The default IP of S-Series VoIP PBX is 192.168.5.150.

This kind of error message, for example, “Your connection is not private” message in Chrome browser, is caused by SSL error. The S-Series VoIP PBX uses HTTPS protocol for login with a local IP address. But SSL certification authority don’t issue certificate for local IP address. There is nothing to worry about and you can proceed to visit the link. If you wish to get the SSL certificate, you need to bind a domain or public IP to the S-Series PBX, and then get the certification from the certification authority. You may need to pay for the certification.

The default administrator account in S-Series VoIP PBX is “admin” and “password”. And for better security, we highly suggest that you change the default password after your first login.

Please click and go to “My Settings” and change the password after you log in the Web GUI.

Input the PBX’s IP address and use the username and password to login. The username and default password are “extension number (e.g. 1000)” and “pass + extension number (e.g. pss1000)”. We highly suggest that you change the default password in extension edit page after your first login.

Go to Settings → PBX → Extensions and change the “User Password” in the extension edit

You could click “Forgot Password?” on the login page to reset your password. The prerequisites for this feature are: 1. The account is tied to an email address. If you are the administrator, you can configure by clicking and “My Settings”.

A. Go to Settings→ General→ Preferences and configure it in the “User Extensions”.

S-Series VoIP PBX can have extensions range from 2 digits to 7digits.

Go to Settings→ PBX→ Extensions, click “add” in the extension edit page. Tick “FXS” in “type” and choose the FXS span port. Notes: 1. Extensions on S-Series VOIP PBX can be “SIP”, “IAX” and “FXS” at the same time. 2. If the extension has ticked “FXS”, you must assign an FXS port for the extension. If the FXS port in the drop-down list is RED, it means the FXS port is being used and you cannot choose this port. If the port is BLACK, it means the port is idle.

A. S-Series VoIP PBX have passed compatibility test with lots of IP phone brands, such as Yealink, Grandstream, Cisco, Polycom, Snom, etc. The detailed configuration manual can be found here: https://www.yeastar.com/auto-provisioning-on-the-yeastar-s-series-voip-pbx/#phonesupported.

1. Make port mapping for communication ports of PBX on router. The default ports are: UDP (SIP): 5060 UDP (RTP): 10000-12000 2. Go to Settings →PBX→General→SIP→NAT. If you use static IP for S-Series VoIP PBX, please select “External IP Address” on “NAT settings” and input the public IP address of S-Series VoIP PBX. If you use dynamic IP for PBX, please input the domain in “External Host” field on the NAT settings page. 3. Got to Settings→ PBX→ Extensions. Choose the extension that needs to be registered remotely, click. Select “Advanced” page, enable “NAT” and “Register Remotely”. 4. Register to this extension configured. For more configuration details please refer to this link: https://support.yeastar.com/hc/en-us/articles/229524467-Configuring-Remote-Extensions

Mobility extension is used to bind mobile number with an extension, so this mobile number will have the corresponding extension’s call privilege, such as: make internal calls, make outgoing calls, and play voicemail unread, etc. When the user uses this mobile number to call into S-Series VoIP PBX, the PBX will treat it as the corresponding extension, and he/she will hear the dial tone. To configure mobility extension, follow the steps below. 1. Click Settings → PBX → Extensions, go to extension edit page. Click “Edit”. 2. Bind a mobile phone number in “Mobile Number” in “Basic” tab. 3. Click “Features”, and go to “Mobility Extension” settings. If you enable mobility extension, when the mobile number configured calls into S-Series VoIP PBX, it can act like this extension number, such as: make internal calls, make outgoing calls, and play voicemail unread, etc. If you enable Ring Simultaneously, the extension and the mobile phone will ring simultaneously. Please note that you must configure the specific prefix to make sure S-Series VoIP PBX can reach this mobile number according to the outbound route. Notes: 1. When a mobility extension makes an internal call, the caller ID presented on the called party’s screen will be the extension number instead of the mobile number. 2. The telephony line/trunk must have caller ID feature. Since PBX recognizes mobile number according to the caller ID detected. 3. Mobility extension feature can’t take effect if administrator has enabled call back feature in inbound routings.

SIP forking is a feature that enables an extension number to be registered by multiple SIP terminals simultaneously; and the amount of terminals can be 5 at most. Click Settings → Add/Edit Extension, and modify Concurrent Registrations.

In S-Series VoIP PBX there is a feature called “Extension Group”. We could custom the extension group by adding the extensions you want to this group and name it. Here’s how to configure the extension group: 1. Go to Settings → PBX → Extension Group, click “Add” in the extension list page. 2. Create an extension group, input the name of the group and choose the extensions to the selected box. 3. After creating the extension group, you could choose the extension group into the selected box on the extension edit page instead of choosing extensions one by one. For example, when you create a queue, you could select an extension group.

S-Series supports auto provisioning of IP phones from Yealink, Cisco, Snom, Panasonic, Mitel, Grandstream, Polycom, Htek, Fanvil, VTech, etc. Please find more details about the brands and models at: https://www.yeastar.com/compatible-sip-terminals/.

Auto provision supported IP phones are all thoroughly tested with S-Series VoIP PBX and are compatible. More details can be found here. IP phones which support standard SIP protocol are also compatible with S-Series VoIP PBX.

Prerequisites Create and edit configuration file according to your needs. The file format should be .cfg or .xml. Note: The configurations and the file name should meet the requirements of the IP phone manufacturers. Generally, the file name is the MAC address of the phone, e.g. 001565212512.cfg. Step 1. Connect your phones and the PBX to the same local network. Step 2. Make sure your phones are ready for auto provisioning. PnP mode Enable PnP if your IP phone supports PnP function. DHCP mode 1) If your IP phone does not support PnP, then enable DHCP. 2) Enable DHCP server on your PBX, set the TFTP address to the IP address of the PBX. Path: Settings > System > Security > Service Note: Make sure PBX is the only one DHCP server in your local network, or auto provisioning would fail. Step 3. Upload the configuration file into PBX. Path: Auto Provisioning > Upload Files > Upload a File. Step 4. Restart IP phones manually. IP phone will obtain configuration file from PBX and finish auto provisioning.

Step 1: Only IP phones which support SIP protocol are allowed to register to S-Series PBX, such as Panasonic KX-TSC11 and Cisco 8800. Step2: Change IP phone’s firmware version to the one that supports SIP protocol. The firmware is usually available at the official website of the phone manufacturer. If you cannot find it, please consult the phone manufacturer. Step 3: Finish registration by entering registration information on IP phone Web GUI.

Configure the time on Auto Provisioning configuration page. Step1. Select and edit the phone. Step2. On the Preference page, set the NTP server as the IP address of the S-Series PBX. Step3. Click “Save” and reboot the phone to take effect.

The configuration details can be found in this knowledge base article.

The configuration details can be found in this knowledge base article.

The configuration details can be found in this LDAP Server Guide.

1. Make sure that you have only one S-Series VoIP PBX in your local network. 2. Make sure that your IP phones and the S-Series PBX are in the same network, and the phones and the S-Series PBX could communicate with each other. 3. Reboot your phones manually. If after that, the function still cannot work, please submit a ticket on Support Portal for help.

No. Auto Provisioning will work only when IP phones and PBX are in the same local area network. Neither VLAN nor VPN can make the function work.

You can do one of the following configurations to disable Auto Provisioning: 1. Delete the IP phones from Auto Provisioning> Device List page. 2. Disable TFTP Sever on the PBX. 3. Uninstall Auto Provisioning application from the PBX.

Yes. Follow the steps below to upgrade your phones. 1. Register your phones to the PBX via Auto Provisioning. 2. Go to Auto Provisioning > Firmware Upgrade to upload the phone’s firmware file. 3. Reboot the phone to take effect. Note: only IP phones that can be auto provisioned by S-Series PBX support this kind of upgrade.

1. Go to Settings → PBX → Trunk, Click to add a VoIP trunk. 2. Select protocol and trunk type. 3. Input the VoIP information which is provided by the ISP. For example: Provider Name: Chinamobile Hostname/IP: the registration IP given by the ISP (e.g. 111.111.111.111) Port: 5060 Domain: The domain server given by ISP (e.g. ims.fj.chinamobile.com) Username: the username given by ISP (e.g. 865921111111@ims.fj.chinamobile.com ) Password: the account password given by the ISP From user: the trunk number given by the ISP (e.g. +86592111111) 4.For more details please refer to S-Series VoIP PBX Administrator Guide.

1. Go to PBX Monitor → Trunk to check the status of the trunk and make sure the trunk is available to use. 2. Go to Settings → PBX → Call Control → Outbound Routes, click in the outbound routes list page. 3. Fill in the details in the edit page and save. For example, you could configure the outbound routes as follows if you want 1000, 1005 and 1006 to use FXO1-7 to dial out. For more details, please refer to the

1. Go to Settings → PBX → Call Control → Inbound Routes, click in the inbound routes list page. 2. In the inbound routes edit page, you should select the trunk to the selected box and choose the destination to direct the call. The time condition can be set up in order to direct the call according to time periods. The example is shown as follows. For more details, please refer to S-Series VoIP PBX Administrator Guide.

1. Go to PBX → Call Control → Time Conditions, and add the conditions or the holiday if you need. 2. Go to Settings → PBX → Call Control → Inbound Routes, click to edit on the inbound routes list page.Tick “Enable Time Condition” and click + , add the time condition and set up the destination. The period which doesn`t match the time condition you create will lead to destination in “[other time]”.

1. Check the feature code on the inbound routes edit page. There are three feature codes in the following picture: “*801”, “*802” and “*800.” “*800” is used to cancel the override and reset the destination to the origin status. 2. Use the extension to press the feature code like “*801” to override the Time Condition “Workday”. 3. The overwritten Time Condition will have a check mark sign before it. 4. If you want to cancel the override and lead calls to the original destinations you have set up, you could use the extension to press the Reset feature code, like “*800” in the example.

S-Series VoIP PBX will automatically record outgoing calls into the AutoCLIP route list. When the called party calls back, the system will lead the call to the extension which dialed this number recently. To set up AutoCLIP: 1. Go to Settings → PBX → Call Control → AutoCLIP Routes, and open the AutoCLIP configuration page. 2. The fields in the AutoCLIP: Record Keep Time: how long each record will be kept in the AutoCLIP List. Digits Match: how many digits from the last digit of the incoming phone number will be used to match the AutoCLIP record. Member trunks: AutoCLIP Route will apply to which trunk and which trunk`s record will be kept in the list.

Go to Settings → PBX → Call Control → AutoCLIP Route, click “View AutoCLIP List” to check the information.

That is because you have unread voicemails, please dial “*2” to listen to it. There will be no beep sound again if all the voicemails are read

A. The blacklist is used to filter the number. If a number is in the blacklist, the user will not be able to dial out or dial in. The whitelist is used to put the number in the trust field. Notes: 1. You can input numbers with patterns in blacklist/whitelist. E.g. if you want to restrict the number starting with 0592 from being dialed, but allow calling the number range of 05925503301-05925503309, you could put “0592550330z” in the whitelist and “0592” in the blacklist. 2. Whitelist has higher priority than blacklist.

You can store the auto recording files in: 1. External storage devices, such as HDD, USB, TF/SD card 2. Network Drive

No, you cannot back up call recording files.

Go to Settings -> System -> Storage -> Auto Cleanup -> Recordings Auto Cleanup, and design max usage of device and longest preservation period.

The file format is .wav.

Yes, you can set up the prompt in Settings -> PBX -> Recording. If it’s an external call (inbound or outbound call through a trunk), the external party will hear the prompt no matter he is the called party or caller. If it’s an internal call, the called party will hear the prompt.

By default, only the super administrator has the permission to manage call recording files. The administrator can go to Settings > System > User Permission to add permission to desired users.

Yes. Go to Settings> PBX> Recordings, select the desired trunk, click Save and Apply. All the calls through the selected trunk will be recorded.

The configuration details can be found this knowledge base article.

S-Series VoIP PBX allows two ways to modify the system prompt: update manually or download online prompt. Manually update: 1. Go to Settings → PBX → Voice Prompts→ System Prompt 2. Browse a local “.tar” file and click “Upload” 3. You could download the system prompt file in the links below Afrikaans American English Arabic Australian English British English Chinese Czech Danish Deutschland Dutch Finnish French Canadian French Greek Hebrew Hungarian Italian Korean Norwegian Persian Polish Portuguese Brazilian Portuguese Russian Slovak Spanish Latin American Spanish Mexican Spanish Swedish Thai Turkish Download online prompt: Go to Settings → PBX → Voice Prompts → System Prompt Click “Download Online Prompt” and choose the prompt you want.

1. Super admin can set voicemail prompts played over all internal extensions. Step1: Go to Settings -> PBX -> Voice Prompts -> Custom Prompts, and upload prompt files to PBX. Step2: Go to Settings -> PBX -> General -> Voicemail -> Greeting Options and select the custom prompt to be applied. Step3: Click “Save” and “Apply”. 2. Extensions users can also set their own voicemail prompt. Step1: Login to user account via Web GUI. Step2: Go to Me-> Extension Settings, and change the voicemail prompt.

File format: “.wav”, “.WAV”, or “.gsm”. File size: smaller than 8MB. Supported wav format: PCM, 8K, 16bit, 128kbps; alaw (g.711), 8k, 8bit, 64kbps; u-law (g.711), 8k, 8bit, 64kbps; gsm, 6.10, 8k, 13kbps.

You can convert audio file format in two ways, by web tool or Wave Pad. Please visit this knowledge base article for details.

There is no duration limit for custom audio files.

Step 1: Delete the applied custom audio file on S-Series VoIP PBX Web interface. Step 2: Dial *2 on an extension. Step 3: Follow the prompt, and press 0->4->2 sequentially.

S412: 9.0-25.6W S20: 1.8-10.6W S50: 2.1-18.5W S100: 5.5-41.2W S300: 6.2-59.1W

S412: No S20: No S50: Yes, need mounting ears (Yeastar supplied) S100: Yes S300: Yes

S412: 4 S20: 2 S50: 4 S100: 8 S300: 12 Note: only one channel can use cellular data of 4G LTE SIM card. And the 4G LTE module must be installed in the first slot of the EX08 Expansion Board in Span1.

No. Storage space does not matter.

Write Speed >= 60MB/s Sandisk Extreme Pro Series Sandisk Extreme Series TOSHIBA EXCERIA Series Samsung Pro Series

If you order GSM/3G/4G modules, your S-Series VoIP PBX will be supplied with the relevant number of antennas, but not antenna extension cable. You need to buy an extension cable if you need it. Please refer to the knowledge base article to see specifications of Yeastar supplied antenna.

GSM Module: SIM800 3G Module: UC15 4G LTE Module: EC20 or EC25

No, the backup file will not work across different models.

You can select to back up the following files in S-Series PBX: 1. System Settings 2. Custom Prompts 3. Call Logs

Yes.

The name of the auto-generated backup files is in the format: model_firmware version_date_time. For example, the file name “S300_30.6.0.6_201711151539” means that the file was generated on S300 with firmware version 30.6.0.6 at 15:39 on November 15, 2017.

No, it’s unnecessary.

There are two ways available: 1. Upload local firmware to S-Series PBX 2. Upgrade through TFTP Server It is easier to upgrade by uploading local firmware. Please refer to the configuration at https://support.yeastar.com/hc/en-us/articles/115011710888-How-to-Do-Firmware-Update-for-S-Series VoIP PBX-PBX.

1. The PBX Center and Auto Provisioning App will be upgraded to the latest version automatically when you upgrade the S-Series’ firmware. 2. For other Apps, you need to upgrade them independently. Note: if you cannot upgrade an application to the latest version, you need to upgrade the S-Series’ firmware first.

1. Create a backup file on the S-Series PBX. 2. Download the backup file to your PC. 3. Reset the S-Series PBX. 4. Restore the backup file to the S-Series PBX. 5. Use the default password (iyeastar) to log in the S-Series PBX via SSH.

Download the firmware file here at https://www.yeastar.com/firmware-download/

The firmware file is encrypted so you can’t open it, but you don’t need to. Just upload the .bin file to your S-Series PBX to upgrade the firmware.

No, it won’t.

Yes, it’s necessary.

No, restoring of backup files is only allowed to be carried out between PBXs with the same firmware version.

Please click “Apply” on the upper-right corner of the page if you want to save the modified configuration. It will take 1-2 seconds to make it work.

S-Series VoIP PBX supports three Ethernet modes: Single, Dual and Bridge. Single: only use LAN port, WAN port will not be in use. Dual: the two ports are being used. But it is necessary to choose the default interface on Network configuration page. Bridge: LAN port will be used for uplink connection. WAN port will be used as bridge for PC connection.

When the Ethernet mode is dual, all the communication packet will go through the default interface except for the routes in the “Static Route”.

1. Go to Settings → System → User Permission, click s-se-faq-trunk-14 on user permission list page. 2. Tick the check box before the permission you want to assign.

1. Go to Settings → PBX → Recording, 2. Set up “Internal Calls Being Recorded Prompt” and “Outbound/Inbound Calls Being Recorded Prompt”, click “Save”.

The default ports your need to forward are: Linkus server port: TCP 8111 & UDP 8111 SIP port: UDP 5060 RTP ports: UDP 10000-12000. Note: TCP 8111 is only required when downloading and playing one touch recordings and voicemails in Linkus Mobile Client.

The built-in music on hold files are distributed under the Creative Commons Attribution-ShareAlike 3.0 license through explicit permission from their authors. The license can be found here.