GCS Grandstream Training
GCS Grandstream Training
GCS Grandstream Training
COMMUNICATION
S
UNIFIED
IP-PBX: UCM series
GCS Certification Training
> Why use VoIP?
> Introduction to CU
> CU Features
>Certification Exam
Why use VoIP?
Yes
VoIP offers many benefits for businesses, one of the most Switching over to VoIP can save time
and money as well increase
attractive is the reduction of the total cost of productivity.
Connect Collaborate
Unified Communication can make a difference
Components of unified
communication
□ Communications: Voice, data,
videos
□ Messaging: Voice, email, video, and IM
□ Video and audio conference systems, web conferencing service in the cloud
Data
Video
Call recording
CDR SIP video call
Codec transcoding Surveillance
Backup system Integration
UCM Applications
V Connecting The World — □
GSAffinity
■ Grandstream's proprietary CTI application. G RAN DSTREAM
view.
■ Support desktop notification by default.
■ Call popup screen. Advanced
Version: 1.0.1.5
■ Call log.
New Line
■ CRM Search
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UCM Applications
GSAffinity – Call Features
□ Dial, outgoing calls.
□ Answer/reject an incoming call.
□ Hold / remove hold.
V Connecting The World □
□ CRM Search
Build a communications network
integrate Security Office
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PSTN
Reception/Lobby
UCM series IP PBX
Conference Room
Main Office
l,XV36IO
GXV367
^
Warehouse/Loading Dock
Executive Office
In trance
Remote Worker
Installation and
Deployment
V UCM6202 Please go to the Change Binding Email page to bind Email address to your account, This Email address will be Apply Changes I English v | (D 1001
used to retrieve your password when you forget it,
i= Basic Information
User Information
Personal Settings
Extensions
Change Information
f Personal Data v
& 1003
30s
No Conference Schedule
• Device information: hardware version, software version, part number, MAC address, uptime.
• Network information: mode (DHCP, static IP, PPPoE), IP address, subnet mask .
• Network Menu: Network Settings for LAN/WAN (DHCP, Static IP, PPPoE).
Factory menu: reset, LCD test patterns, fan mode, LED test patterns, and etc.
step.
Auto provision automatically provides an extension to the device.
• Auto discovery
• Automatic assignment
There are two methods of auto provision: SIP SUBSCRIBE and DHCP Option 66.
For example, when the device boots up, it will send SIP SUBSCRIBE multicast in the LAN. The PBX will find it, create an acco device to
■
Zero Config
nsion Segm...5000 - 6299 Zero Config Extension Segment
Change of password
External
Accounts
Hot Desking:
■ No
Enable Pick Extension:
Network Configuration
Pick Extension Segment: 4000 - 4999 Pick Extension Segment Account 1:
The PBX can automatically discover the new devices by ARP or PING. It can scan the
entire network segment or a single IP address.
■
I 1001 ’|
Time zone
Pick Extension F MAC Address IP Address Extension Version
Account 2:
■
| 1001 " PBX LAN/LAN1 Address: 192.168.2.1
Extensions
Network Se 000B822DD46F 192.168.6.196 1.0.8.9
■
Network Segment: 192.168.2.0- 192.168.2.255
■
grandstream GXP2160 1 6U
Summary
000882836615 192.168.6.185 1.0.8.46 Broadcast IP:
grandstream GXW4004 1 € OR
Zero Config Global Policy Global Templates Model Templates Model Update Zero Config Settings
Basic Settings
Extension Assignment
■Configurable extension segment Auto provision automatically provides an extension to the device.
There are two methods of auto provision: SIP SUBSCRIBE and DHCP Option 66.
Save
Installation and deployment
Zero Config – Discovery
Discovery methods :
/ PING
/ ARP
{ SIP message (NOTIFY)
The discovered devices will be displayed in the list.
MAC Address IP Address Extension Version Vendor model Create Config Options
□ Z®
□
000B822DD46F 192.168.6.196 1.0.8.9 grandstream GXP1405 1
ND
A
m
□ 000B82836615 192.168.6.185 1.0.8.46 grandstream GXP2160 1
ND
A
m
□ 000B829E25D4 192.168.6.169 1.0.0.48 grandstream -- 1
Am
□ 000B825C6927 192.168.6.157 1.0.9.7 grandstream GXP2160 1 NDA m
ND
□ 000B8282C687 192.168.6.238 1.0.4.55 grandstream GXP1630 1
ND
A t
Global Policies
Configuration by Model
The global settings will apply to all Grandstream SIP end devices connected on the same LAN with the UCM,
regardless of the Grandstream device model. It is divided into two levels:
■ Web UI-> Value-added Features->Zero Config->Global Policy
■ Web UI-> Value-added Features- ->Zero Config->Global Templates .
Global template settings have a higher priority than global policy settings, and both will take effect.
Installation and deployment
Zero Config – Global Template
Later, when the user configures the device in Edit Device dialog-
>Advanced Settings , the user can select to use one of the global
template for the device.
Installation and deployment
Zero Config – Global Policy
The categories are: The Global Policy configuration will be applied to all devices. Specific model configurations, if any, will be applied on top
Upgrade
Options
•um2 and
■ Maintenance: configure update, web access, Telnet/SSH access Phone Setting: Firmware Source
Provision
Config Server Path
Maintenance Automatic
■ Network Settings – Configure IP address, QoS and STUN Network Settil Firmware
7em I trust
•uarE
settings. Customization
Rule
supported models .
Save
Installation and deployment
Installation and deployment
Zero Config – Configuration by model
Edit Model Templates: GXV3240
A
□ 000B825C6927 192.168.6.162 1,0.9.9 grandstream GXP2160 1 •
•
Model template settings allow users to apply model - □ 000B826B1958 192.168.6.115 1.03.167 grandstream GXV3240 1
ND
ND
A • A.C.
■ Devices
specific settings to different
created devices. will be
or discovered □ 000B826B1FF7 192.168.6.158 1.03.171 grandstream GXV3240 1
ND
A fir and ®
IP Address:
000B8275CBB8
192.168.6.134
| P1362
value
and advanced
If multiple configuration
templates per modelofare
a device.
created and enabled, © Add New Field
Basic Advanced
the user must select the device template in Edit Device Modify Customize Settings
■Advance
Teleph
Dialog-> Settings.
one
Basic Advanced
Accounts
Hot Desking:
ne
Account 1:
1001 ©Global Policy
Account 2:
• Modify Global Policy 1001
Installation and deployment
Zero Config – Provisioning
■ After going through the five-layer configuration for the device, users could
send NOTIFY to the SIP device and trigger it to start automatic provisioning.
■ Click •h to send the NOTIFY to the device to activate it and download the
configuration file from the URL contained in the NOTIFY message.
■ Ready
Installation and deployment
Zero Config – Reboot Remote Device
The UCM6XXX supports rebooting discovered devices from the zero config page.
MCU
To use this feature, web UI-> Value-added Features -> Zero Config -> Zero Config,
and click the reset button in the Options tab.
Zero Config
Zero Config Settings
Zero Coniig Global Policy Global Templates Model Templates Model Update
Auto Discover Create New Device Delete Selected Devices Modify Selected Devices Reset All Extensions Filter: Scan Results
MAC Address IP Address Extension Version Vendor model Create Config Options
GSWave is a softphone application from Grandstream for Android devices running Android 4.0 or
192.168.2.1 SIP User ID: 1001
Authenticate ID: 1001 Authenticate
Password: t'297eoS1h Name:
2. UCM users who have the email address configured in the extensions will
receive the account registration and LDAP configuration information in the
email.
Server Address : 192.168.2.1
Port: 389
Base : dc=pbx,dc=com
3. In the GSWave app, select “UCM Account (QR_code Scan) to scan and Tis is the QR code of this LDAP config.
register immediately.
Call management
Call Management
SIP Forking – Multiple Registration per Extension
■ UCM supports multiple registration per extension. The user can register from multiple devices to the UCM using a
single account.
Home phone:
Extension 1000
Call Management
Call transference
Call transfer allows a user to transfer calls to another phone. Our UCM supports 3 types of transfers:
■ Seamless Transfer : Seamless transfer allows the user to perform blind transfers using the UCM feature code without
having music on hold presented during the transfer process.
Call Management
DND (DND White List)
Call forwarding is a telephony feature that redirects a telephone call to another destination, which may be, for
example, a mobile, voicemail or another telephone number where the desired called party is available.
Call waiting is a key feature of a telephone system. When a call is placed on hold, the caller is parked and free to
communicate with the person on the other end of the line. Music on hold is usually played for the caller. The person
who placed the caller on hold can perform other tasks during the hold time.
UCM users can set music on hold and upload music files through the web interface, the files must be in 8KHz Mono
Wav format with a maximum size of 30 MB.
UCM users can also record music on hold by initiating a recording call to an extension,
answering the call, and starting recording. Once you hang up and refresh the page, the
new music on hold file will be available in the list.
Call Management
call parking
This feature allows you to put a call on hold, so that it can be retrieved from another phone in the system.
To use call park: If you have an active call on your phone, you can park it in 2 ways:
Someone else on another phone in your system can dial the call park extension to retrieve the call.
If the call is not recovered after the waiting time, it will call back to the phone that parked it.
Call Management Create New Outbound Rule Save
Cancel
Dialing pattern
Safe routes Enable Filter on Source Caller ID
Remove/prepend digits
Back
Privileges/Permissions
Output rule with filter by
Caller ID
Call duration limit
PIN Groups
Blacklist
time condition +Add
No Data
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Outbound Rule Name Pattern • Privilege Level • Options
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Call Management
Call Routing – Entry Rules
Inbound Routes
□ 1 □ 1_______ 1
( () Alarm
Voicemail Presence and DISA Service .
____________
ME
J.
_□
()
Call Barge Callback Fax
follow me
1_________
() N ()
Intercom and k Capture ringing group Call Center
paging
___ Group \____________
/
CU Features
IVR
* External numbers can be set to blacklist/whitelist only when "Dial Trunk" is enabled for IVR.
CU Features
Conference
The conference bridge feature allows multiple people to participate in the
call. Different conference bridges can be defined in the UCM6XXX and have the
following main characteristics:
Max. Number of 25 32 64
participants
CU Features
Multi-conference bridge
UCM A and UCM B are interconnected through a SIP trunk. In the UCM A web user interface, the administrator
can directly invite a conference room on UCM B to connect to the UCM A conference room, essentially
connecting two conference rooms.
UCM6202 A UCM6202B
SIP peer trunk
Room
Conference •d Conference
A
CU Features
Conference agenda
□ Once the user enables Google Calendar settings in the Google service support option, the conference agenda can be
synchronized with Google Calendar.
□ The conference schedule function can be found in the Web UI-> Call Features-> Conference->
Conference Schedule
9 10 11 12 13 f March 2015 14 • ' TodayTomorrow Sunday
1000am Busy 9:00am Busy J I 9X)0am Busy 9:30am Busy 4 • March 2015 New York NY -
25°F/17F — 35'F/29°F 44°F/27°F
Search Calendar (Ctrl ♦
1100am Busy 10:00am Busy 3:15pm Busy HISBusy
1030am M.O.YOUWET.H.FR SA E)
1:30pm Busy 12:00pm Busy 1:00pm
22 Busy
23 24 25 26 27 28 Calendar X 4 Library ■ Main - 428 • Conf Room 1244-0860) x
4:00pm Busy ■ SATURDAY
1 2 3 4 5 ma 7
SUNDAY
Mar 1
MONDAY TUESDAY WEDNESDAY THURSDAY SATURDAY
7
SUNDAY MONDAY TUESDAY WEDNESDAY THURSDAY
8 9 10 11 12 13 14
2 3 4 5
15 16 17 18 19 20 21
22 23 24 25 26 27 28
29 30 31
April 2015
SU MO TU WE TH FR 5A
12 3 4
5 6 7 8 9 10 11
12 13 14 15 16 17 18
19 20 21 22 23 24 25
26 27 28 29 30 1 2
3 4 5 6 7 8 9
4 • My Calendars
J Calendar
a • Rooms
7 Library • Main - 428 * Conf..
16 17 18 19 20 21
11200amBusy 1030am Busy ' 9:00am Busy | P 7100pm Tentative 9:30am Busy
130pm Busy 1:30pm Busy 10:30am Busy 3:15pmBusy 10:30am Busy
12:00pm Busy 1:00pm Busy
26 27 28 24 25
1900am Busy 9:30am Busy | 1300pm Busy
10:00am Busy 1:00pm Busy | 3:00pm Busy
130 pm Busy
315pm Busy
CU Features
CU Features
Voicemail Management
Voicemail is a VoIP telephony feature, which allows the caller to leave a voice message when the called party is
absent or busy on another call.
The called party can then check the stored voice message from:
The user can listen to, delete messages or even change voicemail prompts.
The UCM supports the pickup group feature that allows users to
pick up incoming calls from other extensions if they are in the same
pickup group, by dialing the “Extension Pickup” feature code
(default* 8).
CU Features
Intercom/Paging
Page and Intercom can be used to make an announcement to a group of telephones. Such phones will
answer immediately using the speakerphone.
Paging is a one-way call so people can hear the message but cannot respond.
Cluster
The intercom is a two-way call so people can hear the intercom/paging
message and also respond to the person initiating the call.
■ Whisper
In “Whisper” mode, the monitor extension can hear both parties on the active call, but the monitor audio
can only be heard by the monitored extension. The default feature code is *55 .
■ barge
In “Barge” mode, the monitor extension can talk to both parties on the active call. The call will be set up
similar to a three-way conference. The default feature code is *56 .
CU Features
SIP presence
Custom presence Enter the presence status for this mode in the Web UI.
status
The contact is not available at the moment, please try
Not available
to contact us later.
CU Features
Instant messaging
■ Only SIP terminals compatible with SMS messages will be able to send and receive SMS
messages.
SMS
CU Features
Distinctive tones
□ Users can configure the “Alert-Info" field to select different ringtones for
a specific incoming call.
□ Alert-info can use the available options (Tone 1, Tone 2...) Extension ----------------------------------------
CU Features
Speed Dial speed dial
+Add
Speed dialing allows users to call a certain Extension # speed dial Default Destination Default Destination Options
ND
■ Extension Create New Speed Dial
Total: 5 1 10 / page v | goto —
■ Voicemail
■ Conference
■ voicemail room
group
Enable Destination : Q
■ paging group
Voicemail
Conference Roor
■ DISA IV R
PageGroup
CU Features
Marked by name
Dial by name is a PBX feature that allows a caller to search for a person by their first or last name using the telephone keypad.
When the caller reaches the IVR/Auto Attendant, the IVR will guide the caller to spell the name with digits to find the person in
the Dial by directory
name.. 1. The customer calls the
NET
company number to speak with
Benefits: the sales representative "Jane
Doe"
WOR PSTN
TRUNK
■ The administrator can define the K/
COMPANY PBX: UCM6202
Mark directory by name for 2. UCM6202 dials by name in IVR
greeting:
include only the desired extensions to "Enter the first 3 digits of the name
to search the directory"
reach. This helps block
3. Customer enters 5 (J) 2 (A) 6
calls to employees no (N) LEFONS OF
EMPLOYEES
they receiv thos calls REGISTERED AS EXT
IN THE UCM6202
4. UCM6202 confirms what I
directly. marked
■ The PBX can enable this feature via feature code *11, and cancel it 5
with *12 and inform the caller as soon as the called extension
becomes available. UCM initiates the call
to ext 1000 once
■ Compatible with individual extensions. For example, when Extension A answered, it calls ext
1001
calls Extension B, Busy camp-on can be requested.
2
■ Compatible with SIP trunks. For example, when Extension A in UCM A
calls Extension B in UCM B, Busy Camp-on can be requested. Ext 1001 is
busy or I 4
reject the call
Ext 1001 is placed
available
Extension: 1001
CU Features
Busy Camp-on
The UCM will initiate the callback when the previously unavailable extension becomes available. Use
the following conditions to detect whether the extension is available or not:
□ If extension B was busy when called, extension B is considered available after this call concludes.
□ If extension B rejected the call or the call was not answered, extension B is considered available
after a new call is completed. This means that extension B has to initiate a new call or answer
another incoming call. When the new call concludes, extension B is considered available.
CU Features
DISA
Call center agents are members defined to answer queued calls. Agents can be static
or dynamic.
UCM6208 60
UCM6510 120
Call Center – Web-based call center
□
□
Call waiting
Call transference
To use these
features, there
are 2 methods
D
□
□
3-way conference
Call parking
available:
Using
□
standard
C
function keys or codes on phones
□ Using the control panel
Call Center: position announcement and virtual queue
□ The call center administrator can log in to the UCM user portal with their
extension number and assigned user password. To access the Control
Panel, click “Value-added Features" in the side menu and click “Call
Queue".
Grades:
Only one administrator can be defined per Call Center.
The administrator extension can also be a Call Center agent.
CU Features
Call Center - Control Panel
Agents
Basic Information
f Personal Data
□ Each agent or user who has an extension can now Value-added Features A
Call Queue
* Agents :
abandonment percentage.
Total Information
Received Answered Unanswered Abandoned Transferred % Answ % Unansw % Abandon % Trans Login Logoff Avg Talk A)
■ Announcement Center allows the user to pre-record and store voice notifications in the UCM with an
associated code. The user also needs to create a group with assigned extensions.
■ When the voice prompt code and the group number are dialed together, the specified voice prompt will
be played to all extensions in the group after they answer the call.
CU Features
Call recording
Recordings Storage
■ 55MB usage for about 1 hour
call recording. Enab e auto charge
Crazy :
■ Recording files are in WAV format and can be
played and downloaded from the UCM6XXX series
Web UI
API
CDR API Recording
Integration
ME
with CRM
Integration
with PMS
party
API and 3rd integration
CDR API
□ The UCM has an API for third-party billing, so that API Configuration Files
external billing software can access the CDR in
the PBX.
Enable:
□ REC API is used to access call recording files. This is an API similar to the UCM-compatible CDR API.
□ This is to request call recordings, or a list of recording files within Asterisk directories, that match the parameters
set by the third-party application.
API and 3rd party integration
Asterisk Management Interface -AMI
□ Asterisk Manager Interface (AMI) allows a client program to connect to an Asterisk instance and issue commands
or read events over a TCP/IP transmission.
□ Useful when integrators are trying to track the status of a telephony client within Asterisk or want to initiate
calls.
□ The UCM6XXX supports CRM integration that allows users to connect third-party CRM interfaces with the UCM.
□ UCM currently supports Salesforce and SugarCRM.
• SugarCRM
Allows querying, updating, adding CDR records via SugarCRM
APIs (via SOAP or REST requests) and click-to-dial from
SugarCRM.
• Salesforce
Supports querying, updating, and adding CDR records through
the Salesforce API (via SOAP or REST requests).
API and 3rd party integration
Integration with PMS
■ Property Management System (PMS) is a software application used in
the hospitality industry to automate some hotel functions such as
guest booking, guest details, etc.
■ The system can be divided into three parts:
■ PMS (Property Management System)
■ PMSI (Property Management System Interface)
■ PBX
■ The Grandstream UCM6XXX series has an integrated PMS that offers
the following hospitality features: Check-in, Check-out, set room
status, alarm clock and more.
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GCS Certification
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