UCM630x Audio Series – User Manual

  • Updated on September 21, 2022

Thank you for purchasing Grandstream UCM630xA series IP PBX appliance. The UCM6300A series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on one centralized network, including voice, video calling, voice meeting, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300A series supports up to 1500 users and includes a built-in web meetings and meeting solution that allows employees to connect from the desktop, mobile, GVC series devices and IP phones. It can be paired with the UCM6300A ecosystem to offer a hybrid platform that combines the control of an on-premises IP PBX with the remote access of a cloud solution. The UCM630xA ecosystem consists of the Wave app for desktop and mobile, which provides a hub for collaborating remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections. The UCM6300A series also offers cloud setup and management through GDMS and an TableAPI for integration with third-party platforms. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, meeting and collaboration tools, the UCM6300A series provides a powerful platform for any organization.

Alert

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.

Caution

Please do not use a different power adaptor with the UCM630xA as it may cause damage to the product and void the manufacturer warranty.

Note

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.

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 UCM630xA series.

Interfaces

Analog Telephone FXS Ports

  • UCM6300A: 0 port

  • UCM6302A: 2 ports with lifeline support

  • UCM6304A: 4 ports with lifeline support

  • UCM6308A: 8 ports with lifeline support

Each port supports 3 REN

PSTN Line FXO Ports

  • UCM6300A: 0 port

  • UCM6302A: 2 ports

  • UCM6304A: 4 ports

  • UCM6308A: 8 ports

All ports have lifeline capability in case of power outage

Network Interfaces

Three self-adaptive Gigabit ports (switched, routed or dual card mode) with PoE+

NAT Router

Yes (supports router mode and switch mode)

Peripheral Ports

  • UCM6300A: USB 3.0, and SD card interface

  • UCM6302A: USB 2.0, USB 3.0, and SD card interface

  • UCM6304A/6308A: 2x USB 3.0 and SD card interface

LED Indicators

  • UCM6300A/6302A/UCM6304A: None

  • UCM6308A: Power 1/2, FXS, FXO, LAN, WAN, Heartbeat

LCD Display

  • UCM6300A/6302A/6304A: 320*240 LCD with touch screen for Shortcut Keys and Scroll Bar.

  • UCM6308A: 128x32 dot matrix graphic LCD with DOWN and OK buttons

Reset Switch

Yes, long press for factory reset and short press for reboot

Voice Capabilities

Voice-over-Packet
Capabilities

- LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line
- Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711, NetEQ, FEC 2.0, jitter resilience up to 50% audio packet loss

Voice and Fax Codecs

Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38

QoS

Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

Signalling and Control

DTMF Methods

Inband, RFC4733, and SIP INFO

Provisioning Protocol and
Plug-and-Play

Mass provisioning using AES encrypted XML configuration file, auto-discovery & 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, STUN, SRTP, TLS, LDAP, HDLC, HDLC-ETH, PPP, Frame Relay (pending), IPv6, OpenVPN®

API

Full API available for third-party platform and application integration.

Disconnect Methods

Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect

Security

Media Encryption

SRTP, TLS1.2, HTTPS, SSH, 802.1x

Physical

Universal Power Supply

  • UCM6300A/6302A/6304A: Input: 100 ~ 240VAC, 50/60Hz; Output: DC+12V, 1.5A

  • UCM6308A: 2x DC 12V Power Jack Input: 100~240VAC,50/60Hz; Output: DC+12V, 2A

Dimensions

  • UCM6300A/6302A/6304A: 270mm(L) x 175mm(W) x 36mm(H)

  • UCM6308A: 485mm(L) x 187.2mm(W) x 46.2mm(H)

Environmental

  • Operating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)

  • Storage: 14 - 140ºF / -10 ~ 60ºC, Humidity 10 - 90% (non-condensing)

Mounting

  • Wall mount (Unit will be fixed on the wall using screw) & Desktop for
    UCM6300A/6302A/6304A.

  • Desktop & Rack mount for UCM6308A.

Weight

  • UCM6300A: Unit weight 705g, Package weight 1200g

  • UCM6302A: Unit weight 725g, Package weight 1221g

  • UCM6304A: Unit weight 775g, Package weight 1271g

  • UCM6308A: Unit weight 2540g, Package weight 3465g

Additional Features

Multi-language Support

  • Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish

  • Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Dutch

  • Customizable language pack to support any other languages

Caller ID

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

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/workload, in-queue announcement

Customizable Auto
Attendant

Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

Telephony Operating
System

Based on Asterisk version 16

Maximum Call Capacity

UCM6300A:

  • Users: 250

  • Concurrent calls (G.711): 50

  • Max concurrent SRTP calls: 50

UCM6302A:

  • Users: 500

  • Concurrent calls (G.711): 75

  • Max concurrent SRTP calls: 75

UCM6304A:

  • Users: 1000

  • Concurrent calls (G.711): 150

  • Max concurrent SRTP calls: 120

UCM6308A:

  • Users: 1500

  • Concurrent calls (G.711): 200

  • Max concurrent SRTP calls: 150

Maximum Attendess of Meeting Bridges

  • UCM6300A: 3 Meeting rooms and up to 50 parties

  • UCM6302A: 5 Meeting rooms and up to 75 parties

  • UCM6304A: 7 Meeting rooms and up to 120 parties

  • UCM6308A: 9 Meeting rooms and up to 150 parties

Call Features

Call park, call forward, call transfer, call waiting, caller ID, call record, call history, ringtone, IVR, music on hold, call routes, DID, DOD, DND, DISA, ring group, ring simultaneously, time schedule, PIN groups, call queue, pickup group, paging/intercom, voicemail, call wakeup, SCA, BLF, voicemail to email, speed dial, call back, dial by name, emergency call, call follow me, blacklist/whitelist, voice meeting, eventlist, feature codes, busy camp-on/ call completion, voice control

Wave Mobile App

Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the UCM630xA

Firmware Upgrade

Supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system, It provides a centralized interface to provision, manage, monitor, and troubleshoot Grandstream products

Compliance

  • FCC: Part 15 (CFR 47) Class B, Part 68

  • CE: EN 55032, EN 55035, EN61000-3-2, EN61000-3-3, EN 62368.1, ES 203 021, ITU K.21

  • IC: ICES-003, CS-03 Part I Issue 9

  • RCM: AS/NZS CISPR 32, AS/NZS 62368.1, AS/CA S002, AS/CA S003.1/.2

  • Power adapter: UL 60950-1 or UL 62368-1


Table 1: Technical Specifications

UCM630xA FXS ports lifeline functionality: The UCM630xA 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 UCM630xA 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 UCM630xA series.

Equipment Packaging

Main Case1
Power Adaptor1
Ethernet Cable1
Quick Installation Guide1

Table 2: UCM630xA Equipment Packaging

UCM6300A front and back view

Figure 1: UCM6300A Back View

Figure 2:UCM6300A Front View

UCM6302A front and back view


Figure 3: UCM6302A Back View

Figure 4: UCM6302A Front View

UCM6304A front and back view


Figure 5: UCM6304A Front View

Figure 6: UCM6304A Back View

UCM6308A front and back view


Figure 7: UCM6308A Front View


Figure 8: UCM6308A Back View

Safety Compliances

The UCM630xA series IP PBX complies with FCC/CE and various safety standards. The UCM630xA power adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM630xA package only. The manufacturer’s warranty does not cover damages to the device caused by unsupported power adapters.

Warranty

If the UCM630xA 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 UCM630xA 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 UCM630xA setup process, use the following available interfaces: LCD display, and web portal.

  • The LCD display shows hardware, software, interface status and network information and can be navigated via the Slide control and Touch keys. From here, users can configure basic network settings, run diagnostic tests, and factory reset.
  • 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 and IP address. Upon menu key press timeout (30 seconds), the screen will default back to this information.

  • Menu

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

  • Menu Navigation

Scrolling sown using slide control 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 UCM630x.


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

Table 3: LCD Menu Options

Use the LED Indicators

The UCM6300A/6302A has LED indicators on the network port to display connection status and the following picture shows the other ports status.


Figure 9: Ports Status


The UCM6304A/6308A has LED indicators in the front to display connection status. The following table shows the status definitions.

LED IndicatorLED Status
Power 1/Power 2

 

PoE

LAN

WAN

USB

SD

FXS ports

FXO ports

  • Solid: Connected
  • Fast Blinking: Data Transferring
  • Slow Blinking: Trying to connect
  • OFF: Not Connected

Table 4: UCM6304A/6308A LED Indicators

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 10: UCM630xA 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 can be found on the sticker at the back of the UCM.

By default, the UCM630xA has Redirect From Port 80 enabled. As such, if users type in the UCM630xA 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 under the UCM630xA Web GUI🡪System Settings🡪HTTP Server.

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 11: UCM630xA 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.
  • Other 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 UCM630xA series Web GUI supports English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, Russian, Italian, Polish, German etc.

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


Figure 12: UCM630xA Web GUI Language

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


Figure 13: 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 finalizes the changes. If a modification requires a reboot, a prompt will appear asking to reboot the device.

Setting Up an Extension


Power on the UCM630xA 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 UCM630xA series. This includes settings for the following items: General Settings, HTTP server, network Settings, OpenVPN, DDNS Settings, Security Settings, LDAP server, Time settings, Email settings and TR-069.

General Settings

System administrators can prevent the UCM from making calls and/or writing to the data partition (e.g., CDR, recordings, etc.) once the system reaches a specified threshold of storage usage and CPU usage respectively. These options are located in the System Settings 🡪 General Settings page.


Figure 14: General Settings Interface

General Settings
Device NameConfigure the name of the UCM.
Enable CPU Flow ControlEnables the CPU flow control.
CPU Flow Control ThresholdUsed to set the threshold generated by the CPU Flow Control. When the system CPU reaches the threshold, it will prohibit the new calls.

 

Default value is 90%.

Data Partition Write ThresholdUsed to set a threshold to stop writing data partition. When the disk data partition reaches the threshold configured, the data partition writing will be stopped. Default value is 90%.

Table 5: General Settings Parameters

IM Settings

Cloud IM Service

After enabling Cloud IM, it means that all IM data in Grandstream Wave is stored in the external server Cloud IM, and is no longer stored locally in UCM.GDMS can configure Cloud IM service for UCM devices. At this time, the UCM device synchronizes the configuration item information.


Figure 15: Cloud IM

Cloud IM Service

Enable Cloud IM

If you have purchased the UCM Cloud IM package or purchased the Grandstream IM server, you can configure it.  If you have not purchased it, the configuration will not take effect, but UCM local IM service is allowed. Please note that after enabling this feature, local chat data will not be visible.

Local Proxy

If enabled, the local proxy will be used to forward files and text messages if the IM server cannot be connected to upon Wave login due to certificate issues.

Cloud IM Server Address

The address of the server that provides IM service, you can fill in the address of the Cloud IM server provided by the RemoteConnect package or the IM server address of the GDMS.

Service ID

The service ID of the Cloud IM server.

Key

The Key to the Cloud IM server.

Company Name

Company Name

Trusted User

The trusted user of the cloud IM. Only letters, numbers, and special characters are allowed.

Prepend

As the extension prefix, it is added before the extension number.

Only account details and department information will be synced on local IM and cloud IM. Other configurations such as profile picture, work status and favorite contacts will not be synced, and these are stored in local IM or cloud IM respectively. Therefore, please be aware that when switching between local IM and cloud IM, part of the data cannot be synced and the previously stored data on local IM or cloud IM (depending on which one is switched to) will be retrieved.

IM Server

If Enable IM Server Mode is toggled on, UCM will function only as an IM server. The UCM management portal will remove PBX related services and supports the binding of multiple cross-region UCM devices. The UCM device that wants to bind the IM server address is also bound by turning on the Cloud IM mode, and the IM data in his Grandstream Wave is stored in this IM server.


Figure 16: IM server configuration interface
Company nameThe entered company name
Server AddressThe domain name or IP address of the Cloud IM server.
Service IDThe service ID of the Cloud IM server.
KeyThe Key of the Cloud IM server.
Trusted UserThe trusted user of the cloud IM. Only letter, number, and special characters are allowed.
Bound device information
DepartmentThe department represented by the bound UCM.
MAC AddressMAC address of the bound UCM device.
Dial prefixExtension prefix

Table 7: IM Server parameters

HTTP Server

The UCM630xA’s embedded web server responds to 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.

UCM Web Settings
Redirect From Port 80Toggles automatic redirection to UCM’s web portal from port 80. If disabled, users will need to manually add the UCM’s configured HTTPS port to the server address when accessing the UCM web portal via browser. Default is “Enabled”.
PortSpecifies the port number used to access the UCM HTTP server. Default is “8089”.
Enable IP Address WhitelistIf 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

External HostConfigure a URL and port (optional) used to access the UCM web portal if the UCM is behind NAT.
Grandstream Wave Settings
External HostConfigure a URL and port (optional) used to access the UCM web portal if the UCM is behind NAT.
PortThe port to access Wave Web and Wave Mobile. If behind NAT, please make sure to map the external port to this port.
Certificate Settings
Certificate OptionsSelects 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 KeyUploads 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 CertUploads 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.

DomainEnter the domain to request the certificate for and click on

to request the certificate.

Table 8: HTTP Server Settings

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 UCM630xA 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.

  • UCM630xA 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 UCM6300A, UCM6302A, UCM6304A and UCM6308A, respectively.

Method

Select "Route", "Switch" or "Dual" mode on the network interface of UCM630X Audio Series. The default setting is "Switch".

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

  • Switch: WAN port will be used for the 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 1492.

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.

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 the 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 the WAN port. The default value is 0.

LAN (when Method is set to "Route")

IP Address

Enter the IP address assigned to the 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.

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 the 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 the LAN port. The default value is 0.

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

Default Interface

If "Dual" is selected as "Method", users will need to assign the default interface to be LAN 1 (mapped to UCM6302 Audio Series WAN port) or LAN 2 (mapped to UCM6302 Audio Series LAN port) and then configure network settings for LAN 1/LAN 2. The default interface is LAN 2.

IP Method

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

IP Address

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

Subnet Mask

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

Gateway IP

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

DNS Server 1

Enter the DNS server 1 address for static IP settings.

DNS Server 2

Enter the DNS server 2 address for static IP settings.

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 the 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 the 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: the DHCPv6 server is disabled.

  • Auto: Stateless address auto configuration using NDP protocol.

  • DHCPv6: Stateful address auto configuration using DHCPv6 protocol.

The default setting is Disabled.

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 will need to assign the default interface to be LAN 1 (mapped to UCM630X Audio Series WAN port) or LAN 2 (mapped to UCM630X Audio Series LAN port) and then configure network settings for LAN 1/LAN 2. The default interface is LAN 1.

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.

Network Port Traffic Control

LAN (when Method is set to "Switch")

Enable Network Port Traffic Storm Alert

The UCM will send a an alert notification/email when there is an excessive number of packets in the LAN that impacts the overall performance of the network. 

Note: To enable this feature email or HTTP notification should be set up correctly In Maintenance 🡲 System Events.

Network Port Receiving Traffic Control

You can monitor the traffic in the RX direction on each network port and generate an alarm when the corresponding alarm event is turned on and the set threshold value is exceeded. 

The threshold range is 1 - 1024000 in kbps and 1 - 1000 in mbps.

LAN 1 & LAN 2 (when Method is set to "Dual")

Enable Network Port Traffic Storm Alert

The UCM will send a an alert notification/email when there is an excessive number of packets in the LAN that impacts the overall performance of the network. 

Note: To enable this feature email or HTTP notification should be set up correctly In Maintenance 🡲 System Events.

LAN1 & LAN2

- Network Port Receiving Traffic Control

You can monitor the traffic in the RX direction on each network port and generate an alarm when the corresponding alarm event is turned on and the set threshold value is exceeded. 

The threshold range is 1 - 1024000 in kbps and 1 - 1000 in mbps.

LAN & WAN (When Method is set to Route Mode)

Enable Network Port Traffic Storm Alert

The UCM will send a an alert notification/email when there is an excessive number of packets in the LAN that impacts the overall performance of the network. 

Note: To enable this feature email or HTTP notification should be set up correctly In Maintenance 🡲 System Events.

WAN:

Network Port Receiving Traffic Control

You can monitor the traffic in the RX direction on each network port and generate an alarm when the corresponding alarm event is turned on and the set threshold value is exceeded. 

The threshold range is 1 - 1024000 in kbps and 1 - 1000 in mbps.

LAN:

Network Port Receiving Traffic Control

You can monitor the traffic in the RX direction on each network port and generate an alarm when the corresponding alarm event is turned on and the set threshold value is exceeded. 

The threshold range is 1 - 1024000 in kbps and 1 - 1000 in mbps.

  • Method: Route

When the UCM630xA 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 a sample diagram below.


Figure 17: UCM6302A Network Interface Method: Route
  • Method: Switch

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


Figure 18: UCM6302A 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.


Figure 19: UCM6302A Network Interface Method: Dual

802.1X

IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication mechanism to device before the device can access Internet or other LAN resources. The UCM630xA supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure show UCM630xA use 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet.


Figure 20: UCM630xA Using 802.1X as Client

Figure 21: UCM630xA Using 802.1X EAP-MD5

The following table shows the configuration parameters for 802.1X on UCM630xA. 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, users will also need to upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated from the RADIUS server.

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

 

  • EAP-MD5
  • EAP-TLS
  • EAP-PEAPv0/MSCHAPv2
IdentityEnter 802.1X mode Identity information.
MD5 PasswordEnter 802.1X mode MD5 password information.
802.1X CA CertificateSelect 802.1X certificate from local PC and then upload.
802.1X Client CertificateSelect 802.1X client certificate from local PC and then upload.

Table 10: UCM630xA Network Settings🡪802.1X

Static Routes

The UCM630xA provides users static routing capability that allows the device to use manually configured routes, rather than information only from dynamic routing or gateway configured in the UCM630xA Web GUI🡪System Settings🡪Network Settings🡪Basic Settings to forward traffic. It can be used to define a route when no other routes are available or necessary, or used in complementary with existing routing on the UCM630xA as a failover backup, etc.


  • Click on “Add IPv4 Static Route” to create a new IPv4 static route or click on ”Add IPv6 Static Route” 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.

DestinationConfigure the destination IPv4 address or the destination IPv6 subnet for the UCM630xA to reach using the static route.

 

Example:

IPv4 address – 192.168.66.4

IPv6 subnet – 2001:740:D::1/64

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

 

Example:

255.255.255.0

GatewayConfigure the IPv4 or IPv6 gateway address so that the UCM630xA can reach the destination via this gateway. Gateway address is optional.

 

Example:

192.168.40.5 or 2001:740:D::1

InterfaceSpecify the network interface on the UCM630xA to reach the destination using the static route.

 

LAN interface is eth0; WAN interface is eth1.


Table 11: UCM630xA Network Settings🡪Static Routes

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

The following diagram shows a sample application of static route usage on UCM6304A.


Figure 22: UCM6304A Static Route Sample

The network topology of the above diagram is as below:

  • Network 192.168.69.0 has IP phones registered to UCM6304A LAN 1 address
  • Network 192.168.40.0 has IP phones registered to UCM6304A LAN 2 address
  • Network 192.168.66.0 has IP phones registered to UCM6304A 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 UCM6304A. Therefore, we need configure a static route on the UCM6304A so that the phones in isolated networks can make calls between each other.


Figure 23: UCM6304A Static Route Configuration

Port Forwarding

The UCM network interface supports router function which provides users the ability to do port forwarding. If LAN mode is set to “Route” under Web GUI🡪System Settings🡪Network Settings🡪Basic Settings page, port forwarding is available for configuration.

The port forwarding configuration is under Web GUI🡪System Settings🡪Network Settings🡪Port Forwarding page. Please see related settings in the table below.

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

 

Note:

When it is set to a range, WAN port and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access from WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000 will be port forwarding to LAN port 1000.

LAN IPSpecify the LAN IP address.
LAN PortSpecify the LAN port number or a range of LAN ports.

 

Note:

When it is set to a range, WAN port and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access from WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000 will be port forwarding to LAN port 1000.

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

Table 12: UCM630xA Network Settings🡪Port Forwarding

The following figures demonstrate a port forwarding example to provide phone’s Web GUI access to public side.

  • UCM630xA network mode is set to “Route”.
  • UCM630xA WAN port is connected to uplink switch, with a public IP address configured, e.g. 1.1.1.1.
  • UCM630xA LAN port provides DHCP pool that connects to multiple phone devices in the LAN network 192.168.2.x. The UCM60X is used as a router, with gateway address 192.168.2.1.
  • There is a GXP2160 connected under the LAN interface network of the UCM630xA. It obtains IP address 192.168.2.100 from UCM630xA DHCP pool.
  • On the UCM630xA Web GUI🡪System Settings🡪Network Settings🡪Port Forwarding, configure a port forwarding entry as the figure shows below.

  • Click on

WAN Port: This is the port opened on the WAN side for access purpose.

LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM630xA.

LAN Port: This is the port opened on the GXP2160 side for access purpose.

Protocol Type: We select TCP here for Web GUI access using HTTP.


Figure 24: Create New Port Forwarding

Figure 25: UCM630xA Port Forwarding Configuration

This will allow users to access the GXP2160 Web GUI from public side, by typing in public IP address (example: 1.1.1.1:8088).


Figure 26: GXP2160 Web Access using UCM6302A Port Forwarding

ARP Settings

The ARP settings can be configured under Web GUI🡪System Settings🡪Network Settings🡪ARP Settings

ARP GC Threshold 1Minimum number of entries to keep. Garbage collector will not purge entries if there are fewer than this number. The default value is 128.
ARP GC Threshold 2Threshold when garbage collector becomes more aggressive about purging entries. Entries older than 5 seconds will be cleared when over this number. The default value is 512.
ARP GC Threshold 3Maximum number of non-PERMANENT neighbor entries allowed. Increase this when using large numbers of interfaces and when communicating with large numbers of directly connected peers. The default value is 1024.

Table 13: ARP Settings

OpenVPN®

OpenVPN® settings allow the users to configure UCM630xA to use VPN features, the following table gives details about the various options in order to configure the UCM as OpenVPN Client.


Figure 27: OpenVPN® Feature on the UCM630xA

OpenVPN® Enable

Enable / Disable the OpenVPN® feature.

Configuration Method

Select the 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.

Enable Weak SSL Ciphers

Either to enable the Weak SSL ciphers or not.

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 ‘client.crt’ automatically.

OpenVPN® Client Key 

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

Username

Username used to authenticate into the server.

Password

Password used to authenticate into the server.

DDNS Settings

DDNS setting allows user to access UCM630xA via domain name instead of IP address.

The UCM supports DDNS service from the following DDNS provider:

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

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

  1. Register domain in DDNS service provider. Please note the UCM630xA needs to have public IP access.

Figure 28: Register Domain Name on noip.com
  1. On Web GUI🡪System Settings🡪Network Settings🡪DDNS Settings, enable DDNS service and configure username, password, and host name.

Figure 29: UCM630xA DDNS Setting
  1. Now you can use domain name instead of IP address to connect to the UCM630xA Web GUI.

Figure 30: Using Domain Name to Connect to UCM630xA

Security Settings

The UCM630xA provides users firewall security configurations to prevent certain malicious attack to the UCM630xA system. Users could configure to allow, restrict, or reject specific traffic through the device for security and bandwidth purpose. The UCM630xA also provides Fail2ban feature for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE. To configure firewall settings in the UCM630xA, go to Web GUI🡪System Settings🡪Security Settings page.

Static Defense

Under Web GUI🡪System Settings🡪Security Settings🡪Static Defense page, users will see the following information:

  • Current service information with port, process, and type.
  • Typical firewall settings.
  • Custom firewall settings.

The following table shows a sample current service status running on the UCM630xA.

PortProcessTypeProtocol or Service
7777AsteriskTCP/IPv4SIP
389SlapdTCP/IPv4LDAP
6060zero_configUDP/IPv4UCM630xA zero_config service
5060AsteriskUDP/IPv4SIP
4569AsteriskUDP/IPv4SIP
38563Asteriskudp/ipv4SIP
10000gs_avsudp/ipv4gs_avs
10001gs_avsudp/ipv4gs_avs
10002gs_avsudp/ipv4gs_avs
10003gs_avsudp/ipv4gs_avs
10004gs_avsudp/ipv4gs_avs
10005gs_avsudp/ipv4gs_avs
10006gs_avsudp/ipv4gs_avs
10007gs_avsudp/ipv4gs_avs
10010gs_avsudp/ipv4gs_avs
10012gs_avsudp/ipv4gs_avs
10013gs_avsudp/ipv4gs_avs
10014gs_avsudp/ipv4gs_avs
10015gs_avsudp/ipv4gs_avs
10018gs_avsudp/ipv4gs_avs
10019gs_avsudp/ipv4gs_avs
10020gs_avsudp/ipv4gs_avs
6066Pythonudp/ipv4python
3306Mysqldtcp/ipv4mysqld
45678Pythonudp/ipv4python
8439Lighttpdtcp/ipv4HTTP
8088asterisktcp/ipv4SIP
8888Pbxmidtcp/ipv4pbxmid
25Mastertcp/ipv4master
636Slapdtcp/ipv4SLDAP
4569asteriskudp/ipv6SIP
42050asteriskudp/ipv6SIP
7681Pbxmidtcp/ipv4pbxmid

Table 15: UCM630xA Firewall🡪Static Defense🡪Current Service

For typical firewall settings, users could configure the following options on the UCM630xA.

Ping Defense EnableIf enabled, ICMP response will not be allowed for Ping request. The default setting is disabled. To enable or disable it, click on the check box for the LAN or WAN (UCM630xA) interface.
SYN-Flood Defense EnableAllows the UCM630xA to handle excessive amounts of SYN packets from one source and keep the web portal accessible. There are two options available and only one of these options may be enabled at one time.

 

  • eth(0)LAN defends against attacks directed to the LAN IP address of the UCM630xA.
  • eth(1)WAN defends against attacks directed to the WAN IP address of the UCM630xA.

SYN Flood Defense will limit the amount of SYN packets accepted by the UCM from one source to 10 packets per second. Any excess packets from that source will be discarded.

Ping-of-Death Defense EnableEnable to prevent Ping-of-Death attack to the device. The default setting is disabled. To enable or disable it, click on the check box for the LAN or WAN (UCM630xA) interface.

Table 16: Typical Firewall Settings

Under “Custom Firewall Settings”, users could create new rules to accept, reject or drop certain traffic going through the UCM630xA. To create new rule, click on “Create New Rule” button and a new window will pop up for users to specify rule options.

Right next to “Create New Rule” button, there is a checkbox for option “Reject Rules”. If it is checked, all the rules will be rejected except the firewall rules listed below. In the firewall rules, only when there is a rule that meets all the following requirements, the option “Reject Rules” will be allowed to check:

  • Action: “Accept”
  • Type “In”
  • Destination port is set to the system login port (e.g., by default 8089)
  • Protocol is not UDP

Figure 31: Create New Firewall Rule

Rule NameSpecify the Firewall rule name to identify the firewall rule.
ActionSelect the action for the Firewall to perform.

 

  • ACCEPT
  • REJECT
  • DROP
TypeSelect the traffic type.

 

  • IN

If selected, users will need specify the network interface “LAN” or “WAN” (for UCM630xA) for the incoming traffic.

  • OUT
InterfaceSelect the interface to use the Firewall rule.
ServiceSelect the service type.

 

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

If “Custom” is selected, users will need specify Source (IP and port), Destination (IP and port) and Protocol (TCP, UDP or Both) for the service. Please note if the source or the destination field is left blank, it will be used as “Anywhere”.

Source IP Address and PortConfigure a source subnet and port. If set to “Anywhere” or left empty, traffic from all addresses and ports will be accepted. A single port or a range of ports can be specified (e.g., 10000, 10000-20000).
Destination IP Address and PortConfigure a destination subnet and port. If set to “Anywhere” or left empty, traffic can be sent to all addresses and ports. A single port or a range of ports can be specified (e.g., 10000, 10000-20000).
ProtocolSelect the protocol for the rule to be used.

Table 17: Firewall Rule Settings

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

Dynamic defense is supported on the UCM630xA series. It can blacklist hosts dynamically when the LAN mode is set to “Route” under Web GUI🡪System Settings🡪Network Settings🡪Basic Settings page. If enabled, the traffic coming into the UCM630xA can be monitored, which helps prevent massive connection attempts or brute force attacks to the device. The blacklist can be created and updated by the UCM630xA firewall, which will then be displayed in the web page. Please refer to the following table for dynamic defense options on the UCM630xA.

Dynamic Defense EnableEnable dynamic defense. The default setting is disabled.
Blacklist Update IntervalConfigure the blacklist update time interval (in seconds). The default setting is 120.
Connection ThresholdConfigure the connection threshold. Once the number of connections from the same host reaches the threshold, it will be added into the blacklist. The default setting is 100.
Dynamic Defense Whitelist
Allowed IPs and ports range, multiple IP addresses and port range.

 

For example:

192.168.2.100-192.168.2.105, 1000:9999


Table 18: UCM630xA Firewall Dynamic Defense

The following figure shows a configuration example like this:

  • If a host at IP address 192.168.5.7 initiates more than 20 TCP connections to the UCM630xA it will be added into UCM630xA blacklist.
  • This host 192.168.5.7 will be blocked by the UCM630xA 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 UCM630xA it will not be added into UCM630xA blacklist. It can still establish TCP connection with the UCM630xA.

Figure 32: Configure Dynamic Defense

Fail2ban

Fail2Ban feature on the UCM630xA provides intrusion detection and prevention for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE. Once the entry is detected within “Max Retry Duration”, the UCM630xA will act to forbid the host for certain period as defined in “Banned Duration”. This feature helps prevent SIP brute force attacks to the PBX system.


Figure 33: 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 UCM630X.

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

Within this duration (in seconds), if a host exceeds the max times of retry as defined in "MaxRetry", the host will be banned. The default setting is 600.

MaxRetry

Configure the number of authentication failures during "Max Retry Duration" before the host is banned. The default setting is 5.

Fail2Ban Whitelist

Configure IP address, CIDR mask, or DNS host in the whitelist. Fail2Ban will not ban the host with a matching address in this list. Up to 20 addresses can be added to the list descriptions/comments can be added for each whitelist entry for admin to log what’s the whitelist IP address is for.

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 UCM630X.

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 TCP.

MaxRetry

Configure the number of authentication failures during "Max Retry Duration" before the host is banned. The default setting is 5. Please make sure this option is properly configured as it will override the "MaxRetry" value under "Global Settings".

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

When the number of failed login attempts from an IP address exceeds the MaxRetry number, that IP address will be banned from accessing the Web GUI.

Customer Service System Call Defense

Enable call defense in the customer service system. Off by default.

Listening Port Number

The current service listening port. Default UDP port: 5060, TCP port: 5060, 5061, WebSocket communication port: 8088.

MaxRetry

Set the maximum number of calls allowed in the "time span". The local matching threshold has a higher priority than the global matching threshold. The default setting is 5.

Blacklist

Blacklist

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

SSH Access

SSH switch now is available via Web GUI and LCD. User can enable or disable SSH access directly from Web GUI or LCD screen. For web SSH access, please log in UCM630xA web interface and go to Web GUI🡪System Settings🡪Security Settings🡪SSH Access.

The “Enable SSH access” option is for system debugging. If you enable this option, the system will allow SSH access. The SSH connection also requires the username and password of the super administrator. This option is turned off by default. It is recommended to turn off this option when debugging is not required.

Enable the “Enable remote SSH (via GDMS)” option, the system will allow remote SSH access via the GDMS platform. This option is turned off by default, and it is strongly recommended to turn off this option when remote troubleshooting is not required.


Figure 34: SSH Access
Enable SSH AccessThis option is used for system debugging. Once enabled, UCM will allow SSH access. The SSH connection requires super administrator’s username and password. The default setting is “No”. It is recommended to set it to “No” if there is no need for debugging.
Enable Remote SSH via GDMSIf this option is enabled, remote SSH access will be allowed through the GDMS platform. It is strongly recommended to keep this disabled unless remote troubleshooting is necessary.

Table 20: SSH Access

LDAP Server

The UCM630xA has an embedded LDAP/LDAPS server for users to manage corporate phonebook in a centralized manner.

  • By default, the LDAP server has generated the first phonebook with PBX DN “ou=pbx,dc=pbx,dc=com” based on the UCM630xA user extensions already.
  • Users could add new phonebook with a different Phonebook DN for other external contacts. For example, “ou=people,dc=pbx,dc=com”.
  • All the phonebooks in the UCM630xA 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 UCM630xA, 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 UCM630xA. If the UCM630xA 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 UCM630xA 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 UCM630xA.


Figure 35: LDAP Server Configurations

The UCM630xA 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” 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.

The UCM630xA support secure LDAP (LDAPS) where the communication is encrypted and secure.


Figure 36: Default LDAP Phonebook DN

Figure 37: Default LDAP Phonebook Attributes

LDAP Phonebook

Users could use the default phonebook, edit the default phonebook, add new phonebook, import phonebook on the LDAP server as well as export phonebook from the LDAP server. The first phonebook with default phonebook dn “ou=pbx,dc=pbx,dc=com” displayed on the LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly. 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 38: 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 39: Add LDAP Phonebook

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

Once added, users can select

to edit the phonebook attributes and contact list (see figure below) or select

to delete the phonebook.


Figure 40: 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 41: 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 how a sample file looks like. Please note “Account Number” and “Phonebook DN” fields are required. Users could export a phonebook file from the UCM630xA LDAP phonebook section first and use it as a sample to start with.


Figure 42: 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 UCM630xA 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 UCM630xA LDAP Phonebook, the contacts in the CSV file will be added into the existing phonebook. If the phonebook DN does not exist on the UCM630xA 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 UCM630xA.


Figure 43: LDAP Phonebook After Import

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

  • Export phonebook to your computer from UCM630xA 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 UCM630xA again.


Figure 44: Export Selected LDAP Phonebook

LDAP Settings

Prerequisites to support contacts sync-up to IP Phones, UCM needs to support the following:

1.         If Cloud IM is enabled, UCM can send remote UCM’s contacts to each end device.

2.         Contacts from remote UCM can be synced by Cloud IM or LDAP sync via trunk. The contacts data must be complete and consistent.

3.         If Cloud IM is enabled, the contacts sent from UCM to end device should integrate Cloud IM contacts.

4.         If Cloud IM is disabled, the contacts sent from UCM to end device should only contain contacts on the UCM.

To support contacts sync-up to Wave, it allows Wave to obtain enterprise contacts from Cloud IM or LDAP. On UCM SIP peer trunk, if LDAP sync is enabled, end point can obtain remote UCM extensions’ info via LDAP. Also, it will allow configuring whether to sync up LDAP contacts on Wave so that Wave doesn’t receive duplicate contacts info.

Under UCM webUI🡪 System Settings🡪 LDAP Server, click on “LDAP Settings”, option “Wave enable LDAP phonebook” is available for configuration. If enabled, all Wave users on this UCM will display LDAP contacts. Otherwise, it will not display.

LDAP Settings

Please note the LDAP contacts displayed on Wave will exclude the duplicate contacts from Cloud IM.

Display LDAP Contacts on Wave

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:

Phonebook Name

Enter a name for the phonebook

Server Address

The IP address of the LDAP server

Base DN

Enter the base domain name.

Username

Enter the username used to authenticate into the LDAP server, if authentication is required.

Password

Enter the password used to authenticate into the LDAP server, if authentication is required.

Filter

Enter the filter. Ex: (|(CallerIDName=%)(AccountNumber=%))

Port

Enter the port number. Default port is 389

LDAP Number Attributes

Enter the number attributes for the remote server.

Automatic Update Cycle

If "None" is selected, LDAP phonebooks will not automatically update. Otherwise, LDAP phonebooks will automatically update at 00:00 / 12:00 AM with the selected frequency.

LDAP Name Attributes

Enter the name attributes for the remote server.

Client Type

Choose the client type. For encrypted data transfer please choose LDAPS.

LDAP Client CA Cert

LDAP Client Public Certification

LDAP Client Private Key

LDAP Client Private Certification

The UCM can automatically update the phonebook, by configuring the ‘LDAP Automatic Update Cycle’. Available options are: 1 day/2days/7 days. It is set to ‘None’ by default.

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


Figure 45: 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 UCM
  • 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 35: LDAP Server Configurations].

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

Figure 46: GXP2170 LDAP Phonebook Configuration

AD Client Type

Phonebook Name

Enter a name for the phonebook

Server Address

The IP address of the LDAP server

Base DN

Enter the base domain name.

Username

Enter the username used to authenticate into the LDAP server, if authentication is required.

Password

Enter the password used to authenticate into the LDAP server, if authentication is required.

Filter

Enter the filter. Ex: (|(CallerIDName=%)(AccountNumber=%))

Port

Enter the port number. Default port is 389

AD Attributes

AccountNumber must be included if the default configuration is used.

Automatic Update Cycle

If "None" is selected, LDAP phonebooks will not automatically update. Otherwise, LDAP phonebooks will automatically update at 00:00 / 12:00 AM with the selected frequency.

Host Name

Enter the host name of the remote AD server. 

Time Settings

Automatic Date and Time

The current system time on the UCM630xA can be found under Web GUI🡪System Status🡪Dashboard🡪PBX Status.

To configure the UCM630xA to update time automatically, go to Web GUI🡪System Settings🡪Time Settings🡪Automatic date and Time.

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 UCM630xA for the first time to avoid service interrupt after installation and deployment in production.

Remote NTP ServerSpecify the URL or IP address of the NTP server for the UCM630xA to synchronize the date and time. The default NTP server is pool.ntp.org.
Enable DHCP Option 2If set to “Yes”, the UCM630xA can get provisioned for Time Zone from DHCP Option 2 in the local server automatically. The default setting is “Yes”.
Enable DHCP Option 42If set to “Yes”, the UCM630xA can get provisioned for NTP Server from DHCP Option 42 in the local server automatically. This will override the manually configured NTP Server. The default setting is “Yes”.
Time ZoneSelect the proper time zone option so the UCM630xA can display correct time accordingly.

Table 21: Time Auto Updating

Set Date and Time

To manually set the time on the UCM630xA, go to Web GUI🡪System Settings🡪Time Settings🡪Set Date and Time. The format is YYYY-MM-DD HH:MM:SS.


Figure 47: Set Time Manually

Manually setup time will take effect immediately after saving and applying change in the Web GUI. If users would like to reboot the UCM630xA 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 UCM630xA can be used as an NTP server for the NTP clients to synchronize their time with. To configure the UCM630xA as the NTP server, set “Enable NTP server” to “Yes” under Web GUI🡪System Settings🡪Time Settings🡪NTP Server. On the client side, point the NTP server address to the UCM630xA IP address or host name to use the UCM630xA as the NTP server.

Office Time

On the UCM630xA, the system administrator can define “office time”, which can be used to configure time condition for extension call forwarding schedule and inbound rule schedule. To configure office time, go to Web GUI🡪System Settings🡪Time Settings🡪Office Time. Click on “Add” to create an office time.


Figure 48: Create New Office Time

Start TimeConfigure the start time for office hour.
End TimeConfigure the end time for office hour
WeekSelect the workdays in one week.
Show Advanced OptionsCheck this option to show advanced options. Once selected, please specify “Month” and “Day” below.
MonthSelect the months for office time.
DaySelect the workdays in one month.

Table 22: 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 49: Settings🡪Time Settings🡪Office Time
  • Click on

    to edit the office time.
  • Click on

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

Holiday

On the UCM630xA, the system administrator can define “holiday”, which can be used to configure time condition for extension call forwarding schedule and inbound rule schedule. To configure holiday, go to Web GUI🡪System Settings🡪Time Settings🡪Holiday. Click on “Add” to create holiday time.


Figure 50: Create New Holiday

Name

Specify the holiday name to identify this holiday.

Holiday Memo

Create a note for the holiday.

Month

Select the month for the holiday.

Year

Select the Year for the holiday.

Note: In the "Year" option, select "All" to set annual fixed holiday information.

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.

Time

Select the time on which the holiday starts. 

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 51: Settings🡪Time Settings🡪Holiday

  • Click on

    to edit the holiday.
  • Click on

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

Email Settings

Email Settings

The Email application on the UCM630xA can be used to send out alert event Emails, Voicemail (Voicemail-To-Email) etc. The configuration parameters can be accessed via Web GUI🡪System Settings🡪Email Settings🡪Email Settings.

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

 

  • MTA: Mail Transfer Agent. The Email will be sent from the configured domain. When MTA is selected, there is no need to set up SMTP server for it or no user login is required. However, the Emails sent from MTA might be considered as spam by the target SMTP server.
  • Client: Submit Emails to the SMTP server. A SMTP server is required, and users need login with correct credentials.
DomainSpecify the domain name to be used in the Email when using type “MTA”.
SMTP ServerSpecify the SMTP server when using type “Client”.
Enable SASL AuthenticationEnable SASL Authentication. When disabled, UCM will not try to use the username and password for mail client login authentication. Most of the mail server requires login authentication while some others private mail servers allow anonymous login which requires disabling this option to send Email as normal. For Exchange Server, please disable this option.
UsernameUsername is required when using type “Client”. Normally it is the Email address.
PasswordPassword to login for the above Username (Email address) is required when using type “Client”.
Enable Email-to-FaxMonitors the inbox of the configured email address for the specified subject. If enabled, the UCM will get a copy of the attachment from the email and send it to the XXX extension by fax. The attachment must be in PDF/TIF/TIFF format.
Email-to-Fax Blacklist/WhitelistThe user can enable the Email-to-Fax Blacklist or Email-to-Fax Whitelist.
Email-to-Fax Subject FormatSelect the email subject format to use for emails to fax. XXX refers to the extension that the fax will be sent to. This extension can only contain numbers.
Internal Black/WhitelistEmail address blacklist/whitelist for local extensions.
External Blacklist/WhitelistEmail address blacklist/whitelist for non-local contacts. Separate multiple addresses with semicolon (;) (i.e.”xxx;yyy”).
Fax Sending Success/Failure ConfirmationIf enabled, the UCM will send an email notification to the sender about the fax sending result.
POP/POP3 Server AddressConfigure the POP/POP3 server address for the configured username

 

Example: pop.gmail.com

POP/POP3 Server PortConfigure the POP/POP3 server port for the configured username

 

Example: 995

Display NameSpecify the display name in the FROM header in the Email.
SenderSpecify the sender’s Email address.

 

For example: pbx@example.mycompany.com.

Table 24: Email Settings

The following figure shows a sample Email setting on the UCM630xA, assuming the Email is using 192.168.6.202 as the SMTP server.

Figure 52: UCM630xA Email Settings

Once the configuration is finished, click on “Test”. In the prompt, fill in a valid Email address to send a test Email to verify the Email settings on the UCM630xA.


The Email templates on the UCM630xA can be used for email notification, the configuration parameters can be accessed via Web GUI🡪Settings🡪Email Settings🡪Email Templates.

Email Templates

Users can customize email templates for password reset, voicemail, meeting scheduling, extensions, fax,

meeting report, PMS, CDR, emergency call, missed calls, alert events, call queue statistics and etc.

• Click on

icon to edit the template.

Figure 53: Email Template
Note

The “Multimedia Meeting Schedule” template is improved. Click on “Edit” for this template to view the improved default template.

  • Added “Edge” and “Safari” as supported browser.
  • Added “Download Wave” button for user to download Wave app from: https://fw.gdms.cloud/wave/download/
  • Improved descriptions

Under UCM Web GUI🡪 System Settings🡪 Email Settings🡪 Email Footer Hyperlink, users could edit the text and URL to modify the email footer hyperlink.

Figure 54: Email Footer Hyperlink

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 55: Email Send Log

FieldDescription
Start TimeEnter the start time for filter
End TimeEnter the end time for filter
ReceiversEnter the email recipient, while searching for multiple recipients, please separate then with comma and no spaces.
Send ResultEnter the status of the send result to filter with
Return CodeEnter the email code to filter with
Email Send ModuleSelect the email module to filter with from the drop-down list, which contains:

 

  • All Modules
  • Extension
  • Voicemail
  • Meeting Schedule
  • User Password
  • Alert Events
  • CDR
  • Test
Table 25: Email Log – Display Filter

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

Figure 56: Email Logs

Below are the codes returned when sending emails and their description:

CodeDescription
250Mail sent successfully
501Address format parsing error, 501 will be returned when there are unacceptable characters in the recipient’s email address in MTA mode. Please check if the recipient’s email address format is correct. The “sender” configured on the client is your mail account.
535The user name and password verification in the client mode is incorrect. Please check whether the user name and password are configured correctly.
550Possible reasons:

 

1. The recipient’s mailbox user name does not exist or is in a banned state, please check whether the email recipient is the correct email address.

2. The number of destination addresses sent by the sender exceeds the maximum limit per day and is temporarily blacklisted. Please reduce the sending frequency or try again the next day.

3. The sender’s IP does not pass the SPF permission test of the sending domain. Emails sent in MTA mode may return this error code even if they are sent.

552The sent email is too large or the email attachment type is prohibited
553The sender and the email account are inconsistent, please configure the sender as your email account correctly.
554The email was identified as spam. Please reduce the sending frequency or try again the next day
noneIndicates that there is no return code.

 

If the sending result is “deferred”, the general reason is that the mail service area is configured incorrectly. Please check whether the server configuration is correct.

If the sending result is “bounced”, the general reason is that the receiving email address domain name is wrong, please check whether the email recipient is the correct email address. If it is in MTA mode, please check whether the “domain” is configured to be in the same domain name as the “recipient”.

Table 26: Email Codes

HA

Dual-system hot standby provides a highly reliable and fault-tolerant solution for enterprises using UCM6300 series/UCM6300A series. Based on two UCM devices of the same product model and software version, one of them is in the “Active” working state in real time, and the other is in the “Standby” working state. The daily data on the host server will be synchronized to the standby machine in real time, and the standby machine monitors the running status of the host at all times. When the host fails, including hardware failures and severe software failures, the standby machine will immediately take over the business and enter the “Active” working state, and Upgrade to a host to ensure that the business is not interrupted, and the call will automatically resume.

Before forming a paired HA dual-system hot backup, two UCM devices need to complete their respective network settings. The network mode can only be switching or routing, and the IP type can only be static.

HA settings

The users can configure the HA under System Settings 🡪 HA settings page.

Figure 57: HA Settings

ParameterDescription
High Available EnableEnables/disables the HA functionality. By default, is Disabled.
Force switchAfter clicking the button, the active/standby switch will be enforced.
HA Station TypeThe master and slave static configuration of the device, The real active / standby is decided dynamically by the active / standby.
HA Virtual IPTo carry the service, the main and standby computers should be set the same, and the intranet terminal should register and use the IP address.
HA Peer IPLocal IP address of HA peer device.
HA Peer MAC AddressNeed to specify this peer MAC address while using the UCM RemoteConnect service.
Heartbeat PortThe number of the heartbeat port should be consistent with the peer heartbeat port.
Heartbeat Timeout Period (s)If timeout occurs, services will be transferred over to the Slave UCM.
Software Fault SwitchEnable Software Fault Switch
Hardware Fault SwitchIf issues are detected with the selected connection interfaces, the backup UCM6510 will take over services after the master/slave handover. If not checked, UCM will send only a fault alarm.
Enable IPv6If enabled, HA on UCM can be used with IPv6 while compatible with IPv4.
Table 27: HA Settings parameters

HA Status

Once the HA is configured, the user can view its status under system settings 🡪 HA 🡪 HA Status as shown below

Figure 58: HA Status

HA Log

The user can view the HA log through the system settings 🡪 HA 🡪 HA log page. The HA log effectively records the execution results of past full backup actions, as well as the historical records that triggered the active/standby switchover.

SNMP

UCM63xx supports SNMP in case the system administrator chooses to use third party monitoring tools. These are the options available when setting up SNMP.

SNMP Settings

Figure 59: SNMP Settings

Enable

Tick this box to enable SNMP.

Device Name

Enter the device name.

Location

Enter the location.

Contact Email Address

Enter the email address used to send the SNMP alerts to.

Enable SNMP Trap Proxy

Tick this box to enable a proxy for SNMP Trap.

SNMP Trap Proxy Listening Port

The port number on which the SNMP Trap Proxy is listening on.

SNMP Community

You can also create SNMP communities and affect a certain level of access. An SNMP community is a group created to aggregate many management stations. The community name is used to authenticate and identify these machines in the NMS (Network Management System).

Figure 60: SNMP Community

Name

Enter a name for the community

Access Level

Select an access level:

  • Read Only: The SNMP community will be able only to read SNMP messages.

  • Read/Write: 

SNMP Trap Destinations

SNMP Traps is a very useful feature when there are many network components to manage. Instead of sending requests to all the machines in the network in order to view their SNMP logs risking slowing down or bringing the network to a complete halt, SNMP Traps can be configured so these machines can send unrequested messages to the manager to notify it about critical events and general failures.

Figure 61: SNMP Trap Destinations

Name

Enter a name of your SNMP Trap destination.

IP Address

Enter the SNMP Trap destination's IP address.

Port

Enter the port of the SNMP Trap destination.

Community

Select the community that you want 

Type

Select the type of SNMP:

  • Trapsink: Select this option if you want to send SNMP v1 traps. 

  • Trap2sink: Select this option if you want to send SNMP v2 traps. 

  • Informsink: Select this option if you want to send "Inform" notifications only.

SNMP Version 3

UCM 63xx also supports SNMP v3 in case the system administrator decides to add more security to the monitoring process. SNMP v3 is a very good solution to monitor devices that interface directly with Internet. SNMP v3 offers more security than its predecessors by hashing the authentication information, encrypting the SNMP messages exchanged between the managed devices and the network management system which prevent eavesdropping. Also, it prevents any data tampering which protects the integrity of the data exchanged.

Figure 62: SNMP v3

Name

Set the user's name

Authentication Protocol

Select the authentication protocol:

  • MD5

  • SHA

Authentication Password

Set the authentication password.

Privacy Protocol

Select the protocol to use to encrypt the data

  • DES

  • AES-128

  • AES-192

  • AES-256

Privacy Password

Set the privacy password.

Group Level

Set the group level:

  • Read Only.

  • Read/Write.

SNMP Trap Proxy

Figure 63: SNMP Trap Proxy

Name

Enter a name for the proxy server.

IP Address

Enter the proxy server's IP address.

Port

Enter the proxy server's port. 

TR-069

To configure TR-069 on Grandstream devices, set following parameters:

ParameterDescription
Enable TR-069Toggle it on to enable TR-069. It is enabled by default
ACS URLURL for TR-069 Auto Configuration Servers (ACS), e.g., http://myacs.grandstream.com
TR-069 UsernameACS username for TR-069, must be the same as in the ACS configuration.
TR-069 PasswordACS password for TR-069, must be the same as in the ACS configuration.
Periodic Inform EnableEnables periodic inform. If set to ‘Yes’, device will send inform packets to the ACS.
Periodic Inform IntervalPeriodic time when UCM630xA will send inform packets to TR-069 ACS server.

 

This option is specified in seconds.

ACS Connection Request UsernameThe username for the ACS to connect to UCM.
ACS Connection Request PasswordThe password for the ACS to connect to UCM.
Connection Request PortPort for incoming connection requests.

 

The default value is 7547.

CPE Cert FileThe Cert file for UCM to connect to the ACS via SSL.
CPE Cert KeyThe Cert key for UCM to connect to the ACS via SSL.

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 UCM630xA provides a Plug and Play mechanism to auto-provision the Grandstream SIP devices in a zero-configuration manner by generating XML config file and having the phone to download it within LAN area. This allows users to finish the installation with ease and start using the SIP devices in a managed way.

To provision a phone, three steps are involved, i.e., discovery, configuration, and provisioning. This section explains how Zero Config works on the UCM630xA. The settings for this feature can be accessed via Web GUI🡪Other 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 64: 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 UCM630xA 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 UCM630xA for auto provisioning. Three methods of auto provisioning are used.

Figure 65: UCM630xA Zero Config

  • SIP SUBSCRIBE

When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The UCM630xA 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 UCM630xA and take the new configuration.

  • DHCP OPTION 66

Route mode needs 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 UCM630xA 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 UCM630xA.

  • mDNS

When the phone boots up, it sends out mDNS query to get the TFTP server address. The UCM630xA will respond with its own address. The phone will then send TFTP request to download the XML config file from the UCM630xA.

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

Figure 66: Auto Provision Settings

Enable Zero ConfigEnable or disable the zero-config feature on the PBX. The default setting is enabled.
Enable Automatic Configuration AssignmentBy default, this is disabled. If disabled, when SIP device boots up, the UCM630xA 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 UCM630xA.

 

Note: When disabled, SIP devices can still be provisioned by manually sending NOTIFY from the UCM630xA which will include the XML config file URL for the SIP device to download.

Auto Assign ExtensionIf 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 SegmentClick 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 ExtensionIf 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 SegmentClick 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 WhitelistThis 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.

Table 28: Auto Provision Settings

Please make sure an extension is manually assigned to the phone or “Automatically Assign Extension” is enabled during provisioning. After the configuration on the UCM630xA Web GUI, click on “Save” and “Apply Changes”. Once the phone boots up and picks up the config file from the UCM630xA, it will take the configuration right away.

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🡪Other 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 UCM630xA. 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 UCM630xA Web GUI🡪Other Features🡪Zero Config🡪Auto Provisioning Settings.

Figure 67: 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 Web GUI) are displayed in the list.

Figure 68: 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 ”Import”, 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 69: 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.
    • Extension: Enter the extension number and press Search button.
Figure 70: 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

    to export CSV file containing list of devices. This file can be imported to another UCM to quickly set it up with the original UCM’s devices.
  • Click on

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

All these operations will be detailed on the next sections.

Global Configuration


Global configuration will apply to all the connected Grandstream SIP end point devices in the same LAN with the UCM630xA no matter what the Grandstream device model it is. It is divided into two levels:

  • Global Policy
  • Global Templates

Global Templates configuration has higher priority to Global Policy configuration.

Global Policy

Global Policy can be accessed in Web GUI🡪Other 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.

C:\Users\GSUser\Desktop\S1.JPG
Figure 71: Global Policy Categories

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

Language settings
LanguageSelect the LCD display language on the SIP end device.
Date and Time
Date FormatConfigure the date display format on the SIP end device’s LCD.
Time FormatConfigure the time display in 12-hour or 24-hour format on the SIP end device’s LCD.
Enable NTPTo enable the NTP service.
NTP ServerConfigure the URL or IP address of the NTP server. The SIP end device may obtain the date and time from the server.
NTP Update IntervalConfigure the NTP update interval.
Time ZoneConfigure the time zone used on the SIP end device.
Enable Daylight Saving TimeSelect either to enable or disable the DST.
Table 29: Global Policy Parameters – Localization

Default Call Settings
Dial PlanConfigure 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 FeaturesWhen 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 KeyIf set to “Yes”, pressing the number key “#” will immediately dial out the input digits.
Auto Answer by Call-infoIf 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 TraversalConfigure if NAT traversal mechanism is activated.
User Random PortIf set to “Yes”, this parameter will force random generation of both the local SIP and RTP ports.
General Settings
Call Progress TonesConfigure 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 ModeSelect “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 30: Global Policy Parameters – Phone Settings

LDAP Phonebook
SourceSelect “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 UCM630xA Web GUI🡪System Settings🡪LDAP Server will be applied.
AddressConfigure the IP address or DNS name of the LDAP server.
PortConfigure the LDAP server port. The default value is 389.
Base DNThis is the location in the directory where the search is requested to begin.

 

Example:

  • dc=grandstream, dc=com
  • ou=Boston, dc=grandstream, dc=com
UsernameConfigure the bind “Username” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.
PasswordConfigure the bind “Password” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.
Number FilterConfigure the filter used for number lookups. Please refer to user manual for more details.
Name FilterConfigure the filter used for name lookups. Please refer to user manual for more details.
VersionSelect the protocol version for the phone to send the bind requests. The default value is 3.
Name AttributeSpecify the “name” attributes of each record which are returned in the LDAP search result.

 

Example:

gn

cn sn description

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

 

Example:

telephoneNumber

telephoneNumber Mobile

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

 

Example:

%cn %sn %telephoneNumber

Max HitsSpecify the maximum number of results to be returned by the LDAP server. Valid range is 1 to 3000. The default value is 50.
Search TimeoutSpecify 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 ResultsSpecify whether the searching result is sorted or not. Default setting is No.
Incoming CallsConfigure to enable LDAP number searching when receiving calls. The default setting is No.
Outgoing CallsConfigure to enable LDAP number searching when making calls. The default setting is No.
Lookup Display NameConfigures 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 ServerSelect 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 IntervalConfigure 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 downloadIf set to “Yes”, when XML phonebook is downloaded, the entries added manually will be automatically removed.
Table 31: Global Policy Parameters – Contact List

Upgrade and Provision
Firmware SourceFirmware source via ZeroConfig provisioning could a URL for external server address, local UCM directory or USB media if plugged in to the UCM630xA. Select a source to get the firmware file:

 

  • URL

If select to use URL to upgrade, complete the configuration for the following four parameters: “Upgrade Via”, “Server Path”, “File Prefix” and “File Postfix”.

  • Local UCM Server

Firmware can be uploaded to the UCM630xA 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 UCM630xA 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 UCM630xA and the firmware file must be put under a folder named “ZC_firmware” in the USB storage root directory.

Upgrade viaWhen URL is selected as firmware source, configure upgrade via TFTP, HTTP or HTTPS.
Server PathWhen URL is selected as firmware source, configure the firmware upgrading server path.
File PrefixConfigure the Config Server Path.
Config Server PathWhen 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 43/66If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can be redirected.
Automatic UpgradeIf 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 RuleSpecify how firmware upgrading and provisioning request to be sent.
Zero ConfigSelect either to enable or disable zero config.
Web Access
Admin PasswordConfigure the administrator password for admin level login.
End-User PasswordConfigure the end-user password for the end user level login.
Web Access ModeSelect HTTP or HTTPS as the web access protocol.
Web Server PortConfigure the port for web access.

 

The valid range is 1 to 65535.

RTSP PortConfigure the RTSP Port.
Enable UPnP DiscoverySelect either to enable or disable Enable UPnP Discovery
Login SettingsConfigure the login settings.
User Login TimeoutConfigure User Login Timeout.
Maximum Consecutive Failed Login AttemptsConfigure Maximum Consecutive Failed Login Attempts.
Login Error Lock TimeConfigure Login Error Lock Time.
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 ServerConfigure the URL/IP address for the syslog server.
Syslog LevelSelect the level of logging for syslog.
Send SIP LogConfigure whether the SIP log will be included in the syslog message.
Table 32: Global Policy Parameters – Maintenance

Basic Settings
IP AddressConfigure how the SIP end device shall obtain the IP address. DHCP or PPPoE can be selected.

 

  • 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.

Host NameSpecifies the name of the client. This field is optional but may be required by Internet Service Providers.
Vendor Class IDUsed by clients and servers to exchange vendor class ID.
Account IDEnter the PPPoE account ID.
PasswordEnter the PPPoE Password.
Service NameEnter the PPPoE Service Name.
Advanced Setting
Layer 3 QoSDefine the Layer 3 QoS parameter. This value is used for IP Precedence, Diff-Serv or MPLS. Valid range is 0-63.
Layer 3 QoS For RTPAssign the priority value of the Layer 3 QoS for RTP packets.

 

Valid range is 0 -63.

Layer 3 QoS For SIPAssign the priority value of the Layer 3 QoS for SIP packets.

 

Valid range is 0 -63.

Layer 2 QoS TagAssign the VLAN Tag of the Layer 2 QoS packets.

 

Valid range is 0 -4095.

Layer 2 QoS Priority ValueAssign the priority value of the Layer 2 QoS packets.

 

Valid range is 0-7.

STUN ServerConfigure the IP address or Domain name of the STUN server. Only non-symmetric NAT routers work with STUN.
Keep AliveSelect either to enable or disable Keep Alive.
Keep Alive IntervalSpecify 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.
Register ExpirationSpecify the Register Expiration.
Local SIP PortConfigure Local SIP Port.
Local RTP PortConfigure Local RTP Port.
Auto On-Hook Timer(s)Configure Auto On-Hook Timer(s).
Ring TimeoutConfigure Ring Timeout.
SIP TransportSelect either UDP, TCP or TLS/TCP as SIP transport protocol.
Direct IP CallSelect either to disable or enable Direct IP Call support.
SIP Proxy Compatibility ModeSelect either to disable or enable SIP Proxy Compatibility Mode.
Unregister On RebootSelect either to disable or enable Unregister On Reboot.
Whitelist
WhitelistSelect either to enable or disable Whitelist
SIP Phone Number WhitelistConfigure the SIP Phone Number Whitelist.
Table 33: Global Policy Parameters – Network Settings

Wallpaper
Screen Resolution 1024 x 600Check this option if the SIP end device shall use 1024 x 600 resolution for the LCD screen 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 UCM630xA.

Screen Resolution 800 x 400Check this option if the SIP end device shall use 800 x 400 resolution for the LCD screen 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 UCM630xA.

Screen Resolution 480 x 272Check this option if the SIP end device shall use 480 x 272 resolution for the LCD screen 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 UCM630xA.

Screen Resolution 320 x 240Check this option if the SIP end device supports 320 x 240 resolution for the LCD screen 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 UCM630xA.

Table 34: Global Policy Parameters – Customization

Email Settings
SMTP SettingsCheck 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 1

Email address where alarms notifications will be sent

  • Enable SSL

Enable SSL protocol for SMTP

FTP
FTPCheck 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

Table 35: Global Policy Parameters – Communication Settings

Global Templates

Global Templates can be accessed in Web GUI🡪Other 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 “Add” to add a global template. Users will see the following configurations.

Template NameCreate a name to identify this global template.
Description Provide a description for the global template. This is optional.
ActiveCheck this option to enable the global template.
Table 36: Create New 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 under the global template.


Figure 72: 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 delete 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🡪Other Features🡪Zero Config🡪Global Templates page. Users can click on

    to delete the global template or 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 Other 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 UCM630xA.

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 “Add” to add a model template.
ModelSelect a model to apply this template. The supported Grandstream models are listed in the dropdown list for selection.
Template NameCreate a name for the model template.
DescriptionEnter a description for the model template. This is optional.
Default Model TemplateSelect to assign this model template as the default model template. The value of the option in default model template will be overridden if other selected model template has a different value for the same option.
ActiveCheck this option to enable the model template.
Table 37: Create New 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 to be modified under 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://www.grandstream.com/support/tools


Figure 73: Edit Model Template

  • Click on Save when done. The model template will be displayed on Web GUI🡪Other 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.
  • Click the “Copy Template” button to copy the configuration items of the selected model template to another template, thereby reducing template editing work. Note: The model template only supports copying between devices of the same model.
  • Click the “Import/Export” button to upload/export the model template list in .CSV format.

Model Update

UCM630xA 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 74: OEM Models

Model Template Package List

Templates for most of the Grandstream models are built in with the UCM630xA already. Templates for Grandstream Wave and Grandstream surveillance products require users to download and install under Web GUI🡪Other Features🡪Zero Config🡪Model Update first before they are available in the UCM630xA for selection. After downloading and installing the model template to the UCM630xA, 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 UCM630xA.
Figure 75: Template Management

Upload Model Template Package

In case the UCM630xA 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 package can be manually uploaded from local device through Web GUI. Please contact Grandstream customer support if the model package is needed for manual uploading.

Figure 76: Upload Model Template Manually

Device Configuration

On Web GUI, page Other 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 UCM630xA. 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 “Add” 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 “save” to save the configuration for this device.
Figure 77: Create New Device

Manage Devices

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

Figure 78: 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 79: 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🡪 Other Features🡪Zero Config🡪Global Policy will be applied here. Clicking on “Modify Global Policy” to redirect to page Other 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🡪Other Features🡪Zero Config🡪Model Templates page. Please see default model template option in [Table 37: Create New Model Template].

  1. Model Templates

Select a model template to be used for the device and click on

to add. Multiple model 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 80: 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 https://content.grandstream.com/hubfs/Grandstream_Feb_2021/Zip%20File/config-template.zip?hsLang=en

  • Select multiple devices that need to be modified and then click on ”Update Config” 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 81: 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 82: Modify Selected Devices – Different Models

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🡪Other 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 83: Device List in Zero Config

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

Sample Application

Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1 GXV3275 phone used by customer support supervisor. 3 of the 8 customer support members speak Spanish and the rest speak English. We could deploy the following configurations to provisioning the office phones for the customer support team.

  1. Go to Web GUI🡪Other Features🡪Zero Config🡪Zero Config Settings, select “Enable Zero Config”.
  2. Go to Web GUI🡪Other Features🡪Zero Config🡪Global Policy, configure Date Format, Time Format and Firmware Source as follows.
Figure 84: Zero Config Sample – Global Policy

  1. Go to Web GUI🡪Other Features🡪Zero Config🡪Model Templates, create a new model template “English Support Template” for GXP2170. 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🡪Other Features🡪Zero Config🡪Model Templates, create another model template “Spanish Support Template” for GXP2170. 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🡪Other Features🡪Zero Config🡪Zero Config.
  4. On Web GUI🡪Other 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 85: 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 86: 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 GXV3275 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 GXV3275.
Figure 87: Zero Config Sample – Device Preview 3

  1. Click on “Apply Changes” to apply saved changes.
  2. On the Web GUI🡪Other Features🡪Zero Config🡪Zero Config page, click on

    to send NOTIFY to trigger the device to download config file from UCM630xA.

Now all the 9 phones in the network will be provisioned with a unique extension registered on the UCM630xA. 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 GXV3275 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 manually create new SIP user, go to Web GUI🡪Extension/Trunk🡪Extensions. Click on “Add” and a new window will show for users to fill in the extension information.

Figure 88: 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.

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.

Privilege

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 purposes.

Auth ID

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

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 the 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 voicemail password (digits only) for the user to access the voicemail box. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purposes.

Skip Voicemail Password Verification

When a user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.

Send Voicemail Email Notification

Configures whether to send emails to the extension's email address to notify of a new voicemail.

Attach Voicemail to Email

Configures whether to attach a voicemail audio file to the voicemail notification emails.

Note: When set to “Default”, the global settings in Call Features 🡪 Voicemail 🡪 Voicemail Email Settings will be used.

Keep Voicemail after Emailing

Whether to keep the local voicemail recording after sending them. If set to “Default”, the global settings will be used.

Note: When set to “Default”, the global settings in Call Features 🡪 Voicemail 🡪 Voicemail Email Settings will be used.

Enable Keep-alive

If enabled, an empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is "No".

Keep-alive Frequency

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

Enable SCA

If enabled, (1) Call Forward, Call Waiting, and Do Not Disturb settings will not work, (2) Concurrent Registrations can be set only to 1, and (3) Private numbers can be added in Call Features🡪SCA page.

Emergency CID Name

CallerID name that will be used for emergency calls and callbacks.

Disable This Extension

If selected, this extension will be disabled on the UCM630X.

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

Sync Contact

If enabled, this extension number will be displayed in the Wave contact, otherwise, it will not be displayed, and it cannot be found in the chat, but the user can still dial this number.

User Settings

First Name

Configure the first name of the user. The first name can contain characters, letters, digits, and _.

Last Name

Configure the last name of the user. The last name can contain characters, letters, digits, and _.

Email Address

Fill in the Email address for the user. Voicemail will be sent to this Email address.

User Password

Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purposes.

Language

Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt language under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. The dropdown list shows all the currently available voice prompt languages on the UCM630X. To add more languages to the list, please download the voice prompt package by selecting "Check Prompt List" under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Concurrent Registrations

The maximum endpoints which can be registered into this extension. For security concerns, the default value is 1.

Mobile Phone Number

Configure the phone number for the extension, user can type the related star code for the phone number followed by the extension number to directly call this number.

For example, the user can type *881000 to call the mobile number associated with extension 1000.

Department

Configure the user's department. The department can be configured in User Management->Address Book Management->Department Management.
Job Title: The user's department position.

Contact Privileges

Same as Department Contact Privileges

When enabled, The extension will inherit the same privilege attributed to the department it belongs to.

Contact View Privileges

SIP Settings

NAT

Use NAT when the UCM630X is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is a one-way audio issue, usually it is related to NAT configuration or the Firewall's support of SIP and RTP ports. The default setting is enabled.

Enable Direct Media

By default, the UCM630X 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 UCM630X 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, the SIP INFO message will be used. If "Inband" is selected, a-law or u-law 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”. The “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

When present in 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.

Jitter Buffer

Select the jitter buffer method.

  • Disable: Jitter buffer will not be used.

  • Fixed: Jitter buffer with a fixed size (equal to the value of "jitter buffer size")

  • Adaptive: Jitter buffer with an adaptive size (no more than the value of "max jitter buffer").

  • NetEQ: Dynamic jitter buffer via NetEQ.

Packet Loss Retransmission

Configure to enable Packet Loss Retransmission.

  • NACK

  • NACK+RTX(SSRC-GROUP)

  • OFF

Video FEC

Check to enable Forward Error Correction (FEC) for Video.

Audio FEC

Check to enable Forward Error Correction (FEC) for Audio.

FECC

Configure to enable Remote Camera Management.

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: Only IP addresses in the configured network segments can register to this extension. Press “Add Local Network Address” to add more IP segments.

SRTP Crypto Suite

The following encryption protocols can be used to encrypt an RTP stream.

  • AES_CM_128_HMAC_SHA1_80 (This is the default used protocol)

  • AES_256_CM_HMAC_SHA1_80

  • AEAD_AES_128_GCM

  • AEAD_AES_256_GCM

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.

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”. More details at [PRESENCE].

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 the dropdown list as CFU target.

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

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

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

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

  • Voicemail Group”: Select a voicemail group from the 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 ‘All’, ‘Office Time’, ‘Out of Office Time’, ‘Holiday’, ‘Out of Holiday’, ‘Out of Office Time or Holiday’, ‘Office Time and Out of Holiday’, ‘Specific Time’, ‘Out of Specific Time’, ‘Out of Specific Time or Holiday’, ‘Specific Time and Out of Holiday’.

Notes:

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

  • Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

  • Office Time and Holiday could be configured on page 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 the dropdown list as CFN target.

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

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

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

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

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

The default setting is “None”.

CFN Time Condition

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

Notes:

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

  • Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

  • 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 the dropdown list as CFB target.

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

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

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

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

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

The default setting is “None”.

CFB Time Condition

Select time condition for Call Forward Busy. The available time conditions ‘All’, ‘Office Time’, ‘Out of Office Time’, ‘Holiday’, ‘Out of Holiday’, ‘Out of Office Time or Holiday’, ‘Office Time and Out of Holiday’, ‘Specific Time’, ‘Out of Specific Time’, ‘Out of Specific Time or Holiday’, ‘Specific Time and Out of Holiday’.

Notes: 

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

  • Specific time can be configured under the Specific Time section. Scroll down the add Time Condition for a specific time.

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

Do Not Disturb

If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.

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 under the Specific Time section. Scroll down the add Time Condition for a 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, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.

  • Z match any digit from 1-9.

  • N match any digit from 2-9.

  • X match any digit from 0-9.

FWD Whitelist

Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported.

  • match any digit from 1-9.

  • N match any digit from 2-9.

  • X match any digit from 0-9.

CC Settings

Enable CC

If enabled, UCM630X 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 an ordinary extension.

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

The default setting is “Normal”.

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 that 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.

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 the caller ID number.

External Number

Set the external number to ring simultaneously. ‘-’ is the connection character that 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 RS

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

Monitor privilege control

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

PMS Remote Wakeup Whitelist

Select the extensions that can set wakeup service for other extensions

Selected extensions can set a PMS wakeup service for this extension via feature code.

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 UCM630X. 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 recordings 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 the password when making outbound calls.

Dial Trunk Password

Configure personal password when making outbound calls via the trunk.

Support Hot-Desking Mode

Check to enable Hot-Desking Mode on the extension. Hot-Desking allows using the same endpoint device and logs in using extension/password combination. This feature is used in scenarios where different users need to use the same endpoint device during a different time of the day for instance. 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 the LDAP Phonebook PBX list.
Default is enabled.

Use MOH as IVR ringback tone

If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as a ringback tone instead of the regular ringback tone.

Music On Hold

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

Call Duration Limit

Check to enable and set the call limit the duration.

Maximum Call Duration (s)

The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds

The Maximum Number of Call Lines

The maximum number of simultaneous calls that the extension can have.
0 indicates no limit.

Outgoing Call Frequency Limit

If enabled, if the number of outbound calls exceed the configured threshold within the specified period, further outbound calls will be not be allowed.

Period (m)

The period of outgoing call frequency limit. The valid range is from 1 to 120. The default value is 1.

Max Number of Calls

Set the maximum number of outgoing calls in a period. The valide tange is from 1 to 20. The default value is 5.

Enable Auto-Answer Support

If enabled, the extension will support auto-answer when indicated by Call-info/Alert-info headers.

Call Waiting

Allows calls to the extension even when it is already in a call. This only works if the caller is directly dialing the extension. If disabled, the CC service will take effect only for unanswered and timeout calls.

Stop Ringing

If enabled, when the extension has concurrent registrations on multiple devices, upon incoming call or meeting invite ringing, if one end device rejects the call, the rest of the devices will also stop ringing. By default, it’s disabled.

Email Missed Call Log

If enabled, the log of missed calls will be sent to the extension’s configured email address.

Missed Call Type

If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:

  • Default: All missed calls will be sent in email notifications.

  • Missed Internal Call: Only missed local extension-to-extension calls will be sent in email notifications.

  • Missed External Call: Only missed calls from trunks will be sent in email notifications.

Specific Time
Time ConditionClick to add Time Condition to configure specific time for this extension.
Table 41: SIP Extension Configuration Parameters🡪Specific Time


Table 42:

Follow Me
EnableConfigure to enable or disable Follow Me for this user.
Skip Trunk AuthIf the outbound calls need to check the password, we should enable this option or enable the option “Skip Trunk Auth” of the Extension. Otherwise this Follow Me cannot call out.
Music On Hold ClassConfigure the Music On Hold class that the caller would hear while tracking the user.
Confirm When AnsweringIf enabled, call will need to be confirmed after answering.
Enable DestinationConfigure to enable destination
Default DestinationThe call will be routed to this destination if no one in the Follow Me answers the call.
Use Callee DOD for Follow MeUse the callee DOD number as CID if configured Follow Me numbers are external numbers.
Play Follow Me PromptIf enabled, the Follow Me prompt tone will be played
New Follow Me NumberAdd a new Follow Me number which could be a “Local Extension” or an “External Number”. The selected dial plan should have permissions to dial the defined external number.
Dialing OrderThis is the order in which the Follow Me destinations will be dialed to reach the user.
Table 42: SIP Extension Configuration Parameters🡪Follow Me

Create New IAX Extension

The UCM630xA 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.

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.

Privilege

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 purposes.

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 voicemail password (digits only) for the user to access the voicemail box. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purposes.

Skip Voicemail Password Verification

When a user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.

Send Voicemail Email Notification

Configures whether to send emails to the extension’s email address to notify of a new voicemail.

Attach Voicemail to Email

Configures whether to attach a voicemail audio file to the voicemail notification emails.

Keep Voicemail after Emailing

Only applies if extension-level or global Send Voicemail to Email is enabled.

Disable This Extension

If selected, this extension will be disabled on the UCM630X.

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

User Settings

First Name

Configure the first name of the user. The first name can contain characters, letters, digits, and _.

Last Name

Configure the last name of the user. The last name can contain characters, letters, digits, and _.

Email Address

Fill in the Email address for the user. Voicemail will be sent to this Email address.

User Password

Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purposes.

Language

Select the voice prompt language to be used for this extension. The default setting is “Default” which is the selected voice prompt language under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. The dropdown list shows all the currently available voice prompt languages on the UCM630X. To add more languages to the list, please download the voice prompt package by selecting “Check Prompt List” under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

Mobile Phone Number

Configure the Mobile number of the user.

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.

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: 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.

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”.

Note:

  • “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 a specific time.

  • Office Time and Holiday could be configured on page 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 on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a specific time.

  • 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 a specific time.

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

Do Not Disturb

If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.

DND Time Condition

The time condition of DND. The DND will take effect while the time condition is satisfied.

DND Whitelist

If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.

  • Z match any digit from 1-9.

  • N match any digit from 2-9.

  • X match any digit from 0-9.

FWD Whitelist

Calls from users in the forward whitelist will not be forwarded.

Pattern matching is supported.

  • Z match any digit from 1-9.

  • N match any digit from 2-9.

  • X match any digit from 0-9.

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 the caller ID number.

External Number

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

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 RS

Use the callee’s DOD number as CallerID on Outgoing Forwarding or Ring Simultaneously calls.

Monitor privilege control

Allow call-barging

Members of the list can spy on this extension via feature codes.

Seamless transfer privilege control

Allowed to seamless transfer

Members of the list can seamlessly transfer via feature code.

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 UCM630X, 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 endpoint 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 the password when making outbound calls.

Dial Trunk Password

Configure personal password when making outbound calls via the trunk.

Enable LDAP

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

Music On Hold

Configure the Music On Hold class to suggest to the bridged channel when putting them on hold.

Use MOH as IVR ringback tone

If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as a ringback tone instead of the regular ringback tone.

Call Duration Limit

Check to enable and set the call limit the duration.

Maximum Call Duration (s)

The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds

Email Missed Calls

Send a log of missed calls to the extension’s email address.

Missed Call Type

If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:

  • Default: All missed calls will be sent in email notifications.

  • Missed Internal Call: Only missed local extension-to-extension calls will be sent in email notifications.

  • Missed External Call: Only missed calls from trunks will be sent in email notifications.

Specific Time

Time Condition

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

Follow Me

Enable

Configure to enable or disable Follow Me for this user.

Skip Trunk Auth

If the outbound calls need to check the password, we should enable this option or enable the option “Skip Trunk Auth” of the Extension. Otherwise, this Follow Me cannot call out.

Music On Hold Class

Configure the Music On Hold class that the caller would hear while tracking the user.

Confirm When Answering

If enabled, call will need to be confirmed after answering.

Enable Destination

Configure to enable destination.

Default Destination

The call will be routed to this destination if no one in the Follow Me answers the call.

Use Callee DOD for Follow Me

Use the callee DOD number as CID if configured Follow Me numbers are external numbers.

Play Follow Me Prompt

If enabled, the Follow Me prompt tone will be played.

New Follow Me Number

Add a new Follow Me number which could be a “Local Extension” or an “External Number”. The selected dial plan should have permissions to dial the defined external number.

Dialing Order

This is the order in which the Follow Me destinations will be dialed to reach the user.

Create New FXS Extension

The UCM630xA supports Foreign eXchange Subscriber (FXS) interface. FXS is used when user needs to connect analog phone lines or FAX machines to the UCM630xA.

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.

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.
 

Privilege

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 voicemail password (digits only) for the user to access the voicemail box. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purposes.

Skip Voicemail Password Verification

When a user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.

Send Voicemail Email Notification

Configures whether to send emails to the extension’s email address to notify of a new voicemail.

Attach Voicemail to Email

Configures whether to attach a voicemail audio file to the voicemail notification emails.

Keep Voicemail after Emailing

Only applies if extension-level or global Send Voicemail to Email is enabled.

Emergency CID Name

CallerID name that will be used for emergency calls and callbacks.

Disable This Extension

If selected, this extension will be disabled on the UCM630X.

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

User Settings

First Name

Configure the first name of the user. The first name can contain characters, letters, digits, and _.

Last Name

Configure the last name of the user. The last name can contain characters, letters, digits, and _.

Email Address

Fill in the Email address for the user. Voicemail will be sent to this Email address.

User Password

Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purposes.

Mobile Phone Number

Configure the Mobile number of the user.

Language

Select the voice prompt language to be used for this extension. The default setting is “Default” which is the selected voice prompt language under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings. The dropdown list shows all the currently available voice prompt languages on the UCM630X. To add more languages to the list, please download the voice prompt package by selecting “Check Prompt List” under Web GUI🡪PBX Settings🡪Voice Prompt🡪Language Settings.

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 SNED 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 the 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 the 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 the 3-way calling feature on the user. The default setting is enabled.

Send CallerID After

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

Fax Mode

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

  • None: Disable Fax.

  • Fax Gateway: If selected, the UCM630X can support the conversation and processing of Fax data from T.30 to T.38 or T.38 to T.30. only for FXS ports.

  • Fax Detection: During a call, the fax signal from the user/trunk will be detected, and the received fax will be sent to the email address configured for the user. If an email address has been configured for the user, the fax will be sent to the Default Email Address configured in Fax/T.38->Fax Settings.

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”.

Note:

  • “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 a specific time.

  • Office Time and Holiday could be configured on page 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 on the bottom of the extension configuration dialog. Scroll down the add Time Condition for a specific time.

  • 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 a specific time.

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

Do Not Disturb

If Do Not Disturb is enabled, all incoming calls will be dropped.

All call forward settings will be ignored.

DND Time Condition

The time condition of DND. The DND will take effect while the time condition is satisfied.

DND Whitelist

If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.

  • Z match any digit from 1-9.

  • N match any digit from 2-9.

  • X match any digit from 0-9.

FWD Whitelist

Calls from users in the forward whitelist will not be forwarded.

Pattern matching is supported.

  • Z match any digit from 1-9.

  • N match any digit from 2-9.

  • X match any digit from 0-9.

CC Settings

Enable CC

If enabled, UCM630X will automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason.

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 the caller ID number.

External Number

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

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 RS

Use the callee’s DOD number as CallerID on Outgoing Forwarding or Ring Simultaneously calls.

Hotline

Enable Hotline

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

Hotline Number

Configure the Hotline Number

Hotline Type

Configure the Hotline Type:

  • Immediate Hotline: When the phone is off-hook, UCM630X will immediately dial the preset number

  • Delay Hotline: When the phone is off hook, if there is no dialing within 5 seconds, UCM630X will dial the preset number.

Monitor privilege control

Members of the list can spy on this extension via feature codes.

Seamless transfer privilege control

Allowed to seamless transfer

Members of the list can seamlessly transfer via feature code.

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 UCM630X, 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 endpoint 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 a password when making outbound calls.

Dial Trunk Password

Configure personal password when making outbound calls via the trunk.

Enable LDAP

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

Use MOH as IVR ringback tone

If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as a ringback tone instead of the regular ringback tone.

Music On Hold

Select which Music On Hold class to suggest to the extension when putting the active call on hold.

Call Duration Limit

Check to enable and set the call limit the duration.

Maximum Call Duration (s)

The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds

Email Missed Calls

Send a log of missed calls to the extension’s email address.

Missed Call Type

If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:

  • Default: All missed calls will be sent in email notifications.

  • Missed Internal Call: Only missed local extension-to-extension calls will be sent in email notifications.

  • Missed External Call: Only missed calls from trunks will be sent in email notifications.

Specific Time

Time Condition

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

Follow Me

Enable

Configure to enable or disable Follow Me for this user.

Skip Trunk Auth

If the outbound calls need to check the password, we should enable this option or enable the option "Skip Trunk Auth" of the Extension. Otherwise, this Follow Me cannot call out.

Music On Hold Class

Configure the Music On Hold class that the caller would hear while tracking the user.

Confirm When Answering

If enabled, call will need to be confirmed after answering.

Enable Destination

Configure to enable destination.

Default Destination

The call will be routed to this destination if no one in the Follow Me answers the call.

Use Callee DOD for Follow Me

Use the callee DOD number as CID if configured Follow Me numbers are external numbers.

Play Follow Me Prompt

If enabled, the Follow Me prompt tone will be played.

New Follow Me Number

Add a new Follow Me number which could be a "Local Extension" or an "External Number". The selected dial plan should have permissions to dial the defined external number.

Dialing Order

This is the order in which the Follow Me destinations will be dialed to reach the user.

Batch Add Extensions

Batch Add SIP Extensions


To add multiple SIP extensions, BATCH add can be used to create standardized SIP extension accounts. However, unique extension username cannot be set using BATCH add.

Under Web GUI🡪Extension/Trunk🡪Extensions, click on “Add” and select extension type as SIP extension, and “Select Add Method” as Batch.

General
Create NumberSpecify the number of extensions to be added. The default setting is 5.
Extension IncrementationSelect how much to increment successive extensions. For example, if the value is 2, the extensions will be 1000,1002,1004,…… Note: Up to 3 characters.
ExtensionConfigure the starting extension number of the batch of extensions to be added.
PermissionAssign 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.

VoicemailConfigure 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).
SIP/IAX PasswordConfigure 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 PasswordConfigure Voicemail password (digits only) for the users.

 

  • User Random Password.

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

  • Use Extension as Password.

Enter a password to be used on all the extensions in the batch.

Send Voicemail to EmailSend voicemail messages to the configured email address. If set to “Default”, the global setting will be used. Global settings can be found in Voicemail->Voicemail Email Settings.
Keep Voicemail after EmailingOnly applies if extension-level or global Send Voicemail to Email is enabled.
CallerID NumberConfigure 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.
Skip Voicemail Password VerificationWhen user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.
Enable Keep-aliveIf enabled, the PBX will regularly send SIP OPTIONS to check if host device is online.
Keep-alive FrequencyConfigure the keep-alive interval (in seconds) to check if the host is up.
Disable This ExtensionCheck this box to disable this extension.
Enable SCAIf enabled, (1) Call Forward, Call Waiting and Do Not Disturb settings will not work, (2) Concurrent Registrations can be set only to 1, and (3) Private numbers can be added in Call Features->SCA page.
Emergency Calls CIDCallerID number that will be used when calling out and receiving direct callbacks.
Enable WaveIf enabled, this extension number can register, log in and use Wave normally, otherwise it will not be able to use Wave, but the phone function will still be retained.
Sync ContactIf enabled, this extension number will be displayed in the Wave contact, otherwise it will not be displayed, and it cannot be found in the chat, but the user can still dial this number.
LanguageSelect voice prompt language for this extension. If set to “Default”, the global setting for voice prompt language will be used.
Media
NATUse 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 MediaBy 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 ModeSelect 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, a-law or u-law are required. When “Auto” is selected, RFC4733 will be used if offered, otherwise “Inband” will be used.
Alert-infoWhen present in an INVITE request, the Alert-info header field specifies an alternative ring tone to the UAS.
SRTPEnable/disable SRTP for RTP stream encryption.
Packet Loss RetransmissionConfigure to enable Packet Loss Retransmission.

 

  • NACK
  • NACK+RTX(SSRC-GROUP)
  • OFF
Video FECCheck to enable Forward Error Correction (FEC) for Video.
FECCConfigure to enable FECC
Audio FECCheck to enable Forward Error Correction (FEC) for Audio.
ACL PolicyAccess Control List manages the IP addresses that can register to this extension.

 

  • Allow All: Any IP address can register to this extension.
  • Local Network: Only IP addresses in the configured network segments can register to this extension. Press “Add Local Network Address” to add more IP segments.
Jitter BufferSelect jitter buffer method.

 

  • Disable: Jitter buffer will not be used.
  • Fixed: Jitter buffer with a fixed size (equal to the value of “jitter buffer size”)
  • Adaptive: Jitter buffer with an adaptive size (no more than the value of “max jitter buffer”).
  • NetEQ: Dynamic jitter buffer via NetEQ.
Codec PreferenceConfigure the codecs to be used.
Call Transfer
Presence StatusSelect 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”. More details at [PRESENCE].
Call Forward UnconditionalEnable 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 ConditionSelect 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”.

 

Note:

  • “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.
Call Forward No AnswerConfigure 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.

The default setting is “None”.

CFN Time ConditionSelect 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 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.
Call Forward BusyConfigure 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.

The default setting is “None”.

CFB Time ConditionSelect 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 DisturbIf Do Not Disturb is enabled, all incoming calls will be dropped.

 

All call forward settings will be ignored.

DND Time ConditionSelect 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 WhitelistIf DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.

 

  • Z match any digit from 1-9
  • N match any digit from 2-9
  • X match any digit from 0-9.
FWD WhitelistCalls from users in the forward whitelist will not be forwarded. Pattern matching is supported.

 

  • Z match any digit from 1-9
  • N match any digit from 2-9
  • X match any digit from 0-9.
CC Settings
Enable CCIf enabled, UCM630xA 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 ModeTwo 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”.

CC Max AgentsConfigure 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 MonitorsConfigure 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.
Ring Simultaneously
Ring SimultaneouslyEnable 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 NumberSet 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 SimultaneouslyRing the external number simultaneously along with the extension on the basis of this time condition.
Use callee DOD on FWD or RSUse the DOD number when calls are being diverted/forwarded to external destinations or when ring simultaneous is configured.
Monitor privilege control
Allowed to

 

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 transferAny 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 TimeoutConfigure 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 UCM630xA, 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 3 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 RecordEnable automatic recording for the calls using this extension. The default setting is disabled. The recordings 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 AuthIf ‘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 PasswordConfigure personal password when making outbound calls via trunk.
Enable LDAPIf enabled, the extension will be added to LDAP Phonebook PBX list.
Bind PMS RoomIf enabled, the system will create a room whose room number, by default, will equal the extension number in PMS module. Note: If this room already exists, the configuration of the existing room will be overwritten.
Music On HoldSpecify which Music On Hold class to suggest to the bridged channel when putting them on hold.
Call Duration LimitThe maximum duration of call-blocking.
Maximum Call DurationThe maximum call duration (in seconds). The default value 0 means no limit.
Call WaitingIf disabled, UCM will not invite the extension when it is already in a call and will do the same work as the user is busy.

 

Note: the option only works when the caller dials the extension directly.

Table 53: Batch Add SIP Extension Parameters

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.

General
Create NumberSpecify the number of extensions to be added. The default setting is 5.
Extension IncrementationSelect how much to increment successive extensions. For example, if the value is 2, the extensions will be 1000,1002,1004,……
ExtensionThe extension number associated with this particular user/phone.
PermissionAssign 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.

CallerID NumberConfigure the Caller ID number displayed when dialing calls from this user. Note: The Caller ID usage might be limited by your VoIP provider. In Batch Add Method, “e” means to use the extension as the number.
VoicemailConfigure 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 PasswordConfigure 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 PasswordConfigure Voicemail password (digits only) for the users.

 

  • User Random Password.

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

  • Use Extension as Password.
  • Enter a password to be used on all the extensions in the batch.
Send Voicemail to EmailSend voicemail messages to the configured email address. If set to “Default”, the global setting will be used. Global settings can be found in Voicemail->Voicemail Email Settings.
Keep Voicemail after EmailingOnly applies if extension-level or global Send Voicemail to Email is enabled.
Auto RecordEnable 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 VerificationWhen user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.
Disable This ExtensionCheck this box to disable this extension.
LanguageSelect voice prompt language for this extension. If set to “Default”, the global setting for voice prompt language will be used.
IAX Settings
Max Number of CallsConfigure the maximum number of calls allowed for each remote IP address.
Require Call TokenConfigure 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”.

SRTPEnable/disable SRTP for RTP stream encryption.
ACL PolicyAccess Control List manages the IP addresses that can register to this extension.

 

  • Allow All: Any IP address can register to this extension.
  • Local Network: Only IP addresses in the configured network segments can register to this extension.
Codec PreferenceConfigure the codecs to be used.
Call Transfer
Call Forward UnconditionalEnable and configure the Call Forward Unconditional target number.
CFU Time ConditionSelect 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”.

 

Note:

  • “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.
Call Forward No AnswerConfigure the Call Forward No Answer target number.
CFN Time ConditionSelect 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 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.
Call Forward BusyConfigure the Call Forward Busy target number.
CFB Time ConditionSelect 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 DisturbIf Do Not Disturb is enabled, all incoming calls will be dropped.

 

All call forward settings will be ignored.

DND Time ConditionSelect 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 WhitelistIf DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.

 

  • Z match any digit from 1-9,
  • N match any digit from 2-9,
  • X match any digit from 0-9.
FWD WhitelistCalls from users in the forward whitelist will not be forwarded. Pattern matching is supported.

 

  • Z match any digit from 1-9,
  • N match any digit from 2-9,
  • X match any digit from 0-9.
Ring Simultaneously
Ring SimultaneouslyEnable 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 NumberSet 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 SimultaneouslyRing the external number simultaneously along with the extension on the basis of this time condition.
Use callee DOD on FWD or RSUse the DOD number when calls are being diverted/forwarded to external destinations or when ring simultaneous is configured.
Monitor privilege control
Allowed to

 

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 transferAny 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 TimeoutConfigure 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 UCM630xA, 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 RecordEnable automatic recording for the calls using this extension. The default setting is disabled. The recordings 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 AuthIf ‘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 PasswordConfigure personal password when making outbound calls via trunk.
Enable LDAPIf enabled, the extension will be added to LDAP Phonebook PBX list.
Music On HoldSpecify which Music On Hold class to suggest to the bridged channel when putting them on hold.
Call Duration LimitCheck to enable and set the call limit the duration.
Maximum Call Duration (s)The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds
Table 54: Batch Add IAX Extension Parameters

Batch Extension Resetting Functionality

Users can select multiple extensions and reset their settings to default by pressing the reset button

and confirm the reset functionality. Once done, all settings in Basic Settings page will be restored to default values with the exception of Concurrent Registrations. User voicemail password will be reset to Random Password. User voicemail prompts and recordings will be deleted. User Management settings will also be restored to default with the exception of usernames and custom privileges

Search and Edit Extension

All the UCM630xA 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” or “Delete”. Also, options “Edit”

, “Reboot”

and “Delete”

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.

Note

This is the expected behavior when you reset an extension:

  • All the data and configuration on the user side will be deleted. That includes user information, call history, call recordings, faxes, voice mails, meeting schedules and recordings, as well as chat history. However, the data related to the user will be kept on the UCM side.
  • The extension will be removed from group chats and the messages sent previously by the extension will be kept. However, only other users can search through those messages while the new user of the extension cannot.
  • If the extension was in a meeting schedule, the meeting will still be present. The extension will be removed from the meeting and will not be notified about the meeting.

  • Reboot the user

Click on

to send NOTIFY reboot event to the device which has an UCM630xA extension already registered. To successfully reboot the user, “Zero Config” needs to be enabled on the UCM630xA Web GUI🡪 Other Features🡪Zero Config🡪Zero Config Settings.

  • Delete single extension

Click on

to delete the extension. Or select the checkbox of the extension and then click on “Delete Selected Extensions”.

Notes

This is the expected behavior when you delete an extension:

  • The system will delete all the data of the extension except the CDR and meetings record. All the data on the user side will be erased.
  • The extension will be removed from group chats and the messages sent previously by the extension will be kept. However, only other users can search through those messages while the new user of the extension cannot.
  • If the extension was in a meeting schedule, the meeting will still be present. The extension will be removed from the meeting and will not be notified about the meeting.

  • Modify selected extensions

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

  • Delete selected extensions

Select the checkbox for the extension(s). Then click on “Delete ” to delete the extension(s).

Export Extensions

The extensions configured on the UCM630xA can be exported to csv format file with selected technology “SIP”, “IAX” or “FXS”. Click on “Export Extensions” button and select technology in the prompt below.

Figure 85: Export Extensions

The exported csv file can serve as a template for users to fill in desired extension information to be imported to the UCM630xA.

Import Extensions

The capability to import extensions to the UCM630xA provides users flexibility to batch add extensions with similar or different configuration quickly into the PBX system.

  1. Export extension csv file from the UCM630xA by clicking on “Export Extensions” button.
  2. Fill up the extension information you would like in the exported csv template.
  3. Click on “Import Extensions” button. The following dialog will be prompted.

Figure 86: Import Extensions
  1. 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.
  1. Click on “Choose file to upload” to select csv file from local directory in the PC.
  2. Click on “Apply Changes” to apply the imported file on the UCM630xA.

Example of file to import:

Figure 87: Import File

Field Supported values
ExtensionDigits
TechnologySIP/SIP(WebRTC)
Enable Voicemailyes/no/remote
CallerID NumberDigits
SIP/IAX PasswordAlphanumeric characters
Voicemail PasswordDigits
Skip Voicemail Password Verificationyes/no
Ring TimeoutEmpty/ 3 to 600 (in second)
SRTPyes/no
StrategyAllow All/Local Subnet Only/A Specific IP Address
Local Subnet 1IP address/Mask
Local Subnet 2IP address/Mask
Local Subnet 3IP address/Mask
Local Subnet 4IP address/Mask
Local Subnet 5IP address/Mask
Local Subnet 6IP address/Mask
Local Subnet 7IP address/Mask
Local Subnet 8IP address/Mask
Local Subnet 9IP address/Mask
Local Subnet 10IP address/Mask
Specific IP AddressIP address
Skip Trunk Authyes/no/bytime
Codec PreferencePCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus
PermissionInternal/Local/National/International
NATyes/no
DTMF ModeRFC4733/info/inband/auto
InsecurePort
Enable Keep-aliveYes/no
Keep-alive FrequencyValue from 1-3600
AuthIDAlphanumeric value without special characters
TEL URIDisabled/user=phone/enabled
Call Forward BusyDigits
Call Forward No AnswerDigits
Call Forward UnconditionalDigits
Support Hot-Desking ModeYes/no
Dial Trunk PasswordDigits
Disable This ExtensionYes/no
CFU Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
CFN Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
CFB Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Music On HoldDefault/ringbacktone_default
CC Agent PolicyIf CC is disabled use: never

 

If CC is set to normal use: generic

If CC is set to trunk use: native

CC Monitor PolicyGeneric/never
CCBS Available Timer3600/4800
CCNR Available Timer3600/7200
CC Offer Timer60/120
CC Max AgentsValue from 1-999
CC Max MonitorsValue from 1-999
Ring simultaneouslyYes/no
External NumberDigits
Time Condition for Ring SimultaneouslyAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Time Condition for Skip Trunk AuthAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Enable LDAPYes/no
Enable T.38 UDPTLYes/no
Max ContactsValues from 1-10
Enable WaveYes/no
Alert-InfoNone/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 DisturbYes/no
DND Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Custom Auto answerYes/no
Do Not Disturb WhitelistEmpty/digits
User PasswordAlphanumeric characters.
First NameAlphanumeric without special characters.
Last NameAlphanumeric without special characters.
Email AddressEmail address
LanguageDefault/en/zh
Phone NumberDigits
Call-Barging MonitorExtensions allowed to call barging
Seamless Transfer MembersExtensions allowed to seamless transfer
Table 55: SIP extensions Imported File Example

Field Supported values
ExtensionDigits
TechnologyIAX
Enable Voicemailyes/no
CallerID NumberDigits
SIP/IAX PasswordAlphanumeric characters
Voicemail PasswordDigits
Skip Voicemail Password Verificationyes/no
Ring TimeoutEmpty/ 3 to 600 (in second)
SRTPyes/no
StrategyAllow All/Local Subnet Only/A Specific IP Address
Local Subnet 1IP address/Mask
Local Subnet 2IP address/Mask
Local Subnet 3IP address/Mask
Local Subnet 4IP address/Mask
Local Subnet 5IP address/Mask
Local Subnet 6IP address/Mask
Local Subnet 7IP address/Mask
Local Subnet 8IP address/Mask
Local Subnet 9IP address/Mask
Local Subnet 10IP address/Mask
Specific IP AddressIP address
Skip Trunk Authyes/no/bytime
Codec PreferencePCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2-G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus
PermissionInternal/Local/National/International
NATyes/no
Call Forward BusyDigits
Call Forward No AnswerDigits
Call Forward UnconditionalDigits
Require Call TokenYes/no/auto
Max Number of CallsValues from 1-512
Dial Trunk PasswordDigits
Disable This ExtensionYes/no
CFU Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
CFN Time Condition
CFB Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Music On HoldDefault/ringbacktone_default
Ring simultaneouslyYes/no
External NumberDigits
Time Condition for Ring SimultaneouslyAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Time Condition for Skip Trunk AuthAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Enable LDAPYes/no
Limit Max time (s)empty
Do Not DisturbYes/no
DND Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Do Not Disturb WhitelistEmpty/digits
User PasswordAlphanumeric characters.
First NameAlphanumeric without special characters.
Last NameAlphanumeric without special characters.
Email AddressEmail address
LanguageDefault/en/zh
Phone NumberDigits
Call-Barging MonitorExtensions allowed to call barging
Seamless Transfer MembersExtensions allowed to seamless transfer
Table 56: IAX extensions Imported File Example

Field Supported values
ExtensionDigits
Technology FXS
Analog StationFXS1/FXS2
Enable Voicemailyes/no
CallerID NumberDigits
Voicemail PasswordDigits
Skip Voicemail Password Verificationyes/no
Ring TimeoutEmpty/ 3 to 600 (in second)
Auto Recordyes/no
Fax ModeNone/Fax Gateway/Fax Detection
Skip Trunk AuthYes/no/bytime
PermissionInternal/Local/National/International
Call Forward BusyDigits
Call Forward No AnswerDigits
Call Forward UnconditionalDigits
Call WaitingYes/no
Use # as SENDYes/no
RX GainValues from -30🡪6
TX GainValues from -30🡪6
MIN RX FlashValues from: 30 – 1000
MAX RX FlashValues from: 40 – 2000
Enable Polarity ReversalYes/no
Echo CancellationOn/Off/32/64/128/256/512/1024
3-Way CallingYes/no
Send CallerID After1/2
Dial Trunk Passworddigits
Disable This ExtensionYes/no
CFU Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
CFN Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
CFB Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Music On HoldDefault/ringbacktone_default
CC Agent PolicyIf CC is disabled use: never

 

If CC is set to normal use: generic

If CC is set to trunk use: native

CC Monitor PolicyGeneric/never
CCBS Available Timer3600/4800
CCNR Available Timer3600/7200
CC Offer Timer60/120
CC Max AgentsValue from 1-999
CC Max MonitorsValue from 1-999
Ring simultaneouslyYes/no
External NumberDigits
Time Condition for Ring SimultaneouslyAll 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 LDAPYes/no
Enable Hotline Yes/no
Hotline TypeImmediate hotline/delay hotline
Hotline Numberdigits
Limit Max time (s)empty
Do Not DisturbYes/no
DND Time ConditionAll time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Do Not Disturb WhitelistEmpty/digits
User PasswordAlphanumeric characters.
First NameAlphanumeric without special characters.
Last NameAlphanumeric without special characters.
Email AddressEmail address
LanguageDefault/en/zh
Phone NumberDigits
Call-Barging MonitorExtensions allowed to call barging
Seamless Transfer MembersExtensions allowed to seamless transfer
Table 57: FXS Extensions Imported File Example

The CSV file should contain all the above fields, if one of them is missing or empty, the UCM630xA 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

Once the extensions are created with Email addresses, the PBX administrator can click on button “E-mail Notification” to send the account registration and configuration information to the user. Please make sure Email setting under Web GUI🡪System Settings🡪Email Settings is properly configured and tested on the UCM630xA before using “E-mail Notification”.

When click on ”More” 🡪 “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.

Figure 90: E-mail Notification – Prompt Information

The user will receive Email including account registration information as well as the Grandstream Wave Settings with the QR code:

Figure 91: Account Registration Information


Figure 92: Grandstream Wave Settings and QR Code

Multiple Registrations per Extension

UCM630xA 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 set to 1 for security purpose. Maximum is 10.

Figure 94: Extension – Concurrent Registration

SMS Message Support

The UCM630xA provides built-in SIP SMS message support. For SIP end devices such as Grandstream GXP or GXV phones that supports SIP message, after an UCM630xA 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.

Figure 95: SMS Message Support

EXTENSION GROUPS

The UCM630xA extension group feature allows users to assign and categorize extensions in different groups to better manage the configurations on the UCM630xA. 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.

FXO PortSelect the channel for the analog trunk.

 

  • UCM6300A: 1 channel
  • UCM6302A: 2 channels
  • UCM6304A: 4 channels
  • UCM6308A: 8 channels
Trunk NameSpecify a unique label to identify the trunk when listed in outbound rules, incoming rules and etc.
Advanced Options
SLA ModeEnable 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 AllowedThe 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 AccessThe 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 ReversalIf 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 DelayWhen 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 UCM630xA will use to check on a voltage drop in the line. The default setting is 200. The valid range is 50 to 3000.
Ring TimeoutConfigure 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 UCM630xA considers a non-ringing line with Hangup activity. The default setting is 8000.
RX GainConfigure the RX gain for the receiving channel of analog FXO port. The valid range is from -13.5 (dB) to + 12.0 (dB). The default setting is 0.
TX GainConfigure the TX gain for the transmitting channel of analog FXO port. The valid range is from -13.5 (dB) to + 12.0 (dB). The default setting is 0.
Use CallerIDConfigure to enable CallerID detection.

 

The default setting is “Yes”.

Caller ID SchemeSelect 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 ModeConfigures how faxes to this extension will be handled.

 

  • None: Faxes will not be processed.
  • Fax Gateway: Faxes to this extension will be processed and converted from T.30 to T.38 or vice-versa. FXS/FXO ports only.

The default setting is None.

FXO Dial Delay (ms)Configure the time interval between off-hook and first dialed digit for outbound calls.
Auto RecordEnable 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 TrunkIf selected, the trunk will be disabled and incoming/Outgoing calls via this trunk will not be possible.
DAHDI Out Line SelectionThis 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 if UCM6302A is used would be port 1🡪port 2🡪. 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🡪1🡪2🡪1🡪2🡪…, following the last port being used in case UCM6302A is 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 if UCM6302A is used would be port 2🡪port 1. Every time it will start with port 2 (if it is idle).

The default setting is “Ascend” mode.

Echo Cancellation ModeThe 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: The NLP limits the signal level to the background noise level when active, and the background noise level adjustment is low.
  • High Noise Level Adjustment: The NLP limits the signal level to the background noise level when active, and the background noise level adjustment is high.
  • Noise Masking: The NLP sends sign noise when active, and the background noise level adjustment is high.
  • White Noise Injection: The NLP injects white noise when active. The level corresponds to the background noise level at Sin, and the background noise level adjustment is high.
Direct CallbackAllows 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 DetectionBusy Detection is used to detect far end Hangup or for detecting busy signal. The default setting is “Yes”.
Busy Tone CountIf “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 DetectionCongestion detection is used to detect far end congestion signal. The default setting is “Yes”.
Congestion CountIf “Congestion Detection” is enabled, users can specify the number of congestion tones to wait for. The default setting is 2.
Tone CountrySelect 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 ToneSyntax:

 

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 ToneSyntax:

 

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 DetectionClick on “Detect” to detect the busy tone, Polarity Reversal and Current Disconnect by PSTN. Before the detecting, please make sure there are more than one channel configured and working properly. If the detection has busy tone, the “Tone Country” option will be set as “Custom”.
Table 58: Analog Trunk Configuration Parameters

PSTN Detection

The UCM630xA 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 UCM630xA settings.

  1. Go to UCM630xA Web GUI🡪Extension/Trunk🡪Analog Trunks page.
  2. Click to edit the analog trunk created for the FXO port.
  3. In the 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: UCM630xA FXO Tone Settings

  1. Click on “Detect” to start PSTN detection.
Figure 99: UCM630xA 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: UCM630xA PSTN Detection: Auto Detect

  • If there is only one FXO port connected to PSTN line, use the following settings for auto-detection.
Figure 101: UCM630xA 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 UCM630xA 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 UCM630xA settings.

Detect ModelSelect “Auto Detect” or “Semi-auto Detect” for PSTN detection.

 

  • Auto Detect

Please make sure two or more channels are connected to the UCM630xA 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 UCM630xA 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 UCM630xA 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 ChannelSelect the channel to be detected.
Destination ChannelSelect the channel to help detect when “Auto Detect” is used.
Destination NumberConfigure the number to be called to help the detection.
Dump Call Progress Tone FileChoose whether to save the calling tone file, it is not checked by default.
Table 59: PSTN Detection for Analog Trunk

  • 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 being 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 UCM630xA 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.

  • Click on “Add SIP Trunk” or “Add 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.
https://documentation.grandstream.com/knowledge-base/sip-trunks-guide/

The VoIP trunk options are listed in the table below.


Type

Select the VoIP trunk type.

  • Peer SIP Trunk

  • Register SIP Trunk

Provider Name

Configure a unique label (up to 64 characters) 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 method.
Using TCP requires local TCP support.

Using TLS Requires local TLS support. 

  • UDP

  • TCP

  • TLS

Auto Record

If enabled, calls handled with this extension/trunk will automatically be recorded.

Keep Original CID

Keep the CID from the inbound call when dialing out. This setting will override the “Keep Trunk CID” option. Please make sure that the peer PBX at the other side supports to match user entry using the “username” field from the authentication line.

Keep Trunk CID

If enabled, the trunk CID will not be overridden by the 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 a 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 Number

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 the 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 new name of the caller when the extension has no CallerID Name configured.

Auto Record

If enabled, calls handled with this extension/trunk will automatically be recorded.

Auth ID

Enter the Authentication ID for the "Register SIP Trunk" type.

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.

RemoteConnect Mode

If enabled, the RemoteConnect related parameters will be set synchronously. Please make sure the trunk host is allocated by GDMS or supports TLS.

Limit Concurrent Calls

If enabled and when the number of concurrent calls exceeds any trunk's configured concurrent call thresholds, an alarm notification will be generated. Note: Please make sure the system alert event "Trunk Concurrent Calls" is enabled.

Concurrent Call Threshold

Threshold of all incoming and outgoing concurrent calls through this trunk.

Outgoing Concurrent Calls Threshold

Threshold of all outgoing concurrent calls passing through this trunk.

Incoming Concurrent Calls Threshold

Threshold of all incoming concurrent calls passing through this trunk.

Total Time Limit For Outbound Calls

Enable Total Time Limit For Outgoing Calls

When this setting is activated, the user can set a time limit before calls cannot be initiated through this trunk

Period

This setting defines how long until the time allowed for outgoing calls is reset.


  • Monthly: The time allowed will reset every month.

  • Quarterly: The time allowed will reset every 3 months.

Example: If the time limit has been set to 4320 minutes, the allowed time will always revert back to 4320 after a month or 3 month based on the period configured.

Total Time

Total time allowed in minutes

Table 60: Create New SIP Trunk

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.

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 the “Keep Trunk CID” option. Please make sure that the peer PBX at the other side supports to match user entry using the “username” field from the authentication line.

Keep Trunk CID

If enabled, the trunk CID will not be overridden by the 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 a 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 Number

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

RemoteConnect Mode

If enabled, the RemoteConnect related parameters will be set synchronously. Please make the trunk host is allocated by GDMS or it supports 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 and VP8.

Packet Loss Retransmission

Configure to enable Packet Loss Retransmission.

Audio FEC

Configure to enable Forward Error Correction (FEC) for audio.

Video FEC

Configure to enable Forward Error Correction (FEC) for video.

ICE Support

Toggles ICE support. For peer trunks, ICE support will need to be enabled on the other end.

TURN Relay

TURN servers are used for media NAT traversal and will be prioritized if ICE is also enabled. 

FECC

Configure to enable Far-end Camera Control

SRTP

Enable SRTP for the VoIP trunk. The default setting is "No".

SRTP Crypto Suite

SRTP encryption suite used by UCM for outbound calls. Priority is based on order of configuration.

IPVT Mode

Similar to Enable Direct Media. The PBX will attempt to redirect the RTP media streams to bypass the PBX and to go directly between caller and callee. Primarily for use with trunks to IPVT.

Enable T.38 UDPTL

Enable or disable T.38 UDPTL support.

Special attributes

Carry the ssrc/msid/mid/ct/as/tias/record properties of the SDP. These attributes may cause incompatibility when interconnecting with other devices.

Send PPI Header

If checked, the invite message sent to trunks will contain PPI (P-Preferred-Identity) Header.

Send PAI Header

If checked, the INVITE, 18x and 200 SIP messages sent to trunks will contain P-Asserted-Identity (PAI) header. It is not possible to send both PPI and PAI headers.

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 enabled and "Send PAI Header" is disabled, PAI headers will be preserved as calls pass through the UCM.

Send PANI Header

If checked, the INVITE and REGISTER sent to the trunk will contain P-Access-Network-Info header.

Send Anonymous

If checked, the "From" header in outgoing INVITE message will be set to anonymous.

Outbound Proxy Support

Enable to send outbound signal to the proxy instead of the devices directly.

DID Mode

Configure to obtain the destination ID of an incoming SIP call from SIP Request-line or To header.

GIN Registration

If enabled, the UCM will send a GIN REGISTER (generate implicit numbers).

DTMF Mode

Configure the mode for sending the DTMF.

  • RFC4833 (default): DTMF is transmitted as audio in the RTP stream but is encoded separately from the audio stream. Backwards-compatible with RFC2833. 

  • DTMF is transmitted as audio and is included in the audio stream. Requires alaw/ulaw codecs

  • Info: DTMF is transmitted separately from the media streams.

  • RFC4733_info: DTMF is transmitted through both RFC4733 and SIP INFO

  • Auto: DTMF mode will be negotiated with the remote peer, only supports RFC4733 and inband. RFC4733 will be used by default unless the remote peer does not indicate support.

Enable Heartbeat detection

If enabled, the PBX will regularly send SIP OPTIONS to check if the device is online.

The Maximum Number of Call Lines

The number of current outgoing calls over the trunk at the same time. The default value 0 means no limit.

STIR/SHAKEN

Block Spam Calls.

  • Disabled: Disable STIR/SHAKEN.

  • Outgoing Attest: Enable STIR/SHAKEN attestation for outgoing calls.

  • Incoming Verify: Enable STIR/SHAKEN verification for incoming calls.

  • Both: Enable STIR/SHAKEN for both outgoing and incoming calls 

Enable CC

Check this box to allow the system to automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason. If the Call Waiting is disabled, the CC service will take effect only for unanswered and timeout calls.

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 the “Keep Trunk CID” option. Please make sure that the peer PBX at the other side supports to match user entry using the “username” field from the authentication line. 

Keep Trunk CID

If enabled, the trunk CID will not be overridden by the 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 a 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 the "Register SIP Trunk" type is selected.

The default setting is No.

Allow outgoing calls if registration failure

If enabled outgoing calls even if the registration to this trunk fails will still be able to go through.

Note that if we uncheck the “Need Registration” option, 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 actual domain name where the extension comes from. This can be used to override the “From” Header.

For example, "trunk.UCM630X.provider.com" is the From Domain in From Header: sip:1234567@trunk.UCM630X.provider.com.

From User

Configure the actual username of the extension. This can be used to override the “From” Header. There are cases where there is a single ID for registration (single trunk) with multiple DIDs.

For example, "1234567" is the From User in From Header: sip:1234567@trunk.UCM630X.provider.com.

Username

Enter the username to register to the trunk from the provider when the "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 the "Register SIP Trunk" type.

Auth Trunk

If enabled, the UCM will send a 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.

RemoteConnect Mode

If enabled, the RemoteConnect related parameters will be set synchronously. Please make the trunk host is allocated by GDMS or it supports 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.

Enable Concurrent Call Alert

If enabled, the "Trunk Concurrent Calls" system event will monitor the number of concurrent calls in this trunk. When the number of concurrent calls in a certain period exceeds the set threshold, an alarm message will be generated. Note: Please turn on the alert for the "Trunk Concurrent Calls" event first.

Two-way Concurrent Calls Threshold

Threshold of all incoming and outgoing concurrent calls through this trunk.

Outgoing Concurrent Calls Threshold

Threshold of all outgoing concurrent calls passing through this trunk.

Incoming Concurrent Calls Threshold

Threshold of all incoming concurrent calls passing through this trunk.

Advanced Settings

Codec Preference

Select the audio 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 and ADPCM.

Packet Loss Retransmission
NACK+RTX(SSRC-GROUP)

Configure to enable Packet Loss Retransmission.

Audio FEC

Configure to enable Forward Error Correction (FEC).

Video FEC

Video FEC

ICE Support

Toggles ICE support. For peer trunks, ICE support will need to be enabled on the other end.

TURN Relay

TURN servers are used for media NAT traversal and will be prioritized if ICE is also enabled.

FECC

Remote Camera Management

SRTP

Toggle encryption of RTP streams.

SRTP Crypto Suite

SRTP encryption suite used by UCM for outbound calls. Priority is based on order of configuration.

Enable T.38 UDPTL

Enable or disable T.38 UDPTL support.

Special Attributes

Carry the ssrc/msid/mid/ct/as/tias/record properties of the SDP. These attributes may cause incompatibility when interconnecting with other devices.

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 the 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 to use this feature

Send Anonymous

If checked, the "From" header in the outgoing INVITE message will be set to anonymous.

DOD As From Name

If enabled and "From User" is configured, the INVITE's From header will contain the DOD number.

Note:

Passthrough PAI Header

If checked and the option "Send PAI Header" is not checked, the PAI header will be passthrough from one side to the other side.

Send PANI Header

If checked, the INVITE and REGISTER sent to the trunk will contain the P-Access-Network-Info header.

Access Network Info

The access network information is in the P-Access-Network-Info header.

Send Anonymous

If checked, the "From" header in the outgoing INVITE message will be set to anonymous.

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".

GIN Registration

If enabled, the UCM will send a GIN REGISTER (generate implicit numbers).

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 the 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 a 64-bit codec, i.e., PCMU and PCMA.

  • Auto: Send DTMF using RFC4733 if offered. Otherwise, an inband will be used.

Enable Heartbeat Detection

If enabled, the UCM630X will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is "No".

Heartbeat Frequency

When the "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.

The Maximum Number of Call Lines

The maximum number of concurrent calls using the trunk. The default setting is 0, which means no limit.

STIR/SHAKEN

Configure this feature to prevent call spoofing, robocalls and spam calls.

  • Disabled

  • Outgoing Attest: The UCM will only authenticate the caller ID.

  • Incoming Verify: The UCM will only verify if the caller ID is authenticated

  • Both: The UCM will authenticate the oungoing calls as well as verify if the incoming calls are authenticated.

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 that 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 63: 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 the 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 the "Register IAX Trunk" type is selected.

Password

Enter the password to register to the trunk from the provider when the "Register IAX Trunk" type is selected.

Disable This Trunk

If selected, the trunk will be disabled.

Caller ID Number

The number that the trunk will try to use when making outbound calls.
CID priority from highest to lowest is as follows:

From User (register trunk only) >>> Inbound Call CID (if Keep Original CID is enabled and the call is originally from another trunk) >>> Trunk CID (Keep Trunk CID enabled) >>> DOD CID >>> Extension CID >>> Register Trunk Username (Keep Trunk CID disabled) >>> Global Outbound CID.

Note 1: Certain providers may ignore this CID.

Note 2: If this CID contains an asterisk (*), call recordings from this trunk might be lost when saving them to NAS storage.

CallerID Name

Configure the new name of the caller when the extension has no CallerID Name configured.


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 the 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 UCM630X will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is “No”.

Heartbeat Frequency

When the “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 setting is 0, which means no limit.

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 the 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 the 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 UCM630X will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is “No”.

Heartbeat Frequency

When the “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 setting is 0, which means no limit.

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

Type

Register Trunk

Provider Name

Configure a unique label to identify the trunk when listed in outbound rules and incoming rules.

Host Name 

Enter the IP address or hostname of the VoIP provider's server.

Transport

Configure the SIP Transport method. Using TCP requires local TCP support; using TLS requires local TLS support.

Keep Original CID

Keep CID from the inbound call when dialing out even if option "Keep Trunk CID" is enabled. Please make sure the peer PBX at the other end supports matching user entry using the "username" field from the authentication line.

Keep Trunk CID

Always use trunk CID if specified even if extension has DOD number or CID configured.

NAT

Enable this setting if the UCM is using public IP and communicating with devices behind NAT.
Note 1: This setting will overwrite the Contact header of received messages, which may affect the ability to establish calls when behind NAT. Consider changing settings in PBX Settings 🡪 SIP Settings 🡪 NAT instead.

Disable This Trunk

Check this box to disable this trunk

TEL URI

if "Enabled" option is selected, TEL URI and remove OBP from Route cannot be enabled at the same time. If the phone has an assigned PSTN telephone number, this field should be set to "User=Phone". A "User=Phone" parameter will the 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.

Need Registration

Whether to register on the external server.

Allow outgoing calls if registration fails

Uncheck to block outgoing cakks if registration fails. If "Need Registration" option is unchecked, this settting will be ignored.

CallerID Name

To display the caller ID name of the trunk, you must configure the caller ID number of the trunk.

Trunk Registration Number

The number used to register with the provider server, and the VoIP provider will authenticate the number based on the trunk registration number.

Line Selection Strategy

Linear: Select lines in list order and make Outbound calls. Round Robin: Rotary line selection with memory and making Outboun calls.

AuthTrunk

If enabled, the UCM will send a 401 response to the incming call to authenticate the trunk.

Auto Record

If enabled, calls handled with this extension/trunk will automatically be recorded. 

Direct Callback

Allows external numbers the option to get directed to the extension that last called them.

RemoteConnect Mode

If enabled, RemoteConnect-related options will be automatically configured. Please confirm the trunk has a GDMS-assigned address or supports TLS.

Monitor Concurrent Calls

If enabled, the number of concurrent calls on this trunk will be monitored. If the "Trunk Concurrent Calls" system alert is enabled, alert notifications will be generated if the number of concurrent calls exceeds this trunk's configured concurrent call thresholds.

Concurrent Call Threshold

Threshold of all incoming and outgoing concurrent calls in this trunk.

Outgoing Concurrent Call Threshold

Threshold of all outgoing concurrent calls passing through this trunk.

Incoming Concurrent Call Threshold

Threshold of all incoming concurrent calls passing through this trunk.

Enable Total Time Limit For Outbound Calls

If enabled,  a limit will be placed on the cumulative duration of outbound calls within a specific period. Once this limit has been reached, further outbound calls from this trunk will not be allowed.


WebRTC Trunks

WebRTC, Web Real-Time Communication, is a real time audio/video chatting framework that allows real-time audio/video chatting through the web browser. WebRTC usually does not refer to the web application itself but to the set of protocols and practices bundled with a graphical interface. Our UCM63xx supports creating WebRTC trunks to use exclusively with web application, this allows the users to join calls and meetings just by clicking a link to a web page.

Below is a figure that shows the options to configure when setting up this feature:

Trunk Name

Configure a unqiue label to indetify the trunk when listed in incoming rules.

Disable This Trunk

Tick this box to disable this trunk.

Auto Record

When enabled, calls handled with this extension/trunk will automaticall be recorded.

Enable Concurrent Call Threshold

If you enable this option, an alert notification will be prompted when the number of concurrent calls exceed the number of concurrent calls threshold.

Note: Please make sure that the system alert event "Trunk Concurrent Calls is enabled for this notification to be generated. 

Note: The range of allowed concurrent calls is 1-200.

Incoming Concurrent Call Threshold

Number of all incoming concurrent calls passing through the trunk threshold. 

WebRTC Inbound Link Address

The link on which users can initate calls to this specific trunk.

Important Note

Please note that in order to use WebRTC Trunk feature, you need to have a paid RemoteConnect plan enabled.

Direct Outward Dialing (DOD)

The UCM630xA 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 UCM630xA:

  1. To setup DOD go to UCM630xA 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.
  5. Set the DOD name and If extension number need to be appended to the DID number click on “Add Extension”.
  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
Note

Users can import and export DOD files.

SLA STATION

The UCM630xA 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 UCM630xA is like BLF but SLA is used to monitor external line i.e., analog trunk on the UCM630xA. 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 UCM630xA.

Create/Edit SLA Station

SLA Station can be configured on Web GUI🡪Extension/Trunk🡪SLA Station.

Figure 106: 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 66: SLA Station Configuration Parameters

Station NameConfigure a name to identify the SLA Station.
StationSpecify 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.
Associated SLA TrunksSelect 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 TimeoutConfigure 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 DelayConfigure 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 AccessThis 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 UCM630xA, 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 107: Enable SLA Mode for Analog Trunk


  1. Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.


Figure 108: Analog Trunk with SLA Mode Enabled

  1. On the UCM630xA, 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 109: SLA Example – SLA Station


  1. On the SIP phone 1, configure to register UCM630xA 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 UCM630xA 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 110: 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 UCM630xA, 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 UCM630xA, an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern. This allows different patterns to be dialed through different trunks (e.g., “Local” 7-digit dials through an FXO while “Long distance” 10-digit dials through a low-cost SIP trunk). Users can also set up a failover trunk to be used when the primary trunk fails.

Go to Web GUI🡪Extension/Trunk🡪Outbound Routes to add and edit outbound rules.


  • Click on

    to add a new outbound route.
  • Click on to edit the outbound route.

  • Click on to delete the outbound route.

On the UCM630xA, the outbound route priority is based on “Best matching pattern”. For example, the UCM630xA 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 67: Outbound Route Configuration Parameters

Outbound Rule NameConfigure the name of the calling rule (e.g., local, long_distance, and etc.). Letters, digits, _ and – are allowed.
PatternAll patterns are prefixed by “_” character, but please do not enter more than one “_” at the beginning. All patterns can add comments, such as “_pattern /* comment */”. In patterns, some characters have special meanings:

 

  • [12345-9] … Any digit in the brackets. In this example, 1,2,3,4,5,6,7,8,9 is allowed.
  • N … Any digit from 2-9.
  • . … Wildcard, matching one or more characters.
  • ! … Wildcard, matching zero or more characters immediately.
  • X … Any digit from 0-9.
  • Z … Any digit from 1-9.
  • – … Hyphen is to connect characters and it will be ignored.
  • [] Contain special characters ([x], [n], [z]) represent letters x, n, z.
Disable This RouteAfter 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.
PasswordConfigure the password for users to use this rule when making outbound calls.
Local Country CodeIf your local country code is affected by the outbound blacklist, please enter it here to bypass the blacklist.
Call Duration LimitEnable to configure the maximum duration for the call using this outbound route.
Maximum Call DurationConfigure the maximum duration of the call (in seconds). The default setting is 0, which means no limit.
Warning TimeConfigure the warning time for the call using this outbound route. If set to x seconds, the warning tone will be played to the caller when x seconds are left to end the call.
Auto RecordIf enabled, calls using this route will automatically be recorded.
Warning Repeat IntervalConfigure the warning repeat interval for the call using this outbound route. If set to X seconds, the warning tone will be played every x seconds after the first warning.
PIN GroupsSelect a PIN Group
PIN Groups with Privilege LevelIf enabled and PIN Groups are used, Privilege Levels and Filter on Source Caller ID will also be applied.
Privilege LevelSelect 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 can use this rule.
  • National: Users with National or International level can use this rule.
  • International: The highest level required. Only users with international level can use this rule.
  • Disable: The default setting is “Disable”. If selected, only the matched source caller ID will be allowed to use this outbound route.

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 IDWhen 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 list. 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 CIDAttempt to use the configured outbound route CID. This CID will not be used if DOD is configured.
Send This Call Through Trunk
TrunkSelect the trunk for this outbound rule.
StripAllows 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.

PrependSpecify 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 TrunkFailover trunks can be used to make sure that a call goes through an alternate route when the primary trunk is busy or down. If “Use Failover Trunk” is enabled and “Failover trunk” is defined, the calls that cannot be placed via the regular trunk may have a secondary trunk to go through.

 

UCM630xA 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.

StripAllows 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.

PrependSpecify the digits to be prepended before the call