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Abacus: How do I configure SIP Supplementary Services to send traffic out different physical interfaces even though channels are sequential?


SIP Supplementary Services scripts can be configured with both the originate and terminate channels on the same physical interface on Abacus50Eth and ICG3, but only with a proxy server configured in SUT settings to prevent SIP Signaling from looping back internally on the ICG3 or Abacus50Eth.

If you are not using a Proxy server, you must manually configure SIP Supplementary Services scripts to ensure multiple endpoints, involved in the same call, send traffic out different physical interfaces. 

SIP supplementary services are intended to provide you with complex calling scenarios per RFC 5359 (Session Initiation Protocol Service Examples) and MLPP (Multilevel Precedence and Preemption).  Default scripts are included for approximately 20 scenarios listed by function and RFC section, and users may configure new scripts for supplementary services using the predefined action templates.

SIP Supplementary Services scripts can be accessed and configured using the Script Management window / Supplementary tab, accessible via the CALL SEQUENCE Menu on the Abacus GUI. 

 Abacus SIP Supplementary Services / Script Management Window

SIP Supplementary Services scripts are added the same as standard A calls B (SIP) scripts, by selecting a script in the PARTITION and TIMING window / Scripts tab.  The SCRIPT Management window can be opened using the EDIT button next to the selected script.

Partition and Timing Window / Scripts Tab

By default, all channels assigned to a SIP Supplementary Services script are set as "Originate" channels and cannot be changed.  The number of channels assigned to a supplementary services script must be divisible the the number of channels actually used by the script.  For example, a call forwarding script uses 3 channels, A, B, and C, therefore the number of channels assigned to the set must be divisible by 3. By default, the links are set to "Own" which takes the phone numbers for all channels used from the OWN phone book.

Abacus Partition and Timing window / Association Tab

Channels to be used during a test may be configured on a single physical interface, on the PROTOCOL SELECTION window on the ICG tab, or on multiple physical interfaces.

Abacus Protocol Selection Window / ICG Tab

To manually configure channels to send signaling traffic out different interfaces, use the MAPPING button on the bottom of the SIP Supplementary Services Script Management Window (via the CALL SEQUENCE Menu / Scripts option on the Abacus GUI).

Click the button labeled MAPPING, Select CUSTOM MAPPING from Settings, and move every other channel over to the column labeled "Port 1".  With custom mapping, each endpoint involved in a call will signal out a separate physical interface, ensuring that signaling is not looped back internally on the ICG3 or Abacus50Eth.

 SIP Supplementary Services / Custom channel mapping

With custom mapping configured, the OWN phone book for sets of channels configured for the selected script, will be assigned for channels configured on Port 0 (A and C) then from Port 1 (B).

Phone books are accessed from the PARTITION & TIMING window under the PHONES & ENDPOINTS tab, by clicking the EDIT button next to the Per Set or Global Endpoint configuration.  In the example used above, phone numbers from the OWN phone book will be assigned first to endpoint A, then to endpoint C, and finally to endpoint B, and so on.

OWN Phone book for custom mapped SIP Supplementary Services scripts

Once a test has started, you can verify that signaling is not looping back internally on the Abacus by configuring the Call Tracer to monitor multiple channels involved in the same call script.  The Call Tracer can be accessed from the MONITOR Menu by selecting CALL TRACER. 

Abacus Call Tracer / SIP Supplementary Services call signaling

If SIP Signaling is looping back internally for SIP Supplementary Services, no call signaling will be visible on the Call Tracer.



Product : Abacus,Voice,6.20,6.11 and below,SIP