the duplicate extension in the csv file has different configuration for any options, it will override the configuration for those options in the extension.
  • Click on “Choose file to upload” to select csv file from local directory in the PC.
  • Click on “Apply Changes” to apply the imported file on the UCM6200.

Example of file to import:

Figure 87: Import File

Table 46: SIP extensions Imported File Example

Field

Supported values

Extension

Digits

Technology

SIP/SIP(WebRTC)

Enable Voicemail

yes/no/remote

CallerID Number

Digits

SIP/IAX Password

Alphanumeric characters

Voicemail Password

Digits

Skip Voicemail Password Verification

yes/no

Ring Timeout

Empty/ 3 to 600 (in second)

SRTP

yes/no

Fax Mode

None/Fax Detection

Strategy

Allow All/Local Subnet Only/A Specific IP Address

Local Subnet 1

IP address/Mask

Local Subnet 2

IP address/Mask

Local Subnet 3

IP address/Mask

Local Subnet 4

IP address/Mask

Local Subnet 5

IP address/Mask

Local Subnet 6

IP address/Mask

Local Subnet 7

IP address/Mask

Local Subnet 8

IP address/Mask

Local Subnet 9

IP address/Mask

Local Subnet 10

IP address/Mask

Specific IP Address

IP address

Skip Trunk Auth

yes/no/bytime

Codec Preference

PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus

Permission

Internal/Local/National/International

NAT

yes/no

DTMF Mode

RFC4733/info/inband/auto

Insecure

Port

Enable Keep-alive

Yes/no

Keep-alive Frequency

Value from 1-3600

AuthID

Alphanumeric value without special characters

TEL URI

Disabled/user=phone/enabled

Call Forward Busy

Digits

Call Forward No Answer

Digits

Call Forward Unconditional

Digits

Support Hot-Desking Mode

Yes/no

Dial Trunk Password

Digits

Disable This Extension

Yes/no

CFU Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFN Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFB Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Music On Hold

Default/ringbacktone_default

CC Agent Policy

If CC is disabled use: never

If CC is set to normal use: generic

If CC is set to trunk use: native

CC Monitor Policy

Generic/never

CCBS Available Timer

3600/4800

CCNR Available Timer

3600/7200

CC Offer Timer

60/120

CC Max Agents

Value from 1-999

CC Max Monitors

Value from 1-999

Ring simultaneously

Yes/no

External Number

Digits

Time Condition for Ring Simultaneously

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Time Condition for Skip Trunk Auth

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Enable LDAP

Yes/no

Enable T.38 UDPTL

Yes/no

Max Contacts

Values from 1-10

Enable WebRTC

Yes/no

Alert-Info

None/Ring 1/Ring2/Ring3/Ring 4/Ring 5/Ring 6/Ring 7/ Ring 8/Ring 9/Ring 10/bellcore-dr1/bellcore-dr2/ bellcore-dr3/ bellcore-dr4/ bellcore-dr5/custom

Do Not Disturb

Yes/no

DND Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Custom Auto answer

Yes/no

Do Not Disturb Whitelist

Empty/digits

User Password

Alphanumeric characters.

First Name

Alphanumeric without special characters.

Last Name

Alphanumeric without special characters.

Email Address

Email address

Language

Default/en/zh

Phone Number

Digits

Call-Barging Monitor

Extensions allowed to call barging

Seamless Transfer Members

Extensions allowed to seamless transfer

Table 47: IAX extensions Imported File Example

Field

Supported values

Extension

Digits

Technology

IAX

Enable Voicemail

yes/no

CallerID Number

Digits

SIP/IAX Password

Alphanumeric characters

Voicemail Password

Digits

Skip Voicemail Password Verification

yes/no

Ring Timeout

Empty/ 3 to 600 (in second)

SRTP

yes/no

Fax Mode

None/Fax Detection

Strategy

Allow All/Local Subnet Only/A Specific IP Address

Local Subnet 1

IP address/Mask

Local Subnet 2

IP address/Mask

Local Subnet 3

IP address/Mask

Local Subnet 4

IP address/Mask

Local Subnet 5

IP address/Mask

Local Subnet 6

IP address/Mask

Local Subnet 7

IP address/Mask

Local Subnet 8

IP address/Mask

Local Subnet 9

IP address/Mask

Local Subnet 10

IP address/Mask

Specific IP Address

IP address

Skip Trunk Auth

yes/no/bytime

Codec Preference

PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus

Permission

Internal/Local/National/International

NAT

yes/no

Call Forward Busy

Digits

Call Forward No Answer

Digits

Call Forward Unconditional

Digits

Require Call Token

Yes/no/auto

Max Number of Calls

Values from 1-512

Dial Trunk Password

Digits

Disable This Extension

Yes/no

CFU Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFN Time Condition

CFB Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Music On Hold

Default/ringbacktone_default

Ring simultaneously

Yes/no

External Number

Digits

Time Condition for Ring Simultaneously

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Time Condition for Skip Trunk Auth

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Enable LDAP

Yes/no

Limit Max time (s)

empty

Do Not Disturb

Yes/no

DND Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Do Not Disturb Whitelist

Empty/digits

User Password

Alphanumeric characters.

First Name

Alphanumeric without special characters.

Last Name

Alphanumeric without special characters.

Email Address

Email address

Language

Default/en/zh

Phone Number

Digits

Call-Barging Monitor

Extensions allowed to call barging

Seamless Transfer Members

Extensions allowed to seamless transfer

Table 48: FXS Extensions Imported File Example

Field

Supported values

Extension

Digits

Technology

FXS

Analog Station

FXS1/FXS2

Enable Voicemail

yes/no

CallerID Number

Digits

Voicemail Password

Digits

Skip Voicemail Password Verification

yes/no

Ring Timeout

Empty/ 3 to 600 (in second)

Auto Record

yes/no

Fax Mode

None/Fax Detection

Skip Trunk Auth

Yes/no/bytime

Permission

Internal/Local/National/International

Call Forward Busy

Digits

Call Forward No Answer

Digits

Call Forward Unconditional

Digits

Call Waiting

Yes/no

Use # as SEND

Yes/no

RX Gain

Values from -30🡪6

TX Gain

Values from -30🡪6

MIN RX Flash

Values from: 30 – 1000

MAX RX Flash

Values from: 40 – 2000

Enable Polarity Reversal

Yes/no

Echo Cancellation

On/Off/32/64/128/256/512/1024

3-Way Calling

Yes/no

Send CallerID After

1/2

Dial Trunk Password

digits

Disable This Extension

Yes/no

CFU Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFN Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

CFB Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Music On Hold

Default/ringbacktone_default

CC Agent Policy

If CC is disabled use: never

If CC is set to normal use: generic

If CC is set to trunk use: native

CC Monitor Policy

Generic/never

CCBS Available Timer

3600/4800

CCNR Available Timer

3600/7200

CC Offer Timer

60/120

CC Max Agents

Value from 1-999

CC Max Monitors

Value from 1-999

Ring simultaneously

Yes/no

External Number

Digits

Time Condition for Ring Simultaneously

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Time Condition for Skip Trunk Auth

Enable LDAP

Yes/no

Enable Hotline

Yes/no

Hotline Type

Immediate hotline/delay hotline

Hotline Number

digits

Limit Max time (s)

empty

Do Not Disturb

Yes/no

DND Time Condition

All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time

Do Not Disturb Whitelist

Empty/digits

User Password

Alphanumeric characters.

First Name

Alphanumeric without special characters.

Last Name

Alphanumeric without special characters.

Email Address

Email address

Language

Default/en/zh

Phone Number

Digits

Call-Barging Monitor

Extensions allowed to call barging

Seamless Transfer Members

Extensions allowed to seamless transfer

The CSV file should contain all the above fields, if one of them is missing or empty, the UCM6200 will display the following error message for missing fields.

Figure 88: Import Error

Extension Details

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

  • Extension: Shows the Extension number.
  • Status: Shows the status of the extension.
  • Presence status: Indicates the Presence Status of this extension.
  • Terminal Type: Shows the Type of the terminal using this extension (SIP, FXS…etc.).
  • Caller ID Name: Reveals the Caller ID Name configured on the extension.
  • Messages: Shows the messages stats.
  • IP and Port: The IP address and the ports of the device using the extension.
  • Email status: Show the Email status (sent, to be sent…etc.).
  • Ring Group: Indicates the ring groups that this extension belongs to.
  • Call Queue: Indicates the Cal Queues that this extension belongs to.
  • Call Queue (Dynamic): Indicates the Call Queues that this extension belongs to as a dynamic agent.

Figure 89: Extension Details

E-mail Notification

Users can request for their SIP acccount information via email. The system administrator can select the extensions to send account information emails to and press the Email Notification button in the Extensions page.

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

Figure 90: E-mail Notification – Prompt Information

The selected users will receive emails containing account registration information, LDAP configuration, and QR codes for quick setup on mobile apps.

Figure 91: Account Registration Information and QR Code

Figure 92: LDAP Client Information and QR Code

Multiple Registrations per Extension

UCM6200 supports multiple registrations per extension so that users can use the same extension on devices in different locations.

Figure 93: Multiple Registrations per Extension

This feature can be enabled by configuring option “Concurrent Registrations” under Web GUI🡪Extension/Trunk🡪Edit Extension. The default value is 1 for security purposes. Maximum is 10.

Figure 94: Extension – Concurrent Registration

SMS Message Support

The UCM6200 provides built-in SIP SMS message support. For SIP end devices such as Grandstream GXP or GXV phones that supports SIP message, after an UCM6200 account is registered on the end device, the user can send and receive SMS message. Please refer to the end device documentation on how to send and receive SMS message.

SMS Message support is a new feature added since firmware 1.0.10.x which is built with Asterisk 13.

Figure 95: SMS Message Support

Extension Groups

The UCM6200 extension group feature allows users to assign and categorize extensions in different groups to better manage the configurations on the UCM6200. For example, when configuring “Enable Filter on Source Caller ID”, users could select a group instead of each person’s extension to assign. This feature simplifies the configuration process and helps manage and categorize the extensions for business environment.

Configure Extension Groups

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

  • Click on to create a new extension group.
  • Click on to edit the extension group.
  • Click on to delete the extension group.

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

Figure 96: Edit Extension Group

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

Using Extension Groups

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

Figure 97: Select Extension Group in Outbound Route

Analog Trunks

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

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

Analog Trunk Configuration

The analog trunk options are listed in the table below.

Table 49: Analog Trunk Configuration Parameters

Channels

Select the channel for the analog trunk.

  • UCM6202: 2 channels
  • UCM6204: 4 channels
  • UCM6208: 8 channels

Trunk Name

Specify a unique label to identify the trunk when listed in outbound rules, incoming rules and etc.

Advanced Options

SLA Mode

Enable this option to satisfy two primary use cases, which include emulating a simple key system and creating shared extensions on a PBX. Enable SLA Mode will disable polarity reversal.

Barge Allowed

The barge option specifies whether other stations can join a call in progress on this trunk. If enabled, the other stations can press the line button to join the call.

The default setting is Yes.

Hold Access

The hold option specifies hold permissions for this trunk. If set to “Open”, any station can place this trunk on hold and any other station is allowed to retrieve the call. If set to “Private”, only the station that places the call on hold can retrieve the call.

The default setting is Yes.

Enable Polarity Reversal

If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as “Hangup” on a polarity reversal. The default setting is “No”.

Polarity on Answer Delay

When FXO port answers the call, FXS may send a Polarity Reversal. If this interval is shorter than the value of “Polarity on Answer Delay”, the Polarity Reversal will be ignored. Otherwise, the FXO will Onhook to disconnect the call. The default setting is 600ms.

Current Disconnect Threshold (ms)

This is the periodic time (in ms) that the UCM6200 will use to check on a voltage drop in the line. The default setting is 200. The valid range is 50 to 3000.

Ring Timeout

Configure the ring timeout (in ms). Trunk (FXO) devices must have a timeout to determine if there was a Hangup before the line is answered. This value can be used to configure how long it takes before the UCM6200 considers a non-ringing line with Hangup activity. The default setting is 8000.

RX Gain

Configure the RX gain for the receiving channel of analog FXO port. The valid range is from -12.0 (dB) to + 12.0 (dB). The default setting is 0.

TX Gain

Configure the TX gain for the transmitting channel of analog FXO port. The valid range is from -12.0 (dB) to + 12.0 (dB). The default setting is 0.

Use CallerID

Configure to enable CallerID detection. The default setting is “Yes”.

Caller ID Scheme

Select the Caller ID scheme for this trunk.

  • Bellcore/Telcordia.
  • ETSI-FSK During Ringing
  • ETSI-FSK Prior to Ringing with DTAS
  • ETSI-FSK Prior to Ringing with LR
  • ETSI-FSK Prior to Ringing with RP
  • ETSI-DTMF During Ringing
  • ETSI-DTMF Prior to Ringing with DTAS
  • ETSI-DTMF Prior to Ringing with LR
  • ETSI-DTMF Prior to Ringing with RP
  • SIN 227-BT
  • NTT Japan
  • Auto Detect

If you are not sure which scheme to choose, please select “Auto Detect”. The default setting is “Bellcore/Telcordia”.

Fax Mode

Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38. The default setting is “No”.

Note: If enabled, Fax Pass-through cannot be used.

FXO Dial Delay(ms)

Configure the time interval between off-hook and first dialed digit for outbound calls.

Auto Record

Enable automatic recording for the calls using this trunk. The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.

Disable This Trunk

If selected, the trunk will be disabled and incoming/Outgoing calls via this trunk will not be possible.

DAHDI Out Line Selection

This is to implement analog trunk outbound line selection strategy.

Three options are available:

  • Ascend

When the call goes out from this analog trunk, it will always try to use the first idle FXO port. The port order that the call will use to go out would be port 1🡪port 2🡪port 10🡪port 16. Every time it will start with port 1 (if it is idle).

  • Poll

When the call goes out from this analog trunk, it will use the port that is not used last time. And it will always use the port in the order of port 1🡪2🡪10🡪16🡪1🡪2🡪10🡪16🡪1🡪2🡪10🡪16…, following the last port being used.

  • Descend

When the call goes out from this analog trunk, it will always try to use the last idle FXO port. The port order that the call will use to go out would be port 16🡪port 10🡪port 2🡪port 1. Every time it will start with port 16 (if it is idle).

The default setting is “Ascend” mode.

Echo Cancellation Mode

The Non-Linear Processing (NLP) in echo cancellation helps to remove/suppress residual echo components that could not be removed by the LEC (Line Echo Canceller). Following modes are supported:

  • Default: Slightly reduces the amount of detected background noise.
  • High Noise Level Adjustment: Greatly reduces the amount of detected background noise.
  • Noise Masking: Injects noise to mask echo and greatly reduces the amount of detected background noise.
  • White Noise Injection: Injects variable amounts of white noise based on detected background noise and greatly reduces the amount of detected background noise.

Direct Callback

Allows external numbers the option to get directed to the extension that last called them.

For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination.

Tone Settings

Busy Detection

Busy Detection is used to detect far end Hangup or for detecting busy signal. The default setting is “Yes”.

Busy Tone Count

If “Busy Detection” is enabled, users can specify the number of busy tones to be played before hanging up. The default setting is 2. Better results might be achieved if set to 4, 6 or even 8. Please note that the higher the number is, the more time is needed to Hangup the channel. However, this might lower the probability to get random Hangup.

Congestion Detection

Congestion detection is used to detect far end congestion signal. The default setting is “Yes”.

Congestion Count

If “Congestion Detection” is enabled, users can specify the number of congestion tones to wait for. The default setting is 2.

Tone Country

Select the country for tone settings. If “Custom” is selected, users could manually configure the values for Busy Tone and Congestion Tone. The default setting is “United States of America (USA)”.

Busy Tone

Syntax:

f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];

Frequencies are in Hz and cadence on and off are in ms.

Frequencies Range: [0, 4000)

Busy Level Range: (-300, 0)

Cadence Range: [0, 16383].

Select Tone Country “Custom” to manually configure Busy Tone value.

Default value:

f1=480@-50,f2=620@-50,c=500/500

Congestion Tone

Syntax:

f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];

Frequencies are in Hz and cadence on and off are in ms.

Frequencies Range: [0, 4000)

Busy Level Range: (-300, 0)

Cadence Range: [0, 16383].

Select Tone Country “Custom” to manually configure Busy Tone value.

Default value:

f1=480@-50,f2=620@-50,c=250/250

PSTN Detection

Click on “Detect” to detect the busy tone, Polarity Reversal and Current Disconnect by PSTN. Before the detecting, please make sure there is more than one channel configured and working properly. If the detection has busy tone, the “Tone Country” option will be set as “Custom”.

PSTN Detection

The UCM6200 provides PSTN detection function to help users detect the busy tone, Polarity Reversal and Current Disconnect by making a call from the PSTN line to another destination. The detecting call will be answered and up for about 1 minute. Once done, the detecting result will show and can be used for the UCM6200 settings.

  1. Go to UCM6200 Web GUI🡪Extension/Trunk🡪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 there are two methods to set the busy tone.
  • Tone Country. The default setting is “United States of America (USA)”.
  • PSTN Detection.

Figure 98: UCM6200 FXO Tone Settings

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

Figure 99: UCM6200 PSTN Detection

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

Detect Model: Auto Detect.

Source Channel: The source channel to be detected.

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

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

Figure 100: UCM6200 PSTN Detection: Auto Detect

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

Figure 101: UCM6200 PSTN Detection: Semi-Auto Detect

Detect Model: Semi-auto Detect.

Source Channel: The source channel to be detected.

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

  1. Click “Detect” to start detecting. The source channel will initiate a call to the destination number. For “Auto Detect”, the call will be automatically answered. For “Semi-auto Detect”, the UCM6200 Web GUI will display prompt to notify the user to answer or hang up the call to finish the detecting process.
  2. Once done, the detected result will show. Users could save the detecting result as the current UCM6200 settings.

Table 50: PSTN Detection for Analog Trunk

Detect Model

Select “Auto Detect” or “Semi-auto Detect” for PSTN detection.

  • Auto Detect

Please make sure two or more channels are connected to the UCM6200 and in idle status before starting the detection. During the detection, one channel will be used as caller (Source Channel) and another channel will be used as callee (Destination Channel). The UCM6200 will control the call to be established and hang up between caller and callee to finish the detection.

  • Semi-auto Detect

Semi-auto detection requires answering or hanging up the call manually. Please make sure one channel is connected to the UCM6200 and in idle status before starting the detection. During the detection, source channel will be used as caller and send the call to the configured Destination Number. Users will then need follow the prompts in Web GUI to help finish the detection.

The default setting is “Auto Detect”.

Source Channel

Select the channel to be detected.

Destination Channel

Select the channel to help detect when “Auto Detect” is used.

Destination Number

Configure the number to be called to help the detection.

Note:

  • The PSTN detection process will keep the call up for about 1 minute.
  • If “Semi-auto Detect’ is used, please pick up the call only after informed from the Web GUI prompt.
  • Once the detection is successful, the detected parameters “Busy Tone”, “Polarity Reversal” and “Current Disconnect by PSTN” will be filled into the corresponding fields in the analog trunk configuration.

VoIP Trunks

VoIP Trunk Configuration

VoIP trunks can be configured in UCM6200 under Web GUI🡪Extension/Trunk🡪VoIP Trunks. Once created, the VoIP trunks will be listed with Provider Name, Type, Hostname/IP, Username and Options to edit/detect the trunk.

Note: UCM6200 currently supports up to 200 VoIP trunks.

  • Click on “Create New SIP Trunk” or “Create New IAX Trunk” to add a new VoIP trunk.
  • Click on to configure detailed parameters for the VoIP trunk.
  • Click on to configure Direct Outward Dialing (DOD) for the SIP Trunk.
  • Click on to start LDAP Sync.
  • Click on to delete the VoIP trunk.

For VoIP trunk example, please refer to the document in the following link:

https://documentation.grandstream.com/knowledge-base/sip-trunks-guide/

The VoIP trunk options are listed in the table below.

Table 51: Create New SIP Trunk

Type

Select the VoIP trunk type.

  • Peer SIP Trunk
  • Register SIP Trunk

Provider Name

Configure a unique label (up to 64 character) to identify this trunk when listed in outbound rules, inbound rules etc.

Host Name

Configure the IP address or URL for the VoIP provider’s server of the trunk.

Keep Original CID

Keep the CID from the inbound call when dialing out. This setting will override “Keep Trunk CID” option. Please ensure that the remote peer PBX supports matching user entry via the “username” field from authentication line.

Keep Trunk CID

If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”.

NAT

Turn on this setting when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it is related to NAT configuration or SIP/RTP port support on the firewall.

Disable This Trunk

If checked, the trunk will be disabled.

Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider.

TEL URI

If the trunk has an assigned PSTN telephone number, this field should be set to “User=Phone”. Then a “User=Phone” parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is disabled.

Caller ID

Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored.

Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call:

From user (Register Trunk Only) 🡪 CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID (Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled) 🡪 Global Outbound CID.

Need Registration

Select whether the trunk needs to register on the external server or not when “Register SIP Trunk” type is selected. The default setting is No.

Username

Enter the username to register to the trunk from the provider when “Register SIP Trunk” type is selected.

Password

Enter the password to register to the trunk from the provider when “Register SIP Trunk” is selected.

Auth ID

Enter the Authentication ID for “Register SIP Trunk” type.

Auto Record

Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.

Direct Callback

Allows external numbers the option to get directed to the extension that last called them.

For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination.

Table 52: SIP Register Trunk Configuration Parameters

Basic Settings

Provider Name

Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc.

Host Name

Configure the IP address or URL for the VoIP provider’s server of the trunk.

Transport

Configure the SIP transport protocol to be used in this trunk. The default setting is “UDP”.

  • UDP
  • TCP
  • TLS

SIP URI Scheme When Using TLS

When TLS is selected as Transport for register trunk, users can select between SIP and SIPS URI scheme

Keep Original CID

Keep the CID from the inbound call when dialing out. This setting will override “Keep Trunk CID” option. Please ensure that the remote peer PBX supports matching user entry via the “username” field from authentication line.

Keep Trunk CID

If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”.

NAT

Turn on this option when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it is related to NAT configuration or SIP/RTP port configuration on the firewall.

Disable This Trunk

If selected, the trunk will be disabled.

Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider.

TEL URI

If the trunk has an assigned PSTN telephone number, this field should be set to “User=Phone”. Then a “User=Phone” parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is disabled.

Need Registration

Select whether the trunk needs to register on the external server or not when “Register SIP Trunk” type is selected. The default setting is No.

Allow outgoing calls if registration failure

If enabled, outbound calls can still be made even if registration to the trunk fails. Note: If Need Registration is disabled, this option will be ignored.

CallerID Name

Configure the new name of the caller when the extension has no CallerID Name configured.

From Domain

Configure the domain name of the SIP user account. This will overwrite the domain in the “From” header. Example: If the user account is “1234567”, setting “trunk.UCM6510.provider.com” as the From Domain value will result in the following From header: “sip:1234567@trunk.UCM6510.provider.com”.

From User

Configure the username of the account. This will overwrite the user value in the From header. This is useful for scenarios where a single user account has multiple DIDs. Example: If the domain is “trunk.UCM6510.provider.com”, setting “1234567” as the From User value will result in the following From header: “sip:1234567@trunk.UCM6510.provider.com”.

Username

Enter the username to register to the trunk from the provider when “Register SIP Trunk” type is selected.

Password

Enter the password to register to the trunk when “Register SIP Trunk” is selected.

Auth ID

Enter the Authentication ID for “Register SIP Trunk” type.

Auth Trunk

If enabled, the UCM will send 401 response to the incoming call to authenticate the trunk.

Auto Record

Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.

Direct Callback

Allows external numbers the option to get directed to the extension that last called them.

For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination.

Advanced Settings

Codec Preference

Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p, RTX, OPUS and VP8.

Send PPI Header

If enabled, the SIP INVITE message sent to the trunk will contain PPI (P-Preferred-Identity) header. The default setting is “No”.

Note: “Send PPI Header” and “Send PAI Header” cannot be enabled at the same time. Only one of the two headers can be contained in SIP INVITE message.

PPI Mode

  • Default – Include the trunk’s preferred CID (configured in Basic Settings) in the PPI Header.
  • Original CID – Include the original CID in the PPI Header.
  • DOD Number – Include the trunk’s DOD number in the PPI Header. If no DOD number has been set, the trunk’s preferred CID will be used.

Send PAI Header

If enabled, the SIP INVITE message sent to the trunk will contain PAI (P-Asserted-Identity) header including configured PAI Header. The default setting is “No”.

Note: “Send PPI Header” and “Send PAI Header” cannot be enabled at the same time. Only one of the two headers can be contained in the SIP INVITE message.

PAI Header

If “Send PAI Header” is enabled and “PAI Header” is configured as “123456” for instance, the PAI header in the SIP message sent from the UCM will contain “123456”. If “Send PAI Header” is enabled and “PAI Header” is configured as “empty”, the PAI header in the SIP message sent from the UCM will contain the original CID.

Note: “Send PAI Header” needs to be enabled in order to use this feature.

Only alphanumeric characters are allowed and/or special characters #*-_+. with a limit of 64 characters.

DOD As From Name

If enabled and “From User” is configured, the INVITE’s From header will contain the DOD number.

Passthrough PAI Header

If checked and option “Send PAI Header” not checked, the PAI header will be passthrough from one side to the other side.

Outbound Proxy Support

Select to enable outbound proxy in this trunk. The default setting is “No”.

Outbound Proxy

When outbound proxy support is enabled, enter the IP address or URL of the outbound proxy.

Remove OBP from Route

It is used to set if the phone system will remove outbound proxy URI from the route header. If is set to “Yes”, it will remove the route header from SIP requests. The default setting is “No”.

DID Mode

Configure where to get the destination ID of an incoming SIP call, from SIP Request-line or To-header. The default is set to “Request-line”.

DTMF Mode

Configure the default DTMF mode when sending DTMF on this trunk.

  • Default: The global setting of DTMF mode will be used. The global setting for DTMF Mode setting is under Web GUI🡪PBX Settings🡪SIP Settings🡪ToS.
  • RFC4733: Send DTMF using RFC4733.
  • Info: Send DTMF using SIP INFO message.
  • Inband: Send DTMF using inband audio. This requires 64-bit codec, i.e., PCMU and PCMA.
  • Auto: Send DTMF using RFC4733 if offered. Otherwise, inband will be used.

Enable Heartbeat Detection

If enabled, the UCM6200 will regularly send SIP OPTIONS to the trunk to check connection status.

The default setting is “No”.

Heartbeat Frequency

When “Enable Heartbeat Detection” option is set to “Yes”, configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds.

Maximum Number of Call Lines

The maximum number of concurrent calls using the trunk. The default settings 0, which means no limit.

Fax Mode

Select Fax mode. The default setting is “None”.

  • None: Disable Fax.
  • Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38.

SRTP

Enable SRTP for the VoIP trunk. The default setting is “No”.

CC Settings

Enable CC

If enabled, the system will automatically alert the user when a called party is available, given that a previous call to that party failed for some reason.

CC Max Agents

Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel is allowed to make. The minimum value is 1.

CC Max Monitors

Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time. The minimum value is 1.

Table 53: SIP Peer Trunk Configuration Parameters

Basic Settings

Provider Name

Configure a unique label to identify this trunk when listed in outbound rules, inbound rules and etc.

Host Name

Configure the IP address or URL for the VoIP provider’s server of the trunk.

Auto Record

Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under Web GUI🡪CDR🡪Recording Files.

Keep Original CID

Keep the CID from the inbound call when dialing out, this setting will override “Keep Trunk CID” option. Please ensure that the remote peer PBX supports matching user entry via the “username” field from authentication line.

Keep Trunk CID

If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”.

NAT

Turn on this option when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it is related to NAT configuration or SIP/RTP port configuration on the firewall.

Disable This Trunk

If selected, the trunk will be disabled.

Note:

If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider.

TEL URI

If the trunk has an assigned PSTN telephone number, this field should be set to “User=Phone”. Then a “User=Phone” parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is disabled.

Caller ID

Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored.

Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call :

  • CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID (Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled) 🡪 Global Outbound CID.

CallerID Name

Configure the name of the caller to be displayed when the extension has no CallerID Name configured.

Transport

Configure the SIP transport protocol to be used in this trunk. The default setting is “UDP”.

  • UDP
  • TCP
  • TLS

Direct Callback

Allows external numbers the option to get directed to the extension that last called them.

For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option “Direct Callback” is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination.

Advanced Settings

Codec Preference

Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p,RTX, OPUS and VP8.

DID Mode

Configure where to get the destination ID of an incoming SIP call, from SIP Request-line or To-header. The default is set to “Request-line”.

DTMF Mode

Configure the default DTMF mode when sending DTMF on this trunk.

  • Default: The global setting of DTMF mode will be used. The global setting for DTMF Mode setting is under Web GUI🡪PBX Settings🡪SIP Settings🡪ToS.
  • RFC4733: Send DTMF using RFC4733.
  • Info: Send DTMF using SIP INFO message.
  • Inband: Send DTMF using inband audio. This requires 64-bit codec, i.e., PCMU and PCMA.
  • Auto: Send DTMF using RFC4733 if offered. Otherwise, inband is used.

Enable Heartbeat Detection

If enabled, the UCM6200 will regularly send SIP OPTIONS to the trunk to check connection status. The default setting is “No”.

Heartbeat Frequency

When “Enable Heartbeat Detection” option is set to “Yes”, configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds.

Maximum Number of Call Lines

The maximum number of concurrent calls using the trunk. The default settings 0, which means no limit.

Fax Mode

Select Fax mode. The default setting is “None”.

  • None: Disable Fax.
  • Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38.

SRTP

Enable SRTP for the VoIP trunk.

The default setting is “No”.

IPVT Mode

If enabled, it will allow SDP passthrough to Grandstream IPVideoTalk therefore it will allow calls between the UCM and IPVideoTalk. The default setting is disabled.

Sync LDAP Enable

If enabled, the local UCM6200 will automatically provide and update the local LDAP contacts to the remote UCM6200 SIP peer trunk. In order to ensure successful synchronization, the remote UCM6200 peer also needs to enable this option on the SIP peer trunk.

The default setting is “No”.

Sync LDAP Password

This is the password used for LDAP contact file encryption and decryption during the LDAP sync process. The password must be the same on both UCM6200 peers to ensure successful synchronization.

Sync LDAP Port

Configure the TCP port used LDAP sync feature between two peer UCM6200.

LDAP Outbound Rule

Specify an outbound rule for LDAP sync feature. The UCM6200 will automatically modify the remote contacts by adding prefix parsed from this rule.

LDAP Dialed Prefix

Specify the prefix for LDAP sync feature. The UCM6200 will automatically modify the remote contacts by adding this prefix.

Enable T.38 UDPTL

This option allows to toggle T.38 support for individual trunks.

The default setting is enabled.

CC Settings

Enable CC

If enabled, the system will automatically alert the user when a called party is available, given that a previous call to that party failed for some reason.

CC Max Agents

Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel can make. The minimum value is 1.

CC Max Monitors

Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time.

The minimum value is 1.

Table 54: Create New IAX Trunk

Type

Select the VoIP trunk type.

  • Peer IAX Trunk
  • Register IAX Trunk

Provider Name

Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc.

Host Name

Configure the IP address or URL for the VoIP provider’s server of the trunk.

Keep Trunk CID

If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”.

Username

Enter the username to register to the trunk from the provider when “Register IAX Trunk” type is selected.

Password

Enter the password to register to the trunk from the provider when “Register IAX Trunk” type is selected.

Disable This Trunk

If selected, the trunk will be disabled.

Table 55: IAX Register Trunk Configuration Parameters

Basic Settings

Provider Name

Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc.

Host Name

Configure the IP address or URL for the VoIP provider’s server of the trunk.

Keep Trunk CID

If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured.

The default setting is “No”.

Disable This Trunk

If selected, the trunk will be disabled.

Caller ID

Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored.

Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call :

From user (Register Trunk Only) 🡪 CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID (Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled) 🡪 Global Outbound CID.

CallerID Name

Configure the name of the caller to be displayed when the extension has no CallerID Name configured.

Username

Enter the username to register to the trunk from the provider.

Password

Enter the password to register to the trunk from the provider.

Advanced Settings

Codec Preference

Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p and VP8.

Enable Heartbeat Detection

If enabled, the UCM6200 will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is “No”.

Heartbeat Frequency

When “Enable Heartbeat Detection” option is set to “Yes”, configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds.

Maximum Number of Call Lines

The maximum number of concurrent calls using the trunk. The default settings 0, which means no limited.

Fax Mode

Select Fax mode. The default setting is “None”.

  • None: Disable Fax.
  • Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38.

Table 56: IAX Peer Trunk Configuration Parameters

Basic Settings

Provider Name

Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc.

Host Name

Configure the IP address or URL for the VoIP provider’s server of the trunk.

Keep Trunk CID

If enabled, the trunk CID will not be overridden by extension’s CID when the extension has CID configured. The default setting is “No”.

Disable This Trunk

If selected, the trunk will be disabled.

Caller ID

Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored.

Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call :

CID from inbound call (Keep Original CID Enabled) 🡪 Trunk Username/CallerID (Keep Trunk CID Enabled) 🡪 DOD 🡪 Extension CallerID Number 🡪 Trunk Username/CallerID (Keep Trunk CID Disabled) 🡪 Global Outbound CID..

CallerID Name

Configure the name of the caller to be displayed when the extension has no CallerID Name configured.

Advanced Settings

Codec Preference

Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.263, H.263p and VP8.

Enable Heartbeat Detection

If enabled, the UCM6200 will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is “No”.

Heartbeat Frequency

When “Enable Heartbeat Detection” option is set to “Yes”, configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds.

Maximum Number of Call Lines

The maximum number of concurrent calls using the trunk. The default settings 0, which means no limited.

Fax Mode

Select Fax mode. The default setting is “None”.

  • None: Disable Fax.
  • Fax Detect: Fax signal from the user/trunk during the call can be detected and the received Fax will be sent to the Email address configured for this extension. If no Email address can be found for the user, the Fax will be sent to the default Email address configured in Fax setting page under Web GUI🡪Call Features🡪Fax/T.38.

Trunk Groups

Users can create VoIP Trunk Groups to register and easily apply the same settings on multiple accounts within the same SIP server. This can drastically reduce the amount of time needed to manage accounts for the same server and improve the overall cleanliness of the web UI.

Figure 102: Trunk Group

Once creating the new trunk group and configuring the SIP settings, users can add multiple accounts within the configured SIP server by pressing button and configuring the username, password and authentication ID fields.

Figure 103: Trunk Group Configuration

Direct Outward Dialing (DOD)

The UCM6200 provides Direct Outward Dialing (DOD) which is a service of a local phone company (or local exchange carrier) that allows subscribers within a company’s PBX system to connect to outside lines directly.

Example of how DOD is used:

Company ABC has a SIP trunk. This SIP trunk has 4 DIDs associated to it. The main number of the office is routed to an auto attendant. The other three numbers are direct lines to specific users of the company. Now when a user makes an outbound call their caller ID shows up as the main office number. This poses a problem as the CEO would like their calls to come from their direct line. This can be accomplished by configuring DOD for the CEO’s extension.

Steps to configure DOD on the UCM6200:

  1. To setup DOD go to UCM6200 Web GUI🡪Extension/Trunk🡪VoIP Trunks page.
  2. Click to access the DOD options for the selected SIP Trunk.
  3. Click “Add DOD” to begin your DOD setup
  4. For “DOD Number” enter one of the numbers (DIDs) from your SIP trunk provider. In the example above Company ABC received 4 DIDs from their provider. ABC will enter in the number for the CEO’s direct line. Note: DOD number cannot exceed 32 characters.
  5. Set the DOD name and If extension number need to be appended to the DID number click on “Add Extension”.
    Note: DOD name cannot exceed 32 characters.
  6. Select an extension from the “Available Extensions” list. Users have the option of selecting more than one extension. In this case, Company ABC would select the CEO’s extension. After making the selection, click on the button to move the extension(s) to the “Selected Extensions” list.

Figure 104: DOD extension selection

  1. Click “Save” at the bottom.

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

Figure 105: Edit DOD

DOD can also be assigned to the UCM’s Fax Sending feature. To do this, select “VFAX” from the extension list when creating or editing a DOD number like shown below.

Figure 106: Fax Sending DOD

DOD Import/Export

UCM62xx now supports to import/export DOD CSV file for an easy configuration.

  • To Import DOD CSV file, click on “Import”.

Note: the file must contain and respect the following columns: DOD Number, DOD Name, Add extension, Local Member, LDAP Members.

Figure 107: DOD Import

  • To export the DOD file, users can click on “Export”.

SLA Station

The UCM6200 supports SLA that allows mapping the key with LED on a multi-line phone to different external lines. When there is an incoming call and the phone starts to ring, the LED on the key will flash in red and the call can be picked up by pressing this key. This allows users to know if the line is occupied or not. The SLA function on the UCM6200 is like BLF but SLA is used to monitor external line i.e., analog trunk on the UCM6200. Users could configure the phone with BLF mode on the MPK to monitor the analog trunk status or press the line key pick up call from the analog trunk on the UCM6200.

Create/Edit SLA Station

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

Figure 108: SLA Station

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

Table 57: SLA Station Configuration Parameters

Station Name

Configure a name to identify the SLA Station.

Station

Specify a SIP extension as a station that will be using SLA.

Available SLA Trunks

Existing Analog Trunks with SLA Mode enabled will be listed here.

Selected SLA Trunks

Select a trunk for this SLA from the Available SLA Trunks list. Click on to arrange the order. If there are multiple trunks selected, when there are calls on those trunks at the same time, pressing the LINE key on the phone will pick up the call on the first trunk here.

SLA Station Options

Ring Timeout

Configure the time (in seconds) to ring the station before the call is considered unanswered. No timeout is set by default. If set to 0, there will be no timeout.

Ring Delay

Configure the time (in seconds) for delay before ringing the station when a call first coming in on the shared line. No delay is set by default. If set to 0, there will be no delay.

Hold Access

This option defines the competence of the hold action for one particular trunk. If set to “open”, any station could hold a call on that trunk or resume one held session; if set to “private”, only the station that places the trunk call on hold could resume the session. The default setting is “open”.

Sample Configuration

  1. On the UCM6200, go to Web GUI🡪Extension/Trunk🡪Analog Trunks page. Create analog trunk or edit the existing analog trunk. Make sure “SLA Mode” is enabled for the analog trunk. Once enabled, this analog trunk will be only available for the SLA stations created under Web GUI🡪Extension/Trunk🡪SLA Station page.

Figure 109: Enable SLA Mode for Analog Trunk

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

Figure 110: Analog Trunk with SLA Mode Enabled

  1. On the UCM6200, go to Web GUI🡪Extension/Trunk🡪SLA Station page, click on “Add”. Please refer to section [Create/Edit SLA Station] for the configuration parameters. Users can create one or more SLA stations to monitor the analog trunk. The following figure shows two stations, 1002 and 1005, are configured to be associated with SLA trunk “fxo1”.

Figure 111: SLA Example – SLA Station

  1. On the SIP phone 1, configure to register UCM6200 extension 1002. Configure the MPK as BLF mode and the value must be set to “extension_trunkname”, which is 1002_fxo1 in this case.
  2. On the SIP phone 2, configure to register UCM6200 extension 1005. Configure the MPK as BLF mode and value must be set to “extension_trunkname”, which is 1005_fxo1 in this case.
C:\Users\jingya.tan\Desktop\Capture1.PNG

Figure 112: SLA Example – MPK Configuration

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

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

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

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

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

  • Answering call using LINE key

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

  • Barging-in active call using BLF key

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

  • Hold/UnHold using BLF key

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

Call Routes

Outbound Routes

In the following sections, we will discuss the steps and parameters used to configure and manage outbound rules in UCM6200, these rules are the regulating points for all external outgoing calls initiated by the UCM through all types of trunks: SIP, Analog and Digital.

Configuring Outbound Routes

In the UCM6200, an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern. This allows different patterns to be dialed through different trunks (e.g., “Local” 7-digit dials through an FXO while “Long distance” 10-digit dials through a low-cost SIP trunk). Users can also set up a failover trunk to be used when the primary trunk fails.

Info

UCM6200 currently supports up to 500 Outbound routes.

Go to Web GUI🡪Extension/Trunk🡪Outbound Routes to add and edit outbound rules.

  • Click on to add a new outbound route.
  • Click on to edit the outbound route.
  • Click on to add a DOD to the outbound route
  • Click on to delete the outbound route.

On the UCM6200, the outbound route priority is based on “Best matching pattern”. For example, the UCM6200 has outbound route A with pattern 1xxx and outbound route B with pattern 10xx configured. When dialing 1000 for outbound call, outbound route B will always be used first. This is because pattern 10xx is a better match than pattern 1xxx. Only when there are multiple outbound routes with the same pattern configured.

Table 58: Outbound Route Configuration Parameters

Calling Rule Name

Configure the name of the calling rule (e.g., local, long_distance, and etc.). Letters, digits, _ and – are allowed.

Pattern

  • All patterns are prefixed with the “_”.
  • Special characters:

X: Any Digit from 0-9.

Z: Any Digit from 1-9.

N: Any Digit from 2-9.

.“: Wildcard. Match one or more characters.

!“: Wildcard. Match zero or more characters immediately.

Example: [123459] – Any digit from 1 to 9.

Notes:

  • Multiple patterns can be used. Each pattern should be entered in new line.
  • Users can add comments to the end of patterns to better organize and keep track of complex rules by typing “/*” and “*/” before and after each comment, respectively.
  • Example:

_X.

_NNXXNXXXXX /* 10-digit long distance */

_818X. /* Any number with leading 818 */

Disable This Route

After creating the outbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore. However, the route settings will remain in UCM. Users can enable it again when it is needed.

Password

Configure the password for users to use this rule when making outbound calls.

Call Duration Limit

Once call duration limit is enabled, it will set the maximum duration of call-blocking.

Maximum Call Duration

User can customize the maximum call duration (in seconds) that is allowed for the outbound call. By default, this value is set to 0 means there is no limit for the call duration.

Warning Time

This option will give caller warning when call duration is approaching to its limit. If the warning time is set to ‘y’, the warning tone will be played to caller when y seconds is left to end the call by UCM.

Warning Repeat Interval

Once this option is set to ‘z’, it will repeatedly be warning caller every z seconds after the first warning.

PIN Groups

If selected, the Password, Privilege Level and Enable Filter on Source Caller ID will not take effect.

For more details, refer to [PIN Groups] section.

PIN Groups with Privilege Level

If enabled and PIN Groups are used, Privilege Levels and Filter on Source Caller ID will also be applied.

Password

Configure the password for users to use this rule when making outbound calls.

Privilege Level

Select privilege level for the outbound rule.

  • Internal:

The lowest level required. All users can use this rule.

  • Local:

Users with Local, National, or International level are allowed to use this rule.

  • National:

Users with National or International level are allowed to use this rule.

  • International:

The highest level required. Only users with international level can use this rule.

The default setting is “Disable”. Please be aware of the potential security risks when using “Internal” level, which means all users can use this outbound rule to dial out from the trunk.

Enable Filter on Source Caller ID

When enabled, users could specify extensions allowed to use this outbound route. “Privilege Level” is automatically disabled if using “Enable Filter on Source Caller ID”.

The following two methods can be used at the same time to define the extensions as the source caller ID.

  1. Select available extensions/extension groups from the left to the right. This allows users to specify arbitrary single extensions available in the PBX.
  2. Custom Dynamic Route: define the pattern for the source caller ID. This allows users to define extension range instead of selecting them one by one.
  • All patterns are prefixed with the “_”.
  • Special characters:

X: Any Digit from 0-9.

Z: Any Digit from 1-9.

N: Any Digit from 2-9.

.“: Wildcard. Match one or more characters.

!“: Wildcard. Match zero or more characters immediately.

Example: [123459] – Any digit from 1 to 9.

Note: Multiple patterns can be used. Patterns should be separated by comma “,”. Example: _X. , _NNXXNXXXXX , _818X.

Outbound Route CID

Attempt to use the configured outbound route CID. This CID will not be used if DOD is configured. It is formatted as “name<number>” or “<number>” or “number”.

Send This Call Through Trunk

Use Trunk

Select the trunk for this outbound rule.

Strip

Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk.

Example:

The users will dial 9 as the first digit of a long-distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In this case, 1 digit should be stripped before the call is placed.

Prepend

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.

Use Failover Trunk

Failover Trunk

Failover trunks can be used to ensure calls can still be made in the scenario that the main trunk is congested or down.

UCM6200 support up to 10 failover trunks.

Example: The user’s primary trunk is a VoIP trunk and the user would like to use the PSTN when the VoIP trunk is not available. The PSTN trunk can be configured as the failover trunk of the VoIP trunk.

Strip

Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk.

Example:

The users will dial 9 as the first digit of a long-distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In this case, 1 digit should be stripped before the call is placed.

Prepend

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.

Time Condition

Time Condition

Outbound routes can be configured to use different trunks based on time conditions such as holidays, office times, specific times, etc.

Outbound Blacklist

The UCM6200 allows users to configure blacklist for outbound routes. If the dialing number matches the blacklist numbers or patterns, the outbound call will not be allowed. The outbound blacklist can be configured under UCM Web GUI🡪Extension/Trunk🡪Outbound Routes: Outbound Blacklist.

Users can configure numbers, patterns or select country code to add in the blacklist. Please note that the blacklist settings apply to all outbound routes.

Figure 113: Country Codes

Blacklist Manage

Add Blacklist Rule

Allows to define a rule based on number(s) or pattern(s) as blacklist entry. Pattern rules:

N : Any digit from 2-9

X : Any digit from 0-9

Z : Any digit from 1-9

. : Wildcard, matching one or more characters

! : Wildcard, matching zero or more characters immediately

: Hyphens are used mainly to improve readability and are not involved in pattern matching.

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

Figure 114: Blacklist Import/Export

PIN Groups

The UCM6200 supports pin group. Once this feature is configured, users can apply pin group to specific outbound routes. When placing a call on pin protected outbound routes, caller will be asked to input the group pin number, this feature can be found on the webGUI🡪Extension/Trunk🡪Outbound Routes🡪PIN Groups.

Name

Specify the name of the group

Record In CDR

Specify Whether or not to display the used PIN group and name in a call’s CDR entry.

PIN Number

Specify the PIN that will be required to dial out.

PIN Name

Specify the name of the PIN

Table 59: Outbound Routes/PIN Group

Once user click on the following figure shows to configure the new PIN.

Figure 115: Create New PIN Group

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

Figure 116: PIN Members

Figure 117: Outbound PIN

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

Figure 118: CDR Record

  • Importing PIN Groups from CSV files:

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

  1. Navigate to Extension/Trunk🡪Outbound Routes🡪PIN Groups and click on the “Choose file to upload” button.
Figure 119: Importing PIN Groups from CSV files

  1. Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in error messages such as the one below:
Screenshot (5)
Figure 120: Incorrect CSV File

  1. To ensure a successful import, please follow the format in the sample image below
pingroup_format
Figure 121: CSV File Format

  • The top-left value (A1) is the PIN Group name. In this case, it is “ALPHA”.
  • Row 2 contains the labels for the modifiable fields: pin and pin_name. These values should not be changed and will cause an upload error otherwise.
  • Rows 3+ contain the user-defined values with Column A holding the PINs and Column B holding the PIN names. PIN values must consist of at least four digits.
  • Once the file is successfully uploaded, the entry will be added to the list of PIN Groups.
pingroup_success
Figure 122: CSV File Successful Upload

  • Exporting PIN Groups from CSV files:

Press button under “Options” to download/export PIN Group settings.

Inbound Routes

Inbound routes can be configured via Web GUI🡪Extension/Trunk🡪Inbound Routes.

Note: UCM6200 currently supports up to 5000 Inbound route.

  • Click on to add a new inbound route.
  • Click on “Blacklist” to configure blacklist for all inbound routes.
  • Click on to edit the inbound route.
  • Click on to delete the inbound route.

Inbound Rule Configurations

Trunks

Select the trunk to configure the inbound rule.

Pattern

– All patterns are prefixed with the "_".

– Special characters:

  • X: Any Digit from 0-9.

  • Z: Any Digit from 1-9.

  • N: Any Digit from 2-9.

  • ".": Wildcard. Match one or more characters.

  • "!": Wildcard. Match zero or more characters immediately.

Example: [12345-9] - Any digit from 1 to 9.

– There are two pattern fields: Pattern and CallerID:

  • Pattern: Specifies the pattern that the dialed number must satisfy to use the inbound route. Example: if the pattern is "_2xxx", the dialed number must be a four-digit number that starts with 2 in order for this route to be used.

  • CallerID Pattern: Specifies the CallerID pattern that callers must satisfy to use the inbound route. Example: if the CallerID pattern is "_1234", only callers with a CID of 1234 can use this route.

Notes:

  • Multiple patterns can be used. Each pattern should be entered in new
    line.

  • Users can add comments to the end of patterns to better organize and
    keep track of complex rules by typing “/*” and “*/” before and after each
    comment respectively.

Example:

Pattern:                                                                       CallerID Pattern:
_X.                                                                                 1000
_NNXXNXXXXX /* 10-digit long distance */                 1001
_818X. /* Any number with leading 818 */

Disable This Route

After creating the inbound route, users can choose to enable and disable it.
If the route is disabled, it will not take effect anymore. However, the route
settings will remain in UCM. Users can enable it again when it is needed.

Allowed to seamless transfer

Allows the selected extension to use this function. If an extension is busy,
and a mobile phone is bound to that extension, the mobile phone can pick
up calls to that extension.

Alert-Info

Configure the Alert-Info, when UCM6200 receives an INVITE request, the
Alert-Info header field specifies an alternative ring tone to the UAS.

Special Ringing Tone

This option will allow users to choose the custom ring back tone to play
when caller reaches the route.

Fax Detection

If enabled, fax signals from the trunk during a call will be detected.

Fax Destination

Configures the destination of faxes.

  • Extension: Send the fax to the designated FXS/SIP extension (fax machine) or a FAX extension.

  • Fax to Email: Send the fax as an email attachment to the designated extension's email address. If the selected extension does not have an associated email address, it will be sent to the default email address configured in the Call Features 🡪Fax/T.38 🡪 Fax Settings page.

Note: Please make sure the sending email
address is correctly configured in System Settings🡪Email Settings.

Prepend Trunk Name

If enabled, the trunk name will be added to the caller id name as the displayed caller id name.

Block Collect Calls

If enabled, collect calls will be blocked.

Note 1: Collect calls are indicated by the header “P-Asserted-Service-Info: service-code=Backward Collect Call, P-Asserted-Service-Info: service-code=Collect Call”.

Note 2: There is also a global setting for this option in the SIP Settings -> General Settings page.

Set Caller ID Info

Manipulates Caller ID (CID) name and/or number within the call flow to help identify who is calling. When enabled two field will show allowing to manipulate the CallerID Number and the Caller ID Name.

CallerID Number

Configures the pattern-matching format to manipulate the numbers of incoming callers or to set a fixed CallerID number for calls that go through this inbound route.

  • ${CALLERID(num)}: Default value which indicates the number of an incoming caller (CID). The CID will not be modified.

  • ${CALLERID(num):n}: Skips the first n characters of a CID number, where n is a number.

  • ${CALLERID(num):-n}: Takes the last n characters of a CID number, where n is a number.

  • ${CALLERID(num):s:n}: Takes n characters of a CID number starting from s+1, where n is a number and s is a character position (e.g. ${CALLERID(num):2:7} takes 7 characters after the second character of a CID number).

  • n${CALLERID(num)}: Prepends n to a CID number, where n is a number.

CallerID Name

Default string is ${CALLERID(name)} which means the name of an incoming caller, it is a pattern-matching syntax format.

A${CALLERID(name)}B means Prepend a character ‘A’ and suffix a character ‘B’ to ${CALLERID(name)}.

Not using pattern-matching syntax means setting fix name to incoming caller.

Enable Route-Level Inbound

Gives uses the ability to configure inbound mode per individual route. When enabled two fields will show allowing to set the Inbound mode and the Inbound mode Suffix.

Note: Global inbound mode must be enabled before users can configure route-level inbound mode

Inbound Mode

Choose the inbound mode for this route.

Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If all routes have the option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be affected by the global inbound mode change.

Inbound Mode Suffix

Dial “Global Inbound Mode feature code + Inbound Mode Suffix” or a route’s assigned suffix to toggle the route’s inbound mode.

The BLF subscribed to the inbound mode suffix can monitor the current inbound mode.

Inbound Multiple Mode

Multiple mode allows user to switch between destinations of the inbound rule by feature codes. Configure related feature codes as described in [Inbound Route: Multiple Mode]. If this option is enabled, user can use feature code to switch between different modes/destinations.

Dial Trunk

This option shows up only when “By DID” is selected. If enabled, the external users dialing in to the trunk via this inbound route can dial outbound call using the UCM6200’s trunk.

Privilege Level

This option shows up only when “By DID” is selected.

  • Disable: Only the selected Extensions or Extension Groups are allowed to use this rule, when enabled Filter on Source Caller ID.

  • Internal: The lowest level required. All users are allowed to use this rule, check this level might be risky for security purpose.

  • Local: User with Local level, National or International level are allowed to use this rule.

  • National: Users with National or International Level are allowed to use this rule.

  • International: The highest level required. Only users with international level are allowed to use this rule.

Allowed DID Destination

This option shows up only when “By DID” is selected. This controls the destination that can be reached by the external caller via the inbound route. The DID destination are:

  • Extension

  • Conference

  • Call Queue

  • Ring Group

  • Paging/Intercom Groups

  • IVR

  • Voicemail Groups

  • Fax Extension

  • Dial By Name

  • All

Default Destination

Select the default destination for the inbound call.

  • Extension

  • Voicemail

  • Conference Room

  • Call Queue

  • Ring Group

  • Paging/Intercom

  • Voicemail Group

  • Fax

  • DISA

  • IVR

  • External Number

  • By DID

When “By DID” is used, the UCM6200 will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR, voicemail groups and Fax extension as configured in “DID destination”. If the dialed number matches the DID pattern, the call will be allowed to go through.

  • Dial By Name

  • Callback

  • Announcement

Strip

Specify the number of digits to strip from the beginning of the DID. This is used when “By DID” is selected in “Default Destination”.

Prepend

Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.

Time Condition

Start Time

Select the start time “hour:minute” for the trunk to use the inbound rule.

End Time

Select the end time “hour:minute” for the trunk to use the inbound rule.

Frequency

Select either “By Week” or “By Month” for the time condition.

If “By Week” is selected, select the weekdays that the time condition will be used.

If “By Month” is selected, select the months that the time condition will be used. Next, select either “Week” or “Day”.

(1) If “Week” is selected, choose the weekdays of the month that the time condition will be used.

(2) If “Day” is selected, choose the specific days of the month that the time condition will be used.

Destination

Select the destination for the inbound call under the defined time condition.

  • Extension

  • Voicemail

  • Conference Room

  • Call Queue

  • Ring Group

  • Paging/Intercom

  • Voicemail Group

  • Fax

  • DISA

  • IVR

  • External Number

  • By DID

When “By DID” is used, the UCM6200 will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR, voicemail groups and Fax extension as configured in “DID destination”. If the dialed number matches the DID pattern, the call will be allowed to go through.

Configure the number of digits to be stripped in “Strip” option.

  • Dial By Name

  • External Number

  • Callback

  • Announcement

Table 60: Inbound Rule Configuration Parameters

Inbound Route: Prepend Example

UCM6200 now allows user to prepend digits to an inbound DID pattern, with strip taking precedence over prepend. With the ability to prepend digits in inbound route DID pattern, user no longer needs to create multiple routes for the same trunk to route calls to different extensions. The following example demonstrates the process:

  1. If Trunk provides a DID pattern of 18005251163.
  2. If Strip is set to 8, UCM6200 will strip the first 8 digits.
  3. If Prepend is set to 2, UCM6200 will then prepend a 2 to the stripped number, now the number become 2163.
  4. UCM6200 will now forward the incoming call to extension 2163.
Figure 123: Inbound Route feature: Prepend

Inbound Route: Multiple Mode

In the UCM6200, the user can configure inbound route to enable multiple mode to switch between different destinations. The inbound multiple mode can be enabled under Inbound Route settings.

Figure 124: Inbound Route – Multiple Mode

When Multiple Mode is enabled for the inbound route, the user can configure a “Default Destination” and a “Mode 1” destination for all routes. By default, the call coming into the inbound routes will be routed to the default destination.

SIP end devices that have registered on the UCM6200 can dial feature code *62 to switch to inbound route “Mode 1” and dial feature code *61 to switch back to “Default Destination”. Switching between different mode can be easily done without Web GUI login.

For example, the customer service hotline destination has to be set to a different IVR after 7PM. The user can dial *62 to switch to “Mode 1” with that IVR set as the destination before off work.

To customize feature codes for “Default Mode” and “Mode 1”, click on under “Inbound Routes” page, check “Enable Inbound Multiple Mode” option and change “Inbound Default Mode” and “Inbound Mode 1” values (By default, *61 and *62 respectively).

Figure 125: Inbound Route – Multiple Mode Feature Codes

Inbound Route: Route-Level Mode

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

Figure 126: Inbound Route – Route-Level Mode

Global inbound mode must be enabled before configuring Route-Level Inbound Mode. Additionally, the Mode 1 must be configured as well.

When Route-Level Inbound Mode is enabled, the user can configure a “Default Destination” and a “Mode 1” destination for each specific route. By default, the call coming into this specific inbound route will be routed to the default destination.

Users can toggle the route’s inbound mode by dialing “Global Inbound Mode feature code + Inbound Mode Suffix” and the current inbound route can be monitored by subscribing a BLF to the Inbound Mode Suffix.

For example, Inbound Default Mode feature code is set to *61 and the Inbound Mode suffix for route 1 is set to 1010. To switch the mode of route 1 to Default Mode, users can dial *611010.

Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If all routes have the option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be affected by the global inbound mode change.

Inbound Route: Inbound Mode BLF Monitoring

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

To do this, please refer to the following steps:

  1. Access the UCM web GUI and navigate to Extension/Trunk🡪Inbound Routes.
  2. Click on the button and enable Inbound Multiple Mode.
  3. Edit the subscribe number field to the desired BLF value.

Figure 127: Global Inbound Mode

  1. Configure the BLF value on a phone’s MPK/VPK. As an example, a GXP2140 with the BLF configured will show the Inbound Mode status on its screen once configured. The 777 BLF is lit green, indicating that the current inbound mode is “Default Mode”.

Figure 128: Inbound Mode – Default Mode

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

Figure 129: Inbound Mode – Mode 1

Inbound Route: Import/Export Inbound Route

Users can now import and export inbound routes to quickly set up inbound routing on a UCM or to back up an existing configuration. An exported inbound route configuration can be directly imported without needing any manual modifications.

Figure 130: Import/Export Inbound Route

The imported file should be on CSV format and using UTF-8 encoding, the imported file should contain below columns, and each column should be separated by a comma (It is recommended to use Notepad++ for the imported file creation):

  • Disable This Route: Yes/No.
  • Pattern: Always prefixed with _
  • CallerID Pattern: Always prefixed with _
  • Prepend Trunk Name: Yes/No.
  • Prepend User Defined Name Enable: Yes/No.
  • Prepend User Defined Name: A string.
  • Alert-info: None, Ring 1, Ring 2… User should enter an Alert-info string following the values we have in the Inbound route Alert-Info list.
  • Allowed to seamless transfer: [Extension_number]
  • Fax Detection: No, Yes.
  • Fax Type: Extension, Fax to Email.
  • Fax Destination: [Extension_number] or [Email address]
  • Inbound Multiple Mode: Yes/No.
  • Default Destination: By DID, Extension, Voicemail… User should enter a Default Destination string following the values we have in the Inbound route Default Destination list.
  • Destination: An Extension number, Ring Group Extension…
  • Default Time Condition.
  • Mode 1: By DID, Extension, Voicemail… User should enter a Default Destination string following the values we have in the mode 1 Default Destination list.
  • Mode 1 Destination: An Extension number, Ring Group Extension…
  • Mode 1 Time Condition.

FAX Detection

The UCM can automatically detect fax signaling from FXO ports and forward these calls to specified destinations.

FAX with Two Media

Since UCM’s system is now based on Asterisk 13, fax re-invites that negotiate with multiple codecs are now supported. If a re-invite contains both T.38 and PCMU/PCMA codecs, T.38 will be prioritized.

Blacklist Configurations

In the UCM6200, Blacklist is supported for all inbound routes. Users could enable the Blacklist feature and manage the Blacklist by clicking on “Blacklist”.

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

Figure 131: Blacklist Configuration Parameters

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

Figure 132: Blacklist csv File

Note:

Users could also add a number to the Blacklist or remove a number from the Blacklist by dialing the feature code for “Blacklist Add’ (default: *40) and “Blacklist Remove” (default: *41) from an extension. The feature code can be configured under Web GUI🡪Call Features🡪Feature Codes.

Conference

The UCM6200 supports conference room allowing multiple rooms used at the same time:

  • UCM6202/6204 supports up to 3 conference rooms allowing up to 25 simultaneous PSTN or IP participants.
  • UCM6208 supports up to 6 conference rooms allowing up to 32 simultaneous PSTN or IP participants.

The conference room configurations can be accessed under Web GUI🡪Call Features🡪Conference. In this page, users could create, edit, view, invite, manage the participants and delete conference rooms. The conference room status and conference call recordings (if recording is enabled) will be displayed in this web page as well.

Conference Room Configurations

  • Click on “Create New Conference Room” to add a new conference room.
  • Click on to edit the conference room.
  • Click on to delete the conference room.

Table 61: Conference Room Configuration Parameters

Extension

Configure the conference number for the users to dial into the conference.

Note: The conference extension number can contain up to 64 characters.

Password

When configured, the users who would like to join the conference call must enter this password before accessing the conference room.

Notes:

  • If “Public Mode” is enabled, the password is not required to join the conference room thus this field is invalid.
  • The password must contain at least 4 characters.
  • Conference extension number can no longer be used as the conference password if Strong Password is enabled.

Host Password

Configure the password to join the conference room as administrator. Conference administrator can manage the conference call via IVR (if “Enable Caller Menu” is enabled) as well as invite other parties to join the conference by dialing “0” (permission required from the invited party) or “1” (permission not required from the invited party) during the conference call.

Notes:

  • If “Public Mode” is enabled, the password is not required to join the conference room thus this field is invalid.
  • Password must be at least 4 characters and different than conference extension number.

Enable Caller Menu

If enabled, conference participant could press the * key to access the conference room menu. The default setting is “No”.

Record Conference

If enabled, the calls in this conference room will be recorded automatically in a .wav format file. All the recording files will be displayed and can be downloaded in the conference web page. The default setting is “No”.

Quiet Mode

If enabled, if there are users joining or leaving the conference, voice prompt or notification tone will not be played. The default setting is “No”.

Note:

“Quiet Mode” and “Announce Callers” cannot be enabled at the same time.

Kick Warning Interval (minutes)

If there is only one participant in a conference room, a kick warning prompt will play at the configured interval. If no input from the participant is received after the prompt, he will be automatically kicked out of the conference. The valid range is 1-60 minutes.

Wait For Admin

If enabled, the participants will not hear each other until the conference administrator joins the conference. The default setting is “No”.

Note:

If “Quiet Mode” is enabled, the voice prompt for “Wait For Admin” will not be announced.

Allow User Invite

If enabled, users could press 0 to invite other users (with the users’ permission) or press 1 to invite other users (without the user’s permission) to join the conference. The default setting is “No”.

Note:

Conference administrator can always invite other users without enabling this option.

Announce Callers

If enabled, the caller will be announced to all conference participants when there the caller joins the conference. The default setting is “No”.

Notes:

  • “Quiet Mode” and “Announce Callers” cannot be enabled at the same time.
  • Conference participant names will be announced when joining/leaving the conference even when the conference is on hold..

Public Mode

If enabled, no authentication will be required when joining the conference call. The default setting is “Yes”.

Play Hold Music

If enabled, the UCM6200 will play Hold music when there is only one user in the conference. The default setting is “No”.

Music On Hold

Select the music on hold playlist to be played in conference call. Music On Hold playlist can be set up under Web GUI🡪PBX Settings🡪Music On Hold.

Custom Music On Hold

Select a custom Music On Hold.

Skip Authentication

If enabled, the invitation from Web GUI for a conference room with password will skip the authentication for the invited users. The default setting is “No”.

Conference Settings contains the following options:

Table 62: Conference Settings

Enable Talk detection

If enabled, the AMI will send the corresponding event when a user starts or ends talking.

DSP Talking Threshold

The time in milliseconds of sound above what the dsp has established as base line silence for a user before a user is considered to be talking. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 128.

DSP Silence Threshold

The time in milliseconds of sound falling within the what the dsp has established as base line silence before a user is considered to be silent. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 2500.

Enable Neteq

If enabled, conference audio quality may improve, but system performance will be lowered.

Users can check the talking Caller IDs in conference control page (UCM WebUI🡪Call Features🡪Conference). The image will move up and down when the user is talking.

Figure 133: Conference

Conference Call Operations

Join a Conference Call

Users could dial the conference room extension to join the conference. If password is required, enter the password to join the conference as a normal user, or enter the admin password to join the conference as administrator.

Invite Other Parties to Join Conference

When using the UCM6200 conference room., there are two ways to invite other parties to join the conference.

  • Invite from Web GUI.

For each conference room in UCM6200 Web GUI🡪Call Features🡪Conference, there is an icon for option “Invite a participant”. Click on it and enter the number of the party you would like to invite. Then click on “Add”. A call will be sent to this number to invite it to the conference.

Figure 134: Conference Invitation from Web GUI

  • Invite by dialing 0 or 1 during conference call.

A conference participant can invite other parties to the conference by dialing during the conference call. Please make sure “Allow User Invite” is turned on for the conference room first. Enter 0 or 1 during the conference call. Follow the voice prompt to input the number of the party you would like to invite. A call will be sent to this number to join it into the conference.

If 0 is entered, once the invited party picks up the invitation call, the system will ask the party to accept or reject the invitation.

If 1 is entered, once the invited party picks up the call, they will automatically be brought into the conference.

Note:

Conference administrator can always invite other parties from the phone during the call by entering 0 or 1. To join a conference room as administrator, enter the admin password when joining the conference. A conference room can have multiple administrators.

During The Conference

During the conference call, users can manage the conference from Web GUI or IVR.

  • Manage the conference call from Web GUI.

Log in UCM6200 Web GUI during the conference call, the participants in each conference room will be listed.

  1. Click on to kick a participant from the conference.
  2. Click on to mute the participant.
  3. Click on to lock this conference room so that other users cannot join it anymore.
  4. Click on to invite other users into the conference room.
  • Manage the conference call from IVR.

If “Enable Caller Menu” is enabled, conference participant can input * to enter the IVR menu for the conference. Please see options listed in the table below.

Table 63: Conference Caller IVR Menu

Conference Administrator IVR Menu

1

Mute/unmute yourself.

2

Lock/unlock the conference room.

3

Kick the last joined user from the conference.

4

Decrease the volume of the conference call.

5

Decrease your volume.

6

Increase the volume of the conference call.

7

Increase your volume.

8

More options.

  • 1: List all users currently in the conference call.
  • 2: Kick all non-Administrator participants from the conference call.
  • 3: Mute/Unmute all non-Administrator participants from the conference call.
  • 4: Record the conference call.
  • 8: Exit the caller menu and return to the conference.

Conference User IVR Menu

1

Mute/unmute yourself.

4

Decrease the volume of the conference call.

5

Decrease your volume.

6

Increase the volume of the conference call.

7

Increase your volume.

8

Exit the caller menu and return to the conference.

Note:

When there is participant in the conference, the conference room configuration cannot be modified.

Google Service Settings Support

UCM6200 now supports Google OAuth 2.0 authentication. This feature is used for supporting UCM6200 conference scheduling system. Once OAuth 2.0 is enabled, UCM6200 conference system can access Google calendar to schedule or update conference.

Google Service Settings can be found under Web GUI🡪Call Features🡪Conference🡪Google Service Settings🡪Google Service Settings.

Figure 135: Google Service Settings🡪OAuth2.0 Authentication

If you already have OAuth2.0 project set up on Google Developers web page, please use your existing login credential for “OAuth2.0 Client ID” and “OAuth2.0 Client Secret” in the above figure for the UCM6200 to access Google Service.

If you do not have OAuth2.0 project set up yet, please following the steps below to create new project and obtain credentials:

  1. Go to Google Developers page https://console.developers.google.com/start Create a New Project in Google Developers page.

Figure 136: Google Service🡪New Project

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

Figure 137: Google Service🡪Create New Credential

  1. Use the newly created login credential to fill in “OAuth2.0 Client ID” and “OAuth2.0 Client Secret”.
  2. Click “Get Authentication Code” to obtain authentication code from Google Service.

Figure 138: Google Service🡪OAuth2.0 Login

  1. Now UCM6200 is connected with Google Service.

You can also configure the Status update, which automatically refresh your Google Calendar with the configured time (m). Note: Zero means disable.

Conference Schedule

Conference Schedule can be found under UCM6200 Web GUI 🡪 Call Features 🡪 Conference🡪 Conference Schedule. Users can create, edit, view and delete a Conference Schedule.

  • Click on “Create New Conference Schedule” to add a new Conference Schedule.
  • Click on the scheduled conference to edit or delete the event.

After the user configures UCM6200 with Google Service Settings [Google Service Settings Support] and enables Google Calendar for Conference Schedule, the conference schedule on the UCM6200 can be synchronized with Google Calendar for authorized Google account.

Table 64: Conference Schedule Parameters

Schedule Options

Conference Subject

Configure the topic of the scheduled conference. Letters, digits, _ and – are allowed.

Conference Room

Select a conference room for this scheduled conference.

Conference Password

Conference login password.

Host Password

Host Password.

Kick Time(m)

Set kick time before conference starts. When kick time is reached, a warning prompt will be played for all attendees in the conference room. After 5 minutes, this conference room will be cleared and locked for the scheduled conference to begin.

Note: Kick Time cannot be less than 6 minutes in order to clear the conference room.

Wait for Host

If enabled, conference participants will not hear each other until the host joins the conference.
Note: If Quiet Mode is enabled, the voice prompt for this option will not be played.

Description

The description of scheduled conference.

Repeat

Repeat interval of scheduled conference. By default, set to single event.

Schedule Time

Configure the beginning date and duration of scheduled conference.

Note: Please pay attention to avoid time conflict on schedules in the same conference room.

Meeting Duration

Duration of the conference meeting.

Note: The maximum allowed meeting duration that can be set is 8 hour(s).

Enable Google Calendar

Select this option to synchronize scheduled conference with Google Calendar.

Note: Google Service Setting OAuth2.0 must be configured on the UCM6200. Please refer to section [Google Service Settings Support].

Send email notification

Sends Email notification to the extension.

Conference Administrator

Select the administrator of scheduled conference from selected extensions.

Note: “Public Mode” must be disabled from Conference Room Options tab.

Local Extension

Select available extensions from the list to attend scheduled conference.

Remote Extension

Select available extensions from the remote peer PBX.

Note: “LDAP Sync” must be enabled on the UCM6200 in order to view remote extensions here.

Special Extension

Add extensions that are not in the list (both local and remote list). If the user wishes to add the special extension, please make sure that the UCM’s outbound routes will allow calling that special extension.

Remote Conference

Invite a remote conference.

Conference Room Options

Password

Configure conference room password. Please note that if “Public Mode” is enabled, this option is automatically disabled.

Host Password

Configure the password to join as conference administrator.

Please note that if “Public Mode” is enabled, this option is automatically disabled.

Enable Caller Menu

If this option is enabled, conference participants will be able to access conference room menu by pressing the * key.

Record Conference

If this option is enabled, conference call will be recorded in .wav format. The recorded file can be found from Conference page.

Quiet Mode

If this option is enabled, the notification tone or voice prompt for joining or leaving the conference will not be played.

Note: Option “Quiet Mode” and option “Announce Caller” cannot be enabled at the same time.

Wait For Admin

If this option is enabled, the participants in the conference will not be able to hear each other until conference administrator joins the conference.

Note: If “Quiet Mode” is enabled, voice prompt for this option will not be played.

Allow User Invite

If this option is enabled, the user can:

  • Press ‘0’ to invite others to join the conference with invited party’s permission
  • Press ‘1’ to invite without invited party’s permission
  • Press ‘2’ to create a multi-conference room to another conference room
  • Press ‘3’ to drop all current multi-conference rooms

Note: Conference Administrator is always allowed to access this menu.

Announce Callers

If this option is enabled, when a participant joins the conference room, participant’s name will be announced to all members in the conference room.

Note: Option “Quiet Mode” and option “Announce Caller” cannot be enabled at the same time.

Public Mode

If this option is enabled, no authentication is required for entering the conference room.

Note: Please be aware of the potential security risks when turning on this option.

Play Hold Music

If this option is enabled, UCM6200 will play Hold Music while there is only one participant in the conference room, or the conference is not yet started.

Skip Authentication

If this option is enabled, the invitation from Web GUI via a trunk with password will not require authentication.

Note: Please be aware of the potential security risks when turning on this option.

Cleaner Options

Cleaner Options

Enable Conference Schedules Cleaner

If this option is enabled, conference schedules will be automatically cleaned as configured.

Conference Schedules Clean Time

Enter the clean time (in hours). The valid range is from 0 to 23.

Clean Interval

Enter the clean interval (in days). The valid range is from 1 to 30.

After configuring the scheduled conference, it will be shown under Conference Schedule page as below:

Figure 139: Conference Schedule

Once the conference room is scheduled, at the kick time, all users will be removed from conference room and no extension can join the conference room anymore. At the scheduled conference time, UCM6200 will send INVITE to the extensions that have been selected for conference.

Notes:

  • Please make sure that outbound route is properly configured for remote extensions to join the conference.
  • Once Kick Time is reached, Conference Schedule is locked and cannot be modified.

Contact Group

Users can now quickly invite multiple participants at once to a conference via conference contact groups. Up to 5 contact groups can be created. The maximum allowed number of contacts per group is based on the UCM model’s conference participant limit: 25 for 6202/6204, 32 for 6208, 64 for 6510.

Each contact group must have a password configured, which will be required when inviting the specified contact group to a conference. Additionally, an audio file can be uploaded to each group to be used to announce the contact group name such as “Sales” or “Marketing”. The default announcement for each group is “Conference Contact Group 1”, “Conference Contact Group 2”, etc.

Figure 140: Contact Group Parameters

Contact Group Configurations

  • Click on “Create New Contact Group” to add a new Contact Group.
  • Click on to edit the Contact Group.
  • Click on to delete the Contact Group.

Table 65: Contact Group Parameters

Name

Name associated to the contact group.

Password

Password required to invite the specified contact group to a conference.

Prompt

Audio file that can be uploaded to the group to announce the contact group name such as “Sales” or “Marketing”.

The default announcement for each group is “Conference Contact Group 1”, “Conference Contact Group 2”, etc.

Members

Contacts that needs to be added in each group.

Type

Type of the members to be added, it can be either Extensions or a self-defined number.

Conference Recordings

The UCM6200 allows users to record the conference call and retrieve the recording from Web GUI🡪Call Features🡪Conference🡪Record Conference.

To record the conference call, when the conference room is in idle, enable “Record Conference” from the conference room configuration dialog. Save the setting and apply the change. When the conference call starts, the call will be automatically recorded in .wav format.

The recording files will be listed as below once available. Users could click on to download the recording or click on to delete the recording. Users could also delete all recording files by clicking on “Delate All Recording Files” or delete multiple recording files at once by clicking on “Delete Selected Recording Files” after selecting the recording files.

Figure 141: Conference Recording

Conference Call Statistics

Conference reports will now be generated after every conference. These reports can be exported to a .CSV file for offline viewing. The conference report page can be accessed by clicking on the Call Statistics button on the main Conference page.

Figure 142: Conference Call Statistics

Figure 143: Conference Report on Web

Figure 144: Conference Report on CSV

Video Conference

With the UCM you can easily create, schedule, manage, and join video conference calls, from your desktop or laptop computer. UCM Video conferencing uses WebRTC technology, so all the participants don’t have to download and install any additional software or plugins. If upgrading from firmware that does not have this functionality, system administrators must first toggle on the Enable WebRTC option for extensions that want to use the UCM’s WebRTC video conferencing feature. The video conference configurations can be accessed under Web GUI🡪Call Features🡪Video Conference. Here, users can create and manage video conference rooms and schedule video conferences.

Video Conference Room Configurations

  • Click on “Create New Conference Room” to add a new conference room.
  • Click on to edit the conference room.
  • Click on to delete the conference room.

Table 66: Video Conference room Configuration Parameters

Extension

Configure the conference number for the users to dial into the conference.

Note: Up to 64 characters.

Password

When configured, the users who would like to join the conference call must enter this password before accessing the conference room.

Note:

  • Only digits are allowed.
  • The password has to be at least 4 characters. All repetitive and sequential digits (e.g., 0000, 1111, 1234 and 2345) or common digits (e.g., 111222 and 321321) are not allowed.

Conference Settings

Figure 145: Video Conference Basic settings

Basic Settings

Video Conferencing

This option should be enabled in order to activate the Video Conference feature.

Bind UDP Port

Configure the UDP port number for MCM. The standard UDP port for MCM is 5062.

Packet Loss Retransmission

Configure to enable Packet Loss Retransmission.

FEC

If enabled, the Forward Error Correction (FEC) will be activated. The default setting is “No”.

Enable Talk Detection

If enabled, the AMI will send the corresponding event when a user starts or stops talking.

DSP Talking Threshold

The amount of time (ms) that sound exceeds what the DSP has established as the baseline for silence before a user is considered to be talking. This value affects several operations and should not be changed unless the impact on call quality is fully understood.

DSP Silence Threshold

The amount of time(ms) that sound falls within what the DSP has established as the baseline for silence before a user is considered be silent. This value affects several operations and should not be changed unless the impact on call quality is fully understood.

Conference Schedule

Conference Schedule can be found under UCM Web GUI 🡪 Call Features🡪Video Conference 🡪 Conference Schedule. Users can create, edit, view and delete a Conference Schedule.

  • Click on “Schedule New Conference” to add a new Conference Schedule.
  • Click on the scheduled conference to edit or delete the event.

Table 67: Video Conference Schedule Parameters

Schedule Options

Conference Subject

Configure the name of the scheduled conference. Letters, digits, _ and – are allowed.

Conference Room

Select a conference room for this scheduled conference.

Conference Password

Configure conference room password. Please note that if “Public Mode” is enabled, this option is automatically disabled.

Kick Time(m)

Configure the time before the scheduled conference. When this time is reached, a warning prompt will be played, and all attendees currently in the scheduled conference room will be kicked after 5 mins. The conference room will be locked until the scheduled conference begins. Default value is 10 min.

Start Time

Configure the beginning date and duration of scheduled conference.

Note: Please be mindful to avoid schedule conflicts for the same conference room.

Host

Set the admin of this scheduled conference from the following list of members.

Repeat

Choose when to repeat a scheduled conference.

Local Extension

Select the extensions from the list to attend this scheduled conference.

Remote Extension

The remote extension in the peer PBX connected to the local PBX via LDAP sync.

Special Extension

Add extensions that are not in the list (both local and remote list). If the user wishes to add the special extension, please match the pattern on the outbound route.

Description

Set a description of scheduled conference.

Time Zone

Configure the conference Time Zone.

Once created, the Web GUI will display scheduled conference in Conference Schedule.

Please see figure below:

Figure 146: Video Conference Schedule

Once the conference room is scheduled, at the kick time, all users will be removed from conference room and no extension is allowed to join the conference room anymore. At the scheduled conference time, UCM will send INVITE to the extensions that have been selected for conference.

Notes:

  • Video conferencing can be resource-intensive and may cause performance issues with the UCM when used.
  • To ensure the best experience, please use Google Chrome (v67 or higher) or Mozilla Firefox (v60).

Wave WebRTC Video Calling & Conferencing

Web audio and video calls and conferencing can now be achieved through the UCM’s new WebRTC page. To get started with this new feature, please make sure to:

  1. Navigate to Value-Added Features 🡪 WebRTC and enable WebRTC support.
videoconference1

Figure 147: Enabling WebRTC Feature

  1. Enable the WebRTC on the extensions that would use this feature under Extension / Trunk 🡪 Extensions.
videoconference6

Figure 148: Enabling WebRTC on Extensions

The UCM offers the possibility to log in to an extension via Grandstream Wave Portal using user portal password in addition to SIP registration password, where it offers a sleek interface to host conferences, receive email reminders for scheduled conferences, manage contacts, initiate calls, call transfer, chat functionality and more.

Access the page by adding “/gswave” after the UCM’s server address and port. (e.g. https://my.ucm.com:8089/gswave).

Figure 149: Grandstream Wave Interface

Note: Starting with 1.0.19.27, the registration limit was increased to 300 for UCM62xx. The limit is the same regardless of whether the user is making voice calls or video calls

For more details about the WebRTC feature, please refer to the following guide:

https://documentation.grandstream.com/knowledge-base/wave-video-calling-conferencing-guide/

IPVideoTalk Meetings

UCM extensions can now dial into IPVT (IPVideoTalk) meetings by creating a peer trunk to an IPVT server. However, users must make sure that the IPVT server they are peering to also has a peer trunk to their UCM configured. This setting can be found in Admin Center🡪SIP Trunk Configuration.

Figure 150: IPVT SIP Trunk page

Next, users must create a peer trunk on the UCM to the IPVT server. Enter one of the following addresses based on the desired connection protocol:

  • TCP: pro.ipvideotalk.com:20000
  • TLS: pro.ipvideotalk.com:20001

Figure 151 – Peer Trunk to IPVT

Make sure that the Transport field is either “TCP” or “TLS”. Save and apply changes to create the trunk.

Next, edit the newly created trunk and click on the Advanced Settings tab. Make sure the IPVT Mode option is checked. Otherwise, you may experience audio issues when dialing into IPVT.

Figure 152 – IPVT Mode

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

Figure 153 – IPVT Outbound Pattern

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

Figure 154 – IPVT Outbound Strip

In this example, the Strip field has “3” configured to remove the example unique code *99 from the dialed number before the call is routed out to the IPVT server. Once this outbound route has been created, users can now use a UCM extension to dial IPVT meeting rooms.

Note: An IPVT account can have only 1 SIP trunk peered to it.

IVR

Configure IVR

IVR configurations can be accessed under the UCM6200 Web GUI🡪Call Features🡪IVR. Users could create, edit, view and delete an IVR.

Note: UCM6200 currently supports up to 500 IVR entry.

  • Click on “Create New IVR” to add a new IVR.
  • Click on to edit the IVR configuration.
  • Click on to delete the IVR.

Figure 155: Create New IVR

Table 68: IVR Configuration Parameters

Basic Settings

Name

Configure the name of the IVR. Letters, digits, _ and – are allowed.

Extension

Enter the extension number for users to access the IVR.

Dial other Extensions

If enabled, the caller is allowed to dial extensions other than the ones explicitly defined.

DID Destination

This option shows up only when “By DID” is selected. This controls the destination that can be reached by the external caller via the inbound route. The DID destination are:

  • Extension
  • Conference
  • Call Queue
  • Ring Group
  • Paging/Intercom Groups
  • Voicemail Groups
  • Fax Extension
  • Dial By Name
  • All

Dial Trunk

If enabled, all callers to the IVR is allowed to use trunk. The permission must be configured for the users to use the trunk first. The default setting is “No”.

Permission

Assign permission level for outbound calls if “Dial Trunk” is enabled. The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level. The default setting is “Internal”. If the user tries to dial outbound calls after dialing into the IVR, the UCM6200 will compared the IVR’s permission level with the outbound route’s privilege level. If the IVR’s permission level is higher than (or equal to) the outbound route’s privilege level, the call will be allowed to go through.

Replace Caller ID

If enabled, the UCM will replace the caller display name with the IVR name the caller know whether the call is incoming from a direct extension or an IVR.

Alert-Info

When present in an INVITE request, the alert-Info header field specifies and alternative ring tone to the UAS.

Welcome Prompt

Select an audio file to play as the welcome prompt for the IVR. Click on “Prompt” to add additional audio file under Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt.

Note: Up to 5 welcome prompts can be used.

Digit Timeout

Configure the timeout between digit entries. After the user enters a digit, the user needs to enter the next digit within the timeout. If no digit is detected within the timeout, the UCM6200 will consider the entries complete. The default timeout is 3 seconds.

Response Timeout

After playing the prompts in the IVR, the UCM6200 will wait for the DTMF entry within the timeout (in seconds). If no DTMF entry is detected within the timeout, a timeout prompt will be played. The default setting is 10 seconds.

Response Timeout Prompt

Select the prompt message to be played when timeout occurs.

Invalid Input Prompt

Select the prompt message to be played when an invalid extension is pressed.

Response Timeout Prompt Repeats

Configure the number of times to repeat the prompt if no DTMF input is detected. When the loop ends, it will go to the timeout destination if configured, or hang up. The default setting is 3.

Invalid Input Prompt Repeats

Configure the number of times to repeat the prompt if the DTMF input is invalid. When the loop ends, it will go to the invalid destination if configured, or hang up. The default setting is 3.

Language

Select the voice prompt language to be used for this IVR. The default setting is “Default” which is the selected voice prompt language under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. The dropdown list shows all the current available voice prompt languages on the UCM6200. To add more languages in the list, please download voice prompt package by selecting “Check Prompt List” under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Key Pressing Events

Key Press Event:

Press 0

Press 1

Press 2

Press 3

Press 4

Press 5

Press 6

Press 7

Press 8

Press 9

Press *

Timeout

Invalid

Select the event for each key pressing for 0-9, *, Timeout and Invalid. The event options are:

  • Extension
  • Voicemail
  • Conference Rooms
  • Voicemail Group
  • IVR
  • Ring Group
  • Queues
  • Page Group
  • Fax
  • Custom Prompt
  • Hangup
  • DISA
  • Dial By Name
  • External Number
  • Callback
  • Announcement

Figure 156: Key Pressing Events

IVR Black/Whitelist

In some scenarios, the IPPBX administrator needs to restrict the extensions that can be reached from IVR.
For example, the company CEO and directors prefer only receiving calls transferred by the secretary, some special extensions are used on IP surveillance end points which shouldn’t be reached from external calls via IVR for privacy reason. UCM has now added blacklist and whitelist in IVR settings for users to manage this.

Note: up to 500 extensions are allowed on the black/white list.

To use this feature, log in UCM Web GUI and navigate to Call Features🡪IVR🡪Create/Edit IVR: IVR
Black/White List.

  • If the user selects “Blacklist Enable” and adds extension in the list, the extensions in the list will not be allowed to be reached via IVR.
  • If the user selects “Whitelist Enable” and adds extension in the list, only the extensions in the list can be allowed to be reached via IVR.

Figure 157: Black/White List

Create Custom Prompt

To record new IVR prompt or upload IVR prompt to be used in IVR, click on “Prompt” next to the “Welcome Prompt” option and the users will be redirected to Custom Prompt page. Or users could go to Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page directly.

Figure 158: Click on Prompt to Create IVR Prompt

Once the IVR prompt file is successfully added to the UCM6200, it will be added into the prompt list options for users to select in different IVR scenarios.

Voice Prompt

The UCM6200 supports multiple languages in Web GUI as well as system voice prompt. The following languages are currently supported in system voice prompt:

English (United States), Arabic, Chinese, Dutch, English (United Kingdom), French, German, Greek, Hebrew, Italian, Polish, Portuguese, Russian, Spanish, Catalan, Swedish, Czech and Turkish.

English (United States) and Chinese voice prompts are built in with the UCM6200 already. The other languages provided by Grandstream can be downloaded and installed from the UCM6200 Web GUI directly. Additionally, users could customize their own voice prompts, package them and upload to the UCM6200.

Language settings for voice prompt can be accessed under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. Additionally, UCM6200 allows to customize specific prompt instead of full language package, and it provides ability to upload greeting files for extensions.

Language Settings

Download and Install Voice Prompt Package

To download and install voice prompt package in different languages from UCM6200 Web GUI, click on “Add Voice Prompt Package” button.

Figure 159: Language Settings for Voice Prompt

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

Figure 160: Voice Prompt Package List

Click on to download the language to the UCM6200. The installation will be automatically started once the downloading is finished.

Figure 161: New Voice Prompt Language Added

A new language option will be displayed after successfully installed. Users then could select it to apply in the UCM6200 system voice prompt or delete it from the UCM6200.

Upload Language Package

On the UCM6200, if the user needs to replace some specific customized prompt, the user can upload a single specific customized prompt from Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings instead of the entire language pack.

Figure 162: Upload Voice Prompts Package

The package file should follow below requirements:

Each file uploaded must be under 50MB.

Package structure:

[Package]
├ [voice prompt dir]
│ ├ [... dir]
│ └ [... files]
└ info.txt (containing the language name for display, in UTF8)
  • Language dir name format:
  • Custom dir name format: language_xxx;

For example: If there is a Chinese custom directory named zh_xxx, the custom voice prompt in zh_xxx would be used first, then the Chinese voice prompt zh, then use the default language prompt (en); If not named the format as above, then the custom prompt will be used first, then use the default language prompt (en).

For more details, please refer to:

https://www.grandstream.com/sites/default/files/Resources/ucm_voiceprompt_customization_guide.zip

Custom Prompt

Record New Custom Prompt

In the UCM6200 Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page, click on “Record New IVR Prompt” and follow the steps below to record new IVR prompt.

Figure 163: Record New IVR Prompt

  1. Specify the IVR file name.
  2. Select the format (GSM or WAV) for the IVR prompt file to be recorded.
  3. Select the extension to receive the call from the UCM6200 to record the IVR prompt.
  4. Click the “Record” button. A request will be sent to the UCM6200. The UCM6200 will then call the extension for recording the IVR prompt from the phone.
  5. Pick up the call from the extension and start the recording following the voice prompt.
  6. The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play or delete the recording.

Upload Custom Prompt

If the user has a pre-recorded IVR prompt file, click on “Upload IVR Prompt” in Web GUI🡪PBX Settings🡪Voice Prompt🡪Custom Prompt page to upload the file to the UCM6200. The following are required for the IVR prompt file to be successfully uploaded and used by the UCM6200:

  • PCM encoded.
  • 16 bits.
  • 8000Hz mono.
  • In .mp3 or .wav format; or raw/ulaw/alaw/gsm file with .ulaw or .alaw suffix.
  • File size under 5MB.
  • Filename should not exceed 100 characters.

Figure 164: Upload IVR Prompt

Click on “choose file to upload” to select audio file from local PC and to start uploading. Once uploaded, the file will appear in the IVR Prompt web page.

Download All Custom Prompt

On the UCM6200, the users can download all custom prompts from UCM Web GUI to local PC. To download all custom prompt, log in UCM Web GUI and navigate to PBX Settings🡪Voice Prompt🡪Custom Prompt and click on . The following window will pop up in order to set a name for the downloaded file.

Figure 165: Download All Custom Prompt

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

Username Prompt Customization

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

Upload Username Prompt File from Web GUI

  1. First, users should have a pre-recorded file respecting the following format:
  • PCM encoded / 16 bits / 8000Hz mono.
  • In “.GSM” or “.WAV” format.
  • File size under 5MB.
  • Filename must be set as the extension number with 18 characters max. For example, the recorded file name 1000.wav will be used for extension 1000.
  1. Go under web GUI PBX Settings 🡪 Voice Prompt 🡪 Username Prompt and click on button.
  2. Select the recorded file to upload it and press Save and Apply Settings.
  3. To delete username prompts, either click on the button to delete a single file or select multiple files and click on Delete button to delete multiple prompts at once.

Record Username via Voicemail Menu

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

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

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

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

Voicemail

Configure Voicemail

If the voicemail is enabled for UCM6200 extensions, the configurations of the voicemail can be globally set up and managed under Web GUI🡪Call Features🡪Voicemail.

Figure 166: Voicemail Settings

Table 69: Voicemail Settings

Max Greeting

Configure the maximum number of seconds for the voicemail greeting. The default setting is 60 seconds.

Dial ‘0’ For Operator

If enabled, the caller can press 0 to exit the voicemail application and connect to the configured operator’s extension.

Operator Extension

Select the operator extension, which will be dialed when users press 0 to exit voicemail application. The operator extension can also be used in IVR.

Max Messages Per Folder

Configure the maximum number of messages per folder in users’ voicemail. The valid range 10 to 1000. The default setting is 50.

Max Message Time

Select the maximum duration of the voicemail message. The message will not be recorded if the duration exceeds the max message time. The default setting is 15 minutes. The available options are:

  • 1 minute
  • 2 minutes
  • 5 minutes
  • 15 minutes
  • 30 minutes
  • Unlimited

Min Effective Message Time

Configure the minimum duration (in seconds) of a voicemail message. Messages will be automatically deleted if the duration is shorter than the Min Message Time. The default setting is 3 seconds. The available options are:

  • No minimum
  • 1 second
  • 2 seconds
  • 3 seconds
  • 4 seconds
  • 5 seconds

Note: Silence and noise duration are not counted in message time.

Announce Message Caller-ID

If enabled, the caller ID of the user who has left the message will be announced at the beginning of the voicemail message. The default setting is “No”.

Announce Message Duration

If enabled, the message duration will be announced at the beginning of the voicemail message. The default setting is “No”.

Play Envelope

If enabled, a brief introduction (received time, received from, and etc.) of each message will be played when accessed from the voicemail application. The default setting is “Yes”.

Play from Last

If enabled, UCM will play from the voice message left most recently; if disabled, UCM will play from the earliest left voice message

Allow User Review

If enabled, users can review the message following the IVR before sending.

Voicemail Remote Access

If enabled, external callers routed by DID and reaching VM will be prompted by the UCM with 2 options:

  • Press 1 to leave a message.

To leave a message for the extension reached by DID.

  • Press 2 to access voicemail management system.

This will allow caller to access any extension VM after entering extension number and its VM password.

Note: This option applies to inbound call routed by DID only.

The default setting is “Disabled”.

Forward Voicemail to Peered UCMs

Enables the forwarding of voicemail to remote extensions on peered SIP trunks.

The default setting is “Disabled”.

Voicemail Password

Configures the default voicemail password that will be used when an extension is reset.

Note: Resetting an extension will reset Voicemail Password, Send Voicemail to Email, and Keep Voicemail after Emailing values to default. Previous custom voicemail prompts and messages will be deleted.

Access Voicemail

If the voicemail is enabled for UCM6200 extensions, the users can dial the voicemail access number (by default *97) to access their extension’s voicemail. The users will be prompted to enter the voicemail password and then can enter digits from the phone keypad to navigate in the IVR menu for different options.

Otherwise, the user can dial the voicemail access code (by default *98) followed by the extension number and password in order to access to that specific extension’s voicemail.

Table 70: Voicemail IVR Menu

Main Menu

Sub Menu 1

Sub Menu 2

1 – New messages

3 – Advanced options

1 – Send a reply

2 – Call the person who sent this message

3 – Hear the message envelop

4 – Leave a message

* – Return to the main menu

5 – Repeat the current message

7 – Delete this message

8 – Forward the message to another user

9 – Save

* – Help

# – Exit

2 – Change folders

0 – New messages

1 – Old messages

2 – Work messages

3 – Family messages

4 – Friend messages

# – Cancel

3 – Advanced options

1 – Send a reply

2 – Call the person who sent this message

3 – Hear the message envelop

4 – Leave a message

* – Return to the main menu

0 – Mailbox options

1 – Record your unavailable message

1 – Accept this recording

2 – Listen to it

3 – Re-record your message

2 – Record your busy message

1 – Accept this recording

2 – Listen to it

3 – Re-record your message

3 – Record your name

1 – Accept this recording

2 – Listen to it

3 – Re-record your message

4 – Record temporary greeting

1 – Accept this recording

2 – Listen to it

3 – Re-record your message

5 – Change your password

* – Return to the main menu

Leaving Voicemail

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

In case if the caller is calling from an internal extension, they will be directly forwarded to the extension’s voicemail box.

But if the caller is calling from outside the system and the incoming call is routed by DID to the destination extension, then the caller will be prompted with the choice to either press1 to access voicemail management or press 2 to leave a message for the called extension. This feature could be useful for remote voicemail administration.

Voicemail Email Settings

The UCM6200 can be configured to send the voicemail as attachment to Email. Click on “Voicemail Email Settings” button to configure the Email attributes and content.

Table 71: Voicemail Email Settings

Attach Recordings to

E-Mail

If enabled, voicemails will be sent to user’s Email address. The default setting is “Yes”.

Keep Recordings

If enabled, voicemail will be stored in the UCM6200 after the email is sent. The default setting is “Yes”.

Template for Voicemail Emails

Fill in the “Subject:” and “Message:” content, to be used in the Email when sending to the user.

The template variables are:

  • \t: TAB
  • ${VM_NAME}: Recipient’s first name and last name
  • ${VM_DUR}: The duration of the voicemail message
  • ${VM_MAILBOX}: The recipient’s extension
  • ${VM_CALLERID}: The caller ID of the person who has left the message
  • ${VM_MSGNUM}: The number of messages in the mailbox
  • ${VM_DATE}: The date and time when the message is left

Figure 167: Voicemail Email Settings

Click on “Load Default Settings” button to view the default template as an example.

Configure Voicemail Group

The UCM6200 supports voicemail group and all the extensions added in the group will receive the voicemail to the group extension. The voicemail group can be configured under Web GUI 🡪 Call Features 🡪 Voicemail 🡪 Voicemail Group. Click on “Create New Voicemail Group” to configure the group.

Figure 168: Voicemail Group

Table 72: Voicemail Group Settings

Voicemail Group Extension

Enter the Voicemail Group Extension. The voicemail messages left to this extension will be forwarded to all the voicemail group members.

Name

Configure the Name to identify the voicemail group. Letters, digits, _ and – are allowed.

Voicemail Password

The Voicemail password for the user to check Voicemail messages.

Email Address

The Email address of current user.

Member

Select available extensions from the left list and add them to the right list. The extensions need to have voicemail enabled to be listed in available mailboxes list.

Note: Members selected cannot exceed 30 extensions.

Ring Group

The UCM6200 supports ring group feature with different ring strategies applied to the ring group members. This section describes the ring group configuration on the UCM6200.

Configure Ring Group

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

Figure 169: Ring Group

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

Table 73: Ring Group Parameters

Ring Group Name

Configure ring group name to identify the ring group. Letters, digits, _ and – are allowed.

Extension

Configure the ring group extension.

Members

Select available users from the left side to the ring group member list on the right side. Click on to arrange the order.

LDAP Phonebook

Select available remote users from the left side to the ring group member list on the right side. Click on to arrange the order. Note: LDAP Sync must be enabled first.

Ring Strategy

Select the ring strategy. The default setting is “Ring in order”.

  • Ring simultaneously.

Ring all the members at the same time when there is incoming call to the ring group extension. If any of the member answers the call, it will stop ringing.

  • Ring in order.

Ring the members with the order configured in ring group list. If the first member does not answer the call, it will stop ringing the first member and start ringing the second member.

Music On Hold

Select the “Music On Hold” Class of this Ring Group, “Music On Hold” can be managed from the “Music On Hold” panel on the left.

Custom Prompt

This option is to set a custom prompt for a ring group to announce to caller. Click on ‘Prompt’, it will direct the users to upload the customized voice prompts.

Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom Prompt, where they could record new prompt or upload prompt files.

Ring Timeout for Each Member

Configure the number of seconds to ring each member. If set to 0, it will keep ringing. The default setting is 30 seconds.

Note:

The actual ring timeout might be overridden by users if the phone has ring timeout settings as well.

Auto Record

If enabled, calls on this ring group will be automatically recorded. The default setting is No. The recording files can be accessed from Web GUI🡪CDR🡪Recording Files.

Endpoint Call Forwarding Support

This allows the UCM to work with endpoint-configured call forwarding settings to redirect calls to ring group. For example, if a member wants to receive calls to the ring group on his mobile phone, he would have to set his endpoint’s call forwarding settings to his mobile number. By default, it is disabled.

However, this feature has the following limitations:

    • This feature will work only when call forwarding is configured on endpoints, not on the UCM.
    • If the forwarded call goes through an analog trunk, and polarity reversal is disabled, the other ring group members will no longer receive the call after it is forwarded.
    • If the forwarded call goes through a VoIP trunk, and the outbound route for it is PIN-protected and requires authentication, the other ring group members will no longer receive the call after it is forwarded.
    • If the forwarded call hits voicemail, the other ring group members will no longer receive the call.

Replace Display Name

If enabled, the UCM will replace the caller display name with the Ring Group name the caller know whether the call is incoming from a direct extension or a Ring Group.

Skip Busy Agent

If enabled calls to busy agents will be skipped and sent to the following available ones. Default is enabled.

Enable Destination

If enabled, users could select extension, voicemail, ring group, IVR, call queue, voicemail group as the destination if the call to the ring group has no answer.

Note: Voicemail Password and Email address (limited by 128 characters) are required if voicemail is selected as the destination. Voicemail system will mention if a voicemail is from a ring group.

Default Destination

The call would be routed to this destination if no one in this ring group answers the call.

Figure 170: Ring Group Configuration

Remote Extension in Ring Group

Remote extensions from the peer trunk of a remote UCM6200 can be included in the ring group alongside local extensions. An example of Ring Group with peer extensions is presented in the following:

  1. Create SIP trunks from UCM A to UCM B and vice-versa on the Extension/Trunk->VoIP Trunks page. Additionally, please create the appropriate outbound and inbound routes for these SIP trunks.
  2. Click edit button in the menu , and check if Sync LDAP Enable is toggled on.

Figure 171: Sync LDAP Server option

  1. In the scenario that the LDAP is not synced automatically, users can manually sync LDAP phonebooks by clicking on the LDAP sync button that’s highlighted in the following image.

Figure 172: Manually Sync LDAP Server

  1. Under Ring Groups setting page, click “Add”. Ring Groups can be found under Web GUI🡪Call Features🡪Ring Groups.
  2. If the LDAP phonebook has synced correctly, the remote UCM’s extensions should appear in the Create/Edit Ring Group->LDAP Phonebook list.

Figure 173: Ring Group Remote Extension

Restrict Calls

Restrict calls is a feature that can be used to restrict calls between internal extensions besides those in the Allowed List.

This section describes the configuration of this feature in the Call Features->Restrict Calls page.

Figure 174: Restrict Calls

Configure Restrict Calls

  • Click on “Add” to add a rule for restrict calls.
  • Click on to edit the rule of restrict calls.
  • Click on to delete the rule of restrict calls.

Table 74: Restrict Calls Configuration Parameters

Name

Configure Restrict call’s name.

Restrict Calls between extensions

When enabled, members of the group cannot dial other extension, only the numbers in the Allowed List.

Members

Configure the members that will not be able to call any extensions besides those in the Allowed List.

Allowed list

Select the extensions that the Members list can be able to call.

Paging and Intercom Group

Paging and Intercom Group can be used to make an announcement over the speaker on a group of phones. Targeted phones will answer immediately using speaker. The UCM6200 paging and intercom can be used via feature code to a single extension or a paging/intercom group. This section describes the configuration of paging/intercom group under Web GUI🡪Call Features🡪Paging/Intercom.

Configure Paging/Intercom Group

  • Click on “Add” to add paging/intercom group.
  • Click on to edit the paging/intercom group.
  • Click on to delete the paging/intercom group.
  • Click on “Paging/Intercom Group Settings” to edit Alert-Info Header. This header will be included in the SIP INVITE message sent to the callee in paging/intercom call.

Configure Multicast Paging

Figure 175: Multicast Paging

Table 75: Multicast Paging Configuration Parameters

Name

Configure paging/intercom group name.

Type

Select “Multicast Paging”.

Extension

Configure the paging/intercom group extension.

Multicast IP Address

The allowed multicast IP address range is 224.0.1.0 – 238.255.255.255.

Note: This field appears only when “Type” is set to “Multicast Paging”.

Maximum Call Duration

Specify the maximum call duration in seconds. The default value 0 means no limit.

Custom Prompt

This option is to configure a custom to play to paging callees. Users can now directly upload custom prompts from this page. They do not need to be redirected anymore.

Multicast IP Address

Configure the multicast IP address that endpoints will listen to.

Port

Configure the port for multicast paging.

Note: This field appears only when “Type” is set to “Multicast Paging”.

Configure 2-way Intercom

Figure 176: 2-way Intercom

Table 76: 2-way Intercom Configuration Parameters

Name

Configure paging/intercom group name.

Type

Select “2-way Intercom”.

Extension

Configure the paging/intercom group extension.

Replace Display Name

If enabled, the UCM will replace the caller display name with Paging/Intercom name.

Maximum Call Duration

Specify the maximum call duration in seconds. The default value 0 means no limit.

Custom Prompt

This option is to set a custom prompt for a paging/intercom group to announce to caller. Users can now directly upload custom prompts from this page. They do not need to be redirected anymore.

Members

Select available users from the left side to the paging/intercom group member list on the right.

Paging/Intercom Whitelist

Select which extensions are allowed to use the paging/intercom feature for this paging group.

Configure 1-way Paging

Figure 177 : 1-way Paging

Table 77: 1-way Paging Configuration Parameters

Name

Configure paging/intercom group name.

Type

Select “1-way Paging”.

Extension

Configure the paging/intercom group extension.

Replace Display Name

If enabled, the UCM will replace the caller display name with Paging/Intercom name.

Maximum Call Duration

Specify the maximum call duration in seconds. The default value 0 means no limit.

Custom Prompt

This option is to set a custom prompt for a paging/intercom group to announce to caller. Users can now directly upload custom prompts from this page. They do not need to be redirected anymore..

Members

Select available users from the left side to the paging/intercom group member list on the right.

Paging/Intercom Whitelist

Select which extensions are allowed to use the paging/intercom feature for this paging group.

Configure Announcement Paging

Figure 178: Announcement Paging

Table 78: Announcement Paging Configuration Parameters

Enable

Enable/Disable Announcement Paging.

Name

Configure paging/intercom group name.

Type

Select “Announcement Paging”

Custom Prompt

This option is to set a custom prompt for a paging/intercom group to announce to caller. Users can now directly upload custom prompts from this page. They do not need to be redirected anymore.

Repeat

If enabled, the announcement page will be repeated for the selected weekdays.

Date

Configure Announcement Paging Date.

Include Holidays

If enabled, Announcement Paging will run during holidays.

Time

Configure Announcement Paging Time.

Transmission Method

Configure Announcement Paging transmission method.

Unicast: Depending on members selection

Multicast: Depending on Multicast IP address and Port

Members

Select available users from the left side to the paging/intercom group member list on the right.

Configure Private Intercom

Private Intercom is a new paging type that is meant to be used with Grandstream GSC3510.

http://www.grandstream.com/products/facility-management/intercoms-paging/product/gsc3510

In a private intercom:

  • The initiator can be heard by all parties
  • The initiator can hear only one of the intercom members, which is determined by whose audio is initially detected. Audio from other members cannot be heard until the first responder is done talking.
  • Intercom members can hear only the initiator’s audio and not other intercom members

Figure 179: Private Intercom

Table 79: Private Intercom Configuration Parameters

Name

Configure paging/intercom group name.

Type

Select “Private Intercom”.

Extension

Configure the paging/intercom group extension.

Replace Display Name

If enabled, the UCM will replace the caller display name with Paging/Intercom name.

Maximum Call Duration

Specify the maximum call duration in seconds. The default value 0 means no limit.

Custom Prompt

This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct the users to upload the customized voice prompts.

Note: Users can also refer to the page PBX Settings🡪Voice Prompt🡪Custom Prompt, where they could record new prompt or upload prompt files.

Members

Select available users from the left side to the paging/intercom group member list on the right.

Paging/Intercom Whitelist

Select which extensions are allowed to use the paging/intercom feature for this paging group.

Paging/Intercom Group Settings

Figure 180: Page/Intercom Group Settings

The UCM6200 has pre-configured paging/intercom feature code. By default, the Paging Prefix is *81 and the Intercom Prefix is *80. To edit page/intercom feature code, click on “Feature Codes” in the “Paging/Intercom Group Settings” dialog. Or users could go to Web GUI🡪Call Features🡪Feature Codes directly.

Configure a Scheduled Paging/Intercom

Users can schedule paging/intercom calls by using the Schedule Paging/Intercom page. To schedule, click the Add button on the new page and configure the caller, the group to use, and the time to call out.

Figure 181: Schedule Paging/Intercom page

Table 80: Schedule Paging / Intercom Settings

Caller

Configure the caller ID for the paging / intercom group.

Paging/Intercom Group

Select the paging / intercom group from the list of the available groups.

Start Time

Configure the start time of the scheduled paging / intercom call.

Type

Select the type for the scheduled paging / intercom call. The available types are: Single time or Daily, Weekly basis. Default is “Single”.

Include Holidays

If enabled Paging/Intercom will run during holidays.

Action Status

Display the action status of the scheduled paging / intercom call.

Figure 182: Creating a scheduled paging/intercom call

Call Queue

The UCM provides call center functionality through its Call Queue feature. Static and dynamic agents can be assigned to queue groups to handle varying loads of call traffic. This section describes the configuration of call queues in the Call Features->Call Queue page.

Configure Call Queue

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

Figure 183: Call Queue

UCM6200 supports custom prompt feature in call queue. This custom prompt will active after the caller waits for a period of time in the Queue. Then caller could choose to leave a message/ transfer to default extension or keep waiting in the queue.

To configure this feature, please go to UCM Web GUI🡪Call Features🡪Call Queue🡪Create New Queue/Edit Queue🡪Queue Options🡪set Enable Destination to Enter Destination with Voice Prompt. Users could configure the wait time with Voice Prompt Cycle.

  • Click on “Create New Queue” to add call queue.
  • Click on to edit the call queue. The call queue configuration parameters are listed in the table below.
  • Click on to delete the call queue.

Table 81: Call Queue Configuration Parameters

Basic Settings

Extension

Configure the call queue extension number.

Name

Configure the call queue name to identify the call queue.

Strategy

Select the strategy for the call queue.

  • Ring All

Ring all available Agents simultaneously until one answer.

  • Linear

Ring agents in the specified order.

  • Least Recent

Ring the agent who has been called the least recently.

  • Fewest Calls

Ring the agent with the fewest completed calls.

  • Random

Ring a random agent.

  • Round Robin

Ring the agents in Round Robin scheduling with memory.

The default setting is “Ring All”.

Music On Hold

Select the Music On Hold playlist for the call queue.

Note: Music On Hold playlists can be managed from Web GUI🡪PBX Settings🡪Music On Hold.

Max Queue Length

Configure the maximum number of calls that can be queued at once. This number does not include calls that have been connected with agents. It only includes calls not connected yet. The default setting is 0, which means unlimited. When the maximum value is reached, the caller will be treated with busy tone followed by the next calling rule after attempting to enter the queue.

Wrapup Time

Configure the number of seconds an agent must wait to receive a new call after completing a previous call. If set to 0, an agent can receive a new call as soon as a previous call is completed. Default is 10 seconds.

Retry Time

Configure the number of seconds to wait before ringing the next agent.

Ring Time

Configure the number of seconds to ring an agent before proceeding to ring the next agent. Default is 30 seconds.

Auto Record

If enabled, the calls on the call queue will be automatically recorded. The recording files can be accessed in Queue Recordings under Web GUI🡪Call Features🡪Call Queue.

Enable Record Feature Code

If enabled, call queue members can use recording feature code to pause or restore current queue’s recording when Auto Record is enabled for the queue. Default code for Start/Stop Call Recording is *3.

Welcome Prompt

If enabled, users can upload an audio file that will be played as an initial tone when dialing the queue number.

Satisfaction Survey Prompt

When the agent hangs up, the system will play the prompt to ask the caller to rate their satisfaction scale from 1 to 5, where 5 is the highest.

Max Wait Time

Configure the maximum amount of seconds callers can wait before being disconnected from the queue. Default setting is 60 seconds. If left blank, there will be no limit to the amount of time a caller waits in queue.

Note: It is recommended to configure “Wait Time” longer than the “Wrapup Time”.

Destination

Configure the destination that callers will be redirected to after the configured Max Wait Time has passed. Default is “Hang up”.

Reset Agent Call Counter – Enable

If enabled, the agent call completion count will be reset based on the frequency set in the Repeat and Date/Time fields.

Repeat

Specifies the frequency at which the Agent call counter will be reset.

Destination Prompt Cycle

Configure the frequency to play the Destination Prompt to callers waiting in queue.

Custom Prompt

The custom prompt to play to callers waiting in queue once the Destination Prompt Cycle period has been reached. Callers can press 1 to be redirected to the destination configured in Destination.

Destination

Select the failover destination to redirect callers to after pressing 1 during the custom prompt.

Advanced Settings

  • Virtual Queue
  • Position Announcement
  • Queue Chairman

Refer to Call Center Settings and Enhancements section for detailed information about these features.

Enable Agent Login

Enables quick static agent login/logout via GXP21XX softekys. Supports GXP21XX endpoints on firmware 1.0.9.18 and higher.

Leave When Empty

Configure whether or not callers will be disconnected from the queue based on agent availability.

  • Yes

Callers will be disconnected from the queue if all agents are paused or invalid.

  • No

Never disconnect the callers from the queue when the queue is empty.

  • Strict

Callers will be disconnected from the queue if there are no agents logged in or if all agents are paused or unavailable.

Dial in Empty Queue

Configure whether the callers can dial into a call queue if the queue has no agent. The default setting is “No”.

  • Yes

Callers can always dial into a call queue.

  • No

Callers cannot dial into a queue if all agents are paused or invalid.

  • Strict

Callers cannot dial into the queue if it has no agents logged in or if all agents in it are paused or unavailable.

Failover Destination

Configure the destination to redirect callers to based on the Leave When Empty and Dial in Empty Queue values.

  • Play Sound.
  • Extension.
  • Voicemail.
  • Queues.
  • Ring Group.
  • Voicemail Group.
  • IVR.
  • External Number.

Report Hold Time

If enabled, the UCM will report to the agent the amount of time a caller waited before being connected. Default is disabled.

Replace Display Name

If enabled, the UCM will replace the caller display name with the Call Queue name so that the caller knows the call is incoming from a Call Queue.

Enable Feature Codes

Enable feature codes option for call queue. For example, *83 is used for “Agent Pause”.

Note: Callers can no longer use feature codes in established callbacks.

Dynamic Login Password

If enabled, the configured PIN number is required for dynamic agent to log in. The default setting is disabled.

Alert-Info

Configure the call destination for the call to be routed to if no agent in this call queue answers the call.

Agents

Agents

Go to “Agents” Tab and Select the available users to be the static agents in the call queue. Choose from the available users on the left to the static agents list on the right. Click on 1.PNG (19×19) 2.PNG (20×20) to choose. And use UP and Down arrow to select the order of the agent within the call queue.

Static Agents limitation:

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

Figure 184: Static Agents limit

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

Table 82: Static Agent Limitation

UCM Model

Max Static Agents in Call Queue

UCM6202

22

UCM6204

23

UCM6208

34

UCM6510

75

Click on “Global Queue Settings” to configure Agent Login Extension Postfix and Agent Logout Extension Postfix. Once configured, users could log in the call queue as dynamic agent.

Figure 185: Agent Login Settings

For example, if the call queue extension is 6500, Agent Login Extension Postfix is * and Agent Logout Extension Postfix is **, users could dial 6500* to login to the call queue as dynamic agent and dial 6500** to logout from the call queue. Dynamic agent does not need to be listed as static agent and can log in/log out at any time.

  • Call queue feature code “Agent Pause” and “Agent Unpause” can be configured under Web GUI🡪Call Features🡪Feature Codes. The default feature code is *83 for “Agent Pause” and *84 for “Agent Unpause”.
  • Queue recordings are shown on the Call Queue page under “Queue Recordings” Tab. Click on to download the recording file in .wav format; click on to delete the recording file. To delete multiple recording files by one click, select several recording files to be deleted and click on “Delete Selected Recording Files” or click on “Delete All Recording Files” to delete all recording files.

Global Queue Settings

Dynamic Agent Login Settings

  • Agent Login Code Suffix: Configure the code to dial after the queue extension to log into the queue (i.e queue extension + suffix). If no suffix is configured, dynamic agents will not be able to log in.
  • Agent Logout code Suffix: Configure the code to dial after the queue extension to log out of the queue (i.e queue extension + suffix). If no suffix is configured, dynamic agents will not be able to log out.

Virtual Queue Callback Key settings

  • Call Back Current: Press this feature key to set your current number as callback number.
  • Custom Callback Number: Press this feature key to set a custom callback number.
  • Continue waiting: Press this feature key to continue waiting.

Call Center Settings and Enhancements

UCM supports additional call center features such as virtual call queue and position announcement, allowing callers to know their current position in the queue and providing them the option to either stay in the queue or request for callback once an agent is available.

To configure these, click on the button for a queue group and click on the Advanced Settings tab. The following options are available:

Table 83: Call Center Parameters

Enable Virtual Queue

Enable virtual queue to activate call center features.

Virtual Queue Period

Configure the amount of seconds a caller must wait before the virtual queue prompt is played to them. Default is 20 seconds.

Virtual Queue Mode

Timeout Mode: After the virtual queue period passes, the caller will enter the virtual call queue and be presented with a menu to choose an option, the choices are summarized below :

  • Press * to set current number as callback number.
  • Press 0 to set a callback number different than current caller number.
  • Press # to keep waiting on the call queue.

Note: These options are customizable from the Global Queue Settings page.

DTMF Mode: In this mode, the callers can activate the virtual queue by pressing 2, then they will be presented with the menu to choose an option as below:

  • Press * to set current number as callback number.
  • Press 0 to set a callback number different than current caller number.
  • Press # to keep waiting on the call queue.

Virtual Queue Outbound Prefix

System will add this prefix to dialed numbers when calling back users.

Enable Virtual Queue Timeout

If enabled, agents will have a set amount of time to answer a virtual queue callback.

Write Timeout

Configure the virtual queue callback timeout period in seconds.

Enable Position Announcement

If enabled, callers’ positions in queue will be announced based on the frequency set in Position Announcement Interval.

Note: Queue position will now be announced to the caller upon entering the queue.

Position Announcement Interval

Configure the frequency of the caller position announcement.

Enable Hold time Announcement

Enable the announcement of the estimated hold time to the caller periodically.

Note: Hold time will not be announced if less than one minute.

Queue Chairman

Select the extension to act as chairman of the queue (monitoring).
Note: One queue can have up to 3 chairmen.

Enable Agent Login

Enables quick static agent login/logout via GXP21xx softkeys. Supports GXP21XX endpoint on firmware 1.0.9.18 and higher.

Notes:

  • After enabling the feature, users need to set the option on GXP21XX phone under “Account🡪SIP Settings🡪Advanced Features🡪Special Feature” to “UCM Call Center” to display a softkey on the phone to login/logout.
  • When this option is enabled, dynamic agent login will be no longer supported.
  • In case of concurrent registrations, changing agent status on one phone (login/logout) will be reflected on all phones.

Autofill

Toggles the way UCM handles and distributes call traffic. Please see the section below for more details.

Queue Auto fill enhancement:

In previous UCM firmware, the call queue has a serial type behavior in that the queue will make all waiting callers wait in the queue even if there is more than one available member ready to take calls until the head caller is connected with the member they were trying to get to.

The next waiting caller in line then becomes the head caller, and they are then connected with the next available member and all available members and waiting callers waits while this happens.

Starting from 1.0.14.x, the waiting callers are connecting with available members in a parallel fashion until there are no more available members or no more waiting callers.

For example, in a call queue with linear method, if there are two available agents, when two callers call in the queue at the same time, UCM will assign the two callers to each of the two available agents at the same time, rather than assigning the second caller to second available agent after the first agent answers the call from the first caller.

Queue Statistics

Along with the mentioned call center features, UCM offers detailed call queue statistics, allowing system administrators to make informed decisions regarding call traffic distribution and handling.

Figure 186: Call Queue Statistics

Select the time interval along with the queue(s) and agent(s) to get detailed statistics.

User can download statistics on CSV format by clicking on the download button , also the statistics can be cleared using “Clear Statistics” button .

The statistics can be automatically sent to a specific email address on a preconfigured Period, this can be done by clicking on “Automatic Download Settings” button , and user will be directed to below page where he can configure the download period (Day/Week/Month) and the Email where the statistics will be sent (Email settings should be configured correctly):

Figure 187 : Automatic Download Settings – Queue Statistics

Significantly more information is now available UCM’s queue statistics page. In addition to the information presented in previous firmware, users can now view a call log that displays calls to all agents and queues, a dynamic agent login/logout record, and a pause log. Statistics reports for these new pages can be obtained by pressing the Download button in the top left corner of the Call Queue Statistics page. The reports are in .CSV format and will be packaged into a single tar.gz file upon download.

Agent Details

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

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

Figure 188: Agent details

Login Record

Login Record is a report that shows the timestamps of dynamic agent logins and logouts and calculates the amount of time the dynamic agents were logged in. Dynamic agents are extensions that log in and out either via agent login/logout codes (configured in Global Queue Settings page) or by using the GXP21xx call queue softkey. A new record will be created only when an agent logs out. The following information is available:

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

Figure 189: Login Record

Pause Log

Pause Log is a report that shows the times of agent pauses and unpauses and calculates the amount of time that agents are paused. If an agent is part of several queues, an entry will be created for each queue. An entry will only be created after an agent unpauses. The following information is available:

  • Agentthe extension that paused and unpaused
  • Queue – the queue that the agent is in.
  • Pause Time – the time that the agent paused.
  • Resume Time – the time that the agent unpaused.
  • Pause Duration – the total length of time the agent was paused for.

Figure 190: Pause Log

Global Queue Settings

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

Figure 191: Global Queue Settings

Table 84: Global Queue Settings

Dynamic Agent Login Settings

Agent Login Code Suffix

Configure the code to dial after the queue extension to log into the queue (i.e. queue extension + suffix).
If no suffix is configured, dynamic agents will not be able to log in

Agent Logout Code Suffix

Configure the code to dial after the queue extension to log out of the queue (i.e. queue extension + suffix).
If no suffix is configured, dynamic agents will not be able to log out.

Virtual Queue Callback Key Settings

Call Back Current Number

Press the feature key configured to set your current number as callback number.

Custom Callback Number

Press the feature key configured to set a custom callback number.

Continue Waiting

Press the feature key configured to continue waiting.

Switchboard

Switchboard is a tool that allows system administrators and users to monitor and manage queues. This can be accessed from the Call Features->Call Queue page.

Figure 192: Switchboard Summary

Clicking on one of the queues shown above will bring up the following window:

Figure 193: Call Queue Switchboard

The table below gives a brief description for the main menus:

Table 85: Switchboard Parameters

Waiting

This menu shows the current waiting calls along with the caller id and the option to hang-up call by pressing on the button.

Proceeding

Shows the current established calls along with the caller id and the callee (agent) as well as the option to hang-up, transfer, add conference or barge-in the call.

Agents

Displays the list of agents in the queue and the extension status (idle, ringing, in use or unavailable) along with some basic call statistics and agent’s mode (static or dynamic).

Note: the dashboard will show the number of calls (answered and abandoned) of each agent. For dynamic agents, it will count the number of calls starting from the last login time.

There are three different privilege levels for Call Queue management from the switchboard: Super Admin, Queue Chairman, and Queue Agent.

  • Super Admin – Default admin of the UCM. Call queue privileges include being able to view and edit all queue agents, monitor and execute actions for incoming and ongoing calls for each extension in Switchboard, and generate Call Queue reports to track performance.
  • Queue Chairman – User appointed by Super Admin to monitor and manage an assigned queue extension via Switchboard. The Queue Chairman can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on “Value-added Features” in the side menu and click on “Call Queue”. In the image below, User 1012 is the Queue Chairman appointed to manage Queue Extension 6500 and can see all the agents of the queue in the Switchboard.
    Note: Super Admin can assign 3 Chairmen
  • Queue Agent – User appointed by Super Admin to be a member of a queue extension. A queue agent can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on “Value-added Features” in the side menu and click on “Call Queue”. However, a queue agent can view and manage only his own calls and statistics, but not other agents’ in the queue extension. In the image below, User 1000 is a queue agent and can see only his own information in the Switchboard.
A screenshot of a computer

Description automatically generated

Figure 194: Queue Agent

Pickup Groups

The UCM6200 supports pickup group feature which allows users to pick up incoming calls for other extensions if they are in the same pickup group, by dialing “Pickup Extension” feature code (by default *8).

Configure Pickup Groups

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

  • Click on to create a new pickup group.
  • Click on to edit the pickup group.
  • Click on to delete the pickup group.

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

Figure 195: Edit Pickup Group

Configure Pickup Feature Code

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

The default feature code for call pickup extension is *8, otherwise if the person intending to pick up the call knows the ringing extension they can use ** followed by the extension number in order to perform the call pickup operation.

The following figure shows where you can customize these features codes

Figure 196: Edit Pickup Feature Code

Music On Hold

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

Figure 197: Music On Hold Default Playlist

  • Click on “Create New MOH Class” to add a new Music On Hold playlist.
  • Click on to configure the MOH class sort method to be “Name” or “Random” for the sound files.
  • Click on next to the selected Music On Hold playlist to delete this Music On Hold playlist.
  • Click on to start uploading. Users can upload:
  • Single files with 8KHz Mono Music file, or
  • Music on hold files in a compressed package with .tar, .tar.gz and .tgz as the suffix. The file name can only be letters, digits or special characters -_
  • the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire MoH.
  • Users could also download all the music on hold files from UCM. In the Music On Hold page, click on
    and the file will be downloaded to your local PC.
  • Click on next to the sound file to disable it from the selected Music On Hold Playlist.
  • Click on next to the sound file to enable it from the selected Music On Hold Playlist.
  • Select the sound files and click on to delete all selected music on hold files.

Note: the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire MoH.

The UCM can play selected MOH files to extensions by initiating calls to them from the Manage Music on Hold page

Steps to play the music on hold file:

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

Figure 198: Play Custom Prompt

  1. The selected extension will ring.
  2. Answer the call to listen to the music playback.

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

  1. Click on .
  2. A prompt of confirmation will pop up, as shown below.

Figure 199: Information Prompt

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

Figure 200: Record Custom Prompt

  1. Answer the call and start to record your new music on hold.
  2. Hangup the call and refresh Music On Hold page then you can listen to the new recorded file.

Notes:

Once the MOH file is deleted, there are two ways to recover the music files.

  • Users could download the MOH file from this link:

https://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-moh-opsound-wav-2.03.tar.gz

After downloading and unzip the pack, users could then upload the music files to UCM.

  • Factory reset could also recover the MOH file on the UCM.

Fax Server

The UCM6200 supports T.30/T.38 Fax and Fax Pass-through. It can convert the received Fax to PDF format and send it to the configured Email address. Fax/T.38 settings can be accessed via Web GUI🡪Call Features🡪FAX/T.38. The list of received Fax files will be displayed in the same web page for users to view, retrieve and delete.

Configure Fax/T.38

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

Figure 201: Fax Settings

Table 86: FAX/T.38 Settings

Enable Error Correction Mode

Configure to enable Error Correction Mode (ECM) for the Fax.

The default setting is “Yes”.

Maximum Transfer Rate

Configure the maximum transfer rate during the Fax rate negotiation.

The possible values are 2400, 4800, 7200, 9600, 12000 and 14400.

The default setting is 14400.

Minimum Transfer Rate

Configure the minimum transfer rate during the Fax rate negotiation. The possible values are 2400, 4800, 7200, 9600, 12000 and 14000. The default setting is 2400.

Max Concurrent Sending Fax

Configure the concurrent fax that can be sent by UCM6200. Two modes “Only” and “More” are supported.

  • Only

Under this mode, the UCM6200 allows only single user to send fax at a time.

  • More

Under this mode, the UCM6200 supports multiple concurrent fax sending by the users.

By default, this option is set to “only”.

Fax Queue Length

Configure the maximum length of Fax Queue from 6 to 10.

The default setting is 6.

User Information in Fax Header

If enabled this this will give users the option to send a special header in SIP fax messages.

Fax Header Information

Adds fax header into the fax file.

Default Email Address

Configure the Email address to send the received Fax to if user’s Email address cannot be found.

Note:

The extension’s Email address or the Fax’s default Email address needs to be configured in order to receive Fax from Email. If neither of them is configured, Fax will not be received from Email.

Template Variables

Fill in the “Subject:” and “Message:” content, to be used in the Email when sending the Fax to the users.

The template variables are:

  • ${CALLERIDNUM} : Caller ID Number
  • ${CALLERIDNAME} : Caller ID Name
  • ${RECEIVEEXTEN} : The extension to receive the Fax
  • ${FAXPAGES} : Number of pages in the Fax
  • ${VM_DATE} : The date and time when the Fax is received

Enable Fax Resend

Enables the fax resend option which allow the UCM to keep attempting to send faxes up to a specified amount of times. Additionally, if a fax still fails to send, a Resend button will appear in the File Send Progress list in Value-Added Features🡪Fax Sending to allow manual resending.

Max Resend Attempts

Configures the maximum attempts number to resend the fax.

Receiving Fax

Example Configuration to Receive Fax from PSTN Line

The following instructions describe how to use the UCM6200 to receive fax from PSTN line on the Fax machine connected to the UCM6200 FXS port.

  1. Connect Fax machine to the UCM6200 FXS port.
  2. Connect PSTN line to the UCM6200 FXO port.
  3. Go to Web GUI🡪Extension/Trunk page.
  4. Create or edit the analog trunk for Fax as below.

Fax Mode: Make sure “Fax Mode” option is set to “None”.

Figure 202: Configure Analog Trunk without Fax Detection

  1. Go to UCM6200 Web GUI🡪Extension/Trunk🡪Extensions page.
  2. Create or edit the extension for FXS port.
  • Analog Station: Select FXS port to be assigned to the extension. By default, it is set to “None”.
  • Once selected, analog related settings for this extension will show up in “Analog Settings” section.

Figure 203: Configure Extension for Fax Machine: FXS Extension

Figure 204: Configure Extension for Fax Machine: Analog Settings

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

Figure 205: Configure Inbound Rule for Fax

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

Example Configuration for Fax-To-Email

The following instructions describe a sample configuration on how to use Fax-to-Email feature on the UCM6200.

  1. Connect PSTN line to the UCM6200 FXO port.
  2. Go to UCM6200 Web GUI🡪Call Features🡪Fax/T.38 page. Create a new Fax extension.

Figure 206: Create Fax Extension

  1. Go to UCM6200 Web GUI🡪Extension/Trunk🡪Analog Trunks page. Create a new analog trunk. Please make sure “Fax Detection” is set to “No”.
  2. Go to UCM6200 Web GUI🡪Extension/Trunk🡪Inbound Routes page. Create a new inbound route and set the default destination to the Fax extension.

Figure 207: Inbound Route to Fax Extension

  1. Once successfully configured, the incoming Fax from external Fax machine to the PSTN line number will be converted to PDF file and sent to the Email address fax@domain.local as attachment.

Figure 208: List of Fax Files

FAX Sending

Besides the support of Fax machines, The UCM6200 supports also sending Fax via Web GUI access. This feature can be found on Web GUI🡪Value-added Features🡪Fax Sending page. To send fax, pre-setup for analog trunk and outbound route is required. Please refer to [ANALOG TRUNKS], [VOIP TRUNKS] and [Outbound Routes] sections for configuring analog trunk and outbound route.

After making sure analog trunk or VoIP Trunk is setup properly and UCM6200 can reach out to PSTN numbers via the trunk, on Fax Sending page, enter the fax number and upload the file to be faxed. Then click on “Send” to start. The progress of sending fax will be displayed in Web GUI. Users can also view the sending history is in the same web page.

Figure 209: Fax Sending in Web GUI

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

Figure 210: Fax Send Progress

Note: Only A3, A4, and B4 paper sizes are supported for the Fax Sending.

Busy Camp-On

The UCM6200 supports busy camp-on/call completion feature that allows the PBX to camp on a called party and inform the caller as soon as the called party becomes available given the previous attempted call has failed.

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

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

Presence

UCM supports SIP presence, allowing extensions to advertise their current availability for calls. This presence can be monitored by other users.

Presence is different from BLF in that SIP presence is a status that users can manually set themselves.

Users can change presence status from the webUI:

Admin Portal: Navigate to Extension/Trunk->Extensions->Edit Extension->Features and click on the Presence Status dropdown list.

User Portal: Navigate to Basic Information->Extensions->Features and click on the Presence Status dropdown list.

Figure 211: SIP Presence Configuration

Select which status to set from the presence status selection drop list.

The following options are available:

Table 87: SIP Presence Status

Available

The contact is online and can participate in conversations/phone calls. 

Away

The contact is currently away (ex: for lunch break).

Chat

The contact has limited conversation flexibility and can only be reached via chat.

Do Not Disturb

The Contact is on DND (Do Not Disturb) mode.

Custom Presence Status

Please enter the presence status for this mode on the Web GUI. Up to 64 characters.

Unavailable

The contact is unreachable for the moment, please try to contact later.

Users can also set presence status by dialing the feature code from their phones (*48 by default). The prompt will ask users what presence status they want to set.

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

Figure 212: SIP Presence Feature Code

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

Figure 213: Presence Status CDR

Follow Me

Follow Me is a feature on the UCM6200 that allows users to direct calls to other phone numbers and have them ring all at once or one after the other. Calls can be directed to users’ home phone, office phone, mobile and etc. The calls will get to the user no matter where they are. Follow Me option can be found under extension settings page Web GUI🡪Extension/Trunk🡪Extensions.

To configure follow me:

  1. Choose the extension and click on .
  2. Go to the Follow me tab to add destination numbers and enable the feature.

Figure 214: Edit Follow Me

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

The following table shows the Follow Me configuration parameters:

Table 88: Follow Me Settings

Enable

Configure to enable or disable Follow Me for this user.

Skip Trunk Auth

If external number is added in the Follow Me, please make sure this option is enabled or the “Skip Trunk Auth” option of the extension is enabled, otherwise the external Follow Me number cannot be reached.

Music On Hold Class

Configure the Music On Hold class that the caller would hear while tracking users

Confirm When Answering

By default, it is enabled, and user will be asked to press 1 to accept the call or to press 2 to reject the call after answering a Follow Me call.

If it is disabled, the Follow Me call will be established once after the user answers.

Enable Destination

When enabled, the call will be routed to the default destination if no one in the Follow Me extensions answers the call.

Default Destination

Configure the destination if no one in the Follow Me extensions answers the call. The available options are:

Extension

Voicemail

Queues

Ring Group

Voicemail Group

IVR

External Number

Follow Me Numbers

The added numbers are listed here. Click on to arrange the order. Click on to delete the number. Click on to add new numbers.

New Follow Me Number

Add a new Follow Me number which could be a ‘Local Extension’ or ‘External Number’. The selected dial plan should have permissions to dial the defined external number.

Dialing Order

Select the order in which the Follow Me destinations will be dialed to reach the user: ring all at once or ring one after the other.

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

Table 89: Follow Me Options

Playback Incoming Status Message

If enabled, the PBX will playback the incoming status message before starting the Follow Me steps.

Record the Caller’s Name

If enabled, the PBX will record the caller’s name from the phone so it can be announced to the callee in each step.

Playback Unreachable Status Message

If enabled, the PBX will playback the unreachable status message to the caller if the callee cannot be reached.

Speed Dial

The UCM6200 supports Speed Dial feature that allows users to call specified destinations by pressing only 1-4 digits. This creates a system-wide speed dial access for all the extensions on the UCM6200.

To enable Speed Dial, on the UCM6200 Web GUI, go to page Web GUI🡪Call Features🡪Speed Dial.

User should first click on then create a speed dial extension and assign a destination to it. Supported destinations are Extension, Conference Room, Video Conference Room, Voicemail, Voicemail Group, IVR, Ring Group, Queue, Paging/Intercom Group, Fax, DISA, Dial by Name, Announcement, and External Number.

Figure 215: Speed Dial Destinations

Figure 216: List of Speed Dial

DISA

In many situations, the user will find the need to access his own IP PBX resources, but he is not physically near one of his extensions. However, he does have access to his own cell phone. In this case, we can use what is commonly known as DISA (Direct Inward System Access). Under this scenario, the user will be able to call from the outside, whether it is using his cell phone, pay phone, regular PSTN, etc. After calling into UCM6200, the user can then dial out via the SIP trunk or PSTN trunk connected to UCM6200 as it is an internal extension.

The UCM6200 supports DISA to be used in IVR or inbound route. Before using it, create new DISA under Web GUI🡪Call Features🡪DISA.

  • Click on to add a new DISA.
  • Click on to edit the DISA configuration.
  • Click on to delete the DISA.

Figure 217: Create New DISA

The following table details the parameters to set and configure DISA feature on UCM6200 PBX.

Table 90: DISA Settings

Name

Configure DISA name to identify the DISA.

Password

Configure the password (digit only) required for the user to enter before using DISA to dial out.

Note:

The password must be at least 4 digits.

Permission

Configure the permission level for DISA.

The available permissions are “Internal”, “Local”, “National” and “International” from the lowest level to the highest level.

The default setting is “Internal”.

If the user tries to dial outbound calls after dialing into the DISA, the UCM6200 will compared the DISA’s permission level with the outbound route’s privilege level. If the DISA’s permission level is higher than (or equal to) the outbound route’s privilege level, the call will be allowed to go through.

Response Timeout

Configure the maximum amount of time the UCM6200 will wait before hanging up if the user dials an incomplete or invalid number.

The default setting is 10 seconds.

Digit Timeout

Configure the maximum amount of time permitted between digits when the user is typing the extension. The default setting is 5 seconds.

Allow Hangup

If enabled, during an active call, users can enter the UCM6200 Hangup feature code (by default it is *0) to disconnect the call or hang up directly. A new dial tone will be heard shortly for the user to make a new call. The default setting is “No”.

Replace Display Name

If enabled, the UCM will replace the caller display name with the DISA name.

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

Emergency

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

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

https://www.fcc.gov/mlts-911-requirements

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

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

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

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

  1. Navigate on the web GUI under “Call Features 🡪 Emergency Calls
  2. Click on A close up of a sign

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

Figure 218: Emergency Number Configuration

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

Table 91: Emergency Numbers Parameters

Name

Configure the name of the emergency call.

For example, “emergency911″,”emergency211” and etc.

Emergency Number

Config the emergency service number. For example,”911″,”211″ and etc.

Emergency Level

Select the emergency level of the number. Level “3” means the most urgent.

Disable Hunt on Busy

If this option is not enabled, when the lines of trunks which the coming emergency call routes by are completely occupied, the line-grabbing function will automatically cut off a line from all busy lines so that the coming emergency call can seize it for dialing out. This option is not enabled by default.

Custom Prompt

Configures a custom prompt that will be played to the notified extensions when an emergency call is made. The file can be uploaded from the page “Custom Prompt”. Click “Upload Audio File” to add a custom prompt.

Use Trunks

Select the trunks for the emergency call. Up to five trunks can be selected.

Members Notified

Select the members who will be notified when an emergency call occurs.

Strip

Specify the number of digits that will be Stripped from the beginning of the dialed number before the call is placed via the selected trunk.
Note: Users can strip the same amount of numbers as the emergency number length itself.

Prepend

Specify the digits to be Prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.

Auto Record

When enabled, emergency call will be automatically recorded.

Send Recording File

When enabled recording files will be sent to the configured email address.

Email Address

Configure the email addresses that will receive emergency call notifications.

A screenshot of a cell phone

Description generated with very high confidence

Figure 219: 911 Emergency Sample

Callback

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

The callback feature works as follows:

  1. Configure a new callback on the UCM6200.
  2. On the UCM6200, configure destination of the inbound route for analog trunk to callback.
  3. Save and apply the settings.
  4. The user calls the PSTN number of the UCM6200 using the mobile phone, which goes to callback destination as specified in the inbound route.
  5. Once the user hears the ringback tone from the mobile phone, hang up the call on the mobile phone.
  6. The UCM6200 will call back the user.
  7. The user answers the call.
  8. The call will be sent to DISA or IVR which directs the user to dial the destination number.
  9. The user will be connected to the destination number.

In this way, the calls are placed and connected through trunks on the UCM6200 instead of to the mobile phone directly. Therefore, the user will not be charged on mobile phone services for long distance or international calls.

To configure callback on the UCM6200, go to Web GUI🡪Call Features🡪Callback page and click on . Configuration parameters are listed in the following table.

Table 92: Callback Configuration Parameters

Name

Configure a name to identify the Callback. (Enter at least two characters)

CallerID Pattern

Configure the pattern of the callers allowed to use this callback. The caller who places the inbound call needs to have the CallerID match this pattern so that the caller can get callback after hanging up the call.

Note:

If leaving as blank, all numbers are allowed to use this callback.

Outbound Prepend

Configure the prepend digits to be added at before dialing the outside number. The number with prepended digits will be used to match the outbound route. ‘-’ is the connection character which will be ignored.

Delay Before Callback

Configure the number of seconds to be delayed before calling back the user.

Destination

Configure the destination which the callback will direct the caller to. Two destinations are available:

  • IVR
  • DISA

The caller can then enter the desired number to dial out via UCM6200 trunk.

BLF and Event List

BLF

The UCM6200 supports BLF monitoring for extensions, ring group, call queue, conference room and parking lot. For example, on the user’s phone, configure the parking lot number 701 as the BLF monitored number. When there is a parked call on 701, the LED for this BLF key will light up in red, meaning a call is parked against this parking lot. Pressing this BLF key can pick up the call from this parking lot.

Note:

On the Grandstream GXP series phones, the MPK supports “Call Park” mode, which can be used to park the call by configuring the MPK number as call park feature code (e.g., 700). MPK “Call Park” mode can also be used to monitor and pickup parked call if the MPK number is configured as parking lot (e.g., 701).

Event List

Besides BLF, users can also configure the phones to monitor event list. In this way, both local extensions on the same UCM6200 and remote extensions on the VOIP trunk can be monitored. The event list setting is under Web GUI🡪Call Features🡪Event List.

  • Click on “Create New Event List” to add a new event list.
  • Sort selected extensions manually in the Eventlist
  • Click on to edit the event list configuration.
  • Click on to delete the event list.

Figure 220: Create New Event List

Table 93: Event List Settings

URI

Configure the name of this event list (for example, office_event_list). Please note the URI name cannot be the same as the extension name on the UCM6200. The valid characters are letters, digits, _ and -.

Local Extensions

Select the available extensions/Extension Groups listed on the local UCM6200 to be monitored in the event list.

Remote Extensions

If LDAP sync is enabled between the UCM6200 and the peer UCM6200, the remote extensions will be listed under “Available Extensions”. If not, manually enter the remote extensions under “Special Extensions” field.

Special Extensions

Manually enter the remote extensions in the peer/register trunk to be monitored in the event list. Valid format: 5000,5001,9000

Event List BLF can be used to monitor remote LDAP extensions on VoIP trunks. SIP endpoints will need to support event list BBLF to monitor remote extensions.

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

Notes:

  • To configure LDAP sync, please go to UCM6200 Web GUI🡪Extension/Trunk🡪VoIP Trunk. You will see “Sync LDAP Enable” option. Once enabled, please configure password information for the remote peer UCM6200 to connect to the local UCM6200. Additional information such as port number, LDAP outbound rule, LDAP Dialed Prefix will also be required. Both the local UCM6200 and remote UCM6200 need enable LDAP sync option with the same password for successful connection and synchronization.
  • Currently LDAP sync feature only works between two UCMs.
  • (Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is non-UCM6200 PBX. However, it might not work the other way around depending on whether the non-UCM6200 PBX supports event list BLF or remote monitoring feature.

Dial by Name

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

Dial by Name Configuration

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

Figure 221: Create Dial by Name Group

Figure 222: Configure Extension First Name and Last Name

  1. Name

Enter a Name to identify the Dial By Name group.

  1. Extension

Configure the direct dial extension for the Dial By Name group.

  1. Custom Prompt

This option sets a custom prompt for directory to announce to a caller. The file can be uploaded from the page “Custom Prompt”. Click “Upload Audio File” to add additional record.

  1. Available Extensions/Selected Extensions

Select available extensions from the left side to the right side as the directory for the Dial By Name group. Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group. The extensions here must have a valid first name and last name configured under Web GUI🡪Extension/Trunk🡪Extensions in order to be searchable in Dial By Name directory through IVR. By specifying the extensions here, the administrators can make sure unscreened calls will not reach the company employee if he/she does not want to receive them directly.

  1. Prompt Wait Time

Configure “Prompt Wait Time” for Dial By Name feature. During Dial By Name call, the caller will need to input the first letters of First/Last name before this wait time is reached. Otherwise, timeout will occur, and the call might hang up. The timeout range is between 3 and 60 seconds.

  1. Query Type

Specify the query type. This defines how the caller will need to enter to search the directory.

By First Name: enter the first 3 digits of the first name to search the directory.

By Last Name: enter the first 3 digits of the last name to search the directory.

By Full Name: enter the first 3 digits of the first name or last name to search the directory.

  1. Select Type

Specify the select type on the searching result. The IVR will confirm the name/number for the party the caller would like to reach before dialing out.

By Order: After the caller enters the digits, the IVR will announce the first matching party’s name and number. The caller can confirm and dial out if it is the destination party, or press * to listen to the next matching result if it is not the desired party to call.

By Menu: After the caller enters the digits, the IVR will announce 8 matching results. The caller can press number 1 to 8 to select and call or press 9 for results in next page.

The Dial by Name group can be used as the destination for inbound route and key pressing event for IVR. The group name defined here will show up in the destination list when configuring IVR and inbound route. If Dial by Name is set as a key pressing event for IVR, users can dial star (*) to exit from Dial by Name, and re-enter the IVR to start a new event. The following example shows how to use this option.

Figure 223: Dial By Name Group In IVR Key Pressing Events

Figure 224: Dial By Name Group In Inbound Rule

Username Prompt Customization

Starting from fw 1.0.15.x, the Dial By Name feature can use the recorded name prompt of a user to announce his/her name assigned to the dialed extension. If no name prompt greeting exists, the name will be spelt out like in previous versions.

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

Upload Username Prompt File from Web GUI

  1. First, Users should have a pre-recorded file respecting the following format:
  • PCM encoded / 16 bits / 8000Hz mono.
  • In .GSM or .WAV format.
  • File size under 5MB.
  • Filename must be set as the extension number. For example, the recorded file name 1000.wav will be used for extension 1000.
  1. Go under web GUI PBX Settings 🡪 Voice Prompt 🡪 Username Prompt and click on button.
  2. Select the recorded file to upload it and press Save and Apply Settings.
  • Click on to record again the username prompt.
  • Click on to play recorded username prompt.
  • Select username prompts and press to delete specific file or select multiple files for deletion using the button .

Record Username via Voicemail Menu

Users can also record their username via the voicemail menu.

  • Dial *97 or *98 to access the voicemail menu. If *98 is dialed, enter the desired extension and voicemail password.
  • Upon entering the extension’s voicemail system, dial 0 to enter the recordings menu.
  • Press 3 to start recording the username.

Active Calls and Monitor

The active calls on the UCM6200 are displayed in Web GUI🡪System Status🡪Active Calls page. Users can monitor the status, hang up the call as well as barge in the active calls in real time manner.

Active Calls Status

To view the status of active calls, navigate to Web GUI🡪System Status🡪Active Calls. The following figure shows extension 1000 is calling 1001. 1001 is ringing.

Figure 225: Status🡪PBX Status🡪Active Calls – Ringing

The following figure shows the call between 1002 and 1003 is established.

Figure 226: Status🡪PBX Status🡪Active Calls – Call Established

The black color of the active call means the connection of call time is less than half an hour. It means this call is normal.

Figure 227: Call Connection less than half hour

The yellow color of the active call means the connection of call time is greater than half an hour but less than one hour. It means this call is a bit long.

Figure 228: Call Connection between half an hour and one hour

The red color of the active call means the connection of call time is more than one hour. It means this call could be abnormal.

Figure 229: Call Connection more than one hour

Hang Up Active Calls

To hang up an active call, click on icon in the active call dialog. Users can also click on to hang up all active calls.

Call Monitor

During an active call, click on icon and the monitor dialog will pop up.

Figure 230: Configure to Monitor an Active Call

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

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

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

  • Whisper

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

  • Barge

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

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

Another way to monitor active calls is to dial the corresponding feature codes from an extension. Please refer to [Table 94: UCM6200 Feature Codes] and [Call Recording] section for instructions.

Call Feature Codes

The UCM6200 supports call recording, transfer, call forward, call park and other call features via feature code. This section lists all the feature codes in the UCM6200 and describes how to use the call features.

Feature Codes

Table 94: UCM6200 Feature Codes

Feature Maps

Blind Transfer

  • Default code: #1
  • Enter the code during active call. After hearing “Transfer”, you will hear dial tone. Enter the number to transfer to. Then the user will be disconnected, and transfer is completed.
  • Options:

Disable

Allow Caller: Enable the feature code on caller side only.

Allow Callee: Enable the feature code on callee side only.

Allow Both: Enable the feature code on both caller and callee.

Attended Transfer

  • Default code: *2
  • Enter the code during active call. After hearing “Transfer”, you will hear the dial tone. Enter the number to transfer to and the user will be connected to this number. Hang up the call to complete the attended transfer. In case of the called party does not answer, users could press *0 to cancel the call and retrieve the first call leg.
  • Options:

Disable

Allow Caller: Enable the feature code on caller side only.

Allow Callee: Enable the feature code on callee side only.

Allow Both: Enable the feature code on both caller and callee.

Seamless Transfer

  • Default code: *44 (Disabled by default).
  • Seamless Transfer allows user to perform blind transfer using UCM feature code without having music on hold presented during the transfer process, it minimizes the interruption during transfer, making the process smooth and simple.
  • During an active call use the feature code (*44 by default) followed by the number you want to transfer to in order to perform the seamless transfer.

Disconnect

  • Default code: *0
  • Enter the code during active call. It will disconnect the call.
  • Options:

Disable

Allow Caller: Enable the feature code on caller side only.

Allow Callee: Enable the feature code on callee side only.

Allow Both: Enable the feature code on both caller and callee.

Call Park

  • Default code: #72
  • Enter the code during active call to park the call.
  • Options:

Disable

Allow Caller: Enable the feature code on caller side only.

Allow Callee: Enable the feature code on callee side only.

Allow Both: Enable the feature code on both caller and callee.

Start/Stop Call Recording

  • Default code: *3
  • Enter the code followed by # or SEND to start recording the audio call and the UCM6200 will mix the streams natively on the fly as the call is in progress.
  • Options:

Disable

Allow Caller: Enable the feature code on caller side only.

Allow Callee: Enable the feature code on callee side only.

Allow Both: Enable the feature code on both caller and callee.

Feature Code Digits Timeout

Set the maximum interval (ms) between digits for feature code activation

DND/Call Forward

Do Not Disturb (DND) Activate

  • Default code: *77

Do Not Disturb (DND) Deactivate

  • Default code: *78

Call Forward Busy Activate

  • Default Code: *90
  • Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.

Call Forward Busy Deactivate

  • Default Code: *91

Call Forward No Answer Activate

  • Default Code: *92
  • Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.

Call Forward No Answer Deactivate

  • Default Code: *93

Call Forward Unconditional Activate

  • Default Code: *72
  • Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.

Call Forward Unconditional Deactivate

  • Default Code: *73

Remote Call Forward Enable

  • If enabled, this option will allow specific extensions to dial the remote call forwarding feature codes to set call forwarding for any extension.

Remote Call Forward Busy Enable

  • Default Code: *65
  • Enter the code and follow the voice prompt to set the remote extension number where you want to enable Call Forward Busy and the target destination.

Remote Call Forward Busy Disable

  • Default Code: *651
  • Enter the code and follow the voice prompt to set the remote extension number where you want to disable the Call Forward Busy.

Remote Call Forward No Answer Enable

  • Default Code: *66
  • Enter the code and follow the voice prompt to set the remote extension number where you want to enable Call Forward No Answer and the target destination.

Remote Call Forward No Answer Disable

  • Default Code: *661
  • Enter the code and follow the voice prompt to set the remote extension number where you want to disable the Call Forward No Answer.

Remote Call Forward Unconditional Enable

  • Default Code: *67
  • Enter the code and follow the voice prompt to set the remote extension number where you want to enable Call Forward Unconditional and the target destination.

Remote Call Forward Unconditional Disable

  • Default Code: *671
  • Enter the code and follow the voice prompt to set the remote extension number where you want to disable the Call Forward Unconditional.

Remote Call Forward Whitelist

Only the Extensions selected in this whitelist can configure call forwarding for any extension via feature codes.

Feature Codes

Voicemail Access Code

  • Default Code: *98
  • Enter *98 and follow the voice prompt. Alternatively, dial *98 followed by the extension and pound (#) to immediately access the entered extension’s voicemail box.

My Voicemail

  • Default Code: *97
  • Press *97 to access the voicemail box.

Agent Pause

  • Default Code: *83
  • Pause the agent in all call queues.

Agent Unpause

  • Default Code: *84
  • Unpause the agent in all call queues.

Paging Prefix

  • Default Code: *81
  • To page an extension, enter the code followed by the extension number.

Intercom Prefix

  • Default Code: *80
  • To intercom an extension, enter the code followed by the extension number.

Blacklist Add

  • Default Code: *40
  • To add a number to blacklist for inbound route, dial *40 and follow the voice prompt to enter the number.

Blacklist Remove

  • Default Code: *41
  • To remove a number from current blacklist for inbound route, dial *41 and follow the voice prompt to remove the number.

Call Pickup on Ringing

  • Default Code: **
  • To pick up a call for any extension xxxx, enter the code followed by the extension number xxxx.

Pickup In-call

  • Default Code: *45 (Disabled by default).
  • If “Pickup In-call” feature is enabled, only the extensions added in “Allowed to seamless transfer” in the extension’s Seamless Transfer Privilege Control List” can pick up the call.

Pickup Extension

  • Default Code: *8
  • This code is for the pickup group, which can be assigned for each extension on the extension configuration page.
  • If there is an incoming call to an extension, the other extensions within the same pickup group can dial *8 directly to pick up the call.

Direct Dial Voicemail Prefix

  • Default Code: *
  • This code is for the user to directly dial or transfer to an extension’s voicemail.
  • For example, directly dial *5000 will have to call go into the extension 5000’s voicemail. If the user would like to transfer the call to the extension 5000’s voicemail, enter *5000 as the transfer target number.

Direct Dial Mobile Phone Prefix

  • Default Code: *88
  • If you have the permission to call mobile phone number, use this prefix plus the extension number can dial the mobile phone number of this extension directly.

Call Completion Request

  • Default Code: *11
  • This code is for the user who wants to use Call Completion to complete a call.

Call Completion Cancel

  • Default Code: *12
  • This code is for the user who wants to cancel Call Completion request.

Enable Spy

Check this box to enable spy feature codes.

Disabled by default.

Listen Spy

  • Default Code: *54 (“Enable Spy” needs to be checked)
  • This is the feature code to listen in on a call to monitor performance. Monitor’s line will be muted, and neither party will hear from the monitor’s extension.

Whisper Spy

  • Default Code: *55 (“Enable Spy” needs to be checked)
  • This is the feature code to speak to one side of the call (for example, whisper to employees to help them handle a call). Only one side will be able to hear from the monitor’s extension.

Barge Spy

  • Default Code: *56 (“Enable Spy” needs to be checked)
  • This is the feature code to join in on the call to assist both parties.

Wakeup Service

  • Default Code: *36
  • Dial this code to access UCM wakeup service, you can add, update, activate or deactivate wakeup service.

PMS Wakeup Service

  • Default Code: *35
  • Dial this code to access UCM PMS wakeup service, you can add, update, activate or deactivate PMS wakeup service.

Update PMS Room Status

  • Default Code: *23
  • Use this code with maid code to update PMS room status. Choose the status to set after hearing the prompt, for example: for maid 001 dial *23001 and then 1 after hearing the prompt.

Presence Status

  • Dial this code to set the presence status of the extension.
  • Possible options are:

1:”unavailable”

2:”available”

3:”away”

4:”chat

5:”dnd

6:”userdef”

Dynamic Agent Logout

  • Default Code: *85
  • Use this code to logout the dynamic agent from all queues.

Voicemail Group Access Code

Call voicemail group access code to access group voicemail. If password is required, enter password followed by pound “#” key.

The UCM6200 also allows user to one click enable / disable specific feature code as shown below:

Figure 231: Enable/Disable Feature codes

Parking Lot

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

Figure 232: Parking Lot

User can create a new Parking lot by clicking on button “Create New Parking Lot” :

Figure 233: New Parking Lot

Table 95 : Parking Lot

Parking Lot Extension

  • Default Extension: 700
  • During an active call, initiate blind transfer and then enter this code to park the call.

Parking Lot Name

  • Set a name to the parking lot

Parked Slots

  • Default Extension: 701-720
  • These are the extensions where the calls will be parked, i.e., parking lots that the parked calls can be retrieved.

Use Parklot as Extension

  • If checked, the parking lot number can be used as extension. The user can transfer the call to the parking lot number to park the call. Please note this parking lot number range might conflict with extension range.

Parking Timeout (s)

  • Default setting is 300 seconds and the maximum limit is 99.999 seconds.
  • This is the timeout allowed for a call to be parked. After the timeout, if the call is not picked up, the extension who parks the call will be called back.

Music On Hold Playlists

Select the Music on Hold Class.

Failover Destination

Configures a callback failover destination when the extension that is called back is busy. The call will be routed to the destination number and this reduces the chance of dropping parked calls.

Note: This field cannot exceed 32 characters.

Ring All Callback on Timeout

If enabled, all registered endpoints of the extension will ring when callback occurs. Otherwise, only the original endpoint will be called back.

Note: This option will not be available if Forward to Destination on Timeout is enabled.

Forward to destination on timeout

If enabled, the call will be routed to the configured destination upon timeout. Otherwise, the call will be routed back to the original caller.

Timeout Destination

This option appears once Forward to Destination on Timeout is enabled. Upon park timeout, the call will be routed to the configured destination.

Call Park

The UCM6200 provides call park and call pickup features via feature code.

Park a Call

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

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

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

  • Feature Misc🡪Call Park (Default code 700)

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

Retrieve Parked Call

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

Monitor Call Park CID Name Information (GXP21xx Phones Only)

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

Note: This feature requires Grandstream GXP21xx new firmware support. Please check GXP21xx firmware release information for availability.

Figure 234: Monitored call park CID name

Call Recording

The UCM6200 allows users to record audio during the call. If “Auto Record” is turned on for an extension, ring group, call queue or trunk, the call will be automatically recorded when there is established call with it. Otherwise, please follow the instructions below to manually record the call.

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

Figure 235: Download Recording File from CDR Page

The above recorded call’s recording files are also listed under the UCM6200 Web GUI🡪CDR🡪Recording Files.

Note: Starting with firmware 1.0.20.17, Music on Hold will be also included in the recording.

Figure 236: Download Recording File from Recording Files Page

Enable Spy

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

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

“Enable Spy” allows any user to listen to any call by feature codes. This may result in the leakage of user privacy.

Shared Call Appearance (SCA)

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

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

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

Figure 237: Enabling SCA option under Extension’s Settings

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

Figure 238: SCA Number Configuration

  1. Configure the private number as desired.

Figure 239: SCA Private Number Configuration

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

Figure 240: SCA Options

SCA has various options to change its behavior:

  • Allow Call Retrieve from Another Location – Allows users to retrieve held calls using any device associated with the SCA extension.
  • Alert All Appearances for Group Paging Calls – Alerts all devices associated with the SCA extension.
  • Multiple Call Arrangement – Allows all devices associated with the SCA extension to make different calls at the same time.
  • Allow Bridging between Locations – Allows devices associated with the SCA extension to barge into existing calls of the same SCA group.
  • Bridge Warning Tone – Notification sound that will play when a party barges into the call. Three options are available:
  • None – No notification sound will play
  • Barge-In Only – The notification sound will play once when a party barges in.
  • Barge-In and Repeat – The notification sound will play when a party barges in and will play again after every 30 seconds.
  1. Next, configure the VPK or MPK to Shared for both the main extension and the private number. SCA is now configured for both endpoint devices.

The following table describe the SCA Number configuration setting:

Table 96: Add SCA Private Number

Private Number

Configures the private number for the SCA.

Related Shared Line

Display the related shared line.

Enable This Number

Enable/Disable the private number.

Allow Origination from This Number

Enabling this option will allow calling from this private number. By default, it is enabled.

Allow Termination to This Number

Enabling this option will allows calls to this private number. By default, it is enabled.

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

Table 97: Editing the SCA Number

Shared Line Number

While SCA is enabled, this number will be the same as the extension number.

Allow Call Retrieve from Another Location

Allows remote call retrieval. Must be enabled in public hold. By default, it is enabled.

Alert All Appearances for Group Paging Calls

Allows all SCA group members to ring when the SCA shared number is paged. If disabled, only the SCA shared number will ring when paged. By default, it is disabled.

Multiple Call Arrangement

Allows simultaneous calls in an SCA group. By default, it is disabled.

Allow Bridging between Locations

Allows location bridging for SCA group. Must be enabled when using the Barge-In feature. By default, it is disabled.

Bridge Warning Tone

Configures the notification in the bridge when another party join.

  • None: No notification sound.
  • Barge-In only: Notification sound will play when another party join.
  • Barge-In and Repeat: Notification sound will play when another party joins and repeat every 30 seconds.

By default, it is set to “Barge-In Only”.

Announcement

The Announcement feature (not to be confused with Announcement Paging and Announcement Center) is a feature that allows users to set an unskippable audio file to play to callers before routing them to a configured destination. Announcements can be configured as a destination in the [Inbound Routes] or in [IVR].

To configure the Announcement, users need to follow below steps:

  1. Navigate on the web GUI under “Call Features 🡪 Announcement
  2. Click on A close up of a sign

Description generated with high confidence to add a new Announcement.
  3. Configure the required fields Name, Prompt, Default Destination to be used for the announcement.
  4. Save and apply the configuration.

Figure 241: Announcement settings

The table below gives more description of the configuration parameters when creating Announcement.

Table 98: Announcement Parameters

Name

Configure the name of the Announcement.

Note: Please use letters, digits, _ or – only and no more than 64 characters.

Prompt

Audio file that will be played before ringing the configured default destination.

Note: Sound file must be PCM encoded, 16 bits at 8000Hz mono in mp3/wav format or raw ulaw/alaw/gsm file with .mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.

If uploading a compressed file, the file must have .tar/.tgz/.tar.gz suffix. the file name must contain only letters, numbers or special characters -_. The file size must be less than 30MB.

Filename cannot exceed 100 characters.

Default Destination

Select the destina