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What’s new in Yeastar S-Series PBX Firmware 30.1.0.7

What’s new in Yeastar S-Series PBX Firmware 30.1.0.7

After months of development, the Yeastar S-Series PBX Firmware 30.1.0.7 is now available. This update introduces changes that improve our ability to a number of useful new features and squash pesky bugs, since firmware release 30.0.0.40.

Here is an overview of the changes brought by Yeastar S-Series PBX Firmware 30.1.0.7:

NEW FEATURES

  • New App Conference Panel.

One of the most noticeable changes we made in the new firmware is the new app. Enter S-Series VoIP PBX App Center, you will find the new App Conference Panel. This Conference Panel could realize all conference management based on Web interface. You can initiate and administrate a conference on Web.

  • SLA feature.

Path: Settings > PBX > Call Control > SLA
The Shared Line Appearance (SLA) feature helps users share VoIP trunks and FXO trunks. It also helps monitor the status of the shared line. When an outgoing call is made by the user, all members in the SLA group are informed about the call and will be blocked from this line appearance until the line goes back to idle state or the call is put on hold. When an incoming call is received, all the members are informed of it and may join it depending on the line appearance linked with the SLA extension.

  • New network mode “Single”.

Path: Settings > System > Network > Basic Settings
If you choose Single mode, only LAN port will be used for uplink connection. WAN port is disabled. The default network mode is Single.

  • Notifications on taskbar.

On the upper-right corner, you can see the new added Notification icon and Recourse Monitor icon. Click taskbar-click to check news and alerts, click taskbar-click2 to check the PBX information, network status and storage status.

taskbar
  • Web alert when the PBX power is off or the PBX is off-line.

If the PBX power is off or any problem of the system network, a dialog will display on the webpage to inform you.

  • Added SIP “Jitter Buffer” settings.

Path: Settings > PBX > General > SIP > Jitter Buffer
Jitter is the variation in the time between packets arriving on a VoIP system. These variations can be caused by network congestion, timing drift or route changes. Jitter buffers are used to counter delay or latency, dropped packets, and jitter. They temporarily store arriving packets to minimize jitter and discard packets that arrive too late.
Configure the Jitter Buffer settings on S-Series PBX will improve the call quality through VoIP.

  • Support for importing and exporting extensions.

Path: Settings > PBX > Extensions
You can export an extension file from the PBX and use it as a sample to start with. The file format is csv.

Import and Export Extension on Yeastar S-Series VoIP PBX
  • Support for the following Tone Regions: Turkey, Korea, Serbia, Panama.

Path: Settings > PBX > General > Preferences
We add the new support for the tong regions including Turkey, Korea, Serbia, Panama. Users can select the tone region according to the S-Series PBX location.

  • Support for Spanish Web User Interface language.

Users can switch the web language to Spanish.

  • Conference status on PBX Monitor panel.

Click PBX Monitor, here you can monitor the conference status.

Monitor conference panel- Yeastar S-Series VoIP PBX
  • “Echo Cancellation” setting on GSM/3G trunk and FXO trunk.

Path: Settings > PBX > Trunks > GSM Trunk / FXO Trunk
Check the option “Echo Cancellation” to enable echo cancellation for the trunk.

  • Support for registering SIP trunk with random SIP port.

Path: Settings > PBX > General > SIP > General > Local SIP Port
The random port in the port range will be used when sending packets to SIP server. The default range is 5062-5082.

random SIP port- S-Series VoIP PBX

OPTIMIZATION

  • If you do not change login password and set up an email, the system will remind you to do the configuration every time you log in.
  • Administrator account (admin) Password has a 8-63 character limit; the password must contain uppercase letters, lowercase letters and numbers.
  • Extension account Password has a 6-63 character limit; the following characters are not allowed: &; ” ‘ < > |.
  • Your S-Series PBX’s system prompts would be updated to the newest version automatically after firmware upgrade.
  • If you change the system service port (like FTP, HTTP, SSH etc.), the system will remind you to reconfigure the firewall settings.
  • AutoCLIP routing could match incoming numbers with area codes and special character “+”.
  • Added “SMS to Email” and “Email to SMS” failure records in Event Log.
  • For Logs Auto Cleanup, the “Logs Preservation Duration” setting is for system logs, the “Max Number of Logs” setting is for operation logs.
  • The callback list will display each callback’s detailed destination.
  • Limit the Network Drive number:
    S20 Maximum Network Drive: 2
    S50 Maximum Network Drive: 2
    S100 Maximum Network Drive: 4
    S300 Maximum Network Drive: 4
  • The system will log you off from the web if you do not do any operation on the S-Series PBX web GUI in half an hour.
  • Added CPU temperature information in system logs.

BUG FIXES

  • Fixed the DNS SRV Lookup issue: if the DNS SRV analysis result was a domain, a VoIP trunk registered using the domain would not work.
  • Fixed VoIP Trunk issue: if entering an IP address in the “Hostname/IP” field and entering a domain in the “Domain” field, the trunk could not be registered.
  • Fixed the queue ring strategy issue: if a queue agent has registered his own extension number on multiple phones, incoming calls could only reach one of the agent’s phones.
  • Fixed the compatibility issue with VoIP provider netelip.
  • Fixed the issue that sending email to SMS would fail if the content exceeded the length limit.
  • Fixed the issue that the system would automatically make outbound calls through FXO trunks.
  • Fixed the Event Center issue: WAN port failure was not recorded in the event logs.
  • Fixed the dual mode network issue: if one Ethernet port used VLAN, the other Ethernet port could not work properly.
  • Fixed the IE11 compatibility issue: users could not play recording files on PBX web GUI.
  • Fixed the Holiday issue: the holiday worked based on Time Zone GMT +0.
  • Fixed the E1 trunk issue: the call quality was bad when using the E1 trunk.
  • Fixed the Event Center issue: the event center repeatedly recorded SIP trunk registration failure.
  • Fixed the Call Log issue: if you configured DOD numbers on a SIP trunk, and made outbound calls through the SIP trunk, the “Call From” value would be wrong in call logs.

APPS

The Yeastar S-Series PBX Firmware 30.1.0.7 updates some Apps with new features in APP Center.

  • LDAP Server (Release Note)
    Added support for importing and exporting LDAP contacts.
  • Auto Provisioning (Release Note)
    1. Auto Provisioning supports the following IP phones(check here for all supported phone models and corresponding configuration guide):
    Yealink T19P-E2, T21P-E2, T23G, T23P, T27G, T29G, T40P, T41P, T42G, T46G, T48G, T49G, W52P, W56PFanvil C01, C400, C58/C58P, C600, X3/X3P/X3SP, X5/X5GSnom 320, 710, 715, 720, 725, 760, 765, D305, D315, D345, D375Grandstream GXP1100, GXP1105, GXP1160, GXP1165, GXP1400, GXP1405, GXP1450, GXP1610, GXP1620, GXP1625, GXP1628, GXP1630, GXP2130, GXP2135, GXP2140, GXP2160, GXP2170, GXP2200, GXV3240, GXV3275Htek UC802, UC803, UC804, UC806, UC840, UC842, UC860, UC862, UC924, UC926 Tiptel 3110, 3120, 3130, 3210, 3220, 3230, 3240, 3245, 3275, 3220XL
    1. Fixed the issue that if you reset the system, Auto Provisioning could not work properly with TFTP server.

Check for new update in your S-Series VoIP PBX Web interface, or download the latest firmware here.

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User Login

Instantaneously receive user login credentials via email once the extension has been deployed by the adminstrator.

Log in conveniently via any web browser for access to the user portal.

Integrations

Integrate various CRM software for better customer service and satisfaction.

Enable transcription of audio messages into text with integration of third-party transcription service.

Enable API for integration with various third-party integrations for advanced performance such as call center and hotel reception services.

System

Provision email SMTP and template settings for event notifications to be sent out.

Manage storage allocation and duration of various types of data being stored on the PBX.

Determine the types of event notifications to be sent out, and the parties receiving them.

IVR

Add various audio voice prompts to be played according to different call scenarios.

Add voice prompts on multiple levels for more effecient and precise inbound call routing.

Voicemail

Customize default voicemail settings for each user extension.

Add various voicemail messages for allocation to different call scenarios and users.

Provision specific rules to how voicemail messages are played over the PBX system for better caller experience.

Reports and CDR

All call log activities are stored, with the option of having audio conversations recorded by default or for specific extensions.

Use intuitive visual call reports, to check various call statistics over extensions, trunks, queues, ring groups, and more.

Call reports can be viewed and downloaded directly from the PBX, or scheduled to be sent via email to respective parties.

Security

Restrict a general group or specific IP addresses to prevent unauthorized access to your PBX.

Limit the number of otbound calls users are allowed to make within a time period to protect against toll fraud.

Restrict certain country codes for outbound calls either company-wide or on certain extensions.

User Login

Determine extension ranges for different functions.

Define use of different voice prompts depending on call scenarios or intended function.

Provision basic and advanced SIP settings to be used on the PBX.

Recordings

Record all calls made on specified trunks, including inbound and outbound calls.

Recordings

Record all calls made on specified extensions, conferences, or call queues.

Paging and Intercom

Create different types of paging and intercom broadcast groups for selected extensions.

Set a time schedule to automatically start your broadcast.

Maintenance

Check for and download firmware upgrades to ensure that your PBX is always up-to-date.

Perform back-ups to prevent data and settings from getting lost.

Perform troubleshooting on your own to debug.

Conference

Pre-set conference settings and dial-in passwords for security, and allow internal users to send conference invites to external parties.

Speed Dial

Create multiple speed dial numbers to respective external numbers frequently used for outgoing calls.

Speed dial is activated when the set prefix (default or customized) is dialed.

Ring Group

Create multiple ring groups and determine custom ring strategies and failover settings for each.

Call Queue

Create different call queues with specific ring strategies and failover settings.

Static agents permanently answer calls in queue, while dynamic agents can log in and out. Allocate different extensions to each group.

Determine what actions are to be allowed for Mangers or Agents to manage on the Queue Panel.

Call Control

Have calls received on specific SIP trunks routed to designated extensions or destinations within the PBX.

Set-up dial patterns so that outbound calls are routed to their intended destinations.

Determine how calls are routed and received at different times of a day.

Contacts

Store, retrieve, and share company contact details easily on your PBX.

Organize company contacts into dedicated phonebooks and determine accessibility rights to extension groups for each.

Bulk import a list of contacts instead of having to key details in individually.

Role and Client Permission

Pre-defined user roles within the PBX are conveniently available for allocation, or create your own set of permissions.

Restrict users from viewing specific extensions, departments, or extension groups with Extension Visibility.

Allow or resrict specific users from managing or viewing certain company contacts

Auto Provision

Pre-defined setting parameters specific to respective IP phone models are conveniently available for easy set-up.

IP Phones can be quickly provisioned to your Yeastar PBX with just the pre-defined template and device MAC address.

Trunks

Register multiple SIP trunks on a single PBX as needed.

Different types of trunks can be created, Caller ID can be customized.

Extensions

View, deploy, add, and remove user extensions within your PBX system.

Configure default call and user settings for individual extensions.

Customize allocation of user extensions to specific ring groups.

Portal Login Page

Receive your administrator log-in credentials and set-up guide via email.

Log-in on any web-browser to get started with PBX set-up.

Installation Wizard will guide you through basic configurations.

User Login

Access all company contacts made available to you by the administrator.

Create your own personal contacts list for easy reference and dialing.

Bulk import contact details for convenient data entry.

User Login

Access and manage user settings for softphone application and voicemail.

Manage call routing and presence status for your individual extension.

Retrieve or download your personal call logs and recordings.

User Login

Receive your log-in credentials and set-up guide via email, sent by your admin.

Log-in on any web-browser to get started with PBX set-up.

Configure contacts and personal settings from the web portal.

Auto Provision

Add, remove, or modify IP phone settings from your web interface.

Upgrade IP phone firmware conveniently from your PBX.

Upload various default setting parameters specific to respective IP phone models for greater convenience during set-up.

Contacts

Store, retrieve, and share company contact details easily on your PBX.

Have detailed contact information for your vendors and clients conveniently stored .

Bulk add a list of contacts instead of having to key details in individually.

PBX Monitor

Monitor the status extensions, whether they aren idle, in use, or unavailable.

Know whether your SIP Trunk running properly, or experiencing problems.

Monitor the number of call conferences currently running on your PBX, and their individual durations.

Maintenance

Check for and download firmware upgrades to ensure that your PBX is always up-to-date.

Perform back-ups to prevent data and settings from getting lost.

Perform troubleshooting on your own to debug.

Call Detailed Report

All call log activities are stored, with the option of having audio conversations recorded by default or for specific extensions.

Filter by date, time, or extension number to retrieve specific call logs or recordings

Playback call recordings directly from the PBX or download to store externally.

Event Center

Determine what events and logs are to be recorded and/or have notifications sent.

Specify particular notifications to be sent to respective authorized personnel.

User Permission

Set up vaious groups of users for allocation to certain sets of access rules.

Determine what features a certain group of extensions can or cannot access.

Determine if users can access, play-back, or download call logs and recordings from the pbx.

Security

Restrict a general group or specific IP addresses to prevent unauthorized access to your PBX.

Determine specific the range of SIP ports for connection between your PBX and endpoints.

Restrict country codes for outbound calls either company-wide or on certain extensions.

Voice Prompt

Upload and designate customised audio recordings at multiple levels of a call journey.

Upload recorded music to fill the silence that would be heard by callers on hold.

Upload customised voice prompts to create a professional call experience.

General

Customize general call preferences either company-wide or for individual extensions.

Set-up feature dial codes for convenient user navigation to specific call features.

Determine how voicemails are retrieved, and also upload custom voicemail recordings.

Call Features

Add multiple automated responses to be played at different phases of a call journey.

Designate specific extensions to ring concurrently based on an external caller's dial pattern.

Customize messages and waiting times to keep external callers engaged when put on hold.

Call Control

Have calls received on specific SIP trunks routed to designated extensions within the PBX.

Set-up dial patterns so that outbound calls are routed to their intended destinations.

Determine how calls are routed and received at different times of a day.

Trunks

Register multiple SIP trunks on a single PBX as needed.

Provision SIP trunk settings and dedicate them to particular extensions.

Bulk add DDIs on each SIP trunk, for allocation to an individual or groups of extensions.

Extensions

View, deploy, add, and remove user extensions within your PBX system.

Allocate and deploy specific extension numbers to respective users via email.

Set-up default ring and presence status settings for individual users.

Deal One View

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View cases created for a Deal in One View.

View cases created for a Deal in One View.

Deal One View

View cases created for a Deal in One View.

View cases created for a Deal in One View.

View cases created for a Deal in One View.

Deal One View

View cases created for a Deal in One View.

View cases created for a Deal in One View.

View cases created for a Deal in One View.