SIP Trunk Settings

This topic describes all the settings on a SIP trunk for reference.

Basic settings

Basic
Setting Description
Name Give this trunk a name to help you identify it.
Trunk Status Enable or disable the trunk.
Select ITSP Template Select the country of your ITSP.
Note: If no SIP trunk template is provided for your ITSP, select General.
ITSP Select your ITSP from the list of certified SIP trunk providers.
Detailed Configuration
Setting Description
Trunk Type Select a trunk type:
  • Register Trunk
  • Peer Trunk (DID Based)
  • Peer Trunk (Port Based)
    Note: Yeastar P-Series Cloud Edition supports up to 5 Peer Trunk (Port Based).
  • Peer Trunk (Private Network)
  • Account Trunk
Register Trunk
Transport Select the transport that is provided by the ITSP.
Note: If you select TCP, make sure SIP TCP Port is enabled (Path: PBX Settings > SIP Settings > General > Basic > SIP TCP Port.
Hostname/IP Enter the IP address or the domain of the ITSP.
Port Enter the SIP port provided by the ITSP.
Domain Enter the domain in SIP URI of a specific header like From, To header.
Note: If the domain is not provided by ITSP, enter the same value as Hostname/IP.
Username Enter the username to register to the ITSP.
Password Enter the password that is associated with the username.
Authentication Name Enter the authentication name to register to the ITSP.
Enable Outbound Proxy If the trunk is configured to use an outbound proxy server, when users make outbound calls through this trunk, all the SIP packets will be sent to the outbound proxy server.
Note: Contact your ITSP to check if they support outbound proxy, then configure outbound proxy settings under their guidance.
Peer Trunk (DID Based/Port Based/Private Network)
Transport Select the transport that is provided by the ITSP.
Note: If you select TCP, make sure SIP TCP Port is enabled (Path: PBX Settings > SIP Settings > General > Basic > SIP TCP Port.
Static IP Address The Static IP Address is assigned automatically, this field can not be edited.
Port The port is assigned automatically, this field can not be edited..
Hostname/IP Enter the IP address or the domain of the ITSP.
Port Enter the SIP port provided by the ITSP.
Domain Enter the domain in SIP URI of a specific header like From, To header.
Note: If the domain is not provided by ITSP, enter the same value as Hostname/IP.
Account Trunk
Transport Select the transport for a third-party device to register with.
Note: If you select TCP, make sure SIP TCP Port is enabled (Path: PBX Settings > SIP Settings > General > Basic > SIP TCP Port.
Username Specify a username for the trunk.
Note: The username is regarded as the trunk number.
Password Specify a password that is associated with the username.
Authentication Name Specify an authentication name for a third-party device to register with.

Advanced settings

The advanced settings of VoIP trunk require professional knowledge of SIP protocol. Incorrect configurations may cause calling issues. It is wise to leave the default settings provided on the SIP trunk page. However, for a few fields, you need to change them to suit your situation.

The following settings are included on the Advanced page.

Codec Setting

Each newly created SIP trunk has a default preferred codec list. However, the default codec list may not match the codecs supported by your ITSP. To maximize the quality of calls and the amount of bandwidth used for calls, you can configure your preferred codec list to match the settings that your ITSP supports.

Yeastar P-Series Cloud Edition supports the following codecs:

  • u-law
  • a-law
  • G729A
  • GSM
  • H264
  • H261
  • H263
  • H263P
  • iLBC
  • G722
  • G726
  • SPEEX
  • ADPCM
  • MPEG4
  • VP8
  • Opus
VoIP Setting
Setting Description
DTMF Mode Set the default mode for sending DTMF tones.
  • RFC4733 (RFC2833): DTMF will be carried in the RTP stream in different RTP packets rather 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.
Qualify Enable this option to send SIP OPTION packet to SIP device to check if the device is up.
Enable SRTP Enable or disable SRTP (encrypted RTP) for the trunk.
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.

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.
Ignore 183 Message without SDP This option determines the way PBX handles 183 messages without SDP.
  • Check this option: PBX will not forward 183 messages that don't contain SDP.
  • Uncheck this option: PBX will process all the 183 messages without SDP to those with SDP and forward them.
Call Restriction
Setting Description
Call Restriction Type Specify based on which type of calls to restrict the max concurrent call number of this trunk.
  • Outbound Call: Only outbound calls will be restricted.
  • All: Both outbound calls and inbound calls will be restricted.
Maximum Concurrent Calls Specify the maximum number of concurrent calls allowed in this trunk. The default is Unlimited.

DIDs/DDIs

Direct Inward Dialling (DID), also called Direct Dial-in (DDI), is a service offered by telephone companies. For more information of DID concepts, see DID Number Overview.

Inbound Caller ID Reformatting

When a user calls in the PBX, the trunk provider may send a caller ID that is inconvenient for you to redial directly.

In this case, you can reformat inbound caller ID based on a trunk. The caller ID will be reformatted before it is sent to the called party.

For more information, see Reformat Inbound Caller ID based on a Trunk.

Outbound Caller ID

Outbound caller ID is the phone number or name that is displayed on the called party's device.

You can set up a global outbound caller ID for a trunk or assign caller IDs for extension users.

Note: By default, each trunk has a default phone number that will be displayed on the called party's device. Outbound Caller ID configuration requires support from the trunk provider. Contact your trunk provider first before you configure Outbound Caller ID, or the settings won't take effect and outbound calls may fail.

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.

For more information of outbound caller ID configurations, see Customize Outbound Caller IDs

SIP Headers

The SIP Headers settings require professional knowledge of SIP protocol. Incorrect configurations may cause calling issues. It is wise to leave the default settings provided on the SIP trunk page. However, for a few fields, you need to change them to suit your situation.

The following settings are included on the SIP Headers page.

Inbound Parameters
Setting Description
Get Caller ID From Decide from which header field will the trunk retrieve Caller ID.
Get DID From Different devices or providers may contain DID numbers in different SIP headers. When an inbound call through a SIP trunk reaches the PBX, the PBX needs to retrieve a correct DID number, or the call will fail.

Adjust the setting after analysis of the SIP packets sent from the trunk provider. The following SIP headers are available to select:

  • Follow System

    The trunk will follow the global Get DID From setting.

  • To
  • Invite
  • Diversion
  • 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
  • P-Called-Party-ID
  • P-Preferred-Identity
Outbound Parameters
For outbound calls, you can define the parameters included in the following SIP INVITE headers:
  • From

    A From header contains caller ID and caller ID name, which are defined as the followings in Yeastar P-Series Cloud Edition.

    • From User Part: Indicates caller ID.
    • From Display Name Part: Indicates caller ID name.

    You can define which parameters will be used in these two parts of a SIP From header.

  • Diversion
  • Remote Party ID
  • P-Asserted Identify
  • P-Preferred-Identity

Each SIP header has multiple options to define the parameters. The following tables describe the options.

Note: For different types of SIP trunk, the optional items are different.
Setting Description
[Default] The system selects a parameter by the following priority from top to bottom:
  • Outbound Route Outbound Caller ID
  • Extension's Outbound Caller ID in Trunk
  • Trunk Outbound Caller ID
  • Trunk Username
  • Extension Caller ID
  • The Originator Caller ID
[None] Do not send the parameter with the SIP INVITE packet.
Outbound Route Outbound Caller ID The outbound caller ID configured on the outbound route that is used for the outbound calls.
Extension's Outbound Caller ID in Trunk The extension's associated outbound caller ID with the trunk.
Trunk Outbound Caller ID The global outbound caller ID for the trunk (Trunk > Outbound Caller ID > General).
Trunk Username The username configured on the trunk.
Extension Caller ID The caller ID configured on the extension.
Originator Caller ID The Caller ID of the call originator (the first caller in the case that the call is transferred).
  • If the call originator is an external number, the external number will be taken.
  • If the call originator is an extension, the priority order will be Extension Outbound Caller ID → [Default].
Custom Define a custom value.
Other Settings
Setting Description
User Agent If the ITSP requires User Agent for authentication, enter the User Agent information that is provided by the ITSP.
Realm Realm is a string displayed to users so they know which username and password to use.
Note: If you don't know what to fill in, contact your service provider for further instructions.
Send Privacy ID Whether to send the Privacy ID in SIP header or not. The default is unchecked.
User Phone Whether to add the parameter user=phone as a request line in the header field of the SIP INVITE packet.
Note: Enable this option only when the SIP provider requires.
100rel Whether to support 100rel or not.
Maxptime Select the value of the maxptime used when the PBX sends the INVITE packet.
Note: If you select [Default], PBX will send a corresponding maxptime value according to the codec that is used for the outbound call.
Support P-Early-Media Set whether the P-Early-Media field is included in the INVITE packet.