Cloudpbx Admin Guide en PDF

Admin Guide
Admin Guide
Yeastar Cloud PBX
Version: 81.13.0.15
Date: 2020-03-10
Contents
Admin Guide........................................................................................................................ 5
Extensions........................................................................................................................5
Extension Overview..................................................................................................5
Extension Basic Setup............................................................................................. 6
Extension Groups................................................................................................... 15
Presence.................................................................................................................17
Voicemail.................................................................................................................20
Call Monitoring........................................................................................................29
Call Permission.......................................................................................................31
Extension Settings..................................................................................................32
Contacts......................................................................................................................... 39
Contacts Overview................................................................................................. 39
Manage Company Contacts...................................................................................41
Manage Personal Contacts....................................................................................43
Configure Company Contacts Permissions for Users............................................46
Identify Callers from Contacts................................................................................47
Contacts FAQ......................................................................................................... 48
Trunks............................................................................................................................ 50
Trunk Overview...................................................................................................... 50
VoIP Trunks............................................................................................................ 50
WebRTC Trunks..................................................................................................... 64
Call Control.................................................................................................................... 69
Emergency Numbers..............................................................................................69
Time Conditions......................................................................................................70
Inbound Routes...................................................................................................... 81
Outbound Routes................................................................................................... 99
Outbound Restriction............................................................................................106
Contents | iii
AutoCLIP Routes..................................................................................................108
SLA Stations.........................................................................................................111
Call Features............................................................................................................... 117
IVR........................................................................................................................ 118
Ring Group........................................................................................................... 122
Queue................................................................................................................... 122
Conference........................................................................................................... 128
Call Pickup............................................................................................................130
Call Transfer......................................................................................................... 134
Callback................................................................................................................ 135
Speed Dial............................................................................................................ 136
DISA......................................................................................................................137
Intercom/Paging....................................................................................................138
Call Parking.......................................................................................................... 143
Fax........................................................................................................................ 146
PIN List................................................................................................................. 151
Blacklist/Whitelist.................................................................................................. 153
Call Recording............................................................................................................. 157
Call Recording Overview......................................................................................157
One Touch Record............................................................................................... 157
Auto Recording.....................................................................................................158
Voice Prompts..............................................................................................................166
System Prompt..................................................................................................... 166
Music on Hold (MoH)........................................................................................... 169
Custom Prompt.....................................................................................................171
Set Prompts for Failed Calls................................................................................ 176
System Management...................................................................................................177
System General Settings..................................................................................... 177
Security................................................................................................................. 184
User Permission................................................................................................... 185
Date and Time......................................................................................................186
Email..................................................................................................................... 187
Contents | iv
Auto Cleanup........................................................................................................188
Event Center.........................................................................................................189
Remote Management........................................................................................... 191
API........................................................................................................................ 192
Maintenance.................................................................................................................192
Upgrade Firmware................................................................................................192
Backup and Restore.............................................................................................194
Reboot the PBX................................................................................................... 196
Reset the PBX......................................................................................................197
System Log...........................................................................................................197
Operation Log.......................................................................................................198
Troubleshooting.................................................................................................... 199
PBX Monitor.................................................................................................................203
Resource Monitor........................................................................................................ 204
CDR and Recordings.................................................................................................. 205
Search CDR and Recordings...............................................................................206
Fuzzy Search CDR and Recordings....................................................................207
Download CDR and Recordings.......................................................................... 207
Admin Guide
Admin Guide for Yeastar Cloud PBX.
About this guide

In this guide, we describe every detail on the functionality and configuration of the Yeastar
Cloud PBX. We begin by assuming that you are familiar with networking and other IT
disciplines.
Audience
This guide is for administrators who need to prepare for, configure and operate Yeastar
Cloud PBX.
Extensions
Extension Overview
An extension is a short internal number. Extensions allow users to make and receive calls.
You can assign extensions to every employee in your organization.
Extension types
SIP Extension
A SIP extension is based on SIP protocol.
To use a SIP extension, you need to enter the extension credentials on an IP phone or a
softphone. After the extension is registered on a phone, you can make and receive calls.
Extension format
Yeastar Cloud PBX supports 1-digit to 7-digit extension format. The default extension format
is 4-digit number.
Before you create extensions, you can go to Settings > PBX > General > Preferences >
Extension Preferences > User Extension to change the extension format and range.
Admin Guide | 1 - Admin Guide | 6
Extension Basic Setup

Create Extensions
Extension Creation Overview

Yeastar Cloud PBX supports SIP forking, which enables an extension number to register on
multiple SIP phones simultaneously.
SIP Forking
Yeastar Cloud PBX supports SIP forking, which enables an extension number to be
registered by multiple SIP phones. When a call reaches the extension, all registered phones
will ring simultaneously, and you can take the call from any device easily.
You can configure SIP Forking on the extension configuration page. The value of
Concurrent Registrations limits how many SIP phones the extension can be registered
on.
Note:
• The limit of concurrent registrations is 4.

• By default, if one SIP phone is busy, other SIP phones still can receive calls when calls
reach the extension. To restrict other phones from receiving calls when the extension
is busy, you can enable All Busy Mode for SIP Forking (Settings > PBX > General >
SIP > Advanced).
Create an Extension
Before registering a SIP account on phones, you need to create a SIP account.
1. Go to Settings > PBX > Extensions, click Add.
2. On the Basic page, go to General section, and set the general settings of the
extension.
• Extension: Enter the extension number.

• Caller ID: Enter the caller ID number. The called party will see this caller ID
number when the extension user makes an outgoing call.
• Registration Name: The name used to register a SIP extension.
• Caller ID name: Enter the caller ID name. The called party will see this caller ID
name when the extension user makes an outgoing call.
• Concurrent Registrations: Yeastar Cloud PBX supports to register one extension
number on multiple phones. When a call reaches the extension number, all phones
will ring. The maximum number of concurrent registrations is 4.
• Registration Password: The password is used to register the extension.
3. On the Basic page, go to User Information section, and set the user information.
• Email: Extension user can reset his/her login password, receive voice mails, faxes,
or PBX notifications via this email address.
• User Password: The password is used to log in the PBX or log in Linkus mobile
client. The password is generated randomly by default.
• Prompt Language: The default prompt language is the same as the system
language. If the extension user speaks foreign language, you can set a specific
system prompt.
Note: Before selecting other system prompts, go to Settings > PBX > Voice
Prompts > System Prompt to download online prompts.
• Mobile Number: Extension user can receive the PBX notifications or forwarded
calls on this mobile number.
4. Optional: Click Presence, Features, Advanced, or Call Permission tab to configure

other settings.
5. Click Save and Apply.
Bulk Create Extensions

Yeastar Cloud PBX supports to add SIP extensions in bulk.
1. Go to Settings > PBX > Extensions, click Bulk Add.
2. On the Basic page, go to General section, and configure the following settings:
Note:
• A random Registration Password and a random User Password will be

assigned for each extension.
• Start Extension: Enter the first extension number. The system will create
extensions in bulk starting with the extension number.
• Create Number: Enter the number of extensions that will be created.
• Concurrent Registrations: Yeastar Cloud PBX supports to register one extension
number on multiple phones. When a call reaches the extension number, all phones
will ring.
• Prompt Language: The language of voice prompts. The default prompt language
is the same as the system language. If the extension user speaks foreign
language, you can set a specific system prompt.
Note: Before selecting other system prompts, go to Settings > PBX > Voice
Prompts > System Prompt to download online prompts.
3. Optional: Click Features, Advanced, or Call Permission tab to configure other

settings.
Related tasks
Bulk Edit Extension Names and Emails
Related information
Register a SIP Extension
Register Extensions
Register a SIP Extension

To make calls and receive calls from a SIP extension, you need to register the SIP
extension on an IP phone or soft phone.
1.Gather information of extension registration
For most SIP phones, the following items are needed for the SIP phone to register with
Yeastar Cloud PBX.
• Domain name of PBX

• SIP registration port: The default port is 5060 on Yeastar Cloud PBX.
• Extension information
◦ Extension Number
◦ Registration Name
◦ Registration Password
◦ Caller ID Name
◦ Transport
2.Register the extension on a phone
Log in the phone web interface, fill in and save the required items to register the SIP
extension.
3.Confirm registration status
You can do one of the followings to check if the extension is registered.
• On the phone web interface, check if the status indicates that the extension is
registered.
• Log in PBX web interface, go to PBX Monitor > Extensions to check if the status
shows .
Related tasks
Register Yealink Phone with Yeastar Cloud PBX
Register Htek Phone with Yeastar Cloud PBX
Register Cisco Phone with Yeastar Cloud PBX
Register Fanvil Phone with Yeastar Cloud PBX
Register Snom Phone with Yeastar Cloud PBX
Manage Extensions
Change Extension Range

The default extension range is from 1000 to 5999. Before you create extensions, you can
change the extension range according to your needs.
1. Log in PBX web interface, go to Settings > PBX > General > Preferences >
Extension Preferences.
2. Change the range of User Extensions.
Edit Extensions
After creating extensions, you may need to change extension settings. You can edit an
extension, or edit extensions in bulk.
Edit an Extension
1. Log in PBX web interface, go to Settings > PBX > Extensions.

2. On Extensions page, click beside the extension that you want to edit.
3. Change extension settings according to your needs.
Bulk Edit Extensions

2. On Extensions tab, select the checkbox of desired extensions, and click Edit.
3. Change extension settings according to your needs.
Bulk Edit Extension Names and Emails

To bulk edit the extension names and emails, you need to export the extensions from
Yeastar Cloud PBX first, edit the extension names and email addresses in the CSV file, then
import the file to the PBX.
1. Log in PBX web interface, go to Settings > PBX > Extensions, click Export to export
all the extensions.
2. Edit the CSV file, enter the users' names and email addresses, then save the file.
• fullname: Enter the user's name. The fullname stands for the Caller ID Name.
• email: Enter the user's email address.
3. Import the CSV file to the PBX.
a. Go to Settings > PBX > Extensions, click Import.
b. On the pop-up dialog, click Browse, select your CSV file.
c. Click Import.
Note: You may get an error prompt like the following figure.
d. Click Yes to check the log.
Note: Ignore the error if the Error Cause displays "username[1000]: The
imported record is existing, the record has been overwritten".
4. Check the imported extensions on your PBX.

Delete Extensions
When an employee leaves or an extension is no longer needed, you can delete the
extension from the Yeastar Cloud PBX.
Delete an Extension

2. On Extensions page, click beside the extension that you want to delete.
Bulk Delete Extensions

2. On Extensions page, select the checkbox of desired extensions, and click Delete.
Import/Export Extensions
The extensions configured on Yeastar Cloud PBX can be exported and saved as a
template. You can fill in desired extension information and import the CSV file to PBX again.
Export Extensions

2. Click Export to export the extensions to a CSV file.
Import Extensions
Tip: You can export extensions first, and use the CSV file as a template.

2. Click Import.
3. On the Import Extension page, click Browse to select your CSV file.
4. Click Import.
Send Extension Information

After finishing the extension configurations, you can send the extension information to
the extension users' emails. The extension users can get their extension registration
information, Linkus login information, PBX login information in the email.
1. Log in PBX web interface, go to Settings > PBX > Extensions, click Welcome Email.
2. Select the extensions that you want to send Welcome Email to.
• To send emails to all extensions, select All Extensions.
• To send emails to specific extensions, follow the instruction.
a. Select Selected Extensions.

b. Select the desired extensions from Available box to Selected box.
3. Click Send.
Related tasks
Edit 'Welcome Email' Template
Edit 'Welcome Email' Template

Yeastar Cloud PBX has a default template of the welcome email. You can change the email
subject and email contents according to your needs.
1. Log in PBX web interface, go to Settings > PBX > Extensions, click Welcome Email.
2. On the Send Welcome Email page, click Edit Template.

You will see the description of variables and the default email contents.
Note: The variables in the email contents are unchangeable.
3. Edit the email subject and email contents.

Extension Groups
Create an Extension Group
You can assign and categorize extensions in different groups. Extension groups simplify the
configuration process.
1. Go to Settings > PBX > Extensions > Extension Group, click Add.
2. Set the Name to help you identify the group.
3. In the Available box, select the extensions to the Selected box.

Manage Extension Groups
Edit extension groups

You can edit the group name, add more extensions to the group or remove extensions from
the group.
1. Go to Settings > PBX > Extensions, search and find the desired extension group,
click .
2. Edit the group as you need.
Delete extension group
1. Go to Settings > PBX > Extensions > Extension Group, search and find the desired
extension group, click .
2. Click Yes to confirm the deletion.
Extension Groups Application

You can use the extension groups when you need to assign extensions for outbound routes,
ring groups, queues, etc.
For example, you need to set an outbound route and only allow the Support group members
to make outbound calls through this route. You can simply assign the Support extension
group instead of assigning an extension member one by one. It simplifies the configuration
process.
Presence
Extension Presence Overview
This topic introduces what is presence and how the presence status can benefit the user's
work.
What is Presence
Extension Presence indicates the availability status of an extension. Presence settings are
linked to the Call Forwarding rules and Linkus ring strategy. Different call forwarding rules
and ring strategy can be set for each presence status.
Yeastar Cloud PBX provides five presence statuses:
• Available: The user is online and ready for communication.

• Away: The user is currently away from your office.
• DND: The user doesn't want to be disturb, and you won't receive any calls.
• Lunch Break: The user is currently on lunch break.
• On a Business Trip: The user is currently on a business trip.
How does Presence benefit the user's work?

Presence status and information that are displayed on Linkus clients allows the user to see
the presence of your colleague and instantly know whether the colleague is available, busy,
or away.
Change Presence status to quickly route incoming calls. For example, if the user is at
a meeting and do not want to miss calls, set the status to Away and forward the call to
voicemail. Once the user is ready to receive calls again, switch back to Available.
How to change Presence status?

3 ways to change an extension presence.
• On the Linkus client, extension users can change their own presence status.
• On the Extension Web Portal, Linkus users can change their own presence status.
• On the PBX Web Portal, you can change all extensions presence.
Set Call Forwarding Rules & Presence Status

Set call forwarding rules for each presence status. The call forwarding rules allows the
user to automatically forward an incoming call to voicemail, another extension, or mobile
depending on the extension status.
1. Go to Settings > PBX > Extensions, search and find the desired extension, click .
2. Click the Presence tab.
3. In the Presence drop-down list, select a status to configure.
4. In the Presence Information field, enter the a custom status message to display on
Linkus.
The Linkus users can see whether you are available to communicate.
5. Set call forwarding rules for the Presence status.
a. Select the Call Forwarding conditions:
• Always: All the incoming calls will be forward to the destination.
• No Answer: Only the unanswered calls will be forwarded to the destination.
• When Busy: Only the calls that come in while you are talking on the phone
will be forwarded.
b. Beside the selected forwarding condition, select the forwarding destination.
6. Set the ring strategy for the Presence status.
• Ring First: When a call reaches the extension, which terminal will ring first.
• Ring Secondly: If the incoming call is not answered on the terminals that are
selected as Ring First, the terminals that are selected as Ring Secondly will ring.
Set Call Forwarding Prompt

By default, when the PBX is forwarding an incoming call to another number, the PBX will
play the call forwarding prompt "please hold when I try to locate the person you are calling",
and then play the MoH music when the caller is waiting. You can disable the call forwarding
prompt and change the MoH music to a normal ring tone. In this way, the caller will not
realize that the call is forwarded.
1. Go to Settings > PBX > Voice Prompts > Prompt Preference.
2. Unselect the checkbox of Play Call Forwarding Prompt.
3. In the Music on Hold for Call Forwarding drop-down list, select Ringing Tone.
Activate/Deactivate Call Forwarding

Extension users can dial the Call Forwarding feature codes on their phones to activate or
deactivate Call Forwarding function.
Below are the default call forwarding feature codes and the description of how to use the
feature codes.
Code Action Example
*71 Activate call forwarding ALWAYS

• Dial *71 to forward all calls to voicemail.
• Dial *716000 to forward all calls to extension 6000.
*071 Deactivate call forwarding ALWAYS

• Dial *071 to deactivate call forwarding ALWAYS.
*72 Activate call forwarding WHEN

BUSY • Dial *72 to forward calls (when the user is busy) to
voicemail.
• Dial *726000 to forward calls (when the user is
busy) to extension 6000.
*072 Deactivate call forwarding WHEN

BUSY • Dial *072 to deactivate call forwarding WHEN
BUSY.
*73 Activate call forwarding NO

ANSWER • Dial *73 to forward calls (when the user doesn't
answer) to voicemail.
• Dial *736000 to forward calls ((when the user
doesn't answer) to extension 6000.
*073 Deactivate call forwarding NO

ANSWER • Dial *073 to deactivate call forwarding NO
ANSWER.
Voicemail
Voicemail Overview
Yeastar Cloud PBX integrates a free voicemail system. Voicemail is a modern kind of
answering machine that allows the callers to leave audio messages in case of unavailability.
Enable/Disable Voicemail Function

By default, the voicemail is enabled for all extension users. You can disable the Voicemail
function if the user doesn't need it.
3. Change the Voicemail settings.

• To enable voicemail, select the checkbox of Enable Voicemail.
• To disable voicemail, unselect the checkbox of Enable Voicemail.
Change Voicemail PIN/Password

Extension users can dial voicemail feature code (default *2) on their phones to access their
voicemails. To enhance the extension security, you can change the voicemail PIN on PBX
web interface.
1. Go to Settings > PBX > Extensions.

2. Search and find the desired extension, click .
4. In the Voicemial Access PIN field, enter a numeric PIN/password.
Configure Voicemail to Email

The Voicemail to Email feature of Yeastar Cloud PBX allows extension users to receive
voicemail audio files as email attachments and quicken response time when they are out of
office.
Enable Voicemail to Email

Voicemail to Email function is disabled by default. If an extension user would like to check
voicemail messages via email, you need to enable Voicemail to Email for his/her extension.
Note: To receive voicemail via email successfully, make sure the system email is
working.
1. Go to Settings > PBX > Extensions, select the desired extension, click .
2. Click the Features tab.
3. In the Send Voicemail to Email drop-down list, select an email type.
• Send to user's email: Send voicemail to the extension user's email address.
• Send to custom email: Send voicemail to a custom email address.
Email template of 'Voicemail to Email'

The PBX has a default email template for Voicemail to Email. You can edit the template
according to your needs.
1. Go to Settings > System > Email > Email Templates, click beside Voicemail to
Email.

Check Voicemail Messages

Extension users have multiple ways to check their voicemail messages.
Check Voicemail on Linkus

Log in Linkus, go to Me > Voicemail to check your voicemail.
Check Voicemail on a Phone
• Dial feature code *2 on a phone
A user can dial *2 on his own phone to check voicemail.

• Dial feature code *02 on a phone
A user can dial *02 on other user's phone to enter the voicemail main menu, then enter
his/her extension number and voicemail PIN to check voicemail.
Check Voicemail on Web Page

Extension users can log in the PBX web page to check their own voicemails.
• User name: The extension user's email address.

• Password: The extension's User Password.
Check Voicemail via Email

If voicemail to email is enabled for an extension user, the user can check voicemails in his/
her email box.
Check Voicemail via IVR

If you check the option Dial to Check Voicemail for an IVR; users can access the IVR to
check their voicemails. This solution is for the users who are outside the office to check their
voicemails.
Tip: If the users are using Linkus, they can dial *2 directly to check their voicemails.
Change Voicemail Greetings

You can change the global voicemail greetings for all the extension users or change
voicemail greeting for a specific extension.
Components of a Voicemail Greeting

When an extension user is unavailable, the voicemail greeting consists of 3 audio clips:
Unavailable Prompt + Voicemail Prompt + "Di".
When an extension is busy on a phone, the voicemail greeting consists of 3 audio clips:
Busy Prompt + Voicemail Prompt + "Di"
• Default Unavailable Prompt: The person at the extension XXXX is unavailable.

• Default Busy Prompt: The person at the extension XXXX is busy.
• Default Voicemail Prompt: Please leave your message after the tone, when done hang
up or press the pound key (#)."
Change global voicemail greetings
1. Prepare your custom prompt files, and upload to the PBX.

2. Go to Settings > PBX > General > Voicemail > Greeting Options.
3. Change the global voicemail greetings.
• Busy Prompt: Select the prompt that will be played when the extension is busy.
• Unavailable Prompt: Select the prompt that will be played when the extension is
unavailable.
• Voicemail Prompt: Select the prompt that will be played after Busy or Unavailable
prompt.
Change voicemail greetings for a specific extension

By default, the global busy prompt and global unavailable prompt are applied to all
extensions. If an extension user wants to use her/his personal greetings, you can change
the prompts for the extension.
Note: The greeting prompt file format should be ".wav", ".WAV" or ".gsm" file.
The file size must not be larger than 8MB.
Supported Format: PCM: 8K, 16bit, 128kbps; A-law(g.711): 8k, 8bit, 64kbps; u-law (g.711):
8k, 8bit, 64kbps; gsm: 6.10, 8k, 13kbps.
2. Click the Features tab.
3. Click Browse to upload a prompt file.
Manage Voicemail Messages Centrally

In Yeastar Cloud PBX, you have two options to manage voicemail messages centrally and
efficiently: subscribe BLF keys on a phone to monitor multiple extensions' voicemail status
and receive multiple extensions' voicemail messages from one mailbox.
Monitor voicemail status by BLF keys

By default, an extension's voicemail status cannot be monitored by other users. To monitor
an extension's voicemail status, you need to enable Share Voicemail Status on the
extension.
We take Yealink T27G v69.82.0.20 as an example below to introduce how to monitor

voicemail status of extension 4000 by extension 1000.
1. Enable voicemail status sharing feature of extension 4000.
a. Log in the PBX web interface, go to Settings > PBX > Extensions, edit the
extension 4000.
b. On the extension Features page, enable Share Voicemail Status.
c. Click Save and Apply.

2. Set BLF key to monitor the voicemail status.

a. Log in the IP phone where extension 1000 is registered, go to Dsskey.
b. Set a BLF key to monitor voicemail status of extension 4000.
• Type: Select BLF.

• Value: Enter *2{ext_num}. In this example, enter *24000.
• Line: Select the line where extension 1000 is registered on.
c. Click Confirm.
Result:
• Green BLF LED: The extension 4000 has NO unread voicemail messages.
• Red BLF LED: The extension 4000 has unread voicemail messages.
Receive voicemail from a mailbox

To receive multiple extensions' voicemail messages from one mailbox, you can configure
sending voicemail to a same custom email address for these extensions.
For example, to receive multiple extensions' voicemail messages from the mailbox
voicemial@yeastar.com. Set Send Voicemail to Email to the same custom email address
voicemial@yeastar.com for these extensions.
Global Voicemail Settings

You can change the global voicemail message settings, voicemail playback settings
The global voicemail settings will be applied to all the extensions.
Navigation path: Settings > PBX > General > Voicemail.
Table 1. Global Voicemail settings

Setting Description
Message Options
Max Messages per Folder Each extension user has a Read voicemail folder and an Unread folder.
You can set the maximum number of messages per folder.
Max Message Time Set the maximum time of one message.
Min Message Time Set the minimum time of one message.
Delete Voicemail This function will work if you enable Send Voicemail to Email. If the
voicemail is forwarded to the user's email, PBX will delete voicemails from
the user's voicemail folder.
Ask Caller to Dial 5 By default, when the caller accesses a user's voicemail, PBX starts to
record message automatically. If you want to prompt the caller first, you
can enable this option. The caller needs to dial 5 first, then starts to record
message.
Setting Description
Operator Breakout from If enabled, the users can dial 0 to exit the voicemail destination of an IVR.
Voicemail
Greeting Options
Busy Prompt Select the greeting that will be played when the extension is busy.
Note: To use a custom prompt, you need to upload your audio file to
the Custom Prompt page first.
Unavailable Prompt Select the greeting that will be played when the extension is unavailable.
Voicemail Prompt Select the greeting that will be played before the caller leave a message.
Playback Options
Announce Message Caller ID If enabled, the PBX will announce who left the message.
Announce Message Duration If enabled, the PBX will announce the message duration.
Announce Message Arrival If enabled, the PBX will announce when the message was received.
Time
Allow Users to Review If enabled, the users can review their recorded message, and then send
Messages the messages.
Voicemail Menu
You can dial *2 on your phone to access the voicemail menu. Below is the detailed
voicemail menu.
Call Monitoring
Call Monitoring Overview
Call Monitoring allows authorized users to monitor another extension user's call in real time.
The supervisor can dial "feature code" + "extension number" to monitor the extension user's
call.
Go to Settings > PBX > General, click Feature Code tab.
In the Call Monitor section, you can enable or disable monitor modes, and modify
corresponding feature codes.
Yeastar Cloud PBX supports the following monitor modes:
• Listen (Default code: *90)
Listen mode allows supervisor to listen to a call in real time.
The supervisor can not talk with the monitored extension users.
• Whisper (Default code: *91)
Whisper mode allows supervisor to listen to a call in real time, and talk with the
monitored extension user privately.
The other party can not hear the supervisor's voice.

• Barge-in (Default code: *92)
Barge-in mode allows the supervisor to listen to a call in real time and talk with both
parties.
Configure Call Monitoring

To monitor an extension, you need to set monitor settings for both the supervisors and the
monitored extension users.
1. Enable and select a monitor mode for the supervisor.
a. Go to Settings > PBX > Extensions, click beside the desired extension.
b. On the configuration page, click Features tab.
c. In the Monitor Settings section, select a Monitor Mode for the supervisor.
• Disabled: Not allowed to monitor other extension users' call.

• Extensive: Use any one of listen, whisper, or barge-in mode to monitor.
• Listen: Listen to a call in real time, but you can not talk with the monitored
extension users.
• Whisper: Listen to a call in real time, and talk with the monitored extension
users privately.
• Barge-in: Listen to a call in real time and talk with both parties.
d. Click Save and Apply.
2. Set the extension which will be monitored.
a. Go to Settings > PBX > Extensions, click beside the desired extension.
b. On the configuration page, click Features.

c. On the Monitor Settings section, select the checkbox of Allow Being Monitored.
d. Click Save and Apply.
Call Permission
Set Call Permission of an Extension
On the Extension configuration page, you can set the outbound call permissions for the
extension user.
1. Go to Settings > PBX > Extensions, click beside the desired extension.
2. On the Extension configuration page, click Call Permission tab.
3. Select outbound routes for the extension from Available box to Selected box.
Outbound Routes Permission

Select outbound routes to the Selected box, the extension user will have the permission to
make outbound calls through the selected outbound routes.
Outbound Restriction
• Prohibit Outbound Calls
Select the Outbound Restriction option to prohibit this extension from making
outbound calls.
On the Extensions page, the extension will be locked and the extension status will
show .
Note: If the extension user makes outbound calls over the limit of Outbound
Restriction rule, the extension will also be locked.
• Cancel Restriction for Outbound Calls
Double click the icon or unselect the checkbox of Outbound Restriction to allow
this extension to make outbound calls.
Extension Settings
SIP Extension Settings
This reference describes all settings on a SIP extension.
Basic Settings
Navigation path: Settings > PBX > Extensions, edit a SIP extension on the Basic tab.
General Settings
Settings Descriptions
Extension Enter the extension number.
Caller ID If you set the caller ID number, the called party will
see this caller ID number when the extension user
makes an outgoing call.
Registration Name The name used to register a SIP extension.

Caller ID name If you set the caller ID name, the called party will
see this caller ID name when the extension user
makes an outgoing call.
Concurrent Registrations Yeastar Cloud PBX supports to register one

extension number on multiple phones. When a call
reaches the extension number, all phones will ring.
Registration Password The password is used to register a SIP extension.

The password is generated randomly by default.
User Information Settings
Email Enter the email address. Extension user can reset

his/her login password, receive voice mails, faxes,
or PBX notifications via this email address.
User Password The password is used to log in the PBX or log in

Linkus mobile client. The password is generated
randomly by default.
Prompt Language The language of voice prompts. The default prompt

language is the same as the system language. If the
extension user speaks foreign language, you can
set a specific system prompt.
Mobile Number Enter the mobile number. Extension user can

receive the PBX notifications or forwarded calls on
this mobile number.
Presence Settings
Extension Presence indicates the availability status of a SIP extension. Presence settings
are linked to the Call Forwarding rules and Linkus ring strategy. You can set different call
forwarding rules and ring strategy for each presence status.
Navigation path: Settings > PBX > Extensions, edit a SIP extension under the Presence
tab.
Presence Settings
Settings Description
Presence Set presence status.
Yeastar Cloud PBX provides five presence statuses.
• Available: You are online and ready for

communication.
• Away: You are currently away from your office.
• DND: You don't want to be disturbed, and you
won't receive any calls.
• Lunch Break: You are currently on lunch
break.
• On a Business Trip:You are currently on a
business trip.
Presence Information Add details about your current status.
Call Forwarding Settings
You can forward calls to a specific destination or a specific extension user to avoid missing
calls. Depending on the presence status and your preferences, you can set the PBX to
forward calls to voicemail, extension, mobile number, queue, etc.
Always Forward all calls to the designated destination.
No Answer Only forward the unanswered calls to the

designated destination.
When Busy Only forward the calls that come in while you are
talking on the phone.
Ring Strategy Settings
You can set ring strategy for the following terminals that the SIP extension registered to.
• Extension
• Linkus Mobile Client
• Linkus Desktop Client
Ring First Set which terminal will ring first.
Ring Secondly Set which terminal will ring secondly.
Features Settings
You can configure voicemail, mobility extension, call monitoring, and other settings under
the Features tab.
Navigation path: Settings > PBX > Extensions, edit a SIP extension under the Features
tab.
Voicemail Settings
Enable Voicemail Enable voicemail feature.
Voicemail Access PIN Password used to access voicemail.
Share Voicemail Status Enable this option to share voicemail status of this
extension with other extensions.
Send Voicemail to Email Whether to send voicemail to the designated Email

address or not.
• Disabled: Do not send voicemail to the

designated Email address.
• Send to user's mail: Send voicemail to the
email address of the extension user.
• Send to custom mail: Customize an email
address, and the PBX will send the voicemail to
the designated Email address.
Busy Prompt Set the prompt that will be played when the
extension user is busy in a call.
Unavailable Prompt Set the prompt that will be played when the
extension user is unavailable.
Mobility Extension
Yeastar Mobility Extension allows you to stay in contact with colleagues and clients using
either office phone or mobile phone with the same extension number.
Ring Simultaneously Enable this option to allow both extension and

associated mobile number ring simultaneously
when anyone calls in the extension number.
Enable Mobility Extension Enable this option to allow your mobile number
have the same permission as the office phone when
you use associated mobile number to call in the
PBX.
Mobility Extension
• Set Mobile Number: Set the associated mobile
number.
• Prefix: Set the prefix of the mobile number
according to the outbound route.
Monitor Settings
Call Monitoring allows authorized users to monitor another extension user's call in real time.
Allow Being Monitored Enable this option to allow anyone to monitor the
extension user's ongoing call.
Monitor Mode Decide how you monitor other extension users'

ongoing call.
• Disabled
You can not monitor other extension users'

ongoing call.
• Extensive
Use any one of listen, whisper, or barge-in

mode to monitor other extension user's ongoing
call.
• Listen
Listen to a call in real time, but you can not talk

with the monitored extension users.
• Whisper
Listen to a call in real time, and talk with the

monitored extension users privately.
• Barge-in
Listen to a call in real time and talk with both

parties.
Other Settings
Ring Timeout (s) Set the timeout in seconds. Phone will stop ringing
after timeout.
Max Call Duration (s) Set the maximum call duration in seconds for every
call of this extension.
Note:
The precedence of Max Call Duration(s) (Global

v.s. Extension):
• For internal calls: The Max Call Duration(s)

setting of the caller's extension takes
precedence.
• For outbound calls: The Max Call Duration(s)
setting of the caller's extension takes
precedence.
• For inbound calls: The global Max Call
Duration(s) setting takes precedence.
Send email notification when extension user Enable this option to send email notification when
password is changed extension user password is changed.
Advanced Settings
The advanced settings of SIP extension require professional knowledge of SIP protocol.
Incorrect configurations may cause calling issues. It is wise to retain the default settings
provided on the SIP extension page. However, for a few fields, you need to change them to
suit your situation.
Navigation path: Settings > PBX > Extensions, edit an extension under the Advanced
tab.
VoIP Settings
Qualify Enable this option to send SIP OPTION packet to

SIP device to check if the device is up.
Enable SRTP Enable SRTP for voice encryption.
T.38 Support Enable or disable T.38 fax for the extension.

DTMF Mode Set the default mode for sending DTMF tones.
• RFC4733 (RFC2833): DTMF will be carried in

the RTP stream in different RTP packets.
• Info: DTMF will be carried in the SIP info
messages.
• Inband: DTMF will be carried in the audio
signal.
• Auto: The PBX will detect if the device
supports RFC4733(RFC2833) DTMF. If
RFC4733(RFC2833) is supported, PBX will
choose RFC4733(RFC2833), or the PBX will
choose Inband.
Transport Set the transport protocol.
• UDP
• TCP
• TLS
Enable User Agent Registration Authorization
Enable User Agent Registration Authorization Whether to restrict user agents from registering to
the extension.
User Agent Enter the name of user agent. If the prefix of

the user agent does not match the value, the
registration will fail.
IP Restriction
Enable IP Restriction This option is used for IP access control. Only the
IP address or IP section that matches the settings
can register the extension number.
Permitted IP/Subnet mask Enter the IP address and subnet mask.
• 192.168.5.100/255.255.255.255
In this example, only the device whose IP

address is 192.168.5.100 can register the
extension number.
• 192.168.5.0/255.255.255.0
In this example, only the devices whose

IP section is 192.168.5.0 can register the
extension number.
Call Permission Settings

You can set the outbound call permissions for the SIP extension.
Navigation path: Settings > PBX > Extensions, edit a SIP extension under the Call
Permission tab.
Outbound Routes Set outbound routes for the extension.
Outbound Restriction Enable this option to prohibit this extension from

making outbound calls.
Contacts
Contacts Overview
Yeastar Contacts feature allows you to add external contacts to Company Contacts and
share the Company Contacts with your organization. Each extension user has a Personal
Contacts to create and manage their personal contacts.
Contacts types
Company Contacts
Company Contacts is a phone book that allows you to store a list of external
contacts, such as the company's customers, resellers and partners.
Note: By default, only the PBX administrator can view and manage
Company Contacts. To share Company Contacts with extension users, refer to
Configure Company Contacts Permissions for Users.
Personal Contacts
Personal Contacts is a phone book for each extension user. Users can store a
list of external contacts exclusive to themselves, such as direct customers.
Note: Each user's Personal Contacts is visible only to themselves.
Key features
Sync contacts between Linkus clients and PBX
The contacts information is synced automatically between Linkus clients and

PBX.
Users can manage contacts both on Linkus and PBX web page.
Note: Requirements of Linkus clients:
• Linkus Android Client: 2.9.6 or later.

• Linkus iOS Client: 2.9.10 or later.
• Linkus for Mac: Coming soon.
• Linkus for Windows: Coming soon.
For more information of contacts management, see Manage Company

Contacts and Manage Personal Contacts.
Import and export contacts
Save time and effort by importing and exporting contacts entries.
For more information, see Manage Company Contacts and Manage Personal
Contacts.
Identify incoming calls
The contact's name is displayed for incoming calls to your Linkus, desk phone,
or other softphones if the contact's information is saved in Company Contacts
or Personal Contacts. By knowing who’s calling, the users can handle the calls
efficiently.
For more information, see Identify Callers from Contacts.

Configure Company Contacts permissions for users
Control who can view and manage the Company Contacts.
For more information, see Configure Company Contacts Permissions for

Users.
Contacts limits
The following table shows the maximum number of contacts supported on the PBX.
Contacts type Maximum number
Company contacts (total) 1,000
Personal contacts (per extension) 300
Manage Company Contacts

This topic describes how to add, edit, delete, import, and export company contacts on PBX
web page.
Requirements
Only the PBX administrator and the authorized users can manage Company Contacts.
For more information of Company Contacts permissions, see Configure Company Contacts
Permissions for Users.
Web operations vs. Linkus operations

The authorized users can manage company contacts on both Web and Linkus, but the
operation permissions are different.
For more information of Contacts on Linkus, see Linkus Mobile Help (Me > Settings > Help
& Feedback).
Operations Web Linkus
Add √ √
Edit √ √
Delete √ √
Export √ ×
Import √ ×
Add a company contact
1. Go to Contacts > Company Contacts.

2. Click Add.
3. Enter the contact information.
Note: The First Name, Last Name are required fields, and at least one number is
required.
4. Click Save.
Edit a company contact

2. Select a contact, and click .
3. Edit the contact information.
4. Click Save.
Delete company contacts

2. To delete a single contact, select the contact and click .
3. To delete multiple contacts, select the checkboxes of the desired contacts, and click
Delete.
Export company contacts

2. Click Export.
All the contacts will be exported to a CSV file.
Import company contacts

Before you begin
• Prepare a CSV file
To import contacts, you can export contacts to a CSV file.
Use the CSV file as a template, save your data in the same format. For the data
requirements in the CSV file, see Import Parameters - Contacts.

2. Click Import.
3. In the pop-up dialog, click Browser, and select your CSV file.
4. Click Import.
If the contact data is imported successfully, the web page will display the following
confirmation.
If you get an error prompt like the following figure, click Yes to check the log and
update your data in the CSV file.
Manage Personal Contacts

This topic describes how to add, edit, delete, import, and export personal contacts on PBX
web page.
Web operations vs. Linkus operations

Users can manage personal contacts on both Web and Linkus, but the operation
permissions are different.
For more information of Contacts on Linkus, see Linkus Mobile Help (Me > Settings > Help
& Feedback).
Operations Web Linkus
Add √ √
Edit √ √
Delete √ √
Export √ ×
Import √ ×
Access Personal Contacts

Each extension user has a Personal Contacts phone book.
1. Log in PBX web interface using extension email and password.

• Username: Enter extension email.
• Password: Enter the User Password of extension.
2. On the PBX desktop, select Contacts.
The Personal Contacts is displayed.
Add a personal contact
1. Access Personal Contacts on Web.

2. On the Personal Contacts page, click Add.
3. Enter the contact information.
Note: The First Name, Last Name are required fields, and at least one number is
required.
4. Click Save.
Edit a personal contact

2. Select a contact, and click .
3. Edit the contact information.
4. Click Save.
Delete personal contacts

2. To delete a single contact, select the contact and click .
3. To delete multiple contacts, select the checkboxes of the desired contacts, and click
Delete.
Export personal contacts

2. Click Export.
All the contacts will be exported to a CSV file.
Import personal contacts

Before you begin
• Prepare a CSV file
To import contacts, you can export contacts to a CSV file.
Use the CSV file as a template, save your data in the same format. For the data
requirements in the CSV file, see Import Parameters - Contacts.

2. Click Import.
3. In the pop-up dialog, click Browser, and select your CSV file.
4. Click Import.
If the contact data is imported successfully, the web page will display the following
confirmation.
If you get an error prompt like the following figure, click Yes to check the log and
update your data in the CSV file.
Configure Company Contacts Permissions for Users

By default, only the PBX administrator can view and manage Company Contacts. To share
Company Contacts with your organization, you need to configure Company Contacts
permissions for the users in your organization.
Permissions
The PBX provides two permission levels: View and Manage.
View Company Contacts

The authorized users only have permissions to view the contacts information of
the Company Contacts.
Manage Company Contacts

The authorized users have permissions to view, edit, add, delete, import and
export the contacts of the Company Contacts.
Configure Company Contacts permissions for users
1. Go to Contacts > Settings.

2. To assign View permission to users, configure the section View Company Contacts.
Select the extensions from Available box to Selected box.

3. To assign Manage permission to users, configure the section Manage Company
Contacts.
Select the extensions from Available box to Selected box.
Note: Assign the Manage permission carefully to appropriate users. If a user delete
contacts accidentally, the contacts would be lost.
4. Click Save.
Identify Callers from Contacts

Yeastar Contacts feature allows users to identify incoming callers if the caller information is
saved in the Company Contacts or Personal Contacts.
Requirements
Identifying Caller ID is supported on all endpoints, including Linkus, desktop phone, and
other softphones.
Identifying callers from Company Contacts

Supported for the authorized users who have permissions to view or manage
the Company Contacts.
For more information of the permissions, see Configure Company Contacts

Permissions for Users.
Identifying callers from Personal Contacts
Supported for each extension user.
Priority of Caller ID matching

If an incoming number is stored in Company Contacts, Personal Contacts, and mobile
phone book at the same time, the priority of Caller ID matching is as follows:
1. Mobile phone book

2. Company Contacts
3. Personal Contacts
Configure Caller ID Match
1. Go to Contacts > Settings.

2. Select the checkbox of Enable Caller ID Match.
3. Specify to match the exact caller ID or minimum number of caller ID digits.
• Exact Match: Only when the incoming Caller ID matches exactly your existing
contact number will the contact name be displayed.
• Fuzzy matching: When the last few digits of the incoming Caller ID matches that
of your existing contact number, the contact name will be displayed. The default
value is 7.
4. In the Name Display Format field, select the contact display order.
• First Name Last Name
• Last Name First Name
Example
The contact Dora's phone number 12345678 is saved in Company Contacts.
• Exact Match is selected:
◦ If the incoming caller ID is 12345678, the contact name "Dora" will be displayed.
◦ If the incoming caller ID is +012345678, the contact name will not be displayed.
• Fuzzy matching last 8 digits is configured:
◦ If the incoming caller ID is +012345678, the contact name "Dora" will be displayed.
◦ If the incoming caller ID is 62345678, the contact name "Dora" will not be
displayed.
Contacts FAQ
• Cannot import my contacts
• Can I set a Contacts sub-administrator?
• Will my personal contacts be lost if I uninstall Linkus client?

• Can the administrator or other users see my personal contacts?
• How do extension users view or manage Company Contacts on PBX web page?
• Does IP phone support Contacts feature?
• Will contacts information be saved when I backup the PBX?
• Can I expand the capacity of Company Contacts?
Cannot import my contacts
1. Check if the contacts limit is reached. See Contacts limits.

2. Check if the imported file meets the format requirements: CSV file encoded in UTF-8
without BOM.
Can I set a Contacts sub-administrator?

Yes.
The PBX administrator can go to Settings > Permission to grant Contacts permission for
the desired user.
If the Contacts permission is assigned to an user, the user can do the following operations:
• Manage Company Contacts

• Configure Caller ID Match of Contacts
• Assign Company Contacts permissions to users
Will my personal contacts be lost if I uninstall Linkus client?

The personal contacts won't be lost.
After you create the personal contacts, the contacts is stored in PBX.
Can the administrator or other users see my personal contacts?

No. Personal contacts are visible to the owner.
How do extension users view or manage Company Contacts on PBX web

page?
1. Contact administrator to check if you are allowed to view or manage Company

Contacts.
2. Log in PBX web interface using extension email and password.
• Username: Enter extension email.
• Password: Enter the User Password of extension.
Does IP phone support Contacts feature?

By now, Caller ID match is supported on IP phones, which means that the system can help
you identify callers if their information is saved in Contacts.
You can not retrieve and view Contacts information on IP phones at present. But Contacts
feature will be supported on IP phones soon, and information can be synchronized among
the PBX server, Linkus Clients, and IP phones at that time.
Will contacts information be saved when I backup the PBX?

Yes.
Contacts information is stored on PBX, so it will be automatically saved when you back up
the PBX.
Can I expand the capacity of Company Contacts?

No.
Company Contacts is stored on PBX system disk, so you can not expand the capacity by
adding extra storage device.
Trunks
Trunk Overview
Making and receiving calls between internal extensions is one thing, but if you want to
receive and make calls to the outside world, you need at least a trunk to the outside world.
VoIP Trunks
VoIP Trunks Introduction
VoIP Trunks are phone lines that transmits calls over the Internet. A VoIP provider can
assign a local number to one or more cities or countries and route it to the PBX phone
system. Usually VoIP trunks are cheaper than traditional PSTN trunks.
VoIP Trunk Types

Yeastar Cloud PBX supports the following VoIP trunk types:
• VoIP Register Trunk: Registration based VoIP trunk. VoIP Register Trunk uses the
username and password for registration with SIP providers.
• VoIP Peer Trunk: Uses the IP address & port or domain of PBX for authentication or
connect PBX directly to VoIP provider's dedicated network. Your VoIP provider route
incoming and outgoing calls based on the DID number, PBX port or PBX domain, or
route calls by a private network.
• VoIP Account Trunk
Account Trunk is designed for connection between Yeastar Cloud PBX and other
devices. Yeastar Cloud PBX will act as a VoIP account provider, the other device
should register this account to connect to Yeastar Cloud PBX.
• WebRTC Trunk
A WebRTC (Web Real-Time Communication) trunk is used to set up WebRTC Click

to Call. After you create a WebRTC trunk on the Yeastar Cloud PBX, a link will be
generated automatically.
Create a VoIP Trunk
VoIP Trunk Creation Overview

This topic describes two methods by which to create a VoIP trunk.
VoIP Trunk Creation Methods

Yeastar Cloud PBX supports two methods to create a VoIP trunk.
Create a VoIP Trunk by a Template

Yeastar Cloud PBX supports leading VoIP Service Providers across the globe,
you can use the pre-configured VoIP templates included in Yeastar Cloud PBX
to set up a VoIP trunk quickly and easily.
Check the tested and supported VoIP providers.
For more information, see Create a VoIP Trunk by a Template.

Create a General VoIP Trunk
If your VoIP provider has not undergone an interoperability test by Yeastar, you
can set up a General VoIP trunk.
For more information, see the following topics:
• Create a VoIP Register Trunk - General

• Create a VoIP Peer Trunk - General

• Create a VoIP Account Trunk - General
Create a VoIP Trunk by a Template

If your VoIP trunk provider is tested and supported by Yeastar, you can create a VoIP trunk
by a template.
Procedure
1. Go to Settings > PBX > Trunks, click Add.

2. In the Name field, enter a trunk name.
3. From the Select Country drop down list, select the country that the VoIP provider
operates in.
4. From the ITSP drop down list, select the VoIP provider.
The pre-configured template is applied for the selected VoIP provider.

5. If your trunk is a Register Trunk, complete the following configurations:
a. On the Basic page, configure the following settings:
• Hostname/IP: Enter the IP address or the domain of the VoIP provider.

• Domain: Enter the IP address or the domain of the VoIP provider.
• User Name: Enter the username to register to the VoIP provider.
• Password: Enter the password that is associated with the username.
• Authentication Name: Enter the authentication name to register to the VoIP
provider.
• From User: Enter the same name as User Name.
b. Configure DID settings for the trunk:
To add signal DID:
i. Select Add Signal DID.

ii. Enter the DID Number which is provided by the VoIP provider.
iii. Select the checkbox of DNIS Name, enter a DNIS name for the DID number.
When users call the DID number, the DNIS name will be displayed on ringing
phone.
iv. Click and repeat steps i-ii to add another DID numbers.
To bulk add DIDs:
i. Select Bulk Add DID.

ii. Enter the DID Number Range which is provided by the VoIP provider, click
Add.
iii. Select the checkbox of DNIS Name, enter DNIS name for a DID number.
phone.
6. If your trunk is a Peer Trunk, complete the following configurations:

7. Configure other VoIP trunk settings as your need.
You can check the trunk status in PBX Monitor. If the trunk status shows , the trunk
is ready for use.
Create a VoIP Register Trunk - General

If your VoIP provider is not included in the supported VoIP provider list, and you have got
a VoIP account with user name and password, you can set up a Register Trunk on Yeastar
Cloud PBX.
Assume that you bought a SIP trunk from the VoIP provider, and the trunk information is
displayed as below. We will introduce how to set up a Register Trunk according to the trunk
information.
Provider address abc.provider.com
Protocol SIP
SIP Port 5060
Transport UDP
Username 254258255
Authenticate name 254258255
Password 05JsOmsIS54SYh
Provided DID numbers 5503301 / 5503302 / 5503303

3. In the Select Country drop-down list, select General.
4. In the Trunk Type drop-down list, select Register Trunk.
5. Enter the trunk information that is provided by the VoIP provider:
• Hostname/IP: Enter the IP address or the domain of the VoIP provider
(e.g.,abc.provider.com).
• Domain: Enter the IP address or the domain of the VoIP provider (e.g.,
abc.provider.com).
• User Name: Enter the username to register to the VoIP provider (e.g.,
254258255).
• Password: Enter the password that is associated with the username (e.g.,
05JsOmsIS54SYh).
• Authenticate: Enter the authentication name to register to the VoIP provider (e.g.,
254258255).
• From User: Enter the same name as User Name (e.g., 254258255).
6. Set DID numbers for the trunk#
a. Select Add Single DID.
b. Enter the DID Numbers which is provided by the VoIP provider.
c. Select the checkbox of DNIS Name, enter a DNIS name for the DID number.
phone.
d. Click and repeat steps i-ii to add another DID numbers.

is ready for use.
Related tasks
Add an Outbound Route
Add an Inbound Route
Set up DOD Numbers for VoIP Trunk
Create a VoIP Peer Trunk - General

If your VoIP provider is not included in the supported VoIP provider list, and the ITSP only
provides an IP address or domain for your purchased VoIP account, you can set up a Peer
Trunk on the Yeastar Cloud PBX.

4. In the Trunk Type drop-down list, select a type of Peer trunk.
Note: If you don't know which type to select, contact Yeastar support.
Peer Trunk Type Description
DID-based If the VoIP provider routes incoming calls and outgoing calls based on the DID
number, select DID-based VoIP trunk.
Port-based If the VoIP provider routes incoming calls and outgoing calls based on the SIP
registration port, select Port-based VoIP trunk.
Note: If this type is selected, a specific SIP registration port will be

assigned to the PBX. In this way, the VoIP provide can correctly route the
calls.
Domain-based If the VoIP provider routes incoming calls and outgoing calls based on the PBX
domain name, select Domain-based VoIP trunk.
Private Network If the PBX and the VoIP provider are in the same private network, select
Private-Network based VoIP trunk.
5. Enter the trunk information that is provided by the VoIP provider.

is ready for use.
Related tasks
Set up DOD Numbers for VoIP Trunk
Create a VoIP Account Trunk - General

Create a VoIP Account Trunk on the Yeastar Cloud PBX, and provide this account for the
other device to register. In this way, Yeastar Cloud PBX and the other device are connected.

3. In the Select Country drop down list, select General.
4. In the Trunk Type drop-down list, select Account Trunk.
5. Enter the account information as your need:
• Username: Use the default or change the number.
• Password: Use the default or change the number.
• Authentication Name: Use the default or change the number.
Note: The other device should use the provided trunk information to connect to the
Yeastar Cloud PBX.

After the Account Trunk is registered on the other device, you can check the trunk
status in PBX Monitor. If the trunk status shows , the trunk is ready for use.
Related tasks
Manage VoIP Trunks
Import the VoIP register Trunks

You can create multiple VoIP register trunks by importing a UTF-8 .csv file.
For requirements of the import parameters, see Import Parameters - Trunks.
1. Go to Settings > PBX > Trunks, click Import.

2. Click Download the Template, add the VoIP register trunks information in the template
file.
3. Click Browse to upload the template file, and then click Import.
Edit the VoIP Trunk
1. Go to Settings > PBX > Trunks.

2. Search and find your VoIP Trunk, click .
3. Click the desired tab to edit the VoIP Trunk Settings as your need.
Delete the VoIP Trunk
1. Go to Settings > PBX > Trunks.

2. Search and find your VoIP Trunk, click .
3. Click Yes to confirm the deletion.
VoIP Trunk Settings

When you configure a VoIP trunk, you may need to configure some of the advanced
settings. This reference describes all the settings on a VoIP trunk.
Basic Settings
Navigation path: Settings > PBX > Trunks, edit a trunk on the Basic tab.
Name Give this trunk a name to help you identify it.
Trunk Status Enable or disable the trunk.
Select Country Select the country that the VoIP provider operates in.
Trunk Type Select a trunk type.
Transport Select the transport that is provided by the VoIP provider.

Hostname/IP Enter the IP address or the domain of the VoIP provider.
Domain Enter the IP address or the domain of the VoIP provider.
Username Enter the username to register to the VoIP provider.
Authentication Name Enter the authentication name to register to the VoIP provider.
Password Enter the password that is associated with the username.
From User
Enter a name. All the outgoing calls from this trunk will use this name in From
header of the SIP invite package.
DID Number Direct Inward Dialing number, can be used to distinguish incoming calls.
DNIS Name Dialed Number Identification Service is a telephony service used to identify
which number was dialed.
Bind a DNIS name for a DID number, when users call the DID number, the
DNIS name will be displayed on ringing phone.
Caller ID Number
If you set the caller ID number, when users make outbound calls through this
trunk, the called party will see this caller ID number instead of the calling party's
number.
This feature requires support from the VoIP provider.
Caller ID Name If you set the caller ID name, when users make outbound calls through this
trunk, the called party will see this caller ID name instead of the calling party's
name.
This feature requires support from the VoIP provider.
Enable Outbound Proxy Set the outbound proxy if the VoIP provider needs.
Enable SLA After enabling SLA, users can share this trunk to make outbound calls and
receive inbound calls by BLF keys on their phones. In this way, Inbound Route
settings and Outbound Route settings for the trunk is invalid.
Advanced Settings
The advanced settings of VoIP trunk requires professional knowledge of SIP protocol.
Incorrect configurations may cause calling issues. It is wise to leave the default settings
provided on the VoIP trunk page. However, for a few fields, you need to change them to suit
your situation.
Navigation path: Settings > PBX > Trunks, edit a trunk on the Advanced tab.
VoIP Settings
Qualify Enable this option to send SIP OPTION packet to SIP device to
check if the device is up.
DTMF Mode Set the default mode for sending DTMF tones.
• RFC4733 (RFC2833): DTMF will be carried in the RTP stream in

different RTP packets than the audio signal.
• Info: DTMF will be carried in the SIP info messages.
• Inband: DTMF will be carried in the audio signal.
• Auto: The PBX will detect if the device supports
RFC4733(RFC2833) DTMF. If RFC4733(RFC2833) is supported,
PBX will choose RFC4733(RFC2833), or the PBX will choose
Inband.
Enable SRTP Enable or disable SRTP (encrypted RTP) for the trunk.
Send Privacy ID Whether to send the Privacy ID in SIP header or not.
T.38 Support Enable or disable T.38 fax for this trunk. Enabling T.38 will add the
performance cost.
We suggest that you disable T.38.
User Phone Whether to add the parameter user=phone in the SIP INVITE packet.
Note: Enable this option if the SIP provider requires.
Inbound Parameters
Get DID From Decide from which header field will the trunk retrieve DID header.
• [Follow System]
The trunk will follow the global Get DID From setting.
• TO
• INVITE
• Remote-Party-ID
Note: If this option is selected, but the SIP provider doesn't

support Remote Party ID, the PBX will retrieve DID from INVITE
header.
• P Asserted Identify
• Diversion
• P-Called-Party-ID
• P-Preferred-Identity
Get Caller ID Decide from which header field will the trunk retrieve Caller ID header.
From
• [Follow System]
The trunk will follow the global Get Caller ID From setting.
• From
• Contact
• Remote-Party-ID
• P Asserted Identify
Outbound Parameters
Configure SIP parameters for outbound calls.
• Default: The same as the value in "From".

• Trunk Username: The username you configured for the trunk.
• Extension Number: The extension number.
• DOD Number: The DOD number that you configured to associate with the
extension. If the extension doesn't have an associated DOD number, the
Caller ID Number of the trunk will be taken instead.
• From User: The From User value that you configured for the trunk.
• None: Do not send the parameter with the SIP INVITE packet.
Remote Party ID Select which Remote Party ID value should be contained in the SIP
INVITE headers when making an outbound call.
P Asserted Identify Select which P Asserted Identify value should be contained in the
SIP INVITE headers when making an outbound call.
Diversion Select which Diversion value should be contained in the SIP

INVITE headers when making an outbound call.
P-Preferred-Identity Select which P-Preferred-Identity value should be contained in the

SIP INVITE headers when making an outbound call.
Transfer Parameters
Configure the SIP parameters for transferred calls.
• Default: The same as the value in "From".

• Trunk Username: The username you configured for the trunk.
• Extension Number: The extension number.
• DOD Number: The DOD number that you configured to associate with the
extension. If the extension doesn't have an associated DOD number, the
Caller ID Number of the trunk will be taken instead.
• The Originator Caller ID: The Caller ID Number of the first caller in cases
that the call is transferred.
• From User: The From User value that you configured for the trunk.
• None: Do not send Remote Party ID with the SIP INVITE packet.
Select which From value should be contained in the SIP INVITE

From
headers when the call is transferred.
Select which Diversionvalue should be contained in the SIP INVITE

Diversion
headers when the call is transferred.
Select which Remote Party ID value should be contained in the SIP

Remote Party ID
INVITE headers when the call is transferred.
Select which P Asserted Identify value should be contained in the

P Asserted Identify
SIP INVITE headers when the call is transferred.
Select which P-Preferred-Identity value should be contained in the

P-Preferred-Identity
SIP INVITE headers when the call is transferred.
Other Settings
Maximum Set the maximum number of concurrent calls on the trunk.

Channels
Note: The value 0 means unlimited.
Realm
SIP Realms, also known as domains within SIP networks.
Realm is a component within SIP that is used to authenticate users

within the SIP registration process.
Note: By default, the Realm setting is unnecessary. Contact your

service provider if you want to configure Realm.
Inband Progress This Inband Progress setting applies to the extensions which make
calls through this trunk.
Note: To configure global Inband Progress setting, you need to

contact Yeastar support to configure a custom config file.
• Check this option: PBX will send a 183 Session Progress to the
extension when told to indicate ringing and will immediately start
sending ringing as audio.
• Uncheck this option: PBX will send a 180 Ringing to the extension
when told to indicate ringing and will NOT send it as audio.
Codec Settings
Each new created VoIP trunk has a default preferred codec list. However, the default codec
list may not match the codecs supported by your VoIP provider. In order to maximize the
quality of calls and the amount of bandwidth used for calls, you’ll want to choose and
configure your preferred codec list to match the settings that your VoIP provider supports.
Yeastar Cloud PBX supports the following codecs:
Disabled by default Enabled by default
GSM, G722, G726, ADPCM, H261, H263,

G729, G711 a-law, G711 u-law
H263P, H264, MPEG4, iLBC
Navigation path: Settings > PBX > Trunks, edit a trunk on the Codec tab.
Select Codec
In the Available box, double click a codec, the selected codec will appear in
the Selected box.
Set the Codec Priority
In the Selected box, click a codec, and click to change the priority.
Adapt Caller ID
The incoming caller ID that matches the adaptation pattern will be adapted, so that you can
press the call record directly on your phone call back a number.
For more information, see Change Inbound Caller ID.
Navigation path: Settings > PBX > Trunks, edit a trunk on the Adapt Caller ID tab.
Patterns The following characters have special meanings:
• X matches the numbers 0- 9;

• Z matches the numbers 1-9;
• N matches the numbers 2- 9;
• [12345-9] matches the numbers in the bracket (in this example, 1, 2, 3, 4,5, 6, 7, 8,
9);
• Wildcard matches one or more numbers. E.g. "9011." matches anything starting
with 9011 (excluding 9011 itself);
• Wildcard "!" matches none or more than one numbers. E.g. "9011T matches
anything starting with 9011 (including 9011 itself);
Strip Strip allows you to specify the number of digits that will be stripped from the front of
the Caller ID before the call is displayed. For example, if the incoming Caller ID is
05929999999, but you need to dial number 5929999999 to call back, one digit should
be stripped.
Prepend These digits will be prepended to the Caller ID before the call is displayed. For example,
if the incoming caller ID is 5929999999, but you need to dial digit 0 before the number
to call back, 0 should be prepended.
WebRTC Trunks
WebRTC Click-to-Call
WebRTC (Web Real-Time Communication) is a collection of communications protocols and
application programming interfaces that enable real-time communication over peer-to-peer
connections. Yeastar Cloud PBX supports WebRTC Click-to-Call that allows the website
visitors calling to a pre-configured destination by clicking a link/button the web page.
Supported Concurrent Calls

A WebRTC trunk supports up to 4 concurrent calls.
Supported Web Browser

We have tested the compatibility for the following browsers that support WebRTC
technology.
Note: The failed test is caused by WebRTC not being supported by the web browsers.
Browser Conditions/Limitations
Google Chrome Windows Desktop: √
Mac Desktop: √
Android Phone: √
iOS Phone: ×
Firefox Windows Desktop: √
Opera Windows Desktop: √
Mac Desktop: √
Android Phone: ×
iOS Phone: ×
Safari Safari browser doesn't support WebRTC.

Set up WebRTC Click-to-Call

Create a WebRTC trunk on the PBX, and place the generated link in your website. When a
website visitor clicks the link, a WebRTC call will be established between the visitor and the
pre-configured destination of the PBX.
1. Create a WebRTC Trunk
1. Enable WebRTC feature on the PBX.
Go to Settings > PBX > General > WebRTC, check the option Enable, click Save.

5. In the Trunk Type drop-down list, select WebRTC Trunk.
6. Use the default number or change the Trunk Number.
When a WebRTC call is made through this trunk, the trunk number will be displayed on
the ringing endpoint.
7. Click Save.
A link for the WebRTC trunk is generated in WebRTC Inbound Call Link. You can
place the link on your web page. When your website visitors click the link, they will be
connected to the destination of this WebRTC trunk.
8. On the pop-up dialog, click Copy Now or Copy Later.
2. Set WebRTC Call Destination

Create an inbound route for the WebRTC trunk to route the WebRTC incoming calls. When
the website visitors click to call from the web page, the calls will be routed to the configured
destination.
1. Go to Settings > PBX > Call Control > Inbound Routes, click Add.
2. Set WebRTC call destination.
• Name: Enter a route name.
• Member Trunks: Select the WebRTC trunk to the Selected box.
• Enable Time Condition: Select the checkbox of Enable Time Condition, and
configure time conditions to route the incoming calls based on the time conditions.
• Destination: Select the inbound route destination.
3. Place WebRTC Link on Your Website

Create an HTML button on your website, and set the button link to WebRTC link that is
generated after you creating the WebRTC trunk.
Note: To test the WebRTC Click-to-Call, you can paste the WebRTC link in the web
browser directly.
1. On the WebRTC turnk configuration page, click to copy the WebRTC link.
2. Paste the link your web browser, press Enter.
A dialpad will be displayed on the web page and the call will be connected to your pre-
configured destination.
WebRTC Trunk Settings

When you configure a WebRTC Trunk, you may need to configure some of the advanced
settings. This reference describes all the settings on a WebRTC Trunk.
Basic Settings
Navigation path: Settings > PBX > Trunks, edit WebRTC trunk on the Basic tab.
Name Enter the trunk name.
Trunk Status Enable or disable the trunk.
Trunk Type Select a trunk type.
Use the default number or change the Trunk Number.

Trunk Number
When a WebRTC call is made through this trunk, the trunk number will be
displayed on the ringing endpoint.
WebRTC Inbound Call Link Place the link on your web page. When your website visitors click the link,
they will be connected to the destination of this WebRTC trunk.
Codec Settings
Yeastar Cloud PBX supports a-law and u-law codecs.
Navigation path: Settings > PBX > Trunks, edit WebRTC trunk on the Codec tab.
Select Codec
In the Available box, double click a codec, the selected codec will appear in
the Selected box.
Set the Codec Priority
In the Selected box, click a codec, and click to change the priority.
Maximum Channel Settings

Navigation path: Settings > PBX > Trunks, edit WebRTC trunk on the Advanced tab.
Maximum Channels: Defines the maximum number of concurrent calls allowed in this
trunk.
Call Control
Emergency Numbers
Add an Emergency Number
To ensure that the extension users can make emergency calls at any time, you need to
add emergency numbers on Yeastar Cloud PBX. You can also set an alert to notify the
emergency contacts that an emergency call has been dialed.
Note: Emergency calls have the highest priority. If the trunk used to make emergency
calls is busy, the PBX will terminate the ongoing call, and place the emergency call.
1. Go to Settings > PBX > Emergency Number, click Add.
2. In the Emergency Number field, enter the emergency number.
3. In the Trunk field, set the trunk to make emergency calls.
a. In the drop-down list, select a trunk.
b. Optional: If the selected trunk needs a prepended number before the emergency
number, enter a prepended number in the Prepend field.
For example, if your trunk needs a prepended number 0 before the emergency
number 911, users should dial 0911 to make the emergency call. To comply with
the user's dialing habit, you can set the Prepend as 0. In this way, users can dial
911 as they usually do.
c. Optional: Click to add another trunk.
Note: If the first trunk cannot work properly, the PBX will use the second trunk
to make calls.
4. In the Notification field, select the notification contact.

If someone makes emergency calls through the PBX, the contacts will receive
notification calls on their extensions.
a. In the drop-down list, select a contact.
b. Optional: Click to add another contact.
Manage Emergency Numbers

After you add emergency numbers, you can edit or delete them.
Edit an emergency number
1. Go to Settings > PBX > Emergency Number, click beside the emergency number
that you want to edit.
2. Edit information of emergency number.
Delete an emergency number
1. Go to Settings > PBX > Emergency Number, click beside the emergency number
that you want to delete.
2. In the pop-up window, click Yes to delete the selected emergency number.
3. Click Apply.
Time Conditions
Time Conditions Overview
A Time Condition is a time group, which can be applied to outbound routes and inbound
routes. You can use Time Condition to control calls based on date and time.
What is a Time Condition used for?

A Time Condition contains a time group.
• Apply Time Condition to an Inbound Route
Time Condition is typically used to control the destination of an inbound call based on
the date and time.
You can select a Time Condition and set a corresponding destination for an inbound
route. When a call reaches the PBX, PBX will route the call to the destination when the
current system time matches the time defined in the Time Condition.
• Apply Time Condition to an Outbound Route
You can also apply Time Condition to an outbound route to limit when the users can
use the outbound route.
Set Time Conditions

A Time Condition is a time group, which can be applied to outbound routes and inbound
routes. This topic describes how to set office hours, non-office hours, and holidays on
Yeastar Cloud PBX.
Set office hours

Add a Time Condition according to your office hours. Apply this Time Condition to inbound
routes to route incoming calls during office hours to the corresponding destination.
1. Go to Settings > PBX > Call Control > Time Conditions > Time Conditions, click
Add.
2. In the Name field, enter a name to help you identify it.
3. In the Time field, set the time according to your office time.
4. Click to add another time period.
5. In the Days of Week field, select your office days.
6. If you want to apply the time period(s) to specific dates, select the checkbox of
Advanced Options, and set the month and the days of month.
Note: Advanced Options is disabled by default, which means that the time
period(s) will be applied throughout the year.

Set non-office hours

PBX has a default Time Condition-Other Time. Generally, when you're configuring an
inbound route, you can set one destination for office hours, and set the other destination for
Other Time.
However, you may need to add another Time Condition to route incoming calls to other
destinations due to company's schedule. For example, you want all incoming calls during
lunch break to be routed to the receptionist. In this way, employees can enjoy nap time
without missing any important calls.
In this case, you can add another Time Condition for non-office hours.
1. Go to Settings > PBX > Call Control > Time Conditions > Time Conditions, click
Add.
3. In the Time field, set the time according to your non-office time.
4. Click to add another time period.
5. In the Days of Week field, select your office days.
6. If you want to apply the time period(s) to specific dates, select the checkbox of
Advanced Options, and set the month and the days of month.
Note: Advanced Options is disabled by default, which means that the time
period(s) will be applied throughout the year.
Set holidays
You can add a group of holidays and set a Time Condition destination like an IVR for the
holidays on your inbound route. When a customer calls to your company during holidays,
the PBX will route the call to the IVR and inform your customers that you are on vacation.
1. Go to Settings > PBX > Call Control > Time Conditions > Holiday, click Add.
3. In the Type field, select a type.
• By Date: If the holiday such as Chinese Spring Festival varies every year, select
this type.
• By Month: If the holiday such Chinese National Day always falls on the same
calendar date, select this type.
• By Week: If the holiday such as Thanksgiving Day always falls on the same week,
select this type.
4. In the Start Date field, select the start date of the holiday.
5. In the End Date field, select the end date of the holiday.
Manage Time Conditions

After you create Time Conditions, you can apply them to inbound routes or outbound routes.
You can also edit or delete the Time Conditions.
Apply a Time Condition to an Inbound Route

You can apply a Time Condition to an inbound route to route inbound calls to different
destinations according to your business hours and schedule.
1. Go to Settings > PBX > Call Control > Inbound Routes, click beside the inbound
route that you want to edit.
2. On the Inbound Route page, select the checkbox of Enable Time Condition.
3. Click , and select a Time Condition from the drop-down list.
4. Select destination from the drop-down list.
Inbound calls will be routed to the pre-configured destination if the date and time of the
calls match the time condition.
Apply a Time Condition to an Outbound Route

You can apply a Time Condition to an outbound route to limit when the extension users can
make outbound calls.
1. Go to Settings > PBX > Call Control > Outbound Routes, click beside the
outbound route that you want to edit.
2. On the Outbound Routes page, select the Time Condition which will be applied to the
outbound route.
Only in this time period can extension users make outbound calls via this outbound
route.
Edit a Time Condition
1. Go to Settings > PBX > Call Control > Time Conditions, click beside the Time
Condition that you want to edit.
2. Change Time Condition settings according to your needs.
Delete a Time Condition

After deleting a Time Condition, related configurations of the Time Condition in both inbound
routes and outbound routes will be deleted automatically.
1. Go to Settings > PBX > Call Control > Time Conditions, click beside the Time
Condition that you want to delete.
2. On the pop-up window, click Yes and Apply.
Time Condition Examples

In this topic, we offer you configuration examples of Time Conditions to help you understand
how to set office hours, non-office hours, holidays and apply these Time Conditions to
inbound routes and outbound routes.
Office hours & non-office hours example

Assume that your office hours are Monday - Friday from 9:00 to 18:00, and the lunch break
starts from 12:00 to 13:00.
According to your office hours, you can set two Time Conditions as follows..
• Office hours
• Lunch break
Holiday examples
Yeastar Cloud PBX supports 3 types of holidays.
• Set a Holiday by Date
If date of a holiday varies every year, you can set a holiday by date.
For example, Chinese Spring Festival falls on February 15th-21st. You can set the
holiday as follows.
• Set a Holiday by Month
If a holiday always falls on the same date, you can set a holiday by month.
For example, Christmas falls on December 25th every year. You can set the holiday as
follows.
• Set a Holiday by Week
If a holiday always falls on the same week, you can set a holiday by week.
For example, Thanksgiving Day falls on the 4th week of November. You can set the
holiday as follows.
Route inbound calls based on Time Conditions
On Inbound Route page, enable Enable Time Condition, click to add Time Conditions,
and set corresponding destinations.
For example, the following table is a schedule of Time Conditions for a company.
Time Condition Destination
Office hours IVR
Lunch break Extension 1000
Holiday Holiday IVR
Other time Voicemail
Note: All holidays will be integrated into one Holiday, you don't have to select holidays
one by one from Time Condition on inbound routes.
You can set Time Conditions as follows.
Restrict when to make outbound calls

On Outbound Routes page, select Time Condition, which means that only in this time period
can extension users make outbound calls via this outbound route.
Time Condition Override

The Time Condition Override function is used to switch the inbound call routing against the
Time Condition. An authorised user can dial Time Condition feature code to override the
time condition.
Scenarios
Company A sets day time condition and night time condition in an inbound route with
different destinations.
The staffs occasionally leave early or someone needs to enable the night time condition
manually. In this scenario, the staffs can dial override feature code to override the time
condition.
Time Condition feature code

When you enable and add Time Condition on an inbound route, you will see the default
generated feature code for the Time Condition. If you want to disable Time Condition
Override, dial the Reset feature code *800.
You can go to Settings > PBX > General > Feature Code > Time Condition to change the
feature code prefix.
Set extension permission to override Time Condition

By default, users have no permission to override Time Condition. You can set which
extension users can override Time Condition.
1. Go to Settings > PBX > General > Feature Code > Time Condition, click Set
Extension Permission.
2. Select the desired extensions from Available box to Selected box.

Monitor Time Condition State

You can set a BLF key on your phone to quickly override Time Condition and monitor the
Time Condition state.
We take Yealink T53W v95.0.0.0.0.0.1 as an example to explain how to set BLF keys to
monitor Time Condition state.
1. Set Time Condition Override permission for the extension that is registered on the IP
phone.
a. Log in PBX interface, go to Settings > PBX > General > Feature Code > Time
Condition, click Set Extension Permission.
b. Select the desired extension from Available box to Selected box.
2. Set BLF keys on the phone where the extension is registered.
a. Log in the phone web interface, go to DSS Key > Memory Key.
b. Set Key Type as BLF.
c. Set Key Value as feature code of Time Condition.
d. Select the Line as the extension registered line.
e. Optional: In the Extension field, enter a description of the key.
f. Click Confirm.
The BLF LED will show the Time Condition state.
• Red: The PBX is using this Time Condition; inbound calls go to the destination of
the Time Condition.
• Green: This Time Condition is not in use.
3. Press a BLF key to override Time Condition, the BLF LED turns to red.
You can also log in the PBX web interface, and check the Time Condition state on
configuration page of Inbound Routes. If the state shows , it indicates that the PBX
is using the Time Condition, and route all incoming calls to destination of the Time
Condition.
Inbound Routes
Inbound Route Overview
An inbound route is used to tell the PBX where to route inbound calls based on the caller's
phone number or the DID number. Inbound routes are often used in conjunction with time
conditions and an IVR.
DID routing & Caller ID routing

Yeastar Cloud PBX allows two specific types of inbound routing: DID Routing and Caller ID
Routing. You can set both DID routing and Caller ID routing for an inbound route, or set one
of the routing types.
If you don't specify DID numbers and Caller ID numbers on the inbound route, the inbound
route will match and route all inbound calls to a pre-configured internal destination on the
PBX.
Inbound routes can send inbound calls to destinations as follows:
• Hang up
• Extension
• Extension Range
• Voicemail
• IVR
• Ring Group
• Queue
• Conference
• DISA
• Callback
• Outbound Route
• Fax to Email

To receive external calls on Yeastar Cloud PBX, you need to set up at least one inbound
route.
The PBX has a default inbound route. When users call to the selected trunk, the PBX will
route the call to an IVR. You can delete the default inbound route, then add a new one to
configure settings according to your needs.
1. Go to Settings > PBX > Call Control > Inbound Routes, click Add.
3. Optional: In the DID Pattern field, enter a DID number or a DID pattern if you want to
route inbound calls based on DID numbers.
The PBX will route the call only when the caller dials the matched numbers.
Note: Leave this blank to match calls with any or no DID info.
4. Optional: In the Caller ID Pattern field, enter a Caller ID or a Caller ID pattern if you
want to route inbound calls based on Caller IDs.
The PBX will route the call only when the caller ID number matches the Caller ID
Pattern.
Note: Leave this blank to match calls with any or no caller ID info.
5. In the Member Trunks field, select the desired trunk from Available box to the
Selected box.
The PBX will route the inbound call when the caller calls the number of the selected
trunk.
6. If you allow the inbound calls to be routed to a desired destination without time limit,
configure the following settings:
a. Uncheck the checkbox of Enable Time Condition.

b. Select the Destination.
7. If you allow the inbound calls to be routed to different destinations based on time
condition, configure the following settings:
a. Select the checkbox of Enable Time Condition.
b. Click , select a Time Condition and the destination.
If an inbound call reaches the PBX during the time period, PBX will route the call to
the selected destination.
c. Optional: Click to set another time condition and destination.
d. Set the destination for Other Time.
If an inbound call reaches the PBX beyond the time periods that are defined in the
above Time Conditions, PBX will route the call to the selected destination.
8. Optional: In the Distinctive Ringtone field, enter the ringtone name. Distinctive
Ringtone helps users recognize where the call is from.
Note: Distinctive Ringtone feature needs support from the IP phones.
For example, the IP phone has a ringtone called "Family". You can enter "Family" in
the Distinctive Ringtone field. When a call reaches the IP phone through this inbound
route, the IP phone plays the "Family" ringtone.
9. Optional: Select the checkbox of Enable Fax Detection. PBX will send the fax to Fax
Destination if a fax tone is detected.
• Extension: PBX will send the fax to Fax Destination if a fax tone is detected.
• Fax to Email: PBX will send the fax as an attachment to the specified email
address. An email address can be associated with extensions or be customized
address.
Note: If you want to send fax to email, make sure system email is configured
correctly.
Manage Inbound Routes

After you create inbound routes, you can adjust the priority of the inbound routes. You can
also edit or delete the inbound routes.
Adjust priority of inbound routes

A trunk can be selected to multiple inbound routes. When users call to the selected trunk,
the PBX will route the call through the inbound route with higher priority. You can adjust the
priority of inbound routes according to your needs.
1. Go to Settings > PBX > Call Control > Inbound Routes.

2. Click to adjust the priority of your inbound routes.
Edit an inbound route

2. Click beside the inbound route that you want to edit.
3. Edit the inbound route.
Delete an inbound route

2. Click beside the inbound route that you want to delete.
Import Inbound Routes

You can import inbound routes to quickly set up inbound routing on Yeastar Cloud PBX.
1. Go to Settings > PBX > Call Control > Inbound Routes, click Import.
2. Click Download the Template, add the inbound routes information in the template file.
Note:
• The imported file should be a UTF-8 .csv file.
• For requirements of the import parameters, refer to Import Parameters - Inbound
Routes.
3. Click Browse to upload the template file.

4. Click Import.
Change Inbound Caller ID

By default, the Inbound caller ID on Yeastar Cloud PBX displays the caller's phone number,
you can change the inbound caller ID with Adapt Caller ID feature.
Adapt Caller ID feature is supported on each trunk. Go to Settings > PBX > Trunks, click
Adapt Caller ID tab on the trunk edit page to configure the settings.
Example 1
Company A wants to add a digit 0 to the 11-digit incoming caller ID number that begins with
digit 1 for quick redial purposes.
For example, company A wants to display 012345678910 instead of 12345678910.
In this case, you can configure Adapt Caller ID on trunk 1, and set the rules as follows:
• Patterns: 1.
• Strip: Leave it blank.
• Prepend: 0
Example 2
Company B wants all Xiamen numbers to be displayed as local number without Xiamen
area code (0592) that is received through the trunk 2.
For example, company B wants to display number 5503301 instead of 05925503301.
In this case, you can configure Adapt Caller ID on trunk 2, and set the rules as follows:
• Patterns: 0592.
• Strip: 4
• Prepend: Leave it blank.
Inbound Route Examples
Inbound Route Examples

This topic provides sample configurations that will help you understand DID setting and
Caller ID setting of inbound routes.
Note: The following examples ignore time condition, you can set time condition
Inbound route without limit

Any calls to the selected trunk will be routed to the inbound route destination. You can set
an inbound route as follows:
• Name: Set a name to help you identify it.

• Member Trunks: Select desired trunk(s).
• Destination: Set the destination.
Leave all other fields blank.
Inbound route based on a DID number

If a trunk has multiple DID numbers, you can add multiple inbound routes that based on
different DID numbers. When users dial different DID numbers, they will be routed to
different destinations.
The following example shows an inbound route based on DID number 5503301.
• Name: Set a name to help you identify it. For DID routes, you can set the name as the
DID number, which helps you identify the route.
• DID Pattern: 5503301
• Member Trunks: Select the trunk that has the DID number.
• Destination: Set the destination.
Inbound route based on consecutive DID numbers

If a trunk has multiple consecutive DID numbers, you can quickly set the DID number range
in an inbound route to route calls to different destinations based on the DID numbers.
The following example shows an inbound route based on DID range 5503301-5503305,
which will route calls to extension 1001-1005.

• DID Pattern: 5503301-5503305
• Member Trunk: Select the trunk that has the DID numbers.
• Destination: Select Extension Range, and enter the extension range 1001-1005.
Inbound route based on Caller ID

By default, PBX routes inbound calls without limit. If you set Caller ID Pattern, PBX will
route calls only when the users' caller ID numbers match the Caller ID Pattern.
In the following example, the inbound route will route caller ID numbers that start with digit
1 to the destination. For example, number 532352584 that doesn't start with digit 1 can not
call in the system through this inbound route.
• Caller ID Pattern: 1.
• Member Trunks: Select desired trunk(s).
• Destination: Select a destination.

Inbound route based on Caller ID and DID numbers

If you set both DID pattern and Caller ID pattern for an inbound route, PBX will check if the
DID numbers and the user's caller ID number match the DID pattern and Caller ID pattern.
Only the matched incoming calls can be routed to the pre-configured destination.
In the following example, when users dial 5503301 with phone number starting with digit 1,
the inbound call will be routed to the destination.
• Caller ID Pattern: 1.
• DID Pattern: 5503301
• Member Trunk: Select desired trunk(s).
• Destination: Select a destination.
Route Inbound Calls Based on DID

This topic describes what is DID numbers and how to configure inbound routes on Yeastar
Cloud PBX to route inbound calls based on DID.
DID numbers
DID (Direct Inward Dialing) is a telephone service that allows outside users to reach a
certain destination instead of going to a receptionist or a queue and needing to dial an
extension number.
DID numbers are provided by the trunk provider.
The trunk provider usually assigns a range of numbers to the VoIP trunk or the physical
trunk. There is an extra charge for the DID numbers. Contact your trunk provider for more
information about DID numbers.
Configure DID routing - single DID

Bind a DID number to an inbound destination.
Example:
You purchased two DID numbers from the SIP trunk provider: 5503301 and 5503302.
To route inbound calls to different destinations based on different DID numbers, you can set
up two inbound routes for the two DID numbers.
1. Inbound Route ToSales for DID number 5503301.


• DID Pattern: Enter the DID number 5503301.
• Caller ID Pattern: Leave it blank, which means no limit on caller's Caller ID.
• Member Trunks: Select the trunk that is bound with the DID number.
• Destination: Select the desired destination. When users dial the DID number
5503301, the call will be routed to the destination.
2. Inbound Route ToSupport for DID number 5503302.
• DID Pattern: Enter the DID number 5503302.

• Member Trunks: Select the trunk that is bound with the DID number.
• Destination: Select the desired destination. When users dial the DID number
5503302, the call will be routed to the destination.
Configure DID routing - multiple DIDs

You can assign DID numbers to extension users one by one. When an outside user dials an
DID number, the user can reach a specific extension directly.
Note: The DID numbers should be consecutive DID numbers.
Example: You purchased 10 DID numbers from the SIP trunk provider: 8823201-8823210.
To assign the DID numbers one by one to extension 1001-1010 , you can configure the
inbound route as follows.
• DID Pattern: Enter the DID range 8823201-8823210.

• Member Trunks: Select the trunk that is bound with the DID numbers.
• Destination: Select Extension Range, and enter the extension range 1001-1010.
Note: The number of extensions and DID numbers must be the same.
Route Inbound Calls Based on Caller ID

This topic describes what is Caller ID routing and how to configure inbound routes on
Yeastar Cloud PBX to route inbound calls based on Caller ID.
Caller ID routing
Caller ID (Caller Identification) is a telephone service that displays a caller's phone number
on the called party's phone device before the call is answered.
Caller ID routing allows users to accept or reject calls based on the caller's phone number.
Inbound calls which match the Caller ID pattern on PBX will be routed to the pre-configured
destination. For those unmatched, calls can not be established.
Scenarios
A company is dedicated to offering targeted service for different regions, the company
hopes that the Caller ID of inbound calls can be identified and the calls can be routed to
responsible employees. In this case, you can set Caller ID patterns for inbound routes.
Configuration Example
Company A assigns pre-sales business in France to Rose, and pre-sales business in
America to Mike. Refer to the following table and related configuration figures.
Name Extension Responsible Country Area Code
Rose 1000 France 0033
Mike 2000 America 001
Configure Caller ID pattern for Rose


• Caller ID Pattern: Enter the caller ID pattern 0033..
• Member Trunks: Select the trunk that is bound with the caller ID pattern.
• Destination: Select the desired destination. When a caller calls to the trunk with the
caller ID starting with 0033, the call will be routed to extension 1000.
Configure Caller ID pattern for Mike


• Caller ID Pattern: Enter the caller ID pattern 001..
• Member Trunks: Select the trunk that is bound with the caller ID pattern.
• Destination: Select the desired destination. When a caller calls to the trunk with the
caller ID starting with 001, the call will be routed to extension 2000.
Distinguish Inbound Calls
Distinguish Inbound Calls by Ring Tones

Distinctive ringtone distinguishes calls from different inbound routes. You can set distinctive
ringtones on different inbound routes. When a user hears the ringtone of an incoming call,
he/she may notice the intention of the call.
Note: Distinctive Ringtone feature needs support from the IP phones. We take Yealink
phone as an example.
1. Log in the phone web interface, go to Settings > Ring, select a ringtone and set the
name.
a. In the Internal Ringer Text field, enter the ringtone name.

b. In the Internal Ringer File drop-down list, select a ringtone file.
c. Click Confirm to save the settings.
2. Log in the PBX web interface, go to Settings > PBX > Call Control > Inbound
Routes, select an inbound route to edit.
a. In the Distinctive Ringtone field, enter the ringtone name that is configured on IP
phone.
b. Click Save and Apply.
When a call comes through the inbound route, the phone will play corresponding ringtone.
Distinguish Inbound Calls by DNIS Name

DNIS (Dialed Number Identification Service) is used to identify where the incoming call is
from. You can set different DNIS names for different trunks or set different DID numbers and
DNIS names for a trunk. When external users make outbound calls to PBX, extension users
can identify incoming call by DNIS name.
1. Go to Settings > PBX > Trunks, click beside the trunk that you want to edit.
2. On the Basic page, enter the trunk DID number, and set the DNIS Name for the DID
number.
Note: For VoIP Peer Trunk, click Advanced tab to find the DID settings.
3. If the trunk has another DID number, click to add a DID number and set a DNIS
name.
For example, a VoIP trunk has 3 DID numbers. 5503301 for Support, 5503302 for
Sales, and 5503303 for Marketing. When external users dial a DID number, extension
users can notice the intention by DNIS name displayed on an IP phone.
Make a call to the trunk of the PBX, the user who receives the call will see the incoming
caller ID and the DNIS name of the trunk.
Distinguish Inbound Calls by Caller ID

When inbound calls are routed from a ring group/queue or an IVR, Yeastar Cloud PBX can
display the name of ring group/queue/IVR. When the extension user receives a call from the
ring group/queue/IVR, he/she may notice the intention of the inbound call.
For example:
Set up two Ring Groups according to your organization, one is named as Sales, the other is
named as Support.
You can set up two inbound routes to route incoming calls to different destinations by
different trunks, and enable Distinctive Caller ID feature.
• When external users call to PBX, and IP phones of Sales members ring, "Sales" will be
displayed on IP phones.
• When external users call to PBX, and IP phones of Support members ring, "Support"
will be displayed on IP phones.
1. Go to PBX > General > Preferences, select the checkbox of Distinctive Caller ID.
Outbound Routes
Outbound Route Overview
An outbound route is used to tell the PBX which extension users are allowed to make
outbound calls and which trunk to use for the outbound calls.
How does an outbound route work?

Every time user dials a number, PBX will do the following in strict order:
1. Examine the number user dialed.

2. Compare the dialed number with the pattern that you have defined in route 1.
• If it matches, PBX will route the call out using the associated trunk.
• If it does not match, PBX will match the number with the pattern that you have
defined in route 2, and so on .
Dial Patterns of Outbound Route

This topic describes dial pattern settings of Outbound Route to help you understand and
configure the dial patterns of Outbound Route.
Pattern
A pattern specifies routing rules to route a call based on the digits dialed by a user. The
PBX matches a dial pattern and routes the call out based on the dial pattern.
Pattern Description
X Refers to any digit between 0 and 9.
Z Refers to any digit between 1 and 9.
N Refers to any digit between 2 and 9.
[###] Refers to any digit in the brackets, example [123] would match the numbers 1, 2, or
3.
Range of numbers can be specified with a dash, example [136-8] would match the
numbers 1, 3, 6, 7, and 8.
. Wildcard . matches one or more numbers.
Example 9011. matches any numbers starting with 9011 (excluding 9011 itself).
Wildcard ! matches none or more than one characters.
Example 9011! matches any numbers starting with 9011 (including 9011 itself).
Strip
Strip is an optional setting, it defines how many digits will be stripped from the front of the
dialed number before the call is placed.
Example:
If you set Pattern as 9. and set Strip as 1.
If a user wants to call number 1588902923, he/she should dial 91588902923. The PBX will
strip digit 9 from the dialed number, and call the number 1588902923.
Prepend
Prepend is an optional setting. The prepend will be added to the beginning of a successful
match. If the dialed number matches the Pattern, the prepend will be added to the
beginning of the number before placing the call.
Example:
If a trunk requires 10-digit dialing, but users are more comfortable with 7-digit dialing, you
can prepend a 3-digit area code to all 7-digit phone numbers before the calls are placed.
Related information
Outbound Route Examples

To allow users to make outbound calls through trunks, you need to set up at least one
outbound route on the PBX.
The PBX has a default outbound route with dial pattern X. that allows users to dial any
outgoing numbers. You can delete the default outbound route, then add a new one to
configure settings according to your needs.
1. Go to Settings > PBX > Call Control > Outbound Routes, click Add.
2. On the configuration page, configure an outbound route according to your needs.

• Name: Enter a name to help you identify it.
• Dial Patterns: Used to match the digits that users dial. When the dialed numbers
match a dial pattern, PBX will route the call out through matched outbound route.
Pattern Description
X Refers to any digit between 0 and 9.
Z Refers to any digit between 1 and 9.
N Refers to any digit between 2 and 9.
[###] Refers to any digit in the brackets.
. Wildcard . matches one or more numbers.
! Wildcard ! matches none or more than one characters.

• Member Trunks: Select a trunk to make outbound calls. If the dialed number
matches a dial pattern of the outbound route, PBX will route the call out through
selected trunk.
• Extensions: Select which extensions are allowed to use this outbound route.
• Password: Optional. Set a password for the outbound route. If a password is
set, users are required to enter a password when they try to make outbound calls
through this route.
◦ None: No password is needed.
◦ PIN List: Select a PIN list. Users are required to enter a password in the PIN
list when they try to make outbound calls through this outbound route.
◦ Single Pin: Enter a password. Users are required to enter the password when
they try to make outbound calls through this outbound route.
• Rrmemory Hunt: Optional.
◦ If the feature is enabled, PBX will remember which trunk was used last time,
and then use the next available trunk to call out.
For example, PBX uses the first trunk to call out, then it will use the second
trunk to call out next time.
◦ If the feature is disabled, PBX will use trunks orderly to call out.
• Time Condition: Optional. You can define during which time period can users use
this outbound route. By default, users can call out through the outbound route at
any time.
Note: After you finish the outbound route configurations, you need to check and adjust
the priority of your outbound routes, so that PBX can match and route the call out through
the proper outbound route.
Related concepts
Dial Patterns of Outbound Route
Related information

This topic provides sample configurations that will help you understand dial patterns of
outbound route.
Route Name: Domestic

In Xiamen, China, local numbers are all 7-digit numbers and the numbers do not start with
0, such as 5503305.
For long-distance calls, you need to dial the 4-digit area code and local numbers, such as
0595-5503305. The area code in China is in the format of 0ZXX, the first digit is 0, and the
second digit cannot be 0.
Pattern Strip Prepend Description
90ZXX. 1 Leave it blank.

This is for a long-distance call.
The long-distance number starts with 0, and users

should dial 9 before the number.
Note: Before placing the call, PBX will strip the

leading digit 9.
Example: To call number 05955503303, the user should

dial 905955503303.
9ZXXXXXX 1 Leave it blank.

This is for a local call.
The local number starts with digit 1-9, and users should
dial 9 before the number.
Note: Before placing the call, PBX will strip the

leading digit 9.
Example: To call number 5503301, the user should dial

95503301.
Route Name: Mobile

All mobile phone numbers in China are 11-digit numbers and start with digit 1, such as
15880260666.
1XXXXXXXXXX Leave it blank. Leave it blank. Users can dial the mobile number as they
usually do.
Example: To call number 15880260666, dial

15880260666.
Route Name: International_Call

All international numbers start with digits 00.
00. Leave it blank. Leave it blank. Numbers start with digits 00 will go through this
outbound route.
Example: To call number 16262023379, dial

001626202379.
Import Outbound Routes

You can import outbound routes to quickly set up outbound routing on Yeastar Cloud PBX.
1. Go to Settings > PBX > Call Control > Outbound Routes, click Import.
2. Click Download the Template, add the outbound routes information in the template
file.
Note:
• The imported file should be a UTF-8 .csv file.
• For requirements of the import parameters, refer to Import Parameters - Outbound
Routes.
3. Click Browse to upload the template file.

4. Click Import.
Manage Outbound Routes

After you create outbound routes, you can adjust the priority of the outbound routes. You
can also edit or delete the outbound routes.
Adjust priority of outbound routes

When a user places a call, if the dialed number matches multiple dial patterns, the outbound
route with the highest priority will be used. You can adjust the priority of outbound routes to
route calls through proper outbound routes, greatly saving calling cost for your company.
Note: The route priority is important, especially if there is some overlap. For example,
the number 5503305 matches both a dial pattern of ZXXXXXX and X., the PBX will send the
call through the outbound route with the highest priority.
Example:
When users dial 05503301, both of the two outbound routes match 05503301:
• Outbound Route-Long-distance call: The dial pattern is 0XXXXXXX and uses trunk 1.
• Outbound Route-Local call: The dial pattern is X. and uses trunk 2.
To call 5503301 through trunk 1, you need to prioritize the outbound route of "Long-distance
call"; or PBX will match the outbound route of “Local call" and route the call out using trunk
2.
1. Go to Settings > PBX > Call Control > Outbound Routes.

2. Click the buttons to adjust the priority of your outbound routes.
Note: PBX will match outbound route from top to bottom.
• : Put this outbound route at the top.

• : Move this outbound route upward.
• : Move this outbound route downward.
• : Put this outbound at the bottom.
Edit an outbound route

2. Click beside the outbound route that you want to edit.
3. Edit the outbound route.
Delete an outbound route

2. Click beside the outbound route that you want to delete.
Note: After you delete the outbound route, extension users can not make outbound
calls through this outbound route.
Outbound Restriction
Outbound Restriction Overview
Outbound Restriction is used to limit how many outbound calls extension users can make
within specified time period.
Scenarios
Avoid toll fraud
Most toll fraud is committed from the outside. Hackers may attack the system by registering
to extensions and making outbound calls frequently.
With the Outbound Restriction rules, if extension users make outbound calls over the limited
frequency, the extensions will be blocked and unable to make outbound calls.
Default outbound restriction rule

The PBX has a default rule to limit users to make maximum 5 outbound calls in 1 minute.
You can add another Outbound Restriction rule according to your needs.
Note: We recommend that you keep the default Outbound Restriction rule.
Cancel restriction of outbound calls

If a user makes outbound calls over the limit, the extension will be locked and prohibited
from making outbound calls. On Extensions list, the extension status will display .
Double click the icon , the extension will be able to make outbound calls again.
Add a Rule to Restrict Outbound Calls

The PBX has a default rule to limit users to make maximum 5 outbound calls in 1 minute.
You can add an Outbound Restriction rule to define how many outbound calls the extension
users can make during a period of time.
1. Go to Settings > PBX > Call Control > Outbound Restriction, click Add.
2. On the configuration page, configure an outbound restriction rule according to your

needs.
• Name: Enter a name to help you identify it.

• Time Limit(min): Set time in minutes to limit the number of outbound calls during
the time period.
• Number of Calls Limit: Set the number of outbound calls during the specified
time period. For example, set Time Limit(min) to 5, Number of Calls Limit to 10.
It means if the selected extension users make outbound calls over 10 times in 5
minutes, the extension(s) will be locked and can not make outbound calls.
• Member Extensions: Select extensions which will be restricted by the rule.
Manage Outbound Restriction Rules

After you create restriction rules, you can edit or delete them.
Edit an outbound restriction rule
1. Go to Settings > PBX > Call Control > Outbound Restriction.

2. Click beside the outbound restriction rule that you want to edit.
3. Edit the outbound restriction rule.
Delete an outbound restriction rule
1. Go to Settings > PBX > Call Control > Outbound Restriction.

2. Click beside the outbound restriction rule that you want to delete.
AutoCLIP Routes
AutoCLIP Overview
AutoCLIP (Auto Calling Line Identity Presentation) is an intelligent call matching feature.
You can configure AutoCLIP to route inbound calls to original extensions, which will promote
your customer satisfaction and work efficiency.
Scenarios
Assume sales representatives in your company often make outbound calls to customers for
promotion. More or less, some customers may miss the calls. When customers call back,
the calls are routed to the reception or business auto attendant. Neither reception/business
auto attendant nor the customers know who placed the call.
With AutoCLIP feature, the PBX can redirect the calls to the original extension users who
placed the calls when customers call back.
How does the PBX redirect calls to original extensions?
1. When extension users make outbound calls, the PBX automatically stores the records
to AutoCLIP routing table.
2. When customers call in the PBX, PBX will search the phone numbers from the
AutoCLIP routing table.
• If there're matched records in AutoCLIP routing table, the calls will be routed to
corresponding extensions.
• If there're not matched records in AutoCLIP routing table, the calls will be routed to
the destination specified in inbound routes.
Configure AutoCLIP to Route Inbound Calls to Original Extensions

With AutoCLIP feature on Yeastar Cloud PBX, the PBX can route inbound calls from
customers to original extensions users who placed the calls. This intelligent call matching
feature can greatly improve work efficiency and customer satisfaction.
Note:
• Enable caller ID feature for the trunk that you want to configure AutoCLIP routes, or the
PBX can not distinguish the caller ID and perform AutoCLIP.
• If many extension users make outbound calls to the same external user, PBX will only
match the last extension user that placed the call when the external user calls back.
1. Go to Settings > PBX > Call Control > AutoCLIP Routes.
2. In the Member Trunks field, select the trunk(s) from Available box to the Selected
box.
3. Configure the AutoCLIP settings according to your needs.
• Delete Used Records: Select this option, PBX will perform AutoCLIP as follows:
a. When receiving an external call from customer A, the PBX will search the
record from AutoCLIP list, and redirect the call to the original extension user
that placed the call.
b. PBX will delete the AutoCLIP record.
c. When receiving an external call from customer A again, PBX will always route
the call to the destination specified by the inbound route instead of searching
the record from AutoCLIP list.
d. If extension users of PBX make outbound calls to customer A again, PBX will
generate AutoCLIP record again.
Note: To restrict PBX from routing all inbound calls from a certain
customer to the same extension user, select Delete Used Records.
• Record Keep Time: Set how long records can be kept in AutoCLIP list. If keep
time of a certain record over the value, PBX will automatically delete the record.
• Only Keep Missed Call Records: Select this option. Only unconnected outbound
calls (missed calls on the called party) will be recorded in AutoCLIP list.
• Digit Match: The default value is 7, which means if the digit of caller ID is less
than or equal to 7, the PBX will match the whole phone number with all phone
numbers in AutoCLIP list. If the digit of caller ID over 7, the PBX will match the last
7 digits of phone number with all phone numbers in AutoCLIP list.
Example:
a. Extension user 2000 makes an outbound call to customer 15880270666, and

an AutoCLIP record is generated.
b. When the customer calls in the PBX, the caller ID displays +8615880270666,
where +86 stands for country code. To make sure the PBX can exactly match
the phone number in AutoCLIP list, you should set Digit Match to 11.
c. If the last 11 digits of +8615880270666 exactly match the phone number in
AutoCLIP list, the PBX will route the call to extension 2000.
• Match Outgoing Trunk: Select this option. The PBX will route the call to the
original extension only when the trunk number dialed by external users matches
the trunk that used to place the call earlier.
Example:
Extension user (1000) uses trunk1 to call external user (15880273600). PBX will
route the call to extension (1000) only when the external user (15880273600) calls
the phone number of trunk1.
5. Test AutoCLIP routes.
Extension user uses the trunk with AutoCLIP feature to call external users out.
PBX generates an AutoCLIP record when extension user uses the trunk with AutoCLIP
feature to call external users out. On the AutoCLIP Routes page, click View AutoCLIP List
to view AutoCLIP record.
SLA Stations
SLA Overview
SLA (Shared Line Appearance) feature helps users share and monitor SIP trunks. After
enabling SLA feature for a trunk, the trunk works as the exclusive line for SLA station and is
unavailable in both inbound routes and outbound routes.
SLA trunk refers to the trunk with SLA feature enabled. SLA station refers to an extension
which is bound with a SLA trunk.
• When an SLA station makes an outbound call through SLA trunk, other members
sharing the SLA trunk can monitor the trunk state by BLF keys LED on phone devices.
• When receiving an external call from SLA trunk, all extensions sharing the SLA trunk
will ring.
Note: If Allow Barge feature is enabled on an SLA trunk, all members can place
and join multi-party calls.
SLA Sample Configuration

In a boss-assistant scenario, sometimes assistant needs to answer calls for the boss. So
boss and assistant need to share a trunk. In this topic, we introduce how to configure SLA
trunk and SLA station on Yeastar Cloud PBX based on a boss-assistant scenario.
Assume that the boss’s phone is extension 2000 and the assistant's phone is extension
1000. The shared trunk name is "sipabc" and the trunk number is 5503305.
Note: SLA feature should be used in conjunction with BLF keys on phone devices.
You can set up a shared trunk as follows.
1. Enable SLA feature.
a. Go to Settings > PBX > Trunks, click beside the trunk that you want to enable
SLA.
b. On the Basic page, select Enable SLA and configure the SLA settings.
• Enable SLA: Select this option to enable SLA on the trunk.

• Allow Barge: Optional. Whether to allow other SLA stations that share the
trunk to join the ongoing call by pressing the BLF key on phone devices.
• Hold Access: Whether to allow any SLA stations to retrieve a call that's put
on hold.
◦ Open: Any SLA stations that share the trunk can retrieve the call.
◦ Private: The call can be retrieved only by the SLA station that previously
put the call on hold.
• Failover Destination: The unanswered calls will be routed to the destination.
◦ Hang up
◦ Extension
◦ Voicemail
◦ IVR
◦ Ring Group
◦ Queue
2. Add two SLA stations for the same SLA trunk. One SLA station for the boss's extension
2000, the other SLA station for the assistant's extension 1000.
a. Go to Settings > PBX > Call Control > SLA, click Add.
b. On the SLA Station configuration page, set SLA station for the boss.
• Station Name: Set a name to help you identify it.

• Station: Select the boss's extension 2000.
• Associated SLA Trunks: Select SLA trunk from the Available box to the
Selected box.
• Ring Timeout(s): Set the timeout in seconds. When receiving an inbound call,
the phone of the SLA station will ring until timeout. The default value is 30s.
• Ring Delay(s): Set the time delay in seconds. Phone of the SLA station will
delay ringing after the time defined. The time of Ring Delay(s) can not be
longer than the time of Ring Timeout(s). The default value is 0s.
• Hold Access:Whether to allow any SLA stations to retrieve a call that's put on
hold.
◦ Open: Any SLA stations that share the line can retrieve the call.
◦ Private: The call can be retrieved only by the SLA station that previously
put the call on hold.
d. Repeat steps a to c to set the other SLA station for the assistant.
Note: In the Station field, select the assistant's extension 1000.

3. On the boss's IP phone (extension 2000), configure a BLF key to monitor SLA trunk.
Note: We take an Yealink IP phone as an example.
a. Log in the phone web interface, go to DSS key > Line Key to set a BLF key for the
boss.
b. Select a key to configure.

• Value: Enter {ext_num}_{trunk_name}. In this example, enter 2000_sipabc.
Note:
◦ {ext_num} stands for extension number.
◦ {trunk_name} stands for trunk name.
• Line: Select the line which the extension registers to.

• Extension: Optional. You can enter the key name to help you identify it.
c. Click Confirm.
4. On the assistant's IP phone (extension 1000), configure a BLF key to monitor SLA
trunk.
Note: We take an Yealink IP phone as an example.
a. Log in the phone web interface, go to DSS key > Line Key to set a BLF key for the
assistant.
b. Select a key to configure.

• Value: Enter {ext_num}_{trunk_name}. In this example, enter 1000_sipabc.
Note:
◦ {ext_num} stands for extension number.
◦ {trunk_name} stands for trunk name.
• Line: Select the line which the extension registers to.

• Extension: Optional. You can enter the key name to help you identify it.
c. Click Confirm.
If the configuration is correct, you can see the BLF key LED is on.
• Green: The trunk is available.

• Red: The trunk is busy.
The boss and assistant can share the trunk by SLA.
Related concepts
Share Trunks by SLA
Share Trunks by SLA

After setting up SLA stations on PBX and configuring BLF keys on IP phones, users can
monitor SLA trunks, receive calls from SLA trunks, and make outbound calls through SLA
trunks.
Make outbound calls

SLA station can monitor the status of SLA trunk according to BLF keys status.
Note: For different phone models, there may be some difference in the status of BLF
keys.
• If the BLF key used to monitor SLA trunk turns green, it indicates that the trunk is
available, and the associated SLA station can make outbound calls through this trunk.
To make outbound calls, the SLA station should press BLF key first, and dial the
external number out after hearing a dial tone.
• If the BLF key used to monitor SLA trunk turns red, it indicates that the trunk is in use.
Other SLA stations can not use the trunk to make outbound calls now.
Handle incoming calls

When an external call reaches the SLA trunk, all phones of associated SLA stations will
ring, and BLF keys on phone devices will flash in red. Any SLA stations can answer the call
by pressing BLF keys.
Barge-in an active call

If Allow Barge is enabled for an SLA trunk, other SLA stations are allowed to join an active
call.
When an SLA station is in a call with other users using this SLA trunk, other SLA stations
can join the active call by pressing the BLF key.
Hold and retrieve calls

During the call, the SLA station can press the BLF key to hold and retrieve the call. Whether
an SLA station can retrieve a call or not depends on the Hold Access.
Note: Hold Access of SLA station has a higher priority than the Hold Access of a
trunk.
• If Hold Access is set to Open, other stations that share the trunk can press BLF key to
retrieve the call.
• If Hold Access is set to Private, the call can be retrieved only by the station that
previously put the call on hold.
Related concepts
SLA Sample Configuration
Call Features
IVR
Like most organisations, where possible, we would like to route incoming calls an Auto
Attendant. You can create one or more IVR (Auto Attendant) on the system to achieve it.
When calls are routed to an IVR, the system will play a recording prompting them what
options the callers can enter such as “Welcome to XX, for sales press 1, for Technical
Support press 2”.
Set up an IVR
Set up your own IVR if you need to routing incoming calls via an auto attendant.
1. Go to Settings > PBX > Call Features > IVR, click Add to add an IVR or edit the
default IVR.
2. Edit the Basic settings of the IVR.
• Number: PBX treats IVR as an extension; you can dial this extension number to
reach the IVR from internal extensions.
• Name: Set a name for the IVR.
• Prompt: Use the default IVR prompt or select your custom IVR prompt.
• Prompt Repeat Count: Set how many times the prompt will be played.
• Response Timeout: Set how long the PBX will wait for the caller to operate.
• Digit Timeout: After the user enters a digit, the user needs to enter the next digit
within the timeout.
• Dial Extensions: Whether to allow callers to dial extension numbers via IVR.
• Dial Outbound Routes: Whether to allow callers to dial outbound calls via IVR.
Note: This option is useful if you interconnect two PBXs. The callers can
dial the other PBX's extension number via the IVR. In this solution, you need to
configure the appropriate outbound route and inbound route in both of the two
connected PBXs.
• Dial to Check Voicemail: Whether to allow users to check voicemail via IVR.
Note: This option is for the users who work out of the office. They can call in
the PBX and check their voicemail messages via the IVR.
3. Click Key Press Event tab, set the destination based on callers key presses.
4. On the Key Press Event page, set the Timeout destination and the Invalid
Destination.
• Timeout: If callers do not make an entry within the Prompt Repeat Count, they
will be transferred to the Timeout destination.
• Invalid: If callers enter a digit that is not defined in the IVR, they will be transferred
to the Invalid destination.
Set an IVR Prompt

When users call in the PBX IVR, the users would operate following by the IVR prompt. The
PBX system has one default IVR prompt, you can change the IVR prompt to your audio file.
1. Upload a custom prompt or record a custom prompt on the PBX web interface.
2. Go to Settings > PBX > Call Features > IVR, edit your IVR.
3. Select the Prompt to your custom prompt.
4. Set the Prompt Repeat Count.
Related tasks
Upload a Custom Prompt
Record a Custom Prompt
Convert Audio Files Online
Convert Audio Files via WavePad
Change IVR Prompt Clip

If you need to change one audio clip in the IVR prompt frequently. You can devide your IVR
prompt to mutiple audio clips, and change the desired audio clip when you need to change
the IVR prompt.
For example, your IVR prompt is like the following:
" Thank you for calling Yeastar. We are currently closed in observance of Holiday Name. We
will return on Date. If you got something urgent, please press 1 to contact our support. To
leave a voicemail, please press 2."
The second sentence is what your would change frequently. You can divide the IVR prompt
to 3 clips.
• Clip 1: Thank you for calling Yeastar.

• Clip 2: We are currently closed in observance of Holiday Name. We will return on Date
• Clip 3: If you got something urgent, please press 1 to contact our support. To leave a
voicemail, please press 2.
1. Go to Settings > PBX > Voice Prompts > Custom Prompts, click Upload to upload
your IVR prompt clips.
3. Select the Prompt to the IVR prompt clip1.
4. Click , and select the Prompt to your IVR prompt clip2.
5. Click , and select the Prompt to your IVR prompt clip3.
Next time, when you want to change the IVR prompt, you can change the desired prompt
clip instead of changing the whole IVR prompt.
Dial by Name
You can set the IVR Keypress to "Dial by Name", that will allow the callers to find the person
by entering the first 3 letters of extensions' first name.
To use Dial by Name, you need to do the followings:
• Specify names for extensions on the PBX.
• Better to instruct the callers to use the feature in the IVR prompt.
2. Click Key Press Event tab, set a key action to Dial by Name.
Ring Group
A ring group helps you to ring a group of extensions in a variety of ring strategies. For
example, you could define all the technical support guys' extensions in a ring group and ring
the support guys one by one.
Add a Ring Group
1. Go to Settings > PBX > Call Features > Ring Group, click Add.
2. Configure the ring group.

• Number: Use the default number or change the number.
• Name: Give a name for the ring group to help you identify it.
• Ring Strategy:
◦ Ring All Simultaneously: Ring all the available extensions simultaneously.

◦ Ring Sequentially: Ring each extension in the group one at a time.
• Seconds to ring each member: Define how long the system will wait to ring next
member.
• Members: Select the desired extensions to the Selected box.
• Failover Destination: Define what will happen if none of the members in the ring
group answer the call in the defined time.
Queue
Queues are designed to receiving calls in a call center.
A queue is like a virtual waiting room, in which callers wait in line to talk with the available
agent. Once the caller called in PBX and reached the queue, he/she will hear hold music
and prompts, while the queue sends out the call to the logged-in and available agents. A
number of configuration options on the queue help you to control how the incoming calls are
routed to the agents and what callers hear and do while waiting in the line.
Queue Agents
Yeastar Cloud PBX supports dynamic agents and static agents.
• Static Agent: A static agent always stays in a queue to receive incoming calls.
• Dynamic Agent: A dynamic agent can log in a queue or log out a queue at any time.
On the Queue configuration page, the unselected agents act as dynamic agents.
Add a Queue
Add a simple call queue.
1. Go to Settings > PBX > Call Features > Queue, click Add.
2. Specify a Name and Number for the queue.
3. Select a Ring Strategy for the call.

• Ringing All: Ring All available Agents simultaneously until one answer.
• Least Recent: Ring the Agent which was least recently called.
• Fewest Calls: Ring the Agent with the fewest completed calls.
• Random: Ring a Random Agent.
• Rememory: Round Robin with Memory,Remembers where it left off in the last ring
pass.
• Linear: Rings interfaces in the order specicied in the configuration file.
4. Select Failover Destination, define what should happen if the call does not get
answered by an agent.
5. Select Static Agents for the queue.

• Dynamic agents: A dynamic agent can log in or log out a queue at any time.
• Static agents: A static agents will always stay in the queue.
6. Set the Agent Timeout, define how long the phone should keep ringing before it
considers the call unanswered by that agent.
It is done for a simple call queue, for more information of queue settings, refer to Queue
Settings.
Queue Settings
References of basic queue settings and caller experience settings.
Basic Queue Settings
Option Description
Number Use this number to dial into the queue, or transfer callers to this number to
put them into the queue.
Name Give this queue a brief name to help you identify it.
Password You can require agents to enter a password before they can login to this
queue.
Option Description
Ring Strategy
This option sets the Ringing Strategy for this Queue.
• Ringing All: Ring All available Agents simultaneously until one answer.
• Least Recent: Ring the Agent which was least recently called.
• Fewest Calls: Ring the Agent with the fewest completed calls.
• Random: Ring a Random Agent.
• Rememory: Round Robin with Memory,Remembers where it left off in
the last ring pass.
• Linear: Rings interfaces in the order specicied in the configuration file.
Failover Destination Set the failover destination.
Static Agents
Select static agent of the queue. The static agents will always stay in the
queue.
Note:
• The static agent is not allowed to log in and log out the queue.
• The unselected users are dynamic agents.
Agent Timeout The number of seconds an agent's phone can ring before we consider it a
timeout. If you wish to customize, enter the value in the text box directly.
Ring In Use If set to no, unchecked, the queue will avoid sending calls to members whose
device are known to be “in use”.
Agent Announcement Announcement played to the Agent prior to bridging in the caller.
Retry The number of seconds to wait before trying all the phones again. If you wish
to customize, enter the value in the text box directly.
Wrap-up Time How many seconds after the completion of a call an Agent will have before
the Queue can ring them with a new call .If you wish to customize, enter the
value in the text box directly. Input 0 for no delay.
Call Experience Settings
Caller Settings
Music On Hold Select the “Music on Hold” playlist for this Queue.
Caller Max Wait Time Select the maximum number of seconds a caller can wait in a queue before
being pulled out. If you wish to customize, enter the value in the text box
directly. Input 0 for unlimited.
Leave When Empty If enabled, callers already on hold will be forced out of a queue when no
agents available.
Join Empty If enabled, callers can join a queue that has no agents.
Join Announcement Announcement played to callers once prior to joining the queue.
Caller Settings
Agent ID Announcement Announcement played to the callers to prompt the agent ID. The agent is
who will answer the call.
• [None]: The system will not announce the agent ID.

• [Default]: The system will play the prompt “{extension number} will
be connected. Please wait”. The {extension number} is the extension
number of the agent.
• Custom Prompt: If you choose your custom prompt. The system will
play "{extension number}" + your custom prompt.
Satisfaction Survey Prompt When the agent hangs up, the system will play the prompt to ask the caller
to rate their satisfaction scale.
Caller Position Announcements
Announce Position Announce position of caller in the queue.
Announce Hold Time Enabling this option causes PBX to announce the hold time to the caller
periodically based on the frequency timer. Either yes or no; hold time will be
announced after one minute.
Frequency How often to announce queue position and estimated hold time.
Periodic Announcements
Prompt Select a prompt file to play periodically.
Frequency How often to play the periodic announcements.
Events
Key Once the events settings are configured, the callers are able to press the
key to enter the destination you set. Usually, a prompt should be set on
Periodic Announcements to guide the callers to press the key.
Log in/out a Queue

A dynamic agent can log in or log out a queue at any time.
Log in/out a Queue by Feature Code

Note: If the static agents try to log out a queue, the system will play a prompt "Agent
logged out, goodbye"; But actually, the agent is still in the queue.
• To log in a queue, dial [QUEUE_NUM]*.
For example, dynamic agent 1000 dials 6700* to log in the queue 6700.
• To log out a queue, dial [QUEUE_NUM]**.
For example, dynamic agent 1000 dials 6700** to log out the queue 6700.
• Dial *75[QUEUE_NUM] to log in a queue.
For example, dynamic agent dials *756700 to log in the queue 6700.
• Dial *75[QUEUE_NUM] again to log out a queue.
For example, dynamic agent dials *756700 again to log out the queue 6700.
Log in/out a Queue by BLF Key

A dynamic agent can set a BLF key on his/her IP phone to quickly log in or log out a queue.
For example, on the phone of a dynamic agent, set a BLF key to quickly log in or log out
queue 6700.
The following instructions are based on the Htek UC912 v2.0.4.4.33.
1. Log in the phone web interface, go to Function Keys > Line Key.
2. Set a BLF key to log in or log out queue 6700.
• Type: Set to BLF.

• Value: The BLF key format is *75[QUEUE_NUM]. In this example, set to *756700.
• Account: Select the account that is registered to the extension number of the
agent.
3. Click SaveSet.
Now, the agent can press the BLF key to switch his/her status in the queue.
• When the prompt "agent logged out,goodbye." is played, the agent is logged out of the
queue.
• When the prompt "agent logged in, goodbye." is played, the agent is logged in the
queue.
Monitor Agent Status by BLF

In a call center scenario, a supervisor can set BLF keys to monitor if the agents are in a
specific queue. An agent can also set a BLF key to monitor his own status.
This topic is based on the Htek UC912 v2.0.4.4.33.
Note: Monitoring agent status is supported in the firmware version 30.8.0.8 or later.
We will set a BLF key to monitor if the agent 1001 is in the queue 6700 or not.
1. Log in the phone web interface, go to Function Keys > Line Key.
2. Set a BLF key to monitor extension 1001.

• Value: The BLF key format is *75[EXT_NUM]*[QUEUE_NUM]. In this example, set to
*751001*6700.
• Account: Select the account that has an extension registered to the PBX.
3. Click SaveSet.
Check the BLF LED status:
Note: Different brands of IP phone may have different LED indications.
• Green LED: The agent 1001 is not in the queue 6700.

• Red LED: The agent 1001 is in the queue 6700.
• BLF LED is off: Check if your configurations are correct.
Conference
Conference calls increase employee efficiency and productivity, and provide a more cost-
effective way to hold meetings.
Conference members can dial * to access to the settings options and the admin can kick the
last user out and lock the conference room.
Add a Conference
To make a conference call, you should add a conference on the PBX first.
1. Go to Settings > PBX > Call Features > Conference, click Add.
2. On the configuration page, configure the Conference.

• Number: The extension users need to dial this number to join the conference.
• Name: Set a name for the conference.
• Participant Password: Optional. If the password is set, users need to input the
correct PIN to join this conference.
• Wait for Moderator: If this option is checked, the conference participants could not
hear each other until the moderator joins in the conference.
• Sound Prompt: Select the sound prompt used for the login and logout of
conference members.
• Allow Participant to Invite: Whether to allow the participants to invite users to join
the conference.
• Moderator Password: The moderator doesn't need to enter a password to join the
conference. If a user enter this password to join the conference, he/she will act as
the conference moderator.
• Member Moderators: Select the conference moderators.
Join a Conference
Both the PBX extension users and the external users can join the conference.
1. For the PBX extension users, dial the conference number to join the conference room.
2. For the external users, you need to set the inbound route destination to a conference
first, then the external users call to the PBX, their calls will be routed to the conference.
Conference Voice Menu

During the conference call, the users could manage the conference by pressing * key on
their phones to access voice menu for conference room.
Conference Moderator Voice Menu
1 Mute/ un-mute yourself.
2 Lock /unlock the conference.
3 Eject the last user.
4 Decrease the conference volume.
6 Increase the conference volume.
7 Decrease your volume.
8 Exit the voice menu.
9 Increase your volume.
Conference Users IVR Menu
1 Mute/ un-mute yourself.
4 Decrease the conference volume.
6 Increase the conference volume.
7 Decrease your volume.
8 Exit the voice menu.
9 Increase your volume.
Call Pickup
Call Pickup is a feature that allows a user to answer an incoming call that rings on a
telephone other than the user’s own.
Extension Call Pickup

When a user wants to pick up a call that is ringing at the other extension that is not in
the same pickup group, the user can dial "Extension Pickup feature code (default *04) +
Extension Number" to pick up the call.
Extension Call Pickup Feature Code

The default Extension Call Pickup feature code is *04.
You can change the code on Settings > PBX > General > Feature Code > Extension
Pickup.
Operation
Dial *04[EXT_NUM] to pick up a call.
For example, the ringing extension number is 1000, you should dial *041000 to pick up the
call.
Pick up an Extension's Call by BLF

You can set a BLF key of Extension Call Pickup on your phone. The BLF key will show the
real-time status of the extension. When the extension is ringing, you can press the BLF key
to pick up the call.
We take Yealink T27G v69.82.0.20 as an example below.
1. Set a BLF key to monitor and pick up an extension.
a. Log in the phone web interface, go to Dsskey page.
b. Set the BLF key as below.

• Value: Enter the extension number that you want to monitor.
• Line: Choose the line where your extension is registered on.
• Extension: Enter the feature code of extension pickup. The default code is
*04.
c. Click Confirm.
2. To get notified when the monitored extension has an incoming call, set visual alerts and
audio alerts for the BLF Pickup.
a. On the phone web page, go to Phone > Features > Call Pickup.
b. In the Visual Alert for BLF Pickup, select Enabled.

When a call reaches the monitored extension, you can see the incoming caller ID
on your phone.
c. In the Audio Alert for BLF Pickup, select Enabled.
A “beep” sound will remind you of an incoming call for the monitored extension.
d. Click Confirm.
If your configuration is correct, the BLF LED will turn green.
When the monitored extension has an incoming call, the followings occur on your phone,
press BLF key to pick up the call.
• The phone plays a warning tone.

• The BLF LED turns red.
Group Call Pickup

If extension users are in the same pickup group, they can dial the Group Call Pickup feature
code (default *4) to pick up the group member's incoming call.
Group Call Pickup Feature Code

The default Group Pickup feature code is *4.
You can change the code on Settings > PBX > General > Feature Code > Call Pickup.
Add a Pickup Group

Generally, You can set the extension users who are in the same department in a pickup
group.
1. Go to Settings > PBX > Call Features > Pickup Group, click Add.
2. Set the pickup group.

• Name: Give the group a name to help you identify it.

• Member: Select the desired extensions from Available box to Selected box.
Pick up A Group Member's Call by BLF

You can set a BLF key for Group Call Pickup on your IP phone. When your group member's
phone is ringing, you can press the BLF key to quickly pick up the call.
1. Log in the phone web interface, go to Dsskey page.
2. Set the BLF key as below.
• Type: BLF
• Value: Enter the feature code of group pickup. The default code is *4.
• Label: Set a label that you want to display on the phone screen.
• Line: Choose the line where your extension is registered on.
3. Click Confirm.
If your configuration is correct, the BLF LED will turn green.
Call Transfer
Yeastar Cloud PBX supports Attended Transfer and Blind Transfer, users can dial the
feature code to transfer a call on their phones.
Attended Transfer (Default feature code *3)

An attended transfer, also called consult transfer or warm transfer, is when you speak with
the new person before the call is transferred. You can tell the new person about the caller's
issue and give any background information before transferring the call (without the caller
hearing).
Blind Transfer (Default feature code *03)

A blind transfer is when you transfer the caller to another person without speaking to the
new person first.
Attended Transfer
If you want to tell the new person about the caller's issue and give any background
information before transferring the call, you can choose attended transfer.
Scenario: You (B) are talking with A, then transfer the call to C.
1. During the call with person A, dial *3 on your phone.

You will hear the prompt "transfer" and the dial tone.
2. Dial C's number.

C's phone is ringing. After C answers the call, the call between you and C is
established. In this time, the call between you and A is held.
3. Hang up your call, the call between A and C is established.
Blind Transfer
If you don't need to consult the new person who you want to transfer the call to, you can
perform a blind transfer. Your call will be ended after you transfer the call.
Scenario: You (B) are talking with A, then transfer the call the C.
1. During the call with person A, dial *03 on your phone.

You will hear the prompt "transfer" and the dial tone.
2. Dial C's number and hang up.

C's phone is ringing. After C answers the call, the call between A and C is established.
Callback
Callback feature allows callers to hang up and get called back to the PBX. Callback feature
could reduce the cost for the users who work out of the office using their own mobile
phones.
Set up Callback
Add a Callback rule and set Inbound Route destination to the Callback rule.
Note: Make sure that the Caller ID service is enabled on the callback trunk. If the PBX
cannot recognize the inbound caller ID, callback will fail.
1. Add a Callback rule.
a. Go to Settings > PBX > Call Features > Callback, click Add.
b. On the Callback configuration page, finish the callback settings.
• Name: Set a name for the Callback.

• Callback Through: Select which trunk to use when calling back.
Note: Make sure that you have set up an outbound route for the trunk,
or callback will fail. If the Register-Trunk is used for Callback, make sure the
From User is configured, or callback would fail.
• Delay Before Callback: How long to wait before calling back the caller.
• Strip: Optional. How many digits will be stripped from the call in number
before the callback is placed.
Note: You do not need to configure Strip if the trunk supports calling back
with the Caller ID directly.
For example, user 5503301 calls in the PBX, the caller ID displays 05503301.
To call back to the user, you should set strip 1 digit so that the PBX will call
back to 5503301.
• Prepend: Optional. The digits added before a callback number before the
callback is placed.
Note: You do not need to configure Prepend if the trunk supports calling
back with the Caller ID directly.
For example, user 15880232154 calls in the PBX, the caller ID displays
15880232154. To call back to the long-distance number 15880232154 through
the selected trunk, you should add digit 9 before the number. In this case, set
Prepend to 9.
• Destination: Where the callback will direct the caller to.
2. Set Inbound Route destination to callback.
a. Go to Settings > PBX > Call Control > Inbound Route, edit your inbound route.
b. Set the Inbound Destination to the Callback.
3. Test callback.
Make an inbound call to the PBX trunk, after you hear the ring tone, hangup the call,
the PBX will call back to you.
Speed Dial
Sometimes you may just need to call someone quickly without having to look up his/her
phone number. You can by simply define a shortcut number. You can use Speed Dial
feature to place a call by pressing a reduced number of keys.
Add a Speed Dial Number
1. Go to Settings > PBX > Call Features > Speed Dial, click Add.
2. On the configuration page, configure the Speed Dial.
• Speed Dial Code: Speed dialing number.

• Phone Number: The phone number that you want to call.
Note: You need to add the outbound dial prefix before the phone number if you
want to call an external number.
Speed Dial Example

Assume that you have an outbound route set as below, and you will dial speed number 111
to reach an external number 15990234988 through the route.
You need to set the Speed Dial as below:
Dial *99111 on your phone to call the number 15990234988. *99 is the default feature code
for speed dial.
DISA
DISA (Direct Inward System Access) allows users outside the office to make calls through
the PBX's trunks. For the staffs who are outside the office, they can use DISA feature to
take advantage of lower long-distance rates that are provided by the PBX trunks.
Set up DISA
Add a DISA and set the Inbound Route destination to DISA.
1. Add a DISA.
a. Go to Settings > PBX > Call Features > DISA, click Add.
b. On the DISA configuration page, finish the DISA configurations.

• Name: Set the DISA name.

• Password: Set password for the DISA.
• Response Timeout: The maximum amount of time it will wait before hanging
up if the user has dialled an incomplete or invalid number.
• Digit Timeout: The maximum amount of time permitted between digits.
• Member Outbound Routes: Select the outbound routes that can be
accessed from the DISA.
2. Set Inbound Route destination to DISA.
a. Go to Settings > PBX > Call Control > Inbound Route, edit your inbound route.
b. Set the Inbound Destination to the DISA.
3. Test DISA.
a. Make an inbound call to the PBX, you will get a dial tone after inputting a correct
DISA pin code.
b. Dial the external number that you want to call.
Intercom/Paging
The Paging and Intercom features allow you to make an announcement to a group of
extensions. The called parties do not need to pick up the handset as the audio will be
played via the phone speakers.
Set up 1-Way Paging

Paging is used to make an announcement over the speakerphone to a phone or group of
phones. The called parties will not ring, but instead answer immediately into speakerphone
mode.
Note: Paging is typically one way for announcements only.
1. Go to Settings > PBX > Call Features > Paging/Intercom, click Add.
2. Set a 1-Way paging group.
• Number: Use the default or specify an number for the paging group.
• Name: Enter a name for the paging group.
• Type: Choose 1-Way Paging.
• Dial * to Answer: If this option is checked, the paging group members can dial * to
talk to the paging initiator.
Note: When a member dials *, the group announcement will terminate, and
the member who dials * can have a private call with the paging initiator.
• Member: Choose the group members to the Selected box.

When you dial the paging group number, the members in the group will hear the
announcement.
Set up 2-Way Intercom

2-way intercom is used to make a multi-party conference. The called parties will
automatically answer the call into speakerphone mode and join the conference.
Note: Intercom allows all users in the group to talk and be heard by all.
1. Go to Settings > PBX > Call Features > Paging/Intercom, click Add.
2. Set a 2-Way intercom group.
• Number: Use the default or specify an number for the intercom group.
• Name: Enter a name for the intercom group.
• Type: Choose 2-Way Intercom.
• Dial * to Answer: If this option is checked, the intercom group members can dial *
to talk to the intercom initiator.
Note: When a member dials *, the group announcement will terminate, and
the member who dials * can have a private call with the intercom initiator.
• Member: Choose the group members to the Selected box.

When you dial the intercom group number, the members in the group will automatically join
the conference by speakerphone mode.
Set up 1-Way Multicast Paging

Multicast Paging allows you to easily and quickly broadcast instant audio announcements to
phone users who are listening to the same multicast IP address of the PBX.
When you make a Multicast Paging, the PBX sends Real-time Transport Protocol (RTP)
streams to the IP phones without involving SIP signaling. The phones that will receive the
RTP streams don't need to register SIP extensions.
Note:
• The IP phone that will receive 1-way multicast paging should support Multicast Paging
feature.
• The Multicast Paging is one-way audio call.
1. Set a 1-way Multicast Paging on the PBX.
a. Go to Settings > PBX > Call Features > Paging/Intercom, click Add.
b. Set a 1-Way multicast paging.
• Number: Use the default or specify an number for the paging group.
• Name: Enter a name for the paging group.
• Type: Choose 1-Way Multicast Paging.
• IP of Multicast Channel: Enter the multicast IP address and port (e.g.
224.255.255.255:1000).
Note: The range of multicast IP address is 224.0.0.0 - 239.255.255.255.
2. Set Multicast Paging on each of your IP phone.

In the following, we take Yealink T27G as an example.
a. Log in the phone web interface, go to Directory > Multicast IP.
b. In the Multicast Listening section, enter the same multicast IP address and port
of the PBX.
c. Click Confirm.
When you dial the paging group number, the members in the group will automatically
answer the call into speakerphone mode.
Note: If the multicast paging doesn't work, check the following:
• Multicast IP address and port are in the correct range.

• If the PBX and IP phones are in different IP segments (e.g. PBX is in 192.168.5.X IP
segment, IP phones are in 192.168.3.X IP segment), check if your router supports IP
Multicast in different IP segments.
Make an Announcement to a Specific User

Extension users can dial the intercom feature code to make an intercom to a specific
extension, the called party can respond immediately without picking up the handset.
The default Intercom feature code is *5.
Note: In this way, the audio is two way, both the caller and called party can hear each
other.
Extension user 2000 makes an intercom call to extension user 1000.
1. Dial *51000 on the phone of extension 2000.
The call on extension 1000 will be answered automatically.

Call Parking
Call Parking is a feature that allows you to suspend a call for an extended period of time
and then retrieve that call from any extension.
Scenarios
During a call with clients, extension users may need to check information somewhere else.
In such case, extension users can park the call temporarily and retrieve the call by any
extensions when getting things done.
Settings of Call Parking

Go to Settings > PBX > General > Feature Code > Call Parking, you can modify the
feature code, parking extension range, and parking time.
We provide default settings of call parking as follows.
Call Parking The default feature code is *6. During a call, dial *6 on your phone, the
system will automatically assign a parking slot number to the call.
Directed Call Parking The default feature code is *06. During a call, dial "06+parking slot number",
the call will be parked to the designated parking slot number.
Parking Extension Range Specify the range of parking extension where a call will be parked. The
default value is 6900-6999.
Note: The rang of parking extension must be different from existing

extension ranges (Settings > PBX > General > Preferences > Extension
Preferences).
Parking Timeout (s) Specify the time that a call can be parked before it is retrieved by other
extensions. The default value is 60s.
Note: Parking Timeout must be longer than 30s.
Timeout Destination
If a parked call hasn't been retrieved before the parking timeout, PBX will
route the call to the designated destination.
• Original Parker:The call will be routed to the user who parks this call.
• Extension: Te call will be routed to the designated extension number.
• Extension's Voicemail: The call will be routed to the designated
extension's voicemail.
• Custom Number: The call will be routed to the designated number.
Call Parking (Default feature code: *6)

You can dial the feature code of Call Parking to get the parking slot number, then dial the
parking slot number on another phone to retrieve the call.
Example:
1. During a call, dial *6 on your phone, the system will prompt you that the parking slot
number is 6900.
2. Dial 6900 on another phone to retrieve the call.
Direct Call Parking (Default feature code: *06)

If you get a parking slot number from your administrator, you can dial the “feature code of
Direct Call Parking+parking slot number" to park the call to the slot.
Example:
1. During a call, dial *066900 to park the call to slot 6900.

2. Dial 6900 on another phone to retrieve the call.
Park Calls by BLF

You can set a BLF key of Call Parking on your phone. The BLF key will show the real-time
status of the parking slot. If the parking slot is vacant, you can press the BLF key to park a
call to the parking slot.
1. Log in the phone web interface, go to Dsskey page.
2. Set the BLF key as below.

• Value: Enter the parking slot number.
• Line: Select the line where your extension is registered on.
• Extension: Enter the feature code of Direct Call Parking. The default code is *06.
3. Click Confirm.
• When the parking slot is vacant, the BLF LED is green.

Press the BLF key to park a call to the parking slot.

• When the parking slot is occupied, the BLF LED is red.
Configure Call Parking Caller ID

By default, when you retrieve a parked call, the call-park slot number (e.g. 6900) will be
displayed on the phone. To display the original caller ID of the user who you were talking to,
you need to configure SIP settings to get caller ID from Remote- Party-ID SIP header.
1. On PBX, enable Send Remote Party ID.

a. Go to Settings > PBX > General > SIP > Advanced.
b. Check the option Send Remote Party ID option.
2. On the IP phone that you will use to retrieve a parked call, configure the Caller ID
Source.
Note: We take Yealink T29G v46.83.0.50 as an example below.

a. Log in the phone web interface, go to Account > Advanced.

b. In the Account, select the account where the extension is registered.
c. In the Caller ID Source field, select RPID-FROM.
d. Click Confirm.
Test call parking. When you retrieve the parked call from the IP phone, the phone screen
will display the parking slot number for 1 or 2 seconds, then display the original caller ID.
The following call flow shows how the IP phone gets caller ID when a user retrieves a
parked call.
1. A user dials parking slot number 6900 on IP phone to retrieve a parked call.
2. PBX sends a Re-INVITE packet that contains Remote-Party-ID.
3. The IP phone gets the caller ID from the Remote-Party-ID header.
Fax
Yeastar Cloud PBX supports Fax over IP. You can send or receive a fax via a physical fax
machine or receive a fax over the network.
What is T.38 Fax over IP?

T.38 is a protocol for sending faxes over a voice over IP (VoIP) network or the Internet in
real time.
T.38 protocol defines the transport of data (a fax) between PSTN fax terminals through a fax
gateway, between two Internet-aware fax terminals, or from a PSTN fax terminal through
a fax gateway to an Internet-aware fax terminal. A T.38 stream is sometimes referred to as
Fax over IP (FoIP).
PSTN fax terminals traditionally use the T.30 protocol to send analog data. To exchange
analog fax data with a PSTN terminal over the Internet, the T.38 protocol first converts
analog data into digital data. The protocol then converts the data back to analog on the
receiving end if the receiver is a PSTN fax terminal.
T.38 Fax Settings

If the Fax over IP doesn't work, you can go to Settings > PBX > General > SIP > T.38 to
change the T.38 settings.
• No Re-invite SDP Add T.38 Attribute
If this option is enabled, no T.38 attributes will be added in re-invite SDP packet.
• Error Correction
Error Correction Mode (ECM) for the Fax.
• T.38 Max BitRate
T38 Max Bit Rate.
Fax to Email
Fax to Email feature helps you receive faxes on your smart phone or computer. Yeastar
Cloud PBX will convert the received fax and forward it to an extension user's email.
Steps to Configure 'Fax to Email'
1. Configure the PBX System Email.
Make sure the PBX system email works, or the PBX cannot forward the received faxes
to an extension user's email.
2. Check if the extension user's email is configured.
3. Configure the destination of your inbound route.
• If you want to receive fax via fax detection, set the Destination to IVR, and set Fax
Destination to Fax to Email.
• If you want to receive fax through a private trunk, set the Destination to Fax to
Email.
Receive Fax through a Dedicated Trunk

You can assign one or more trunks to receive faxes, and tell your customers to send faxes
to the dedicated trunk number.
1. Go to Settings > PBX > Call Control > Inbound Route, click Add.
2. On the configuration page, select the dedicated trunk to the Selected box.
Note: The selected trunk will be used to receive faxes only, users cannot make
audio calls through the selected trunk.
3. Set the Destination to Fax to Email.
Users can dial the number of the dedicated trunk, then send fax to the PBX.
Receive Fax via Fax Detection

If you want to receive calls and also receive faxes through a trunk, you can set fax detection
on your inbound route.
1. Go to Settings > PBX > Call Control > Inbound Route, configure your inbound route.
2. Select the trunk to the Selected box.
3. Set the Destination to IVR.
4. Check the option Enable Fax Detection.
5. Set the Fax Destination to Fax to Email.
Edit 'Fax to Email' Template

The PBX has a default email template for Fax to Email. You can edit the template
1. Go to Settings > System > Email > Email Template, click beside Fax to Email.
On the Edit Template page, the description of variables and the default email contents
are displayed.
Note: The variable names are unchangeable.
PIN List
PIN List is used to manage lists of PINs (numerical passwords) that can be used to access
restricted features such as outbound route and DISA.
Add a PIN list
1. Go to Settings > PBX > Call Features, click More to display more call features.
2. Click PIN List.
3. On the Add PIN List page, configure the following settings:
• Name: Set a name for the PIN list.

• Record In CDR: When a PIN code has been used, whether to display the PIN
code in the relevant CDR.
• PIN List: Enter the PIN codes. Press Enter key to add multiple PIN codes.
Apply a PIN list

You can apply a PIN list to an outbound route or a DISA to restrict users dialling outbound
calls. When a PIN list is applied to an outbound route or a DISA, users need to dial the
correct PIN to place the outbound calls.
Blacklist/Whitelist
Yeastar Cloud PBX allows you to blacklist and whitelist IP addresses. This article briefly
introduces the definitions and basic settings of blacklist and whitelist, and provides related
configuration examples.
What is Blacklist and Whitelist

We briefly introduce the definitions of blacklist and whitelist as follows.
• Blacklist
The blacklist is used to filter phone numbers. If a phone number is added to the
blacklist, the system blocks incoming or outgoing calls for the phone number.
• Whitelist
The whitelist is used to add trusted phone numbers. If a phone number is added to the
whitelist, the system allows incoming or outgoing calls for the phone number.
Note: The whitelist has a higher priority than the blacklist.
Blacklist/Whitelist Setting
Yeastar Cloud PBX supports system blacklist/whitelist and personal blacklist/whitelist. You
can set a global system blacklist/whitelist to apply to all extensions. Extension users can
also log in the PBX web interface by their accounts, and set blacklist/whitelist for their own
extensions.
• System Blacklist and Whitelist
Log in the PBX web interface as an administrator, and go to Settings > PBX > Call
Features > Blacklist/Whitelist to set blacklist and whitelist.
Yeastar Cloud PBX supports to block or allow three types of numbers:

◦ Inbound: If blacklist type is set to Inbound, the number can not call in the system;
if whitelist type is set to Inbound, the number can call in the system.
◦ Outbound: Extension users can not call the number whose blacklist type
is Outbound; extension users can call the number whose whitelist type is
Outbound.
◦ Both: Neither inbound calls nor outbound calls are allowed for the number whose
blacklist type is Both; both inbound calls and outbound calls are allowed for the
number whose whitelist type is Both.
• Personal Blacklist and Whitelist

Log in the PBX web interface by extension accounts, the extension users can view the
system blacklist and whitelist that is set by the administrator.
Note: Extension users can add personal blacklist and whitelist for their extensions
according to their needs.
• Blacklist/Whitelist Priority
Priority of blacklist/whitelist: system whitelist > system blacklist> personal whitelist >
personal blacklist.
Blacklist Example
We demonstrate a few examples of blacklist as follows.
Prohibit inbound calls from external numbers
For example, 10086 and 1008611 are not allowed to call in PBX. You can add the two
numbers to blacklist as follows.
Prohibit inbound calls and outbound calls
For example, 10086 and 1008611 are not allowed to call in PBX, and all extensions on PBX
are not allowed to call out 10086 and 1008611.
Prohibit selected extensions or extension groups from calling certain numbers
• Prohibit extension group (Sales) from calling 10086 and 1008611.
Note: You can add an extension group in advance for quick selection.
• Prohibit all extensions from calling 10086 and 1008611.

• Prohibit extensions from calling numbers with specified extension format
For example, prohibit extension group (sales) from calling R&D team (all extension
numbers are in the format 5XXX).
Whitelist Example
The whitelist has a higher priority than the blacklist, so you can use whitelist to filter trusted
phone numbers from blacklist, and allow inbound/outbound calls for the phone numbers.
For example, assume you've added 5XXX (extension numbers of R&D team) to blacklist to
prohibit sales from calling R&D teams, but you want to allow sales to call extension 5001. In
this case, you can add 5001 to whitelist as follows.
Call Recording
Call Recording Overview

Yeastar Cloud PBX supports One Touch Recording and Auto Recording.
One Touch Recording
One Touch Reording, also known as On-demand Recording, allows users to
dial *1 on their phones to record calls at any time.
For more detail of One Touch Recording, refer to One Touch Record.
Auto Recording
Auto Recording is a feature that enables the PBX to automatically record
internal calls, external calls, and conference calls.
For more detail of Auto Recording, refer to Auto Recording.
One Touch Record

During a call, you can dial the One Touch Record feature code to start recording the call;
dial the feature code again to stop the recording.
One Touch Record Feature Code

The default One Touch Record feature code is *1.
You can change the code via Settings > PBX > General > Feature Code > One Touch
Record.
One Touch Record Prompt

By default, when a user dials *1 to record the call, the PBX will not play prompt to notify the
other party that the call is being recorded.
To set One Touch Record prompt:
1. Go to Settings > PBX > Voice Prompts > Prompt Preferences.

2. In the One Touch Record Start Prompt field, select a custom prompt.
When an extension user dial *1 to record the call, PBX will play the prompt to the other
party.
3. In the One Touch Recording End Prompt field, select a custom prompt.
When an extension user dial *1 to stop recording the call, PBX will play the prompt to
the other party.
Auto Recording
Auto Recording is a feature that enables the PBX to automatically record internal calls,
external calls, and conference calls.
Set up Auto Recording

To use Auto Recording on your PBX, contact your PBX provider to buy the Recording
capacity. By default, you can enjoy 500-minute recording time for free.
Set up Call Recording for Internal Calls
1. Go to Settings > PBX > Recording, check the option Enable Recording of Internal
Calls.
2. Set the recording announcement for internal calls.
a. Upload a custom prompt to the PBX or record a custom prompt on the PBX.
b. Set Internal Call Being Recorded Prompt to your custom prompt.
The PBX will notify the called party that the call is being recorded.
3. In the Record Extensions section, select extensions to the Selected box.
The selected extensions will be recorded.
Set up Call Recording for External Calls
1. Go to Settings > PBX > Recording, set the recording announcement for external calls.
a. Upload a custom prompt to the PBX or record a custom prompt on the PBX.
b. Set custom prompt for outbound calls, inbound calls, and callback calls.
• Outbound Calls Being Recorded Prompt: If the external call (outbound) has
enabled call recording, this prompt will notify the external party that the call is
being recorded.
• Inbound Calls Recorded Prompt: If the external call (inbound) has enabled
call recording, this prompt will notify the external party that the call is being
recorded.
• Callback Calls Being Recorded Prompt: If the external call (callback) has
enabled call recording, this prompt will notify the external party that the call is
being recorded.
2. In the Record Trunks section, select trunks to the Selected box.
The calls through the selected trunks will be recorded.
Note: If you have selected extensions in the Record Extensions section, the
extensions' calls will be recorded no matter which trunks are used.
Set up Call Recording for Conference Calls
1. Go to Settings > PBX > Recording.

2. In the Record Conferences section, select conferences to the Selected box.
The selected conferences will be recorded.
Pause/Resume Auto Recording

During an external call, the extension user can pause the Auto Recording and then resume
the Auto Recording to avoid the sensitive personal information such as credit card details
being recorded.
When you play the recording files, the paused part will be absent.
The default feature code to pause and resume Auto Recording is *00.
You can change the code in Settings > PBX > General > Feature Code > Auto
Recording Switch.
During an external call, the extension user can dial feature code to pause and resume call
recording.
1. Dial *00 to pause the call recording.

2. Dial *00 again to resume the call recording.
Related tasks
Monitor Auto Recording Status
Monitor Auto Recording Status

When you pause and resume the Auto Recording during a call, you may need to know if the
call recording state is switched successfully or not. You can set a BLF key on your IP phone
to monitor the auto recording status of your current call.
This topic is based on the Yealink T41S version 66.84.0.10.
1. Log in the phone web interface, go to Dsskey > Line Key.
2. Set a BLF key to monitor your own extension.

In this example, your extension number is 1000, and the extension 1000 is registered
on the phone Line 1.

• Value: The BLF key format is *00{extension_number}.
In this example, set to *001000.

• Label: Optional. The label will be displayed on the phone screen.
• Line: Choose the Line where your extension number is registered.
3. Click Confirm.
When the monitored extension is being recorded, the BLF LED will turn red.
When the monitored extension is not in a call or the Call Recording is paused, the BLF LED
will turn green.
Related tasks
Pause/Resume Auto Recording
Auto Clean up Recording Files

When the recording capacity limit is reached, the PBX will automatically delete the oldest
recording files. When 80% of the maximum recording capacity is reached, the PBX will send
an email notification to you.
Enable 'Auto Cleanup Reminder'

To get informed of the recording usage, you can enable Auto Cleanup Reminder.
1. Go to Settings > Event Center > Event Settings > System, enable Notification and
Record for Auto Cleanup Reminder.
• Record: The event of Recording Auto Cleanup will be recorded in Event Log.
• Notification: When the recording capacity is about to be reached, the PBX will
send notification email to the Notification Contacts.
2. To modify the email template, click .
Check Recording Usage

Go to Resource Monitor > Recording Usage to check your Recording usage.
Manage Call Recording Files

Go to CDR and Recordings to search, play, download, or delete the recording files.
Search Recording Files
1. Set the search criteria Time.

2. Enable Include Recording Files to filter the records that have associated recording
files.
3. Optional: Set other search criteria.
4. Click Search.
Download a Recording File
Click behind a recording log to download the recording file.

Play a Recording File
Click to play the recording file on web or play to an extension.
Delete a Recording File
Click behind a recording log to delete the recording file.
Grant Recording Permissions to Users

By default, only the super administrator has permission to manage the call recording files.
The super administrator can grant recording permission to extension users and allow the
users to play, download, and delete recording files.
1. Go to Settings > System > User Permission, click Add.
2. In the User drop-down list, select a user whom you want to grand permissions to.
3. Set Set Privilege As.

• Custom: All permissions are disabled by default.
• Administrator: All permissions are enabled by default.
4. Click CDR and Recordings tab, and grant Recording Permission to the user.
5. Set which extensions' recording files are allowed to play, download or delete.
• All Extension: The user can manage all the extensions' recording files.
• Selected Extensions: The user can manage only the selected extensions'
recording files.
6. Click Save.
When the user log in the PBX User Portal, he/she will have permission to manage
recording files.
Voice Prompts
System Prompt
The default system prompt language is English. You can change the global system prompt,
and if an extension user works in a foreign language, you can set a different system prompt
for the user.
Change System Prompt

Yeastar have stored all the supported system prompts online. You can check the supported
system prompts on the PBX web page, and download an online system prompt file, then
change to the desired system prompt.
1. Go to Settings > PBX > Voice Prompts > System Prompt.
2. Click Download Online Prompt.
3. On the Download Online Prompt page, select your desired system prompt, click to
download the file.
After the file is downloaded, you can see the system prompt in Prompt List.
4. Set the downloaded system prompt as the default system prompt.
Customize System Prompt

You can upload your own system prompts to the PBX, so that users can hear the
customized system prompts.
Contact Yeastar support to record your own system prompts.
1. Go to Settings > PBX > Voice Prompts > System Prompt.
2. In the Upload System Prompts section, click Browse to choose the system prompt
file.
Note: Upload the .tar file that is provided by Yeastar, or the system prompt won't
work.
3. Click Upload.
If the file is uploaded successfully, you can see the prompt file in the Prompt List.
4. Set the uploaded system prompt as the default system prompt.

Change an Extension's System Prompt

If a user works in a foreign language, you can set a different system prompt for the
extension user.
1. Download a system prompt for the extension user.
a. Go to Settings > PBX > Voice Prompts > System Prompt.
b. Click Download Online Prompt.
c. On the Download Online Prompt page, select your desired system prompt, click
to download the file.
After the file is downloaded, you can see the system prompt in Prompt List.
2. Go to Settings > PBX > Extensions, select the desire extension, click .
3. On the Basic page, set the Prompt Language.

Music on Hold (MoH)

Music on hold (MoH) is the business practice of playing recorded music to fill the silence
that would be heard by callers who have been placed on hold.
The PBX has a default MoH playlist, you can add MoH playlists and upload music files to
the PBX.
Note: The default MoH files are distributed under the Creative Commons Attribution-
ShareAlike3.0 license through explicit permission from their authors.
Add a Custom MoH Playlist

You can add a custom MoH playlist and upload your audio files to the PBX.
1. Go to Settings > PBX > Voice Prompts > Music on Hold, click Create New Playlist.
2. On the configuration page, set the playlist name and the playlist order, click Save.
3. On the Music On Hold page, choose the new created playlist.
4. Click Browse to choose an audio file from your local PC, then click Upload.
Note: The uploaded file should meet the audio file requirements.
5. Repeat the step 4 to add another audio file.

You can see the uploaded audio files in the MoH list.
Related concepts
Requirements of Custom Audio Files
Related tasks
Change the MoH Playlist
Change the MoH Playlist

To change the MoH playlist, you need to first add a MoH playlist and upload your audio files
to the PBX.
2. Select a MoH playlist from the drop-down menu of Music On Hold.
The PBX will play the select MoH playlist when a user is held in a call.
Related tasks
Add a Custom MoH Playlist
Custom Prompt
The default voice prompts and announcements in the system are suitable for almost every
situation.
However, you may want to use your own voice prompt to make it more meaningful and
suitable for your case. In this case, you need to upload a custom prompt to the system or
record a new prompt and apply it to the place you want to change.
Requirements of Custom Audio Files

You can upload your audio file to the PBX, the audio file should meet the following
requirements.
Option Requirement
File Format WAV, wav, or gsm file.
• gsm 6.10 8kHz, Mono, 1Kb/s

• alaw 8kHz, Mono, 1Kb/s
• ulaw 8kHz, Mono, 1Kb/s
• pcm 8kHz,Mono,16Kb/s
File Name Should NOT contain special characters.

Option Requirement
File Size Smaller than 8MB.
1. Go to Settings > PBX > Voice Prompts > Custom Prompts, click Upload.
2. On the configuration page, click Browse to choose your audio file.
Note: The uploaded file should meet the audio file requirements.
3. Click Upload to start uploading the file.
After the file is uploaded, you can see the file on the Custom Prompts page.

You can use an extension to record custom prompts.
1. Go to Settings > PBX > Voice Prompts > Custom Prompts, click Record New.
2. On the configuration page, set the prompt name and select an extension to record the
prompt.
3. Click Save.
The selected extension will ring.
4. Record your prompt on the phone. When done, press the # key or hang up.
5. Refresh the Custom Prompts page, you can see the saved prompt file.
You can click to play the prompt, and decide whether to save it or not. If you are not
satisfied with the prompt, click to record again.
Related tasks
Play a Custom Prompt
Play a Custom Prompt

After you upload a custom prompt or record a custom prompt, you can select an extension
to play the prompt.
Note: We recommend that you play your custom prompts before you apply the custom
prompts to IVR, MoH, or other places.
1. Go to Settings > PBX > Voice Prompts > Custom Prompts.
2. In the Custom Prompts list, choose a prompt, click .
3. On the configuration page, choose an extension to play the prompt.

4. Click Play.
The selected extension will ring.
5. Pick up the phone to listen to the prompt.
Related tasks

WavePad is audio editing software, you can convert audio files via WavePad, then upload
the audio files to your PBX.
1. Launch WavePad, open your audio file.
2. Click File > Save File As.
3. Set the Save as type to .wav or .gsm, click Save.
4. For the .wav type, set the encoder options according to the requirements of custom
audio files, click OK.
Related tasks

You can quickly convert your audio files via G711 File Converter online.
1. Visit g711.org.
2. Click Browse to upload your audio file.
3. Set the Output Format.

We recommend BroadWorks Classic or Asterisk Standard.
4. Click Submit to start converting the file.

Set Prompts for Failed Calls

A user may fail to make outbound calls due to many reasons, such as the trunk is busy, no
trunk available, or invalid number. You can set different prompts to inform the user why the
call fails.
2. Set the prompts for different type of failed calls.
• Invalid Phone Number Prompt: The PBX will play the prompt when the dialed
number is invalid.
• Busy Line Prompt: The PBX will play the prompt when the trunk used is busy.
• Dial Failure Prompt: The PBX will play the prompt if no trunk is available to call
out.
System Management
System General Settings

The system general settings can be applied globally to Yeastar Cloud PBX
System Preference
Configure the preferences settings that will be applied globally to the system.
Go to Settings > PBX > General > Preferences to configure the system preferences.
General Preference
Table 2. Descriptions of General Preference
Option Description
Max Call Duration Select the global maximum call duration.
Note:
The precedence of Max Call Duration(s) (Global v.s.

Extension):
• For internal calls: The Max Call Duration(s) setting of the

caller's extension takes precedence.
• For outbound calls: The Max Call Duration(s) setting of
the caller's extension takes precedence.
• For inbound calls: The global Max Call Duration(s) setting
takes precedence.
Attended Transfer Caller ID The Caller ID that will be displayed on the recipient's phone.
For example, Phone A (transferee) calls Phone B (transfer),
and Phone B transfers the call to Phone C (recipient). If set to
Transfer, the Caller ID displayed will be Phone B's number; if
set to Transferee, Phone A's number will be displayed.
Flash Event Set which event will be triggered by pressing the hook flash:
• 3-way Calling
• Call Transfer
Virtual Ring Back Tone Once enabled, when the caller calls out with cellular trunks,
the caller will hear the virtual ring back tone generated by the
system before the callee answers the call.
Option Description
Distinctive Caller ID When the incoming call is routed from Ring Group, Queue or
IVR, the Caller ID would display where it comes from.
Match Route Permission When Seizing If checked, when users seize a line to place an outbound
a Line call, the call will succeed only when the route permission is
matched.
FXO Mode
Select a mode to set the On Hook Speed, Ringer Impedance,
Ringer Threshold, Current Limiting, TIP/RING voltage,
adjustment, Minimum Operational Loop Current, and AC
Impedance as predefined for your country's analog line
characteristics.
The default setting is FCC for USA.
Tone Region Select your country or nearest neighboring country to enable

the default dial tone, busy tone, and ring tone for your region.
DTMF Duration Set the duration of a DTMF tone on the FXO trunk.
DTMF Gap Set the interval between each DTMF tone on the FXO trunk.
Extension Preference
Below are default extension ranges. You can change the extension range according to your
needs.
Note: PBX treats Ring Group, Paging Group, Conference, Queue as extensions.
Extension users can dial the extension numbers to reach them directly.
Extension Type Default Range
User Extensions 1000 - 5999
Account Trunk 6100 - 6199
Ring Group Extensions 6200 - 6299
Paging Group Extensions 6300 - 6399
Conference Extensions 6400 - 6499
IVR Extensions 6500 - 6599
Queue Extensions 6700 - 6799
Feature Code
Feature codes are used to enable and disable certain features available in the Yeastar
Cloud PBX. Extension users can dial feature codes on their phones to use that particular
feature.
Go to Settings > PBX > General > Feature Code to view or change the feature code
settings.
• Feature Code Digit Timeout: The timeout to input next digit. The default is 4000 ms.
Default Feature Codes
Recording
One Touch Record *1
Call Recording Switch *00
Call Forwarding
Reset to Defaults *70
Enable Forward All Calls *71
Disable Forward All Calls *071
Enable Forward When Busy *72
Disable Forward When Busy *072
Enable Forward No Answer *73
Disable Forward No Answer *073
Voicemail
Check Voicemail *2
Voicemail for Extension **
Voicemail Main Menu *02
Transfer
Blind Transfer *03
Attended Transfer *3
DND
Enable Do Not Disturb *74
Disable Do Not Disturb *074
Call Pickup
Call Pickup *4
Extension Pickup *04
Time Condition
Time Condition Override *8
Intercom
Recording
Intercom *5
Call Monitor
Listen *90
Whisper *91
Barge-in *92
Call Parking
Call Parking *6
Directed Call Parking *06
Parking Extension Range 6900-6999
SIP Settings
The SIP configurations require professional knowledge of SIP protocol, incorrect
configuration may cause calling issues on the SIP extensions and SIP trunks.
Go to Settings > PBX > General > SIP to configure the SIP settings.
SIP General Settings
Option Description
UDP Port UDP Port used for SIP registrations. The default is 5060.
TCP Port TCP Port used for SIP registrations. The default is 5060.
Registration Timers
Max Registration Time Maximum duration (in seconds) of incoming registrations and
subscriptions. The default is 3600 seconds.
Min Registration Time Minimum duration (in seconds) of incoming registration and
subscriptions. The default is 60 seconds.
Qualify Frequency How often to send SIP OPTIONS packet to SIP device to check if the
device is up. The default is 30 per second.
Outbound SIP Registrations
Registration Attempts The number of registration attempts before giving up (0 for no limit).
Default Incoming/Outgoing
Registration Time Default duration (in seconds) of incoming/outgoing registration. The
default is 120 seconds.
Note: The actual duration needs to minus 10 seconds from the

value you filled in.
Option Description
Subscription Timer
Max Subscription Time Maximum duration (in seconds) of incoming subscriptions. The default is
3600 seconds.
Min Subscription Time Minimum duration (in seconds) of incoming subscriptions. The default is
90 seconds.
SIP Codec
A codec is a compression or decompression algorithm that used in the transmission of
voice packets over a network or the Internet.
Codec Selection
Yeastar Cloud PBX supports G711 a-law, u-law, GSM, H261, H263, H263P,
H264, SPEEX, G722, G726, ADPCM, G729A, MPEG4, and iLBC.
Note:
• You need to choose at least one same code on the PBX and on your
phones, or there may be a problem of the call.
• If you want to make video calls, you need to select H261, H263, H263P,
H264 or MPEG4 codec on the PBX and on your phones.
iLBC Settings
The iLBC codec supports two modes: 20ms and 30ms frame length modes,
To get better voice quality, you need to set the iLBC mode according to your
SIP endpoints.
Note: Linkus uses iLBC 20ms mode. When Linkus is enabled, this option is
switched to 20ms mode automatically.
TLS Settings
Option Description
Enable TLS Check the checkbox to enable TLS.
TLS Port TLS Port used for SIP registrations. The default is 5061.
TLS Client Method Specify protocol for outbound client connections. The default is sslv2.
Session Timer
A periodic refreshing of a SIP session that allows both the user agent and proxy to
determine if the SIP session is still active.
Option Description
Session-timers
Choose the session timers mode on the system:
• No: Do not include “timer” value in any field

• Supported: Include “timer” value in Supported header
• Require: Include “timer” value in Require header
• Forced: Iclude “timer” value in both pportednd equired header.
The default is Supported.
Session-Expires The max refresh interval in seconds.
Min-SE The min refresh interval in seconds, it must not be less than 90.
Qos
QoS (Quality of Service) is a major issue in VoIP implementations. The issue is how to
guarantee that packet traffic for a voice or other media connection will not be delayed or
dropped due interference from other lower priority traffic.
When the network capacity is insufficient, QoS could provide priority to users by setting the
value.
Option Description
ToS SIP Type of Service for SIP packets.
ToS Audio Type of Service for RTP audio packets.
ToS Video Type of Service for RTP video packets.
Cos SIP Class of Service for SIP packets.
Cos Audio Class of Service for RTP audio packets.
Cos Video Class of Service for RTP video packets.
T.38
Adjust T.38 settings if T.38 Fax don't work.
Option Description
No T.38 Attributes in Re-invite SDP If this option is selected, SDP re-invite packet will not contain T.38
attributes.
Option Description
Error Correction Enable or disable Error Correction for the fax.
T.38 Max BitRate Adjust the max BitRate for T.38 fax.
Advanced SIP Settings
Option Description
User Agent Change the User-Agent field.
Send Remote Party ID Whether to send Remote-Party-ID in SIP header or not.
Note: This configuration only take effects on internal calls.

To set up for external calls, configure the Advance settings of SIP
trunk.
Send P Asserted Identify Whether to send P-Asserted-Identify in SIP header or not.

trunk.
Send Diversion ID Whether to send Diversion in SIP header or not.
If this option is selected, the Diversion value will be extension

number.

trunk.
Support Early Media Whether to support Early Media or not.
All Busy Mode for SIP Forking

• Check this option: When one of the terminals that register the
same extension number is busy in a call, the other terminals will
not receive calls.
• Uncheck this option: When one terminal is busy, the other
terminals will still be able to make and receive calls.
Option Description
Inband Progress This Inband Progress setting applies to all the extensions.
Note: To configure global Inband Progress setting, you need to

contact Yeastar support to configure a custom config file.
• Check this option: PBX will send a 183 Session Progress to

the extension when told to indicate ringing and will immediately
start sending ringing as audio.
• Uncheck this option: PBX will send a 180 Ringing to the
extension when told to indicate ringing and will NOT send it as
audio.
Get Caller ID From Decide the system will retrieve Caller ID from which header field.
Get DID From Decide the system will retrieve DID from which header field.
Note: If Remote-Party-ID is selected but the SIP trunk doesn't

support this, the system will retrieve DID fron INVITE header.
100rel Whether to support 100rel or not.
Support Message Request Whether to support SIP Message Request or not.
Maxptime Select or enter the Maxptime value.
Security
Blocked IP Address
The PBX will block an IP address for too many failed login attempts, too many failed
registration attempts, or too many failed authentications for Auto Provisioning.
The blocked IP addresses would be listed in the Blocked IP Address table. If a trusted IP
address was blocked by the PBX, you can go to Settings > System > Security > IP Auto
Defense > Blocked IP Address to delete the IP address.
Service
All the PBX service statuses and ports are displayed on the security Service page.
Go to Settings > System > Security > Service to configure the service settings.
Option Description
Auto Logout Time (min) After the set time of inactivity, the session will automatically log out. The
default time is 15 minutes.
SIP UDP Port SIP registration port. The default SIP UDP port is 5060.
Enable SIP TCP Whether to enable SIP TCP or not. The default port is 5060.
Enable SIP TLS Whether to enable SIP TLS or not. The default port is 5061.
User Permission
By default, the extension users can log in the system and check their own settings and
CDR. You can set different permission to the users according to their roles and duty.
User Types on the PBX

Super Admin
Super Admin has the highest privilege. The super administrator can access all pages on S-
Series Web and make all the configurations on the system.
• Username: admin
Administrator or Custom User

Administrator or Custom User is created by the Super Admin. The Super Admin sets the
privileges for those users according to their roles and duty.
• Username: The extension number or the email address of the extension user.
Note:
• Administrator and Custom User can have the same permission. The different
between the two role type:
◦ Administrator: All permissions are enabled by default.

◦ Custom User: No permission is enabled by default.
• Administrator and Custom User do not have permission to configure User

Permission.
Configure User Permission

To grand more privilege for a user or change the user's privilege, you need to configure the
User Permission on PBX.
Scenarios
In the following scenarios, you may need to add permissions for the extension users
according to their roles.
• For an HR, he/she may need the permission to add extension, configure extension's
outbound route privilege when there are new staffs.
• For a supervisor, he/she will have permission to check the CDR and recordings, and
have no permission to configure the system or other extensions.
Procedures
1. Log in the PBX web interface by the super admin account, go to Settings > System >
User Permission, click Add.
2. On the configuration page, select the User.
3. Set the Set Privilege As.
• Administrator: All the permissions are enabled for the user by default.
• Custom User: No permission is enabled for the user by default.
4. Click the Settings, CDR and Recordings, Monitor, Application, Contacts, and
Others tabs, and check or uncheck the relevant options for the user.
Results: When the user logs in the PBX web interface by the extension user account,
he/she can access the permitted configuration page.
Date and Time

To ensure that the time of logs and CDR is consistent with your local time , you need to
adjust the date and time of the PBX.
On the Date & Time configuration page, you can see the current time of the PBX.
You can set the PBX time to be synchronized with a NTP server or set the time manually.
Change the PBX Time
1. Go to Settings > System > Date & Time.

2. Select your current and correct Time Zone.
3. Check the option Daylight Saving Time if you need it in your place.
4. Click Save.
5. Reboot the PBX to take effect.
Email
The system email can be used to reset password, send voicemail to email, send alert event
emails, and send fax to email. To make these features work, you need to set up the PBX
system email.
Set up System Email
1. Go to Settings > System > Email to set up the system email.

• Sender Email Address: Enter an available email address.

• Email Address or Username: If the email server supports for User Name, enter
user name. If not,enter the email address.
• Password: Enter the login password of the email address.
• Outgoing Mail Server (SMTP): Enter the outgoing mail server and port according
to the email server.
• Incoming Mail Server (POP3): Enter the incoming mail server and port according
to the email server.
• Enable TLS: Enable or disable TLS during transferring/submitting your Email to
another SMTP server.
Note: For Gmail or Exchange server, you need to enable TLS.
• STARTTLS: If you enable TLS, the STARTTLS is enabled by default . If the mail
server doesn't support STARTTLS, do not select this option.
2. Click Test to check if the email works.
3. Click Save to save the email settings.
Auto Cleanup
Auto Cleanup is a feature that can auto clean your CDR, logs, voicemails, one-touch
recordings periodically.
Table 3. Configuration Parameters of Auto Cleanup

CDR Auto Cleanup
Max Number of CDR Set the maximum number of CDR that should be retained. The old CDR
will be deleted when the threshold is reached.
CDR Auto Cleanup
CDR Preservation Duration Set the maximum number of days that CDR should be retained.
Voicemail and One Touch Recording Auto Cleanup
Max Number of Files Set the maximum number of voicemail and one touch recording files
that should be retained. The old CDR will be deleted when the threshold
is reached.
Files Preservation Duration Set the maximum number of minutes that voicemails and one touch
recordings should be retained.
Logs Auto Cleanup
Max Size of Total Logs Limit the total size of pbxlog files in syslog.
The old logs will be deleted when the threshold is reached.
Logs Preservation Duration

Set the maximum number of days that system logs should be retained
respectively.
Max Number of Logs

Set the maximum number of event logs and operation logs that should
be retained. The old logs will be deleted when the threshold is reached.
Event Center
You can set the PBX to send notifications when specific events or errors occur, notifying you
via email.
For example, the system can automatically send a notification when the network connection
is lost, VoIP trunk registration is failed, storage volume is running out of space, or the
administrator password is changed.
Event Settings
Go to Settings > Event Center > Event Settings to configure the event settings.
• Record
indicates that Record function is enabled. When the event occurs, the PBX will
record the event in Event Log.
indicates that Record function is disabled. When the event occurs, the PBX will
NOT record the event in Event Log.
• Notification
indicates that Notification function is enabled. When the event occurs, the PBX will
send notification to the Notification Contacts.
indicates that Notification function is disabled. When the event occurs, the PBX will
NOT send notification to the Notification Contacts.
• Edit Notification
Click to edit the template of notification email.
Event Log
Go to Settings > Event Center > Event Log to search and check event logs.
Add Notification Contacts

You can set the PBX to send notifications when specific events or errors occur, notifying you
via email.
1. Go to Settings > Event Center > Notification Contacts, click Add.
2. On the configuration page, choose a contact and set the notification method.
• Choose Contact: Choose an extension user or choose Custom to add an

external contact.
• Notification Method: Select how to notify the contact when the event occurs.
◦ Email: The PBX will send notifications to the email address of the contact.
◦ Call Extension: The PBX will call the extension number of the contact when
the event occurs.
◦ Call Mobile: The PBX will call the mobile number of the contact when the
event occurs.
• Email: If you choose Notification Mode to Email, you need to set the email
address of the contact.
Remote Management
Yeastar Remote Management provides an affordable, low maintenance solution for easily
deploying Yeastar VoIP PBX and VoIP gateways across multiple locations, reducing
complexity and providing deep visibility and control.
Compatibility
The following Yeastar products supports Remote Management feature:
• Yeastar Cloud PBX: 81.4.0.X or later

• Yeastar S-Series VoIP PBX: 30.6.0.20 or later
• Yeastar TA1600/TA2400/TA3200 V3
Remote Management Guide

How to manage Yeastar products on the Remote Management platform, refer to the
Remote Management Guide.
API
Yeastar Cloud PBX provides API interfaces for you to integrate a third-party software or
device.
Compatibility
API feature is supported on Yeastar Cloud PBX v81.4.0.8 or later.
API Guide
For more information of API, refer to Yeastar Cloud PBX- API guide.
In the API guide, we introduces how to enable and configure API on Yeastar Cloud PBX,
and provides API references.
Maintenance
Maintenance gives you access to upgrade PBX firmware, check logs and troubleshooting.
Upgrade Firmware
Note:
• Back up the PBX configurations before you start to update the PBX firmware.
• If “Reset configuration to Factory Defaults” is enabled, the system will reset to factory
default settings after upgrading.
• When update the firmware, please don’t turn off the power. Or the system will get
damaged.
Related tasks
Create a Backup File
Upgrade Firmware
You can check for new version immediately or schedule automatic firmware check, if the
PBX has a new released version, upgrade the PBX firmware with just one click.
Note: Make sure that the PBX can access the Internet, or the upgrade will fail.
Check firmware and upgrade immediately
1. Go to Maintenance > Upgrade.

2. Click Check for new version to check for new firmware immediately.
If a new version is detected, you can click New check the release notes and decide
whether to upgrade or not.
Schedule automatic update
1. Go to Maintenance > Upgrade.

2. Select one of the following options:
• Never check for updates
This option disables Automatic Updates.

• Check for updates and let me choose whether to upgrade
This option notifies you that there are updates available. It requires user interaction
to download them and install them.
•
If a new version is detected, you can click New check the release notes and decide
whether to upgrade or not.
Backup and Restore

Go to Maintenance > Backup and restore, then you can back up all configurations of
PBX. Once backed up, back up file will be displayed in the list. You can upload backup file
from local client to PBX, or you can choose from backup list and restore.
Create a Backup File

You can create a backup file of the PBX settings on the PBX web interface.
Note:
• For PBX version before 81.5.0.7, you can not back up one-touch recording files and
voicemail files.
• For PBX version 81.5.0.7 or later, you can back up one-touch recording files and
voicemail files.
1. Go to Maintenance > Backup and Restore, click Backup.

2. Set the File Name.

The default file name contains the PBX model, firmware version, and backup date.
3. In the Memo field, enter notes for the backup file.
4. Select which configurations and files to back up.
5. Click Save.
The created backup file will appear on the Backup and Restore page.
Upload a Backup File

You can select a backup file from your local PC, and upload the file to the PBX.
Note: The file format is .bak and the file name should not contain special characters.
1. Go to Maintenance > Backup and Restore, click Backup.

2. Click Browse, and select your backup file to upload.
3. In the Memo field, enter notes for the backup file.
4. Click Upload.
The uploaded backup file will appear on the Backup and Restore page.
Restore a Backup File

After restore a backup file, the current configurations on your PBX will be OVERWRITTEN
with the backup data.
Note:
• You cannot restore a backup file that is downloaded from a different PBX model.
• If a backup file is created from a newer version of PBX, you cannot restore this backup
file. For example, restore a backup file (v30.7.0.35) to PBX (v30.6.0.16) would not
work.
• You can restore a backup file that is created from a older version of PBX. For example,
restore a backup file (v30.6.0.16) to PBX(v30.7.0.35) would work.
1. Go to Maintenance > Backup and Restore.
2. Choose a backup file, click .

A pop-up window will appear at the bottom-right of the web page.
3. Click Yes to reboot the PBX.

The PBX starts to restore data from the backup file.
Reboot the PBX

Reboot the PBX immediately on the PBX web interface or schedule auto reboot to keep the
system running smoothly.
Note: When the PBX is rebooting, all the on-going calls will be terminated.
Reboot the PBX Immediately
1. Go to Maintenance > Reboot, click Reboot.
Schedule Auto Reboot
1. Go to Maintenance > Reboot, check the option Enable Auto Reboot.

2. Set the frequency and time of auto reboot.

3. Click Save.
Reset the PBX

If you want to erase all the configurations on your PBX, you can reset the PBX to the factory
defaults.
1. Go to Maintenance > Reset, click Reset.
2. Note down the new activation code.

You need to use the new activation code to activate the PBX after reset.
3. Enter the verification code.
4. Click Reset.
System Log
The PBX automatically trace the PBX information, notices, warnings, errors, debug logs,
and web logs, then generate log files. You can download the system logs on the PBX web
interface, and check the logs.
Go to Maintenance > System Log to trace real-time logs or download the generated
system logs.
System Log Settings

The PBX traces different levels of log.
• Information: Basic information.

• Notice: NOTICE information.
• Warning: WARNING information.
• Error: ERROR information.
• DTMF: DTMF information.
• Time Log: Add time stamp of system logs.
• Debug: Select the following checkboxes to decide which type of debug logs to trace:
◦ Enable SIP Debug

◦ Enable RTP Debug
System Log
The PBX generates system logs everyday. The system logs are compressed into a tar file.
You can check the system logs on the System Log page.
Click to download the log file and open the log file by Notepad++ or other editor
software to check the logs.
The PBX provides the following kinds of system logs:
• PBX firmware version

• AMI logs
• API logs
• Asterisk guard logs
• App logs
• Module update logs
• Linkus Cloud Service logs
• SSH connection logs
• PnP logs
• Web logs
Operation Log
The PBX records all the users' operations, and keep the logs in Operation Log.
Go to Maintenance > Operation Log to search and check the operation logs.
Troubleshooting
Yeastar Cloud PBXEthernet Capture Tool, IP Ping and Traceroute can be used to debug
and capture packets.
Access the PBX via SSH

To debug the system, you can establish a temporary SSH connection on the PBX, and
access the PBX via SSH to check the logs.
1. Establish SSH connection.
a. Go to Maintenance > SSH Connection, click Establish Connection.
b. On the Time Settings dialog box, select the time period in the Timeout drop-down
menu, click OK.
Note: When the connection times out, the SSH tunnel will be closed, you will
not be able to access PBX via SSH.
The SSH connection information is displayed on the SSH Connection page.

2. Use PuTTY to access the PBX via SSH.
• Host Name (or IP address): Enter the PBX domain.

• Port: Enter the SSH port.
• Connection Type: Choose SSH.
3. To get more logs in the window, set the Lines of scrollback to a larger value, click
Apply.
4. Enter the username and password to access the PBX.

• login as: Enter support.
• password: Enter the SSH password.
Tip: After copying the SSH password, right click on the Putty interface to paste
password.
Capture Ethernet Packets

When there is a problem on the VoIP extensions or trunks, you can use the Ethernet
Capture Tool to capture Ethernet packet, and download the packet to analyze it.
1. Go to Maintenance > Troubleshooting > Ethernet Capture Tool.
2. Click Start.
The PBX will start to capture the Ethernet packet. During this time, you should
duplicate the problem of your VoIP trunks or extensions.
3. Click Stop to stop capturing packets.
4. Click Download to download the captured packet.
Decompress the .tarfile and use Wireshark software to open the packet file.
Ping IP Address
A ping utility sends test messages from the local client to a remote target over the TCP/IP
network connection. You can use IP Ping tool to test if the PBX can access the target IP
address.
1. Go to Maintenance > Troubleshooting > IP Ping.
2. In the Host field, enter the target domain name or IP address.
3. Click Start and check the result.
4. Click Stop to stop ping.
Traceroute
Traceroute is a common diagnostic tool for displaying the route (path) and measuring transit
delays of packets across a network.
1. Go to Maintenance > Troubleshooting > Traceroute.
2. In the Host field, enter the target domain name or IP address.
3. Click Start and check the result.

4. Click Stop to stop traceroute.
PBX Monitor
The PBX monitors the status of Trunks, Extensions, Concurrent Call, Conference.
You can log in the PBX web interface, go to PBX Monitor to check the real-time status of
your trunks, extensions, and conferences.
VoIP Trunk Status

Table 4.
Status Description
Registered
Registering
• Unreachable
• Registration failed, caused by:
◦ wrong password
◦ wrong authentication name
◦ wrong user name
◦ transport type inconsistent
Concurrent Call
Check the maximum supported concurrent calls and the real-time concurrent calls on the
PBX.
Monitor Conference
Check how many conferences are created on the PBX, and monitor the status of the
conferences.
Resource Monitor
Monitor the CPU usage, memory usage, disk utilization and network flow.
You can go to Resource Monitor to check the information or click the shortcut icon at the
right-top corner.
Information
Check the basic information of the PBX.
• Product
• Serial Number
• Hardware Version
• Software Version
• System Time: The current time on the PBX.
• Uptime: The system up time since the last reboot.
• Extensions/Max Extensions: The number of added extensions/Maximum number of
extensions allowed to be added
Network
Check the status of local network, cellular network, and VPN network.
Storage Usage
Check the usage of local storage in the PBX.
Recording Usage
CDR and Recordings

You can check CDR and auto recordings on the PBX web interface. CDR (Call Detail
Record) is a data record that contains various attributes of the call, such as time, duration,
call status, source number, and destination number, etc.
Searching Criteria
You can search CDR and recordings by the following criteria:
• Time: Set the start date and the end date to filter the call logs that are in the date
duration.
• Call From: The number or the name of the caller.
• Call To: The number or the name of the callee.
• Call Duration: The time between the call started and the call ended. Enter a value to
filter the call logs that have call duration equal or greater than this value.
• Talk Duration: The time between the call answered and the call ended. Enter a value
to filter the call logs that have talk duration equal or greater than this value.
• Status: Call status, including "answered", "no answered", "busy", "failed", and "has
voicemail".
• Communication Type: Communication type, including "internal", "inbound",
"outbound", "callback", "PBX warning call", "transfer", and "multisite interconnect".
• Include Recording Files: Check the option if you want to filter the calls that had been
recorded.
Search CDR and Recordings

1. Log in the PBX web interface, go to CDR & One Touch Recording.
2. Set the Time to filter the call logs during the date duration.
3. If you want to search recording files, check the option Include Recording Files.
4. Set other searching criteria.
5. Click Search.
The filtered call logs will display.
Fuzzy Search CDR and Recordings

By default, you need to enter an exact and complete phone number in the relevant
searching criteria, or you cannot get the search result. If you cannot remember the exact
number or the name, you can use Fuzzy Search feature.
1. Go to CDR & One Touch Recording.
2. Set the Time to filter the call logs during the date duration.
3. Enter a desired number or letters in Call From field or Call To field.
4. Check Number Fuzzy Search.
5. Set other searching criteria.
6. Click Search.
The call logs that match the fuzzy searching will display.
Download CDR and Recordings

You can download the searched CDR or recording files to your local PC.
1. Go to CDR & One Touch Recording.

2. Search the CDR and Recordings.
3. To download the searched CDR, click Download CDR.
4. To download the searched recording files, click Download Recordings.

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Admin Guide

About this guide

A SIP extension is based on SIP protocol.

Extension Basic Setup

Extension Creation Overview

• The limit of concurrent registrations is 4.

1. Go to Settings > PBX > Extensions, click Add.

• Extension: Enter the extension number.

4. Optional: Click Presence, Features, Advanced, or Call Permission tab to configure

5. Click Save and Apply.

Bulk Create Extensions

1. Go to Settings > PBX > Extensions, click Bulk Add.

• A random Registration Password and a random User Password will be

3. Optional: Click Features, Advanced, or Call Permission tab to configure other

4. Click Save and Apply.

Register a SIP Extension

1.Gather information of extension registration

• Domain name of PBX

2.Register the extension on a phone

3.Confirm registration status

You can do one of the followings to check if the extension is registered.

Change Extension Range

1. Log in PBX web interface, go to Settings > PBX > Extensions.

Bulk Edit Extensions

1. Log in PBX web interface, go to Settings > PBX > Extensions.

Bulk Edit Extension Names and Emails

3. Import the CSV file to the PBX.

a. Go to Settings > PBX > Extensions, click Import.

b. On the pop-up dialog, click Browse, select your CSV file.

d. Click Yes to check the log.

4. Check the imported extensions on your PBX.

1. Log in PBX web interface, go to Settings > PBX > Extensions.

Bulk Delete Extensions

1. Log in PBX web interface, go to Settings > PBX > Extensions.

1. Log in PBX web interface, go to Settings > PBX > Extensions.

1. Log in PBX web interface, go to Settings > PBX > Extensions.

Send Extension Information

a. Select Selected Extensions.

Edit 'Welcome Email' Template

2. On the Send Welcome Email page, click Edit Template.

Note: The variables in the email contents are unchangeable.

3. Edit the email subject and email contents.

4. Click Save and Apply.

2. Set the Name to help you identify the group.

3. In the Available box, select the extensions to the Selected box.

4. Click Save and Apply.

Manage Extension Groups

Edit extension groups

Delete extension group

Extension Groups Application

Yeastar Cloud PBX provides five presence statuses:

• Available: The user is online and ready for communication.

How does Presence benefit the user's work?

How to change Presence status?

Set Call Forwarding Rules & Presence Status

Set Call Forwarding Prompt

1. Go to Settings > PBX > Voice Prompts > Prompt Preference.

2. Unselect the checkbox of Play Call Forwarding Prompt.