Telephony Integration field descriptions

BASIC CONFIGURATION

Name	Description
Basic Configuration
Switch Integration Type	Messaging uses SIP integration. The SIP SPECIFIC CONFIGURATION section is available only for SIP integration.
IP Address Version	The version of the IP address.

SIP SPECIFIC CONFIGURATION

Name	Description
Transport Method	The transport method the telephony server uses for SIP signaling. The transport method of the application server and the telephony server must match. Choices are: TCP (not encrypted). Use port 5060. TLS (encrypted). Use port 5061.
Far-end Connections	The total number of SIP proxy servers. Valid far-end connections are: Direct (no proxy). Select 1. Session Manager. Select a maximum of 12. SIP Enablement Services. Select 1. AudioCodes gateway. Select a maximum of 4. Other application servers for the same site. The number of Connection fields expand to match the number you select.
Connection x	The IP address and the port number for each far-end connection. If you select more than one far-end connection, the system accepts incoming calls from any of the servers on the connection list. The system attempts to place outgoing calls to the first server that accepts a connection in the order listed. MWI only uses one connection regardless of the number of sites or servers in your deployment.
Messaging Address	The IP address for this near-end application server is always a read-only field. The port number is typically one of the following: TCP = 5060 TLS = 5061
SIP Domain	The domain names for the application server and the far-end connection must match. Example: sip.example.com
Messaging Ports	The number of network ports reserved for Messaging: The range of the Call Answer Ports is 2 to 100. The read-only Transport field displays the number of ports available for transfer operations. This number is the difference between the number of trunks and the number of call answer ports. A typical ratio between call answer and transfer ports is 80:20.
Switch Trunks	The number of trunk members for messaging on the telephony server. If the telephony server is configured for multiple signal groups, enter the sum of all trunk members in all groups. The minimum is the number of call answer ports plus one. The maximum is the sum of the number of call answer and transfer ports. Ensure that you do not exceed 120 telephony server trunk members. The Transfer Ports field is read-only. The system uses the following formula to calculate the number of transfer ports: Switch Trunks Total – Call Answer Ports = Transfer Ports. The number in the Switch Trunks Total field must match the number of trunk members on the telephony server.
Advanced Options: When the Advanced Options are hidden, the system displays the Show Advanced Options button. After clicking Show Advanced Options, the system displays the following table, and the button changes to Hide Advanced Options.
Quality Of Service	This information is only relevant if your IP network infrastructure supports the Quality of Service feature. You can accept the default values or enter new ones. However, the values must match the number in the network region of the telephony server that is used by the Messaging signaling group. The range for both fields is 0 to 63. Call Control PHB sets the quality of service level for call control messages. Audio PHB sets the quality for audio streams.
UDP Port Range	The range of port numbers used by the User Datagram Protocol (UDP) for the Real Time Protocol (RTP). The default range is 8000 through 10000. You can change the Start value. The system uses the number of available trunks to calculate the End value. Ensure that the range of ports that you allocate to UDP does not conflict with ports used for other purposes.
Media Encryption	The type of Secure Real-time Transport Protocol (SRTP) media encryptions that the telephony server uses. This field is optional.
SIP INFO for DTMF	Support SIP INFO messages for DTMF out-of-band. Options are: Ignore: Ignores all SIP INFO DTMF digits in the signaling stream. Accept: Accepts all incoming SIP INFO messages for the two formats and interprets the messages as if the messages had been received in the RTP stream as RFC 2833 compliant digits. Outbound DTMF are sent as SIP INFO messages with application type dtmf-relay with a specified duration of 250ms. The default value is Ignore. For more information on SIP INFO message formats, see Support for SIP INFO messages on SIP connections.
Media Encryption During CapNeg	Select use of media encryption, if capability negotiation (CapNeg) is present in the Session Description Protocol (SDP). Options are: Enabled: This is the default value. Disabled: In this option, Messaging automatically changes the value in the Media Encryption field to None. Configure the value in this field as Disabled only if required for a specific telephony integration. For more information, see the configuration notes.
Supported Header includes “replaces”	For endpoints to properly reflect their capabilities in the SIP headers, the supported header must include the replaces value. This allows Messaging to communicate effectively with a specific telephony integration. Options are: no: This is the default value. yes Configure the value in this field as yes, only in case this is required for a specific telephony integration. For more information, refer to the configuration notes.
Monitor Far-end OPTIONS messages	Messaging proactively monitors the SIP OPTIONS messages that the far-end sends. If Messaging does not receive a SIP OPTIONS message from the far-end within the time specified in the Proactive Interval field, then Messaging considers the far-end as down or unreachable. The options are: yes no The default value is no. If you want Messaging to proactively monitor the SIP OPTIONS messages from the far-end, click yes. If set to yes, you must set the value in the Proactive Interval field to the interval at which the far-end is configured for sending the OPTIONS message. The value you enter in the Proactive Interval field must be in seconds.
Inactive Link Actions	This field is available only if the value in the Monitor Far-end OPTIONS messages field is set to yes. The options are: Alarm Only Close Connections If set to Alarm Only, Messaging generates an alarm whenever an expected OPTIONS message does not arrive within the Proactive Interval period plus 30% of that time. On the next receipt of SIP OPTIONS or the next incoming call, the system clears the alarm. If set to Close Connections, in addition to generating an alarm, Messaging closes all incoming connections, dropping any calls that might be active at that point.