External trunk settings

There are three types of external trunks: BYOC Carrier and BYOC PBX (for BYOC Cloud), and Premises External SIP (for BYOC Premises). When you configure an external trunk, you configure the basic settings described in the Create a trunk under BYOC Cloud and Create a trunk under BYOC Premises articles. Depending on your needs, you may also 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 of 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 are noted in the description.

Setting	Description
External Trunk Name	Use this box to assign the external trunk a descriptive name. This name identifies this trunk when you need to select an external trunk on the various telephony configuration pages in Genesys Cloud.
Type	Use this list to select the type of external trunk that you want to create. There are three choices: BYOC Carrier BYOC PBX Premises External SIP For more information on creating trunks, see Create a trunk under BYOC Cloud and Create a trunk under BYOC Premises.
Type	BYOC Carrier and BYOC PBX only When you select one of the BYOC trunks, you see a second Type list. If you select the BYOC Carrier trunk, you need to select the type of carrier you want to use. You can select either Generic BYOC Carrier or Verizon BYOC Carrier If you select the BYOC PBX trunk, you need to select the type of PBX you want to use. you can select Generic BYOC PBX.
Managed By	Use this switch to specify who is authorized to manage the External Trunk configuration going forward. The default setting is Everyone, which includes telephony admins and the provider. However, you can select the Provider Only option to limit control to just the provider.
Trunk State	Use this switch to change the operational state of the external trunk. The default setting is In-Service.
Protocol	Use this list to choose the trunk transport protocol variant. There are three choices for the trunk transport protocol: UDP TCP TLS In most cases, you select UDP as the Protocol.
Listen Port	Premises External SIP only Use this box to specify the trunk transport listen port. Common values for this setting in Genesys Cloud are 5061 for TLS and 5060 for UDP and TCP.

Setting

Description

External Trunk Name

Use this box to assign the external trunk a descriptive name. This name identifies this trunk when you need to select an external trunk on the various telephony configuration pages in Genesys Cloud.

Type

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

BYOC Carrier
BYOC PBX
Premises External SIP

For more information on creating trunks, see Create a trunk under BYOC Cloud and Create a trunk under BYOC Premises.

Type

BYOC Carrier and BYOC PBX only

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

If you select the BYOC Carrier trunk, you need to select the type of carrier you want to use. You can select either Generic BYOC Carrier or Verizon BYOC Carrier
If you select the BYOC PBX trunk, you need to select the type of PBX you want to use. you can select Generic BYOC PBX.

Managed By

Use this switch to specify who is authorized to manage the External Trunk configuration going forward. The default setting is Everyone, which includes telephony admins and the provider. However, you can select the Provider Only option to limit control to just the provider.

Trunk State

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

The default setting is In-Service.

Protocol

Use this list to choose the trunk transport protocol variant.

There are three choices for the trunk transport protocol:

In most cases, you select UDP as the Protocol.

Listen Port

Premises External SIP only

Use this box to specify the trunk transport listen port.

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

Inbound

Setting	Description
Number Plan Site	Use this setting to identify the site with the number plan that you want to use. To improve inbound call handling, Genesys Cloud allows you to identify the site with the number plan that you want to use. The site that you select controls which number plan is used, both for transforms on inbound calls and for subsequent outbound transfers for calls that never flow through another site.
Inbound SIP Termination Identifier	BYOC Carrier and BYOC PBX Use this box to specify your termination identifier. Genesys Cloud requires that a unique identifier exists in the INVITE to associate inbound calls with the appropriate Genesys Cloud 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. When 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 Genesys Cloud organization’s region. If you select Generic BYOC Carrier, the Inbound SIP Termination Identifier is a static configuration that should be populated with a regionally unique identifier. This identifier will be used to direct traffic from external Carriers to the organization. If you select Verizon BYOC Carrier, the Inbound SIP Termination Identifier is a static configuration that should be populated with a regionally unique identifier. This identifier will be used to direct traffic from Verizon to the organization. If you select Generic BYOC PBX, the Inbound SIP Termination Identifier is a static configuration that should be populated with a regionally unique identifier. This identifier will be used to direct traffic from third-party PBXs to the organization.
Inbound SIP Termination Header	BYOC Carrier and BYOC PBX Use this box to specify the name of the SIP header the carrier will populate with the inbound SIP termination identifier value on inbound calls. You only use this setting when none of the standard identification methods are supported by your carrier. If you select Generic BYOC Carrier, the Inbound SIP Termination Header is a custom field that contains the Termination Identifier value for inbound calls to Genesys Cloud. If you select Verizon BYOC Carrier, the Inbound SIP Termination Header is by default set to X-VZ-CSP-Customer-Identifier. If you select Generic BYOC PBX, the Inbound SIP Termination Header is a custom field that contains the Termination Identifier value for inbound calls to Genesys Cloud.
DNIS Replacement Routing	BYOC Carrier and BYOC PBX 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 Replacement 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. When you begin typing an identifier in one of the outbound SIP boxes, 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 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 is used by default. Note: IP addresses added to the Outbound SIP Servers or Proxies section are placed automatically on the SIP Access Control Allow list. BYOC Carrier and BYOC PBX When you add multiple SIP servers or intermediate proxies, Genesys Cloud randomly selects a server from the list for each outbound call. Premises External SIP When adding multiple SIP servers or intermediate proxies, use the arrows to put the highest priority server at the top of the list. For outbound calls, Genesys Cloud attempts to use the server at the top of the list first but will cycle thru the list until it finds an available server.
Digest Authentication	Use this switch to enable or disable Digest Authentication. When enabled, digest authentication provides extra layer of security to the process of granting outbound requests. If you enable digest authentication, you need to specify the associated realm, the user name, 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 authenticates outbound requests.
User Name	Use this box to specify the user name 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.

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. When you begin typing an identifier in one of the outbound SIP boxes, 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 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 is used by default.

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

BYOC Carrier and BYOC PBX

When you add multiple SIP servers or intermediate proxies, Genesys Cloud randomly selects a server from the list for each outbound call.

Premises External SIP

When adding multiple SIP servers or intermediate proxies, use the arrows to put the highest priority server at the top of the list. For outbound calls, Genesys Cloud attempts to use the server at the top of the list first but will cycle thru the list until it finds an available server.

Digest Authentication

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

If you enable digest authentication, you need to specify the associated realm, the user name, 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 authenticates outbound requests.

User Name

Use this box to specify the user name 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
Caller Address	Use this box to specify the telephone number that you want to appear in the caller ID information for outbound calls. Enter this number in E.164 format, which includes the plus sign (+) and the country code.
Caller Name	Use this box to specify the name that you want to appear in the caller ID information for outbound calls.
Prioritized Caller Selection	Use this list to add and prioritize the locations from which Genesys Cloud can pull caller ID information For more information, see Use the Prioritized Caller Selection feature to configure caller ID information.
Suppress User Name	Use this list specify how you want to handle Caller ID and Caller name when the call source is User DID. For more information, see Use the Prioritized Caller Selection feature to configure caller ID information.

Availability

Setting	Description
Availability	Premises External SIP only Use this switch to enable or disable the ability to send availability requests. The default setting is Disabled. Warning: Enabling the Availability setting may disrupt the priority of the SIP Servers or Proxies list you configured for your trunk.
Interval	Premises External SIP only Use this box to enter the time interval in seconds before the availability request should be sent.

Setting

Description

Availability

Premises External SIP only

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

The default setting is Disabled.

Warning: Enabling the Availability setting may disrupt the priority of the SIP Servers or Proxies list you configured for your trunk.

Interval

Premises External SIP only

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

Registration

Setting	Description
Registration	Premises External SIP only Use this switch to enable or disable registration. The default setting is Disabled.
Expiration Interval	Premises External SIP only Use this box to specify the number of seconds to wait before the registration expires.
Address of Record	Premises External SIP only Use one of these options to specify the URI or the user portion of the registration. (The user portion is combined with the Edge Interface IP.) By default Genesys Cloud automatically generates the address of record from the Edge network interface. However, you can enter a custom address.

Setting

Description

Registration

Premises External SIP only

Use this switch to enable or disable registration.

The default setting is Disabled.

Expiration Interval

Premises External SIP only

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

Address of Record

Premises External SIP only

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

By default Genesys Cloud automatically generates the address of record from the Edge network interface. However, you can enter a custom address.

SIP Access Control

Setting	Description
Use Source Address	Premises External SIP only Use this switch to determine whether ACL matching uses 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	Premises 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	Premises 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.
PBX Passthrough	Use this switch to enable or disable PBX Passthrough. Enabling the PBX passthrough feature allows your PBX to pass calls intended for the PSTN directly through Genesys Cloud. Warning: Do not enable for trunks with PSTN access or you are at a high risk for toll fraud. For more information, see Configure PBX passthrough for a BYOC trunk.

Setting

Description

Use Source Address

Premises External SIP only

Use this switch to determine whether ACL matching uses 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

Premises 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

Premises 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.

PBX Passthrough

Use this switch to enable or disable PBX Passthrough. Enabling the PBX passthrough feature allows your PBX to pass calls intended for the PSTN directly through Genesys Cloud.

Warning: Do not enable for trunks with PSTN access or you are at a high risk for toll fraud.

For more information, see Configure PBX passthrough for a BYOC trunk.

General

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: Settings in Architect can override this language.

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: Settings in Architect can override this language.

Calls

Setting	Description
Max Calls	Premises External SIP only Enter a value in this box to either limit or expand the maximum number of combined active inbound and outbound calls that are allowed on this trunk. Notes: On an external trunk, the Max Calls setting is a per trunk setting. Max Calls is the amount of calls the trunk on a particular Edge can handle. Setting Max Calls to 0 (zero) will configure the trunk to accept an unlimited number of calls. If you set Max Calls to 0 (zero), be sure to take into account the concurrent call capacity for your Edges. Setting the Max Calls to 0 (zero) does not increase the Edge’s concurrent call capacity. For more information, see Concurrent call capacity for Edge models. For more information, see Configure maximum call settings.
Max Concurrent Calls	BYOC Carrier and BYOC PBX Select an option to either limit or expand the maximum number of combined active inbound and outbound calls that are allowed on this trunk. The default setting is Unlimited. Genesys has optimized the Max Concurrent Calls setting to be set to Unlimited. As such, Genesys recommends leaving the Max Concurrent Calls set to Unlimited. However, if you or your carrier determine that setting a limit on the number of concurrent calls is required, then you should select Limited to and enter a value in the box. When selecting the Limited option, Genesys recommends that you carefully evaluate the number of calls that you want to be able to handle concurrently. For more information, see Configure maximum call settings.
Max Call Rate	Use this box to specify the average number of inbound and outbound calls (call arrival/initiation) that can be handled by this trunk per time period. You can specify this number as a decimal or a fraction. For example: 40/5s: This would specify that a maximum of 40 calls can be handled by this trunk every 5 seconds.
Max Dial Timeout	Use this box to specify the maximum number of seconds for a delay before an outgoing call attempt is aborted. Note: The Max Dial Timeout setting has a maximum of 2 minutes on BYOC Cloud trunks.
Max Calls Reason Code	Use this box to specify a custom SIP response code that you want the trunk to return when Max Call threshold is exceeded. By default, a trunk returns a 503 Service Unavailable reason code if the Max Call threshold is exceeded. Using the Max Calls Reason Code setting. you can set the reason code to any SIP response code in the 4xx, 5xx, or 6xx range. This allows you to configure a response code, such as 486 Busy Here, that more accurately indicates that the Max Call threshold has been exceeded.

Setting

Description

Max Calls

Premises External SIP only

Enter a value in this box to either limit or expand the maximum number of combined active inbound and outbound calls that are allowed on this trunk.

Notes:

On an external trunk, the Max Calls setting is a per trunk setting.
Max Calls is the amount of calls the trunk on a particular Edge can handle.
Setting Max Calls to 0 (zero) will configure the trunk to accept an unlimited number of calls.
- If you set Max Calls to 0 (zero), be sure to take into account the concurrent call capacity for your Edges. Setting the Max Calls to 0 (zero) does not increase the Edge’s concurrent call capacity. For more information, see Concurrent call capacity for Edge models.

For more information, see Configure maximum call settings.

Max Concurrent Calls

BYOC Carrier and BYOC PBX

Select an option to either limit or expand the maximum number of combined active inbound and outbound calls that are allowed on this trunk.

The default setting is Unlimited. Genesys has optimized the Max Concurrent Calls setting to be set to Unlimited. As such, Genesys recommends leaving the Max Concurrent Calls set to Unlimited.

However, if you or your carrier determine that setting a limit on the number of concurrent calls is required, then you should select Limited to and enter a value in the box. When selecting the Limited option, Genesys recommends that you carefully evaluate the number of calls that you want to be able to handle concurrently.

For more information, see Configure maximum call settings.

Max Call Rate

Use this box to specify the average number of inbound and outbound calls (call arrival/initiation) that can be handled by this trunk per time period.

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

40/5s: This would specify that a maximum of 40 calls can be handled by this trunk every 5 seconds.

Max Dial Timeout

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

Note: The Max Dial Timeout setting has a maximum of 2 minutes on BYOC Cloud trunks.

Max Calls Reason Code

Use this box to specify a custom SIP response code that you want the trunk to return when Max Call threshold is exceeded.

By default, a trunk returns a 503 Service Unavailable reason code if the Max Call threshold is exceeded. Using the Max Calls Reason Code setting. you can set the reason code to any SIP response code in the 4xx, 5xx, or 6xx range. This allows you to configure a response code, such as 486 Busy Here, that more accurately indicates that the Max Call threshold has been exceeded.

Transport

Setting	Description
Transport DSCP Value	Premises 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.
	When an outbound call made on this line fails, the trunk receives a Failure SIP Response, which contains either a SIP reason code, a Q.850 cause code, or both. Genesys Cloud compares the codes found in the Failure SIP Response against the codes you configure and determines how to proceed. If the Failure SIP Response has only a reason code and that code matches an item in the Retryable Reason Code list on your trunk, the call will be retried on the next trunk, if there is one. If the Failure SIP Response has a reason code and a cause code and those codes match items in both the Retryable Reason Code and Retryable Cause Code list on your Trunk, the call will be retried on the next trunk, if there is one. If the Failure SIP Response has a reason code and a cause code and neither of them match the Retryable Reason Code or Retryable Cause Code list on your trunk, the call is terminated.
Retryable Reason Codes	Premises External SIP only Use the box to enter a list of valid SIP reason codes. You can specify individual reason codes or ranges of reason codes, separated by commas. By default, Genesys Cloud 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	Premises External SIP only Use this box to enter a list of valid Q.850 cause codes. You can specify individual reason codes or ranges of reason codes, separated by commas. By default, Genesys Cloud 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

Setting

Description

Transport DSCP Value

Premises 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.

When an outbound call made on this line fails, the trunk receives a Failure SIP Response, which contains either a SIP reason code, a Q.850 cause code, or both. Genesys Cloud compares the codes found in the Failure SIP Response against the codes you configure and determines how to proceed.

If the Failure SIP Response has only a reason code and that code matches an item in the Retryable Reason Code list on your trunk, the call will be retried on the next trunk, if there is one.
If the Failure SIP Response has a reason code and a cause code and those codes match items in both the Retryable Reason Code and Retryable Cause Code list on your Trunk, the call will be retried on the next trunk, if there is one.
If the Failure SIP Response has a reason code and a cause code and neither of them match the Retryable Reason Code or Retryable Cause Code list on your trunk, the call is terminated.

Retryable Reason Codes

Premises External SIP only

Use the box to enter a list of valid SIP reason codes. You can specify individual reason codes or ranges of reason codes, separated by commas.

By default, Genesys Cloud 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

Premises External SIP only

Use this box to enter a list of valid Q.850 cause codes. You can specify individual reason codes or ranges of reason codes, separated by commas.

By default, Genesys Cloud 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	Premises 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. Note: You cannot use the TCP Connection Timeout setting with the Release Link Transfer setting.
TCP Connection Idle Timeout	Premises External SIP only Use this box to specify the number of seconds that a TCP connection can remain idle before being automatically closed.

Setting

Description

TCP Connection Timeout

Premises 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.

Note: You cannot use the TCP Connection Timeout setting with the Release Link Transfer setting.

TCP Connection Idle Timeout

Premises 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	Premises 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	Premises 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	Premises External SIP only Use this list to choose which SSL 0r TLS method version to use. Available choices are: SSL v2.3 (This version is no longer considered secure.) SSL v3 TLS v1 TLS v1.1 TLS v1.2 The default is TLS v1.2
Ciphers	Premises External SIP only Use this list to select and build a preferred order list of TLS ciphers. Available choices are: TLS_RSA_WITH_AES_256_GCM_SHA384 TLS_RSA_WITH_AES_256_CBC_SHA256 TLS_RSA_WITH_AES_256_CBC_SHA 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	Premises 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:

Setting

Description

Mutual Authentication

Premises 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

Premises 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

Premises External SIP only

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

SSL v2.3 (This version is no longer considered secure.)
SSL v3
TLS v1
TLS v1.1
TLS v1.2

The default is TLS v1.2

Ciphers

Premises External SIP only

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

Available choices are:

TLS_RSA_WITH_AES_256_GCM_SHA384
TLS_RSA_WITH_AES_256_CBC_SHA256
TLS_RSA_WITH_AES_256_CBC_SHA
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

Premises 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:

Identity

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

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 Genesys Cloud’s capability to apply header privacy information. When enabled, an agent can use 67 to request privacy. More specifically, this prevents Genesys Cloud from sending the actual header information (typically the contact address) along with the call. Instead, Genesys Cloud 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 Genesys Cloud’s capability to apply user privacy information. When enabled, an agent can use 67 to request privacy. More specifically, this prevents Genesys Cloud from sending ANI information with the call. Instead, Genesys Cloud 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. Notes: When using *67 to apply user privacy information, do not preface the country code with a plus (+) sign. Check with your local regulators and your carrier for specific information about privacy settings.

Setting

Description

Apply Header Privacy

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

When enabled, an agent can use *67 to request privacy. More specifically, this prevents Genesys Cloud from sending the actual header information (typically the contact address) along with the call. Instead, Genesys Cloud 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 Genesys Cloud’s capability to apply user privacy information.

When enabled, an agent can use *67 to request privacy. More specifically, this prevents Genesys Cloud from sending ANI information with the call. Instead, Genesys Cloud 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.

Notes:

When using *67 to apply user privacy information, do not preface the country code with a plus (+) sign.
Check with your local regulators and your carrier for specific information about privacy settings.

Calling

Setting	Description
Address Transformation	Use the controls in this section to enter regular expressions and build an ordered list of regular expressions to reformat addresses. The expressions is 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. Note: As you fill in the fields in the Calling section, keep in mind that the Calling address is the number that is calling and that it only applies to outbound calls. For more information, see Transform outbound addresses with regular expressions.
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 to send.
Address Omit + Prefix	Use this switch to enable or disable the ability to exclude the plus (+) prefix in the outgoing origination address to send. 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. Note: As you fill in the fields in the Called section, keep in mind that the Called address is the number that was called and that it only applies to outbound calls. For more information, see Transform outbound addresses with regular expressions.
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.
Asserted Identity	Select the Send asserted identity header check box to enable sending of the asserted identity header on outbound calls. Enabling this will allow you to choose what information will be placed in the asserted identity header of every call made from this trunk. Under Get asserted identity header from, you can choose to set the asserted identity header or dynamically. Selecting Caller address sets the asserted identity header dynamically. This setting uses the name and address of the originating calling party in the asserted identity header. Selecting Custom data sets the asserted identity header statically. This setting will pull the data from the Caller Address and Caller Name in the Outbound – Calling section, insert it into and Name and URI boxes, and then use it in the asserted identity header.

Media

Setting	Description
DSCP Value	Premises 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 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. Note: An extended period of time is defined as 5 minutes for normal calls or 12 hours for media that is sent in one direction (not send and receive). The default setting is Enabled.

Setting

Description

DSCP Value

Premises 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.

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.

Note: An extended period of time is defined as 5 minutes for normal calls or 12 hours for media that is sent in one direction (not send and receive).

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.

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
Record calls on this trunk	Use this check box to enable or disable recording. The default setting is disabled.
Require user consent before recording	Use this check box to enable or disable the user consent requirement. The default setting is disabled.
Consults	Use this check box to enable or disable the recording of the private conversation between an agent and a supervisor in a call consult scenario. The default setting is disabled, which means the consult recording is not captured.
Holds	Use this check box to continue or to suppress the trunk recording when the call is put on hold by the agent or the customer. The default setting is disabled, which means the recording is suppressed when the call is on hold.
External bridged transfers	Use this check box to continue or to terminate the trunk recording on an external transfer that result in external to external connected calls. The default setting is disabled, which means the recording is terminated upon external bridged transfer.
Level the volume on both sides of the conversation	Use this check box to enable or disable automatic level control for the recordings. The default setting is disabled.
Audio Format/Codec	Use this list to select the audio codec to use for recording. The available choices are: G726-32 GSM L16 (Uncompressed Linear 16-bit PCM) Opus PCMA (G.711 A-law) PCMU (G.711 u-law) Truespeech Note: If you want to have your recordings transcribed, you must select one of the following audio codecs: PCMU (G.711 u-law) PCMA (G.711 A-law) L16 (Uncompressed Linear 16-bit PCM) Opus
Dual channel	Use this check box to enable or disable dual channel recording. The Dual channel setting is only available if you select one of the following audio formats: PCMU (G.711 u-law) PCMA (G.711 A-law) L16 (Uncompressed Linear 16-bit PCM) Opus When you enable the Dual channel setting, the system saves each channel of the recording in a separate stream. The system uses audio channel 0 to save the external participant recording and audio channel 1 to save the internal participant recording. The default setting is disabled. Note: If you want to have your recordings transcribed, enable the Dual channel setting.
Play periodically while recording	Use this check box to enable or disable the beep tone feature. The default setting is disabled.
Number of Tones	The Recording Beep Tone has two modes of operation: Single and Dual. Single allows for a single tone to be configured and injected into voice audio. Dual allows for two tones to be configured and injected back to back into voice audio. The default setting is Single.
Customize Tones	Click Customize Tones if you want to configure the way that the tones play. When you do, you’ll see the Customize Tones dialog. To specify how long in milliseconds the beep tone plays, use the Duration slider. To specify the rate in hertz at which the beep tone occurs, use the Frequency slider. To specify the beep tone volume in decibels, use the Volume slider.
Play Beep Tone Every	Use the slider to specify how often in seconds that the beep tone plays.

Setting

Description

Record calls on this trunk

Use this check box to enable or disable recording.

The default setting is disabled.

Require user consent before recording

Use this check box to enable or disable the user consent requirement.

The default setting is disabled.

Consults

Use this check box to enable or disable the recording of the private conversation between an agent and a supervisor in a call consult scenario.

The default setting is disabled, which means the consult recording is not captured.

Holds

Use this check box to continue or to suppress the trunk recording when the call is put on hold by the agent or the customer.

The default setting is disabled, which means the recording is suppressed when the call is on hold.

External bridged transfers

Use this check box to continue or to terminate the trunk recording on an external transfer that result in external to external connected calls.

The default setting is disabled, which means the recording is terminated upon external bridged transfer.

Level the volume on both sides of the conversation

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

The default setting is disabled.

Audio Format/Codec

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

G726-32
GSM
L16 (Uncompressed Linear 16-bit PCM)
Opus
PCMA (G.711 A-law)
PCMU (G.711 u-law)
Truespeech

Note: If you want to have your recordings transcribed, you must select one of the following audio codecs:

PCMU (G.711 u-law)
PCMA (G.711 A-law)
L16 (Uncompressed Linear 16-bit PCM)
Opus

Dual channel

Use this check box to enable or disable dual channel recording. The Dual channel setting is only available if you select one of the following audio formats:

PCMU (G.711 u-law)
PCMA (G.711 A-law)
L16 (Uncompressed Linear 16-bit PCM)
Opus

When you enable the Dual channel setting, the system saves each channel of the recording in a separate stream. The system uses audio channel 0 to save the external participant recording and audio channel 1 to save the internal participant recording.

The default setting is disabled.

Note: If you want to have your recordings transcribed, enable the Dual channel setting.

Play periodically while recording

Use this check box to enable or disable the beep tone feature.

The default setting is disabled.

Number of Tones

The Recording Beep Tone has two modes of operation: Single and Dual. Single allows for a single tone to be configured and injected into voice audio. Dual allows for two tones to be configured and injected back to back into voice audio.

The default setting is Single.

Customize Tones

Click Customize Tones if you want to configure the way that the tones play. When you do, you’ll see the Customize Tones dialog.

To specify how long in milliseconds the beep tone plays, use the Duration slider.
To specify the rate in hertz at which the beep tone occurs, use the Frequency slider.
To specify the beep tone volume in decibels, use the Volume slider.

Play Beep Tone Every

Use the slider to specify how often in seconds that the beep tone plays.

Protocol

Header / Invite

Setting	Description
Conversation Headers	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 Disabled.
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 include on an outbound call.
Request Target Address	Premises 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 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.

Setting

Description

Conversation Headers

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 Disabled.

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 include on an outbound call.

Request Target Address

Premises 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

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

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 Note: If you do not specify an encoding format, Genesys Cloud assumes the Hex encoding format.
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.

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

Note: If you do not specify an encoding format, Genesys Cloud assumes the Hex encoding format.

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

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, Genesys Cloud 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 applies.
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 has the same header name, Genesys Cloud determines which UUI data value to send based on the Priority setting. If you select Low Priority, Genesys Cloud always chooses the dynamic UUI data from Architect or Scripter. If you select High Priority, Genesys Cloud always chooses 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 is sent.

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, Genesys Cloud 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 applies.

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 has the same header name, Genesys Cloud determines which UUI data value to send based on the Priority setting.

If you select Low Priority, Genesys Cloud always chooses the dynamic UUI data from Architect or Scripter.
If you select High Priority, Genesys Cloud always chooses 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 is 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 Genesys Cloud 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.

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 Genesys Cloud 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 only affects an Architect call flow using the Transfer to Number action when an external party (Transferee) is transferred to another external party (Target). After the transfer completes, the Transferee and Target connect and Genesys Cloud releases all telephony resources. For more information, see Set up a Transfer to Number action and Transfer to Number action. Notes: You cannot use the Release Link Transfer setting with the TCP Connection Timeout setting. For Release Link Transfer to work, the inbound call must arrive on the same trunk on which the outbound transfer is being requested.

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 only affects an Architect call flow using the Transfer to Number action when an external party (Transferee) is transferred to another external party (Target). After the transfer completes, the Transferee and Target connect and Genesys Cloud releases all telephony resources.

For more information, see Set up a Transfer to Number action and Transfer to Number action.

Notes:

You cannot use the Release Link Transfer setting with the TCP Connection Timeout setting.
For Release Link Transfer to work, the inbound call must arrive on the same trunk on which the outbound transfer is being requested.

Outbound

Setting	Description
Custom SIP headers	BYOC Carrier/Generic BYOC Carrier and BYOC PBX/Generic BYOC PBX If your SIP device requires additional information to correctly process calls, you can use the Custom SIP headers panel to add custom SIP headers and their values to the SIP INVITE. These headers are attached to every call sent out on the trunk. You can add multiple headers. BYOC Carrier/Genesys Cloud BYOC Verizon If you select BYOC Carrier/Genesys Cloud BYOC Verizon, the Custom SIP headers section is preconfigured with three custom headers for Verizon: X-VZ-CSP-Domain X-VZ-CSP-Customer-Identifier X-VZ-CSP-Leg-Type You need to work with your Verizon representative to find the correct values to enter into theValue boxes for X-VZ-CSP-Domain and X-VZ-CSP-Customer-Identifier headers. The Value box for the X-VZ-CSP-Leg-Type header is automatically filled in with the value of “agent” so leave this box as is. For more information, see Configure Custom SIP headers.
Header	Use this box to specify the SIP header name.
Value	Use this box to specify the static SIP value.
Location conveyance	Use this check box to convey a geolocation when using Enhanced 911 HTTP-Enabled Location Delivery (HELD). For more information, see Configure HTTP Enabled Location Delivery (HELD) for E911

Diagnostic

Setting	Description
Media Capture	Use this switch to enable or disable media capture. The default setting is Disabled. The Media Capture setting is enabled while you are working with Genesys Cloud Technical Support personnel. Enabling it generates 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 Genesys Cloud 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 Genesys Cloud organization and have the PCI DSS setting enabled, then you cannot enable media capture – the Media Capture setting is not available. For more information, see Enable Media Capture.
Protocol Capture	Premises External SIP only Use this switch to enable or disable protocol capture. The default setting is Disabled. The Protocol Capture setting is enabled while you are working with Genesys Cloud Technical Support personnel. Enabling it generates 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 Genesys Cloud Technical Support. Warnings: Protocol diagnostics do not capture the following SIP method messages: OPTIONS, REGISTER, SUBSCRIBE, and NOTIFY. Enabling protocol capture can degrade performance and affect QoS. Protocol capture logs all data entered into the system, including data entered via Secure IVR flows. This data 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 Genesys Cloud organization and have the PCI DSS setting enabled, then you cannot enable protocol capture – the Protocol Capture setting is not 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.

Setting

Description

Media Capture

Use this switch to enable or disable media capture.

The default setting is Disabled.

The Media Capture setting is enabled while you are working with Genesys Cloud Technical Support personnel. Enabling it generates 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 Genesys Cloud 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 Genesys Cloud organization and have the PCI DSS setting enabled, then you cannot enable media capture – the Media Capture setting is not available.

For more information, see Enable Media Capture.

Protocol Capture

Premises External SIP only

Use this switch to enable or disable protocol capture.

The default setting is Disabled.

The Protocol Capture setting is enabled while you are working with Genesys Cloud Technical Support personnel. Enabling it generates 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 Genesys Cloud Technical Support.

Warnings:

Protocol diagnostics do not capture the following SIP method messages: OPTIONS, REGISTER, SUBSCRIBE, and NOTIFY.
Enabling protocol capture can degrade performance and affect QoS.
Protocol capture logs all data entered into the system, including data entered via Secure IVR flows. This data 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 Genesys Cloud organization and have the PCI DSS setting enabled, then you cannot enable protocol capture – the Protocol Capture setting is not 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.

Custom

The Custom option is designed to allow Genesys Cloud Customer Care personnel to alter an External Trunk configuration for troubleshooting or special circumstances. Only enter custom property settings as directed by Genesys Cloud 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 accepts any characters you enter into the box. When you select Number, the Value box only accepts numeric characters. When you select List, the Value box only accepts data entered as a comma-separated list. You can enter numbers and letters enclosed in quotes (“a”,”b”)

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 accepts any characters you enter into the box.
When you select Number, the Value box only accepts numeric characters.
When you select List, the Value box only accepts data entered as a comma-separated list. You can enter numbers and letters enclosed in quotes (“a”,”b”)