External trunk settings

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.

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.

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.

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 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 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 Connection Timeout

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

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

Mutual Authentication

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

The default setting is Disabled.

SIPS URI scheme

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

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

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

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 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

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. 

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.

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

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

Address Digits Length

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.

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. 

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

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

Address Digits Length

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.

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

• audio/g722
• audio/g729
• audio/PCMA (g711 A-Law)
• audio/PCMU (g711 µ-Law)
• audio/opus

SRTP Cipher Suite List

• 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 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.

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.

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. 

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.

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

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

• None
• Diversion Header

Asserted Identity Header

• P-Asserted -identity
• First Diversion Entry
• Remote-Party-ID

Max Diversion Entries

Request Target Address

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

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.

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.

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.

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.

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.

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

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.

For more information, see Enable Media Capture.

Protocol Capture

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.

For more information, see Enable Protocol Capture.

Capture Until