External trunk settings


There are three types of external trunks: BYOC Carrier, BYOC PBX and External SIP/BYOC Premises. When you configure an external trunk, you need to configure several basic settings. Depending on your needs, you may also need to configure some of the more advanced settings. This reference describes all the settings that you find on the Create/Edit External Trunk page.

Note: While most these settings apply to all types of external trunks, some settings apply only to certain types of trunks. Those settings that are trunk type specific will be noted in the description.
Setting Description
External Trunk Name Use this box to assign the external trunk a descriptive name. This is the name that you will use to identify this trunk when you need to select an external trunk on the various telephony configuration pages in PureCloud.

Type

Use this list to select the type of external trunk that you want to create. There are three choices:

  • BYOC Carrier
  • BYOC PBX
  • External SIP

Type

BYOC Carrier and BYOC PBX only

When you select one of the BYOC trunks, you’ll see a second Type list.

  • If you select the BYOC Carrier trunk, you’ll need to select the type of carrier you want to use.
  • If you select the BYOC PBX trunk, you’ll need to select the type of PBX you want to use.

Trunk State

Use this switch to change the operational state of the external trunk.

The default setting is In-Service.

Protocol

External SIP, BYOC Carrier and BYOC PBX

Use this list to choose the trunk transport protocol variant.

  • UDP
  • TCP 
  • TLS (Not available for BYOC Carrier or BYOC PBX) 

In most cases, you’ll select UDP as the Protocol.

Listen Port

External SIP only

Use this box to specify the trunk transport listen port.

Common values for this setting in PureCloud are 5061 for TLS and 5060 for UDP and TCP.

Inbound / Termination

Setting Description

Inbound SIP Termination Identifier

BYOC Carrier and BYOC PBX only

Use this box to specify your termination identifier. PureCloud requires that a unique identifier exists in the INVITE to associate inbound calls with the appropriate PureCloud organization’s resources.

By default you can use the FQDN or the TGRP method for formatting your SIP INVITES.

To make it easy to understand how to format inbound INVITES, the user interface provides dynamic help. As soon as you begin typing an identifier in the Inbound SIP Termination Identifier box, the Inbound Request-URI Reference panel appears. This panel provides you with invite formatting examples that include your termination identifier and the address of your PureCloud organization’s region.

DNIS Routing

BYOC Carrier and BYOC PBX only

If your carrier’s requirements prevent you from using either the FQDN or the TGRP method for formatting your SIP INVITES, you can enable DNIS Routing. You can then format your SIP INVITES using the DNIS method. The same dynamic help is available.

For more information, see Configure SIP routing for a BYOC Cloud trunk.

Outbound

Setting Description

Outbound SIP Termination FQDN

Use this box to specify the FQDN portion of the outbound INVITE Request-URI.*

Outbound SIP TGRP Attribute

Use this box to specify the TGRP parameter of the outbound INVITE Request-URI.*

TGRP Context-ID 

If you specify an Outbound SIP TGRP Attribute, you must use this box to specify the trunk context parameter of the Request-URI.*

Outbound SIP DNIS

Use this box to specify the DNIS value for the outbound call attempt.*

* To make it easy to understand how to format outbound INVITES, the user interface provides dynamic help. As soon as you begin typing an identifier in on of the outbound SIP boxes box, the Outbound Request-URI Reference panel appears. This panel provides you with invite formatting examples that include your termination identifier.

For more information, see Configure SIP routing for a BYOC Cloud trunk.

SIP Servers or Proxies

Use the controls in this section to create a prioritized list of SIP servers or intermediate proxies to which all outgoing requests should be sent regardless of the destination address in the request. You can also add a port number.

If do not specify a port number, the inbound listen port will be used by default.


External SIP only

IP addresses added to the Outbound SIP Servers or Proxies section are automatically placed on the SIP Access Control Allow list.

Digest Authentication

Use this switch to enable or disable Digest Authentication. When enabled, digest authentication provides additional layer of security to the process of granting outbound requests.

If you enable digest authentication, you’ll need to specify the associated realm, as well as the username, and password to use as the authentication credentials.

The default setting is Disabled.

Realm

Use this box to specify the domain name of the SIP realm that is used to authenticate outbound requests.

User Name

Use this box to specify the username to use to authenticate the call.

Password

Use this box to specify the password to use to authenticate the call.

By default the password is masked, but you can select the Show Password check box to see the password in plain-text.

Calling

Setting Description

Address

Use this box to specify the number that you want to appear as the caller ID for outbound calls. Enter this number in E.164 format, which includes the plus sign (+) and the country code.

In most cases, you’ll enter the main contact number for your call center.

Address Override Method

Use this list to choose the address override method. The two address override methods include:

  • Always: This setting always uses the number specified in the Address box for the caller ID.
  • Unassigned DID: This setting only uses the number specified in the Address box for the caller ID, when no DID number is assigned to the user’s phone.

The default setting is Always.

Name

Use this box to specify the name that you want to appear in the caller ID for outbound calls.

In most cases, enter the name of your company.

Note: If you specify a Calling Name, be aware that it takes precedence over the Calling Party Name setting in a queue. In other words, the Calling Party Name you configure for a queue never appears if a Calling Name is configured on a trunk. For more information, see Create and configure queues.

Name Override Method

Use this list to choose the name override method. The two name override methods include:

  • Always: This setting always uses the origination calling name specified in the Name box for the caller ID.
  • Unassigned DID: This setting only uses the origination calling name specified in the Name box for the caller ID, when no name is assigned to the user’s phone.

The default setting is Always.

Availability

Setting Description

External SIP only

Use this switch to enable or disable the ability to send availability requests.

The default setting is Disabled.

Interval

External SIP only

Use this box to enter the time interval in seconds before the availability request should be sent.

Registration

Setting Description

External SIP only

Use this switch to enable or disable registration.

The default setting is Disabled.

Expiration Interval

External SIP only

Use this box to specify the number of seconds to wait before the registration expires.

Address of Record

External SIP only

Use one of these options to specify the URI or the user portion of the registration. (The user portion will be combined with the Edge Interface IP.)

SIP Access Control

Setting Description

Use Source Address

External SIP only

Use this switch to determine whether ACL matching should use the SIP messaging source address (Yes) or the VIA header originating address (No).

The default setting is Yes.

Allow the Following Addresses

Use the controls in this section to enter and build a list of IP or CIDR addresses to which you want to allow SIP access.

Always Deny the Following Addresses

External SIP only

Use the controls n this section to enter and build a list of IP or CIDR addresses to which you want to deny SIP access.

Allow All

External SIP only

Select this check box if you want to allow access to any IP or CIDR address.

Note: Allowing all addresses is a security risk.

External Trunk Configuration


Setting Description

Call Draining

Use this switch to enable or disable call draining. Call draining is designed to allow calls to complete normally when a Trunk is taken out of service.

The default setting is Enabled.

Language

Use this list to choose the language that you want to use for all calls that come in on this trunk.

Note: This language can be overridden by settings in Architect.

Calls

Setting Description

Max Calls

Use this box to specify the maximum number of combined active inbound and outbound calls that are allowed on this trunk.

Max Call Rate

Use this box to specify the average number of calls per time period that are allowed on this trunk. This rate applies to both inbound and outbound calls.

You can specify this number as a decimal or a fraction. For example:

  • 40/5s: 40 calls/5 seconds
  • 3.5/2m: 3.5 calls/2 minutes
  • 2.5/1h:  2.5 calls/1 hour

Max Dial Timeout

Use this box to specify the maximum number of seconds for a delay before an outgoing call attempt is aborted.

Asserted Identity

Setting Description

Use this switch to enable or disable asserted identity. This allows you to choose how you want the authenticated name and address to display.

The default setting is Disabled.

URI

Use this box to specify an alternate URI that the protocol can use to identify itself. (As opposed to its calling address.)

Name

Use this box to specify an alternate name that the protocol can use to identify itself. (As opposed to its caller ID.)



Setting Description

Transport DSCP Value

External SIP only

Use this list to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP and RTCP packets.

The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP and RTCP packet. The range of values available is 00 (0,000000) through 3F (63, 111111). The default value is 18 (24,011000) CS3.

Retryable Reason Codes

Use the box to enter a list of valid SIP reason codes. If an outbound call that is made on this line returns one of the SIP reason codes in this list, then that call is retried on the next line.

You can specify individual reason codes or ranges of reason codes, separated by commas.  

By default, PureCloud automatically enters a list of the most common retryable codes in the Retryable Reason Codes field.

The default code list: 500-599

Retryable Cause Codes

Use this box to enter a list of valid Q.850 cause codes. If an outbound call that is made on this line returns one of the Q.850 cause codes in this list, then that call is retried on the next line.

You can specify individual reason codes or ranges of reason codes, separated by commas.  

By default, PureCloud automatically enters a list of the most common retryable codes in the Retryable Cause Codes field.

The default code list:

1-5,25,27,28,31,34,38,41,42,44,46,62,63,79,91,96,97,99,100,103

TCP Settings

Setting Description

TCP Connection Timeout

External SIP only

Use this box to specify the number of seconds to delay before marking the TCP Connection to the remote IP address as failed and marking the port as unreachable.

TCP Connection Idle Timeout

External SIP only

Use this box to specify the number of seconds that a TCP connection can remain idle before being automatically closed.

TLS Settings

Setting Description

Mutual Authentication

External SIP only

Use this switch to enable or disable the mutual authentication requirement when negotiating the TLS handshake.

The default setting is Disabled.

SIPS URI scheme

External SIP only

Use this switch to enable or disable the sending of a SIPS URI scheme when it is configured for the TLS transport protocol.

The default setting is Disabled.

Method

External SIP only

Use this list to choose which SSL 0r TLS method version to use. Available choices are:

  • SSL v2.3
  • SSL v3
  • TLS v1
  • TLS v1.1
  • TLS v1.2

Ciphers

External SIP only

Use this list to select and build a preferred order list of TLS ciphers. 

Available choices are:

  • TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA384
  • TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA
  • TLS_DHE_RSA_WITH_AES_256_CBC_SHA256
  • TLS_DHE_RSA_WITH_AES_256_CBC_SHA
  • TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA256
  • TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA
  • TLS_DHE_RSA_WITH_AES_128_CBC_SHA256
  • TLS_DHE_RSA_WITH_AES_128_CBC_SHA
  • TLS_RSA_WITH_AES_128_GCM_SHA256
  • TLS_RSA_WITH_AES_128_CBC_SHA256
  • TLS_RSA_WITH_AES_128_CBC_SHA

Subject Alternative Names

External SIP only

Use this box to specify and build a list of subject alternative names to use for the secure interface.

Valid values must begin with one of the following prefixes:

  • DNS:
  • IP:
  • URI:
  • email:



Inbound

Setting Description

Identity Type

Use this list to select the type of address you want to use for inbound identity. Available choices are:

  • From
  • First Diversion Entry
  • Last Diversion Entry
  • Remote-Party-ID
  • P-Asserted-Identity

Outbound

Setting Description

Apply Header Privacy

Use this switch to enable or disable PureCloud’s capability to apply header privacy information.

When enabled, an agent can use *67 to request privacy. More specifically, this prevents PureCloud from sending the actual header information (typically the contact address) along with the call. Instead, PureCloud replaces the actual header information with the word Anonymous.

When disabled, an agent cannot use *67. Disable this setting if you do not want agents to represent your organization as private on a public telephony network.

The default setting is Enabled. 

Apply User Privacy

Use this switch to enable or disable PureCloud’s capability to apply user privacy information. 

When enabled, an agent can use *67 to request privacy. More specifically, this prevents PureCloud from sending ANI information with the call. Instead, PureCloud replaces the actual ANI with sip:anonymous@anonymous.invalid.

When disabled, an agent cannot use *67. Disable this setting if you do not want agents to represent your organization as private on a public telephony network.

The default setting is Enabled. 

Calling

Setting Description
Address Transformation Use the controls in this section to enter regular expressions and build an ordered list of regular expressions to essentially reformat addresses. The expressions will be applied in the order in the list. If the address matches the match expression, the format expression will be applied. You can add up to three entries to the list.
Match Regular Expression

Use this box to enter the match regular expression – the pattern that you want to search for in the external trunk number.

Format Regular Expression

Use this box to enter the format regular expression – the format that you want to use to display the result.

Address Digits Length

Use this box to specify the number of trailing digits from the outgoing origination address that is to be sent.

Address Omit + Prefix

Use this switch to enable or disable the ability to exclude the plus (+) prefix in the outgoing origination address that is to be sent.

The default setting is Enabled. 

Called

Setting Description
Address Transformation Use the controls in this section to enter regular expressions and build an ordered list of regular expressions to match and format addresses. The expressions will be applied in the order in the list. If the address matches the match expression, the format expression will be applied. You can add up to three entries to the list.
Match Regular Expression

Use this box to enter the match regular expression – the pattern that you want to search for in the external trunk number.

Format Regular Expression

Use this box to enter the format regular expression – the format that you want to use to display the result.

Address Digits Length

Use this box to specify the number of trailing digits from the outgoing destination address that is to be sent.

Address Omit + Prefix

Use this switch to enable or disable the ability to exclude the plus (+) prefix in the outgoing destination address that is to be sent.

The default setting is Enabled. 



Setting Description

DSCP Value

Use this list to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP and RTCP packets.

The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP and RTCP packet. The range of values available is 00 (0,000000) through 3F (63, 111111). The default value is 2E (46 101110) EF.

Media Method 

Use this list to choose the method that you want to use to offer an SDP (Session Description Protocol) to the other participant when making an outgoing call.  The offer proposes the set of media streams and codecs along with the IP addresses and ports to use. 

There are three choices for the Media Method:

  • Normal: Use the normal method, which sends an SDP Offer in the initial SIP INVITE request.
  • Delayed: Use the delayed method, which waits for an SDP Offer in a response before sending an SDP Answer.
  • Auto: Allow the system to choose between using the normal or the delayed method.

Preferred Codec List

Use the controls in this section to choose and build a preferred list of codecs. Available choices are:
  • audio/g722
  • audio/g729
  • audio/PCMA (g711 A-Law)
  • audio/PCMU (g711 µ-Law)
  • audio/opus

Note: PCMU and PCMA are also known as the g711 codec (the PCM stands for Pulse Code Modulation). PCMU (µ-Law) is primarily for use in North America and PCMA (A-Law) is primarily for use in other countries outside North America.

SRTP Cipher Suite List

Use the controls in this section to choose and build a preferred list of SRTP cipher suites to offer or allow in response. Available choices are:
  • AES_CM_128_HMAC_SHA1_32
  • AES_CM_128_HMAC_SHA1_80
  • AES_CM_192_HMAC_SHA1_32
  • AES_CM_192_HMAC_SHA1_80
  • AES_CM_256_HMAC_SHA1_32
  • AES_CM_256_HMAC_SHA1_80

Ringback

Use this switch to enable or disable the line ringback.

When enabled, this setting controls if a ringback should be generated and sent to the incoming trunk when a 18x response message that does not include an SDP is received\relayed from the outbound call.

The default setting is Enabled. 

Disconnect on Idle RTP

Use this switch to enable or disable the ability to disconnect a call when RTP is not received for an extended period of time.

The default setting is Enabled. 

DTMF Settings

Setting Description

DTMF Payload

Use this box to specify the payload type value to use when the DTMF Method type is RTP Events. Valid range is 96–127.  The default value is 101.

Valid only when DTMF Method value is set to RTP Events. 

DTMF Method

Use the list to select the method to use to transmit dual-tone multifrequency (DTMF) signaling. The default value is RTP Events.

There are three choices for the DTMF Method:

  • RTP Events: Enables out-of-band processing of events from the RTP stream (RFC 2833 or 4733).
  • In-band Audio: Enables the processing, detection, and synthesis of events from the audio codec stream.
  • None: Don’t use a DTMF method.

Recording

Setting Description

Line Recording

Use this switch to enable or disable line recording.

The default setting is Disabled. 

Audio Format

Use this list to select the audio format to use for recording. The available choices are:

  • audio/PCMU
  • audio/PCMA
  • audio/L16
  • audio/G726-32
  • audio/GSM
  • audio/x-truespeech
  • audio/opus

Dual Channel

Use this switch to enable or disable dual channel recording, which saves each channel in a separate stream.

The default setting is Disabled. 

Note: Dual channel recording only works with certain codecs. If you are using one of the following codecs, the Dual Channel recording setting is available:
  • PCMU
  • PCMA
  • L16
  • Opus

Automatic Level Control

Use this switch to enable or disable automatic level control for the recordings.

The default setting is Disabled. 

Continue on External Transfer

Use this switch to enable or disable the ability to continue active recordings on an external transfer that result in external to external connected calls.

The default setting is Disabled. 

Consent Required

Use this switch to enable or disable the requirement that a user must give consent before a trunk recording can begin.

The default setting is Disabled. 



Header / Invite

Setting Description

Conversation Headers

External SIP only

Use this switch to enable or disable the ability to insert the custom conversation header: “x-inin-cnv” with the UUID value into SIP messages.

The default setting is Enabled. 

From Header Hostname

Use one of these options to specify the name to replace default host name value in the From header on a SIP INVITE.

Routing Address

Use this list to choose which field in the inbound SIP INVITE request that you want to use for routing decisions. There are two choices:

  • To Header
  • Request-URI

Diversion Method

Use this list to choose how you want diversion information delivered to the remote end in the outbound SIP INVITE request. There are two choices:
  • None
  • Diversion Header

Asserted Identity Header

Use this list to choose how you want identity information delivered to the remote end in the outbound SIP INVITE request. There are three choices:
  • P-Asserted -identity
  • First Diversion Entry
  • Remote-Party-ID

Max Diversion Entries

Use this box to specify the maximum number of diversion entries to be included on an outbound call.

Request Target Address

External SIP only

Use this box to specify the target address to use for routing outbound SIP requests if the present takes precedence over the Request-URI.

Request URI Override

BYOC Carrier and BYOC PBX only

Use this box to specify the destination value for the Request-URI and the TO header, but still send calls to the destinations indicated in the Outbound SIP Servers list.

User to User Information (UUI)

Setting Description

UUI Passthrough

Use this switch to enable or disable the sending of UUI data for outgoing calls.

UUI is used to send small amounts of data along with call information between applications by embedding that data inside the SIP header. UUI data can be received and sent by Architect and Scripter on call flows. For more information, see Set UUI Data action.

The default setting is Disabled. 

Note: When enabling UUI Passthrough, keep in mind that there is no added security for the UUI data. As such, any sensitive data that could be transmitted via UUI should be encrypted at the client. 

Header: Type

Use this list to choose the type of UUI header information you want to use. There are three choices:

  • x-UserToUser: This is the Audiocodes proprietary header, which only includes the data. It does not use the protocol discriminator nor any of the other standard parameters and depending on the encoding you select, it is in the format: 

x-User-to-User: hexdata

x-User-to-User: ascii

  • User-To-User: This is the general header, which requires the use of the protocol discriminator in the format:

User-to-User: XXhexdata;encoding=hex;purpose=isdn-uui;content=isdn-uui

where XX is the protocol discriminator. 

  • User-To-User PD Attribute: This is the type that some gateways use where the protocol discriminator is specified separately in the format:

User-to-User: hexdata;pd=XX;encoding=hex;purpose=isdn-uui;content=isdn-uui

where XX is the protocol discriminator. 

Header: Encoding Format

Use this list to select the encoding format for the header. There are two choices:

  • hex
  • ascii

Header: Protocol Discriminator

Use this box to specify the two-digit Hexadecimal protocol discriminator. You can specify any integers, lowercase letters from a-f, and uppercase letters from A-F.

Note: If you select the X-UserToUser header type, this field is not available.

Static User Data

Setting Description
Static UUI

Use the Static UUI switch to enable or disable support for sending static UUI data. 

When you are sending UUI data with outgoing calls, you usually specify the UUI data in either Architect or Scripter. Specifying UUI data in that manner is the dynamic method. However, if you want to send the exact same UUI data with every outgoing call (static method), you can enable the Static User Data setting. You then specify the static UUI data in the following fields.

The default setting is Disabled. 

Header: Name

Use this box to specify a name for the header.

You can use a name that is the same as one of the standard header type names (X-UserToUser, User-to-User, or User-to-User PD Attribute) or you can specify a custom name for the header. 

If you specify both dynamic UUI data and static UUI data and each have different header names, PureCloud sends both the dynamic and static UUI data in the outgoing call.

Note: If you use one of the standard header type names, the descriptions/rules listed above in the User to User Information (UUI) | Header: Type section apply.

Header: Value

Use this box to specify the UUI data to place in the header using the appropriate format.

For example: 

00TestData;encoding=ascii;purpose=isdn;content=isdn-uui

Note: If you specify User-To-User as the header name, you must manually prepend the two characters to the value to act as the protocol discriminator.

Header: Priority

Use this list to specify a priority level for choosing the UUI data source.

If you enable the Enable Static User Data setting but may also specify UUI data in either Architect or Scripter, and the UUI data from both have the same header name, PureCloud will determine which UUI data value to send based on the Priority setting. 

  • If you select Low Priority, PureCloud will always choose the dynamic UUI data from Architect or Scripter.
  • If you select High Priority, PureCloud will always choose the static UUI data you specify here.

Note: If you select Low Priority and no dynamic UUI data is specified, then both the empty dynamic UUI data and the static UUI data will be sent.

Take Back and Transfer

Setting Description

Enable Take Back and Transfer

Use this switch to enable or disable the Take Back and Transfer feature.

The default setting is Disabled. 

When you enable the Take Back and Transfer feature, you enable the inbound REFER method. This allows an active call between a PureCloud agent (Transferee) and an external party A (Transferor) to be transferred by external party A and another external party B (Target). Once the transfer is complete, the Transferee and Target are connected and the Transferor releases all telephony resources.

Release Link Transfer

Setting Description

Enable Release Link Transfer

Use this switch to enable or disable the Release Link Transfer feature.

The default setting is Disabled. 

When you enable the Release Link Transfer feature, you enable the outbound REFER method. This setting allows an active call between a PureCloud agent (Transferor) and an external party A (Transferee) to transfer by the PureCloud agent to another external party (Target). After the transfer completes, the Transferee and Target connect and PureCloud releases all telephony resources; unless the call needs to be recorded.



Setting Description

Media Capture

External SIP, BYOC Carrier, and BYOC PBX

Use this switch to enable or disable media capture.

The default setting is Disabled. 

The Media Capture setting is designed to be enabled while you are working with PureCloud Technical Support personnel. Enabling it will generate an HPAA Packet File Format (HPAACAP) file that contains live packet streams that can be used for diagnostic and troubleshooting purposes. Therefore, you should only enable the Media Capture setting as directed by PureCloud Technical Support.

Warnings:
  • Enabling media capture can degrade performance and affect QoS.
  • Media capture will log all data entered into the system, including data entered via Secure IVR flows. This could include sensitive data that should not be exposed or captured. As such, if your organization is using Secure IVR, you should not enable the Media Capture setting.
  • If you are a PCI-compliant PureCloud organization and have the PCI DSS setting enabled, then you will not be able to enable media capture – the Media Capture setting will not be available.

For more information, see Enable Media Capture.

Protocol Capture

External SIP only

Use this switch to enable or disable protocol capture.

The default setting is Disabled. 

The Protocol Capture setting is designed to be enabled while you are working with PureCloud Technical Support personnel. Enabling it will generate a PCAP file that contains protocol-specific network information that can be used for diagnostic and troubleshooting purposes. Therefore, you should only enable the Protocol Capture setting as directed by PureCloud Technical Support.

Warnings:
  • Enabling protocol capture can degrade performance and affect QoS.
  • Protocol capture will log all data entered into the system, including data entered via Secure IVR flows. This could include sensitive data that should not be exposed or captured. As such, if your organization is using Secure IVR, you should not enable the Protocol Capture setting.
  • If you are a PCI-compliant PureCloud organization and have the PCI DSS setting enabled, then you will not be able to enable protocol capture – the Protocol Capture setting will not be available.

For more information, see Enable Protocol Capture.

Capture Until

Use the calendar and clock controls to specify how long you want to collect data.



The Custom option is designed to allow PureCloud Customer Care personnel to alter an External Trunk configuration for troubleshooting or special circumstances. You should only enter custom property settings as directed by PureCloud Customer Care.
Setting Description
Property Name The name to assign to the custom property.
Data Type 

The data type for the custom property.

The available data types include:

  • Boolean
  • Text
  • Number
  • List
Value

The value to assign the custom property.

The data allowed in the Value field changes depending on the Data Type selected:

  • When you select Boolean, the Value box changes to a list containing True and False.
  • When you select Text, the Value box will accept any characters you enter into the box.
  • When you select Number, the Value box will only accept numeric characters.
  • When you select List, the Value box will only accept data entered as a comma-separated list. You can enter numbers and letters enclosed in quotes (“a”,”b”)