Yeastar Cloud PBX SIP Trunk

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VoIP Trunks Introduction

VoIP Trunks are phone lines that transmit 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.

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.

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:
    1. 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.
    2. Configure DID settings for the trunk:

      To add signal DID:

      1. Select Add Signal DID.
      2. Enter the DID Number which is provided by the VoIP provider.
      3. 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.

      4. Click  and repeat steps i-ii to add another DID numbers.

      To bulk add DIDs:

      1. Select Bulk Add DID.
      2. Enter the DID Number Range which is provided by the VoIP provider, click Add.
      3. Select the checkbox of DNIS Name, enter DNIS name for a DID number.

        When users call the DID number, the DNIS name will be displayed on ringing phone.

  6. If your trunk is a Peer Trunk, complete the following configurations:
    • 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.
  7. Configure other VoIP trunk settings as your need.
  8. Click Save and Apply.

    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
  1. Go to Settings > PBX > Trunks, click Add.
  2. In the Name field, enter a trunk name.
  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：
    1. Select Add Single DID.
    2. Enter the DID Numbers which is provided by the VoIP provider.
    3. 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.

    4. Click  and repeat steps i-ii to add another DID numbers.

  7. Configure other VoIP trunk settings as your need.
  8. Click Save and Apply.

    You can check the trunk status in PBX Monitor. If the trunk status shows , the trunk is ready for use.

 

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.

  1. Go to Settings > PBX > Trunks, click Add.
  2. In the Name field, enter a trunk name.
  3. In the Select Country drop-down list, select General.
  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.
    • 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.
  6. Configure other VoIP trunk settings as your need.
  7. Click Save and Apply.

    You can check the trunk status in PBX Monitor. If the trunk status shows , the trunk is ready for use.

 

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.

  1. Go to Settings > PBX > Trunks, click Add.
  2. In the Name field, enter a trunk name.
  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.

  6. Configure other VoIP trunk settings as your need.
  7. Click Save and Apply.

    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.

 

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.
  4. Click Save and Apply.

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.

Settings Description
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
Settings Description
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
Settings Description
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 From Decide from which header field will the trunk retrieve Caller ID header.

  • [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.
Settings Description
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.
Settings Description
From Select which From value should be contained in the SIP INVITE headers when the call is transferred.
Diversion Select which Diversionvalue should be contained in the SIP INVITE headers when the call is transferred.
Remote Party ID Select which Remote Party ID value should be contained in the SIP INVITE headers when the call is transferred.
P Asserted Identify Select which P Asserted Identify value should be contained in the SIP INVITE headers when the call is transferred.
P-Preferred-Identity Select which P-Preferred-Identity value should be contained in the SIP INVITE headers when the call is transferred.
Other Settings
Settings Description
Maximum Channels Set the maximum number of concurrent calls on the trunk.

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, H263P, H264, MPEG4, iLBC, opus G729, G711 a-law, G711 u-law

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.

Settings Description
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 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.

  2. Go to Settings > PBX > Trunks, click Add.
  3. In the Name field, enter a trunk name.
  4. In the Select Country drop-down list, select General.
  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. Click Save and Apply.

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.

Settings Description
Name Enter the trunk name.
Trunk Status Enable or disable the trunk.
Trunk Type Select a trunk type.
Trunk Number

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.
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.

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Next: Yeastar Cloud PBX Call control

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