Zone

The name of the zone where the SIP connection is configured. Select the name of the zone from the drop-down box.

Name

The unique identifier for this SIP connection on the Experience Portal system.

The name can be up to 32 alphanumeric characters. Do not use any special characters.

Enable

Whether this SIP connection is available for use by the Experience Portal system.

The default is Yes, which means the connection is available.

Proxy Transport

The IP Protocol used by the SIP connection.

The options are:

• TCP

• TLS

Address

The address of the proxy server.

This must be a valid network address in the form of a fully qualified hostname or an IP address.

This field is enabled only when you select the Proxy Server option.

Port

The port used by the proxy server.

The default for TCP is 5060, and the default for TLS is 5061.

This field is enabled only when you select the Proxy Server option.

Priority

When you configure more than one proxy server, this field determines the order in which outbound calls are sent to the list of proxy servers.

Calls are sent to the proxy server with the lowest priority value first. If this proxy server fails, calls are sent to the proxy server with the second lowest priority value. This continues up the proxy server list in priority order until either the call succeeds or the list is exhausted.

Enter a number in the range 0 to 65535. The default is 0.

This field is enabled only when you select the Proxy Server option.

Weight

When you add more than one proxy server with the same priority value, this field determines the relative chances of which proxy server is used for an outbound call. The proxy server with the highest weight has the greatest odds of receiving a call.

For example, if proxy servers 1 and 2 are assigned a priority of 1, and weight of 4 and 6 respectively, then proxy server 1 has a 40% (4/(4+6)) chance of receiving a call while proxy server 2 has a 60% (6/(4+6)) chance.

Enter a number in the range 0 to 65535. The default is 0.

This field is enabled only when you select the Proxy Server option.

Remove

Removes the proxy server.

This field is enabled only when you select the Proxy Server option.

Additional Proxy Server

Adds additional proxy addresses and ports.

You cannot use the same proxy server address and port for adding another proxy server.

This field is enabled only when you select the Proxy Server option.

DNS Server Domain

This is the domain name under which the SIP proxy list is configured in the DNS server.

The entry must be a valid hostname.

Ensure that the DNS server domain is configured to retrieve the ordered list of available server records which can be used to handle calls.

This field is enabled only when you select the DNS Server Domain option.

Listener Port

The port used by the Listener.

The default for TCP is 5060, and the default for TLS is 5061.

SIP Domain

The domain in which the SIP connection is configured. The SIP domain must match the domain name of the connected proxy (that is the domain name in SIP URIs for incoming calls).

* (asterisk) means that all calls are routed to this trunk.

P-Asserted-Identity

The assumed identity used to determine the service class and the restriction permissions class for the SIP connection.

For Communications Manager, this should map to an extension configured on the switch.

Maximum Redirection Attempts

The number of redirection attempts allowed before the call is considered to have failed. The MPP redirects a call when it receives a 302 response code from an INVITE request. This response code indicates that the endpoint which received the call has moved to another location, and the call should be redirected to the new location. The call continues to be redirected until either no further 302 response is received or the retry count is exhausted.

Enter a number in the range 0 to 100. The default is 0.

Redirect attempt is disabled when the number in this field is set to 0.

Consultative Transfer

If a connection cannot be established, Consultative Transfer allows Experience Portal to regain control of the call.

The following options determine the SIP messages used for a VXML Consultative Transfer:

• INVITE with REPLACES: When Experience Portal receives INVITE with REPLACES in the SIP message, it establishes a secondary call to the transfer destination to:

  • Determine availability

  • Ensure that the destination answers within the established timeout

  The secondary call is then merged with the primary call that is being transferred. Experience Portal sends a request to the transferee for an INVITE message with a Replaces header that contains the information necessary to take control over the primary call. Experience Portal controls the progress of the call in case the response of the second call is not positive.

  Note:

  This option requires the transfer destination to support the INVITE with Replaces SIP message.

• REFER: With this option, the transferee determines the entire process of establishing the new call to the transfer destination. If the transfer destination is unavailable or does not respond to the call, the transferee sends the call to Experience Portal.

The default option is INVITE with REPLACES.

SIP Reject Response Code

The response code that is sent to a SIP proxy when all SIP resources for an MPP are in use.

The options are:

• ASM (503): Sends a request to ASM to call another MPP that might be available.

• SES (480): Sends a request to SES to call another MPP that might be available.

• Custom: Allows custom response codes to be set for interoperability with other proxies and media gateways that might require a different response code to initiate a similar operation as stated above.

T1

Timer T1 is a general estimate of the maximum round trip time for SIP packets between the MPP and the proxy. It is used to determine the minimum retransmit interval for SIP messages.

The default value is 250 millisecond(s).

Enter a number in the range of 10 to 8000 millisecond(s).

T2

Timer T2 is the maximum retransmit interval for SIP messages. The T1 and T2 values are used together in an algorithm that backs off message retransmits in case of congestion.

The default value is 2000 millisecond(s).

Enter a number in the range of 10 to 8000 millisecond(s).

B and F

Timers B and F are the transaction timeouts for INVITE and non-INVITE requests, respectively. They determine the amount of wait time before a SIP request is aborted, when no response is received.

The default value is 4000 millisecond(s).

Enter a number in the range of 500 to 180000 millisecond(s).