Yeastar Cloud PBX System Management

You are here:
Estimated reading time: 10 min

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

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.
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 a Line If checked, when users seize a line to place an outbound 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
Busy Camp-on
Enable Busy Camp-on *79
Disable Busy Camp-on *079
Time Condition
Time Condition Override *8
Intercom
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.
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, opus 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.
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.

Note: This configuration only take effects on internal calls. To set up for external calls, configure the Advance settings of SIP 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.

Note: This configuration only take effects on internal calls. To set up for external calls, configure the Advance settings of SIP 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.
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.
Enable uaCSTA connection If this option is enabled, the PBX will use uaCSTA (User Agent Computer Supported Telecommunications Application) to remotely control the IP Phone via Linkus Desktop Client CTI. Your IP Phone needs to support uaCSTA standard to use this function.

 

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 SettingsCDR and RecordingsMonitorApplicationContacts, and Others tabs, and check or uncheck the relevant options for the user.
  5. Click Save and Apply.

    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.

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 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.
  3. Click Save and Apply.

 

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.

 

Previous: Yeastar Cloud PBX Contacts
Next: Yeastar Cloud PBX Maintenance

Install linkus on mobile & desktop, next make some calls!

Don't worry, a free trial is a free trial

Book an online demo appointment with our sales specialists.

Don't worry, it's a free consultation and no-obligation to purchase.