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.
Protocol Select the protocol that is provided by the VoIP provider.
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 is not needed.

User Phone Whether to add the parameter user=phone in the SIP INVITE packet.
Note: Enable this option if the SIP provider requires.
DID Settings
Settings Description
DID Number Direct Inward Dialing number, can be used to distinguish incoming calls.
Note: For Register Trunk, if the trunk DID number is different from the trunk authentication name, you need to enter the DID number.
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.

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.
Progress Inband Enable or disable Progress Inband.
  • Enable Progress Inband:

    When “RING” event is requested, always send 180 Ringing (if it hasn’t been sent yet) followed by 183 Session Progress and in-band audio.

  • Disable Progress Inband:

    Send 180 Ringing if 183 has not yet been sent establishing audio path. If audio path is established already (with 183) then send in-band ringing.

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 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 Adapt 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.