SIP phone trunk settings
When you configure a phone trunk for SIP phones, you’ll need to configure several basic settings. Depending on your requirements, you may also need to configure some of the more advanced settings. This reference describes all the settings that you’ll find on the Create/Edit Phone Trunk page for SIP phones.
Setting | Description |
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Phone Trunk Name |
Use this field to assign the phone trunk a descriptive name. |
Type |
Use this drop down to select SIP as the phone trunk type, |
Trunk State |
Use this switch to change the operational state of the phone trunk. The default setting is In-Service. |
Protocol |
Use this drop-down to choose the trunk transport protocol variant. There are three choices for the trunk transport protocol:
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Listen Port |
Use this field to specify the trunk transport listen port. Common values for this setting are 8061 for TLS and 8060 for UDP and TCP. |
Registration
If your SIP server or proxy server provides support for registration, then the status will be updated accordingly if the trunk ever becomes unavailable.
Setting | Description |
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Max Registration Rate |
Use this field to specify the average number of REGISTER requests per time period that are allowed on this trunk. You can specify this number as a decimal or a fraction. For example:
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SIP Access Control
The access control list that you construct here only apply to inbound calls.
Setting | Description |
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Use Source Address |
Use this switch to determine whether ACL matching should use the SIP messaging source address (Yes) or the VIA header originating address (No). The default setting is Yes. |
Allow the Following Addresses |
Use the controls in this section to enter and build a list of IP or CIDR addresses to which you want to allow SIP access. Notes:
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Always Deny the Following Addresses |
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. Notes:
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Allow All |
Select this check box if you want to allow access to any IP or CIDR address. Note: Allowing all addresses is a security risk.
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Connection Configuration
Setting | Description |
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Language |
Use this drop-down to choose the language that you want to use for all calls that come in on this trunk. Note: This language can be overridden by settings in Architect.
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Calls
Setting | Description |
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Max Calls |
Use this field to specify the maximum number of combined active inbound and outbound calls that are allowed on this trunk. |
Max Call Rate |
Use this field to specify the average number of calls per time period that are allowed on this trunk. This rate applies to both inbound and outbound calls. You can specify this number as a decimal or a fraction. For example:
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Max Dial Timeout |
Use this field to specify the maximum number of seconds for a delay before an outgoing call attempt is aborted. |
Setting | Description |
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Transport DSCP Value |
Use this drop-down 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). |
Retryable Reason Codes |
Use the field to enter a list of valid SIP reason codes. If an outbound call that is made on this line returns one of the SIP reason codes in this list, then that call is retried on the next line. You can specify individual reason codes or ranges of reason codes, separated by commas. By default, 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 |
Use this field to enter a list of valid Q.850 cause codes. If an outbound call that is made on this line returns one of the Q.850 cause codes in this list, then that call is retried on the next line. You can specify individual reason codes or ranges of reason codes, separated by commas. By default, 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 |
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TCP Connection Timeout |
Use this field 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 field to specify the number of seconds that a TCP connection can remain idle before being automatically closed. |
TLS Settings
Setting | Description |
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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 drop-down to choose which SSL 0r TLS method version to use. Available choices are:
The default setting is TLS v1.2. |
Far-End NAT Traversal |
Use this switch to enable or disable the Far-End NAT Traversal features for the remote user agent. For example, you would use this with SRTP Symmetric Latching of media streams. The default setting is Disabled. For more information, see FENT overview. |
Ciphers |
Use this drop-down to select and build a preferred order list of TLS ciphers. Available choices are:
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Subject Alternative Names |
Use this field 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:
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Setting | Description |
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DSCP Value |
Use this drop-down 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 drop-down 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:
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Preferred Codec List |
Use the controls in this section to choose and build a preferred list of codecs. Available choices are:
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 of North America.
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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:
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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. This setting is designed to account for intentional hold/conference scenarios. 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 |
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DTMF Payload |
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 drop-down 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:
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Header / Invite
Setting | Description |
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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. |
Inbound Digest Authentication
Setting | Description |
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Use this switch to enable or disable the ability to challenge Inbound requests. If you enable digest authentication, you’ll need to specify the associated authorization methods, realm, user name, and password. The default setting is Disabled. |
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Authorization Methods |
Use this field to select the SIP methods that should be challenged using SIP digest authentication. Available choices are:
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Realm |
Use this field to specify the domain name for the SIP realm that is used to authenticate the SIP challenge response. |
User Name |
Use this field to specify the user name that is authorized to access this trunk when receiving credentials from an external user agent client. |
Password |
Use this field to specify the password associated with the username that is authorized to access this trunk when receiving credentials from an external user agent client. By default the password is masked, but you can select the Show Password check box to see the password in plain-text. |
Setting | Description |
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Require Authentications |
Use this switch to enable or disable the requirement that an HTTP request must be authenticated by either a digest or MTLS before being processed. The default setting is Disabled. |
Prefer HTTP Redirects |
Use this switch to enable or disable a preference for HTTP redirects over a proxy when the configuration data is not found on the local Edge. The default setting is Disabled. |
Require HTTPS |
Use this switch to enable or disable the requirement that HTTPS forces an HTTP redirect if there is an available HTTPS server. The default setting is Enabled. |
External Provision Server |
Use this field to specify the URI to an external provision server to redirect requests for phone configuration when the phone’s configuration is not found. |
HTTP Port |
Use this field to specify the port on which the HTTP receiver should listen. |
HTTPS Port |
Use this field to specify the port on which the HTTPS receiver should listen. |
Setting | Description |
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Media Capture |
Use this switch to enable or disable media capture. The default setting is Disabled. The Media Capture setting is designed to be enabled while you are working with Genesys Cloud Technical Support personnel. Enabling it will generate a 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:
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Protocol Capture |
Use this switch to enable or disable protocol capture. The default setting is Disabled. The Protocol Capture setting is designed to be enabled while you are working with Genesys Cloud Technical Support personnel. Enabling it will generate a PCAP file that contains protocol specific network information that can be used for diagnostic and troubleshooting purposes. Therefore, you should only enable the Protocol Capture setting as directed by Genesys Cloud Technical Support. Warnings:
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Capture Until |
Use the calendar and clock controls to specify how long you want to collect data. |
Phone System Logging (Syslog)
Setting | Description |
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Syslog |
Use this switch to enable or disable the ability of the the Edges connected to this trunk to receive syslogs from phones The default setting is Disabled. Note: You must also enable System Logging on the Base Setting or Phone configuration settings for your phones. For more information, see Find your phone’s configuration settings.
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Syslog Port |
Use this field to specify the port on which Edges connected to this trunk will listen for information on receiving syslogs from phones. The default setting is port 514. |
Setting | Description |
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Property Name | The name to assign to the custom property. |
Data Type |
The data type for the custom property. The available data types include:
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Value |
The value to assign the custom property. The data allowed in the Value filed changes depending on the Data Type selected:
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