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 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):
|
| 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:
|
| 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.
| 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 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 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.
- iLBC Settings
- The iLBC codec supports two modes: 20ms and 30ms frame length modes,
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:
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 |
|
| 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.
|
| 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 to delete the IP address.
Service
All the PBX service statuses and ports are displayed on the security Service page.
Go to 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
- Log in the PBX web interface by the super admin account, go to , click Add.
- On the configuration page, select the User.
- 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.
- Click the Settings, CDR and Recordings, Monitor, Application, Contacts, and Others tabs, and check or uncheck the relevant options for the user.
- 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
- Go to .
- Select your current and correct Time Zone.
- Check the option Daylight Saving Time if you need it in your place.
- Click Save.
- Reboot the PBX to take effect.
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
- Go to 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.
- Click Test to check if the email works.
- 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 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 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.
- 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.
