Routing script tutorial:Mini Development Guide: Difference between revisions

 contacts = params[ :contacts ]
 contacts = {
     :index=>"3",
     :list=>[
        {:called_number=>"6660", :priority=>"1000", :is_number_ported=>"0", :sip_uri=>"", :raw_data=>"", :expiration=>"3600"},
        {:called_number=>"6661", :priority=>"1000", :is_number_ported=>"0", :sip_uri=>"sip:6661@192.168.215.127", :raw_data=>"<sip:6661@192.168.215.127>", :expiration=>"3600"}
        {:called_number=>"6662", :priority=>"1000", :is_number_ported=>"0", :sip_uri=>"sip:6662@192.168.215.128", :raw_data=>"<sip:6662@192.168.215.128>", :expiration=>"3600"},
        {:called_number=>"6663", :priority=>"1000", :is_number_ported=>"0", :sip_uri=>"sip:6663@192.168.215.129", :raw_data=>"<sip:6663@192.168.215.129>", :expiration=>"3600"},
        {:called_number=>"6664", :priority=>"1000", :is_number_ported=>"0", :sip_uri=>"sip:6664@192.168.215.150", :raw_data=>"<sip:6664@192.168.215.150>", :expiration=>"3600"}
     ],
     :source_indexes=>"nil,0,0,0,2"
 }

params[:contacts][:list] contains the contact log. Each contact within the list has the following fields:
- :called_number - the called number
- :is_number_ported - if the called number has been ported (for SIP: if the npdi parameter is present)
- :ported_number - the called number that was ported (for SIP: the rn parameter value, if available)
- :sip_uri - the SIP URI of the contact, without the contact-params section (without the expires and q contact parameters)
- :raw_data - the raw data representing the contact in the signaling protocol. For SIP, this is the full SIP URI (including the expires and q contact parameters).
- :priority - the priority of the contact [0-1000]
- :expiration - the expiration time in seconds of the contact
params[:contacts][:index] contains the index of the contact that is currently being routed.
params[:contacts][:source_indexes] contains a comma-separated list of indexes from params[:contacts][:list]. Each index represents the contact from which the contact in the list was obtained from.

To get more information, see:

SIP Redirection Contacts Parameters in a Call Flow

Connected number

Insert a connected number in the answer message of the call flow.

Routing script example:

   bridge = params[ :bridge ]
   bridge [ :connected_number ] = "3335577"
   bridge [ :connected_number_noa ] = :national_number
   bridge [ :connected_number_npi ] = :private
   bridge [ :connected_number_presentation ] = :allowed
   bridge [ :connected_number_screening ] = :pass

Terminating calls

In release 2.8, it is now possible to terminate a call through the routing scripts. The reason code must be specified in params[:bridge][:reason]. The :terminate hash must be created and copied into params:

 terminate = {}
 params[:terminate] = terminate

The following fields can then be set in :terminate:

:sip_header
:contacts # list of contacts as described in the redirection section
:isup_raw
:isup_raw_variant
:redirecting_number
:redirecting_number_noa
:redirecting_number_npi
:redirecting_number_presentation
:redirecting_number_reason
:redirecting_number_counter
:redirecting_number_indicator
:original_called_number
:original_called_number_noa
:original_called_number_npi
:original_called_number_presentation
:original_called_number_reason
:original_called_number_counter

Reason values

Check here for Termination Reason Cause codes:

Termination Reason Cause codes

Example to refuse an incoming call leg.

 raise RoutingException, :no_route

Reason cause strings available inside routing scripts:

List of Q.850 reason causes:

 :unallocated_number
 :no_route_to_network
 :no_route_to_destination
 :send_special_tone
 :misdialled_trunk_prefix
 :channel_unacceptable
 :call_awarded_in_established_channel
 :preemption
 :reattempt
 :qor_ported_number
 :normal_call_clearing
 :user_busy
 :no_user_responding
 :no_answer_from_user
 :subscriber_absent
 :call_rejected
 :number_changed
 :redirection
 :exchange_routing_error
 :non_selected_user_clearing
 :destination_out_of_order
 :address_incomplete
 :facility_rejected
 :response_to_status_enquiry
 :normal_unspecified
 :no_circuit_available
 :network_out_of_order
 :frame_mode_out_of_service
 :frame_mode_connection_operational
 :temporary_failure
 :switching_equipment_congestion
 :access_information_discarded
 :requested_circuit_not_available
 :precedence_call_blocked
 :resource_unavailable
 :quality_of_service_not_available
 :requested_facility_not_subscribed
 :outgoing_calls_barred
 :outgoing_calls_barred_within_cug
 :incoming_calls_barred
 :incoming_calls_barred_within_cug
 :bearer_cap_not_authorized
 :bearer_cap_not_available
 :inconsistency_access_info
 :service_not_available
 :bearer_cap_not_implemented
 :channel_type_not_implemented
 :requested_facility_not_implemented
 :only_restricted_digital_info
 :service_not_implemented
 :invalid_call_reference
 :channel_does_not_exist
 :call_identity_does_not_exist
 :call_identity_in_use
 :no_call_suspended
 :call_has_been_cleared
 :user_not_member_of_cug
 :incompatible_destination
 :non_existant_cug
 :invalid_transit_network
 :invalid_message_unspecified
 :mandatory_ie_missing
 :message_type_non_existent
 :message_not_compatible_with_call_state
 :ie_non_existent
 :invalid_ie_content
 :msg_not_compatible_with_call_state
 :recovery_on_timer_expiry
 :parameter_non_existent_passed_on
 :message_with_non_recognized_parameters_discarded
 :protocol_error
 :interworking_unspecified

List of toolpack reason causes:

 :toolpack_normal                       or :normal
 :toolpack_resource_error               or :resource_error
 :toolpack_timeout                      or :timeout
 :toolpack_no_route                     or :no_route
 :toolpack_call_collision               or :call_collision
 :toolpack_sync_drop                    or :sync_drop
 :toolpack_signaling_error              or :signaling_error
 :toolpack_locally_rejected             or :locally_rejected
 :toolpack_interface_not_available      or :interface_not_available
 :toolpack_reset_in_progress            or :reset_in_progress
 :toolpack_adapter_reject               or :adapter_reject
 :toolpack_missing_or_invalid_ie        or :missing_or_invalid_ie
 :toolpack_incoming_only                or :incoming_only
 :toolpack_system_configuration_changed or :system_configuration_changed
 :toolpack_resource_no_more_available   or :resource_no_more_available
 :toolpack_incompatible_media           or :incompatible_media
 :toolpack_resource_allocation_failed   or :resource_allocation_failed
 :toolpack_data_path_not_available      or :data_path_not_available
 :toolpack_local_congestion             or :local_congestion
 :toolpack_authorization_required       or :authorization_required
 :toolpack_call_divert_is_not_allowed   or :call_divert_is_not_allowed

List of SIP reason causes:
Reason causes starting with a digit must use the following syntax (can't use : as prefix).

 '300_multiple_choices'
 '301_moved_permanently'
 '302_moved_temporarily'
 '305_use_proxy'
 '380_alternative_service'
 '400_bad_request'
 '401_unauthorized'
 '402_payment_required'
 '403_forbidden'
 '404_not_found'
 '405_method_not_allowed'
 '406_not_acceptable'
 '407_proxy_authentication_required'
 '408_request_timeout'
 '409_conflict'
 '410_gone'
 '413_request_entity_too_large'
 '414_request_URI_too_long'
 '415_unsupported_media'
 '416_unsupported_URI_scheme'
 '420_bad_extension'
 '421_extension_required'
 '422_session_timer_too_small'
 '423_interval_too_brief'
 '429_referrer_identity_error'
 '480_temporary_unavailable'
 '481_call_or_transaction_does_not_exist'
 '482_loop_detected'
 '483_too_many_hops'
 '484_address_incomplete'
 '485_ambiguous'
 '486_busy_here'
 '487_request_terminated'
 '488_not_acceptable_here'
 '489_bad_event'
 '491_retry_after'
 '500_server_internal_error'
 '501_not_implemented'
 '502_bad_gateway'
 '503_service_unavailable'
 '504_server_timeout'
 '505_version_unsupported'
 '513_message_too_large'
 '600_busy_everywhere'
 '603_decline'
 '604_not_exist_anywhere'
 '606_not_acceptable'

NAP status and other NAP information

All the status fields of the NAPs are provided for use by the routing scripts.
For developpers: see the nap status provider for more details on which fields are available in the CEngineStatTransNap.hpp file.

Notice: These values may change between major release.

Routing script call attribute name Description
--------------------------------------------------------------------------------------------
"name" NAP name.
"signaling_type" Signaling type (SS7, ISDN, CASR2, SIP)
"profile" Profile name.
"sip_destination_ip" Destination IP address.
"sip_destination_port" Destination IP port.
"sip_transport_type" SIP transport type (:udp, :tcp, or :tls) (Toolpack 3.1 and more)
"inst_incoming_call_cnt" Instantaneous Count of incoming calls.
"inst_outgoing_call_cnt" Instantaneous Count of outgoing calls.
"available_cnt" Number of available circuits or channels.
"unavailable_cnt" Number of unavailable circuits or channels.
"availability_percent" Percentage of available circuits or channels.
"usage_percent" Percentage of used circuits or channels.
"unused_shared_percent" Percentage of used circuits or channels of this NAP available to make new calls with (taking into account shared with other NAPs)
"total_incoming_call_cnt" Total Count of incoming calls.
"global_asr_percent" Global calculated ASR percentage.
"total_outgoing_call_cnt" Total Count of outgoing calls.
"last_24h_asr_percent" Last 24 hours calculated ASR percentage.
"last_24h_outgoing_call_cnt" Last 24 hours outgoing calls.
"current_hour_asr_percent" Current hour calculated ASR percentage.
"current_hour_outgoing_call_cnt" Current hour outgoing calls.
"last_hour_asr_percent" Last hour calculated ASR percentage.
"last_hour_outgoing_call_cnt" Last hour outgoing calls.
"poll_remote_proxy" Remote proxy polling enabled
"is_available" Remote proxy actually available or not
"time_since_polling" Time since the last availibility polling
"time_available_seconds" Number of seconds since the NAP is available
"time_unavailable_seconds" Number of seconds since the NAP is unavailable
"register_to_proxy" Register to proxy enabled
"registered" Actually registered or not
"time_since_refresh" Time since the last refresh
"time_registered_seconds" Number of seconds since the NAP is registered
"time_not_registered_seconds" Number of seconds since the NAP is not registered
"asr_stats_incoming_struct" Detailed Answer-Seizure Rate incoming statistics.
{
"global_asr_percent" Global calculated ASR percentage.
"total_call_cnt" Total count of calls.
"total_accepted_call_cnt" Total count of accepted calls (not dropped due to congestion or rate-limiting).
"total_answered_call_cnt" Total count of answered calls.
"last_24h_asr_percent" Last 24 hours calculated ASR percentage.
"last_24h_call_cnt" Last 24 hours count of calls.
"current_hour_asr_percent" Current hour calculated ASR percentage.
"current_hour_call_cnt" Current hour count of calls.
"last_hour_asr_percent" Last hour calculated ASR percentage.
"last_hour_call_cnt" Last hour count of calls.
}
"asr_stats_outgoing_struct" Detailed Answer-Seizure Rate outgoing statistics.
{
"global_asr_percent" Global calculated ASR percentage.
"total_call_cnt" Total count of calls.
"total_accepted_call_cnt" Total count of accepted calls (not dropped due to congestion or rate-limiting).
"total_answered_call_cnt" Total count of answered calls.
"last_24h_asr_percent" Last 24 hours calculated ASR percentage.
"last_24h_call_cnt" Last 24 hours count of calls.
"current_hour_asr_percent" Current hour calculated ASR percentage.
"current_hour_call_cnt" Current hour count of calls.
"last_hour_asr_percent" Last hour calculated ASR percentage.
"last_hour_call_cnt" Last hour count of calls.
}
"mos_struct" Detailed Mean Opinion Score statistics.
{
"last_24h_ingress" Last 24 hours calculated MOS for incoming RTP packets.
"last_24h_egress" Last 24 hours calculated MOS for outgoing RTP packets.
"current_hour_ingress" Current hour calculated MOS for incoming RTP packets.
"current_hour_egress" Current hour calculated MOS for outgoing RTP packets.
"last_hour_ingress" Last hour calculated MOS for incoming RTP packets.
"last_hour_egress" Last hour calculated MOS for outgoing RTP packets.
}
"network_quality_struct" Detailed network quality statistics.
{
"last_24h_ingress" Last 24 hours network quality percentage for incoming RTP packets.
"last_24h_egress" Last 24 hours network quality percentage for outgoing RTP packets.
"current_hour_ingress" Current hour network quality percentage for incoming RTP packets.
"current_hour_egress" Current hour network quality percentage for outgoing RTP packets.
"last_hour_ingress" Last hour network quality percentage for incoming RTP packets.
"last_hour_egress" Last hour network quality percentage for outgoing RTP packets.
}

The nap status is part of a substructure and will be a hash containing all subfield elements.

Example to access NAP information and NAP status of the current call

To find the incoming nap in a routing script, we can do this in before_filter or after_filter and use these lines to create a symbol:

   incoming_nap = params[:call][:nap].to_sym
   log_trace 1, "incoming_nap = " + incoming_nap.inspect

To get the configured NAP list, you can do this:

   nap_lists = params[:naps]
   log_trace 1, "nap_lists = " + nap_lists.inspect

From the list above, you can find how many calls you have on this NAP:

   log_trace 1,"Incoming NAP call count=" + nap_lists[incoming_nap][:asr_stats_incoming_struct][:total_call_cnt].inspect

Or which network the call is coming from:

   log_trace 1,"Incoming NAP signaling type=" + nap_lists[incoming_nap][:signaling_type].inspect

If you are using a "NAP Columns" custom parameter (Create New NAP Column), you can get the information (since incoming_nap is a symbol):

   network_type = nap_lists[incoming_nap][:network_type]
   log_trace 1, "For incoming_nap = " + incoming_nap.inspect + " network_type is: " + network_type.inspect

If you want to modify something according to nap information, you may need to do a loop like this:

   nap_lists.each do |nap_list,nap_info|
     log_trace 1, "NAP: " + nap_info[:name].inspect + " network_type: " + nap_info[:network_type].inspect
   end

Telephony Services

In release 2.10 and the above, telephony services (CNAM Request) can be manage from the routing engine in the following format:

 params[:telephony_services].each do |service|  -> Array of telephony services
   service[:name]                               -> Customer telephony service name
   service[:type]                               -> For now only "CNAM Request"
   service[:enabled]                            -> Indicate if the service is enabled (true) or not (false)
                                                   (Only the telephony service define in the profile associated to the NAP is enabled)
                                                   (The others telephony services define in others profiles are disabled)
                                                   (If we are in the case where we return in the routing script with a response, it is important to set :enabled to false in order to avoid repeating the same query)
   serviceParams = service[:params]
   serviceParams[:return_to_script]             -> Indicates to Gateway if we must return to the routing script after receiving the CNAM response
                                                   (It is important to set back to false this field to avoid an infinite loop)
   serviceQuery = service[:query]
   serviceQuery[:phone]                         -> 10 digits of calling number from the incoming call to send to the CNAM server
   serviceQuery[:timeout]                       -> Timeout in millisecond to wait a CNAM response from CNAM server
                                                   (Default value from profile configuration)
   serviceResponse = service[:response]
   serviceResponse[:success]                    -> Indicates if we received a good CNAM response from the CNAM Server (Only present if :return_to_script is set to true)
   serviceResponse[:caller_name]                -> The caller name received in the CNAM response from the CNAM Server (Only present if :return_to_script is set to true)

Custom user context

The routing script may save per-call information within the call context, that will be available if routing is called again later during the call flow.

Cases where routing is called multiple times for the same call are: - Call transfer requests - SIP redirect requests - Radius Authorization result - Announcement server with digit collection

The routing script can save a recursive hash of attributes here:

 params[:user_context]

For example

 params[:user_context] = { "SomeKey" => "Some value I want to retrieve upon next routing for this call", "OtherVal" => { "subkey" => "subval" } }

Upon first call to routing script, params[:user_context] will be nil. Upon subsequent calls to routing script, it will contain whatever the script had stored upon previous call (or nil if it was not set)

Note: This feature is available starting from release 2.9.85, 2.10.31 and 3.0.15 (in respective branches 2.9, 2.10 or 3.0)

Routing Script Tests

The Web portal features a tool for Testing Scripts. The user must enter parameters to simulate the incoming call and after pressing the Test button, will output selected routes and numbers. You do not need to activate the new routes, or the new scripts to use this test tool: It can be used to test the routing scripts and routing table before activating it. This is available in the Routing Scripts section of the Web portal.

Test parameters

@call_params

That variable should contain a hash of call parameters that will be passed to the routing script. This is equivalent to the incoming call parameters.

@nap_list

A list of the hash containing the nap statuses. This is equivalent to the nap statuses at the time the call is to be routed.

The nap list is hashed by the nap names in UPPERCASE. It is important to consider this when creating new dynamic route or nap attributes that may nap names that will be used to fetch a status.

@params

A hash of hashes containing parameters. This hash contains bridge parameters and other kind of parameter groups may be added in the future.

 @params = {
    :bridge => {:announcement_tone, "announcement.wav"},
    :contacts => {
      :index=>"1",
      :list=>[
         {:called_number=>"6660", :priority=>"1000", :is_number_ported=>"0", :sip_uri=>"",
          :raw_data=>"", :expiration=>"3600"},
         {:called_number=>"6661", :priority=>"1000", :is_number_ported=>"0", :sip_uri=>"sip:6661@192.168.215.127", 
          :raw_data=>"<sip:6661@192.168.215.127>", :expiration=>"3600"}
      ],
      :source_indexes=>"nil,0"
    }
 }

Back to Routing Script Tutorial.

@@ Line 40: / Line 40: @@
 [[Routing Script Tutorial: DNS Query|Learn how to send DNS Query requests to DNS servers in routing scripts.]]
-== Call diversion options ==
+== Call Diversion Options ==
 [[Routing Script Tutorial: Call Diversion Options|Learn how control a call flow with diversion options.]]
-== Call transfer requests ==
+== Call Transfer Requests ==
-Toolpack allows that [[Call transfer]] requests are relayed from one leg to the other, or to process them locally (making another outgoing call to replace the call that requested the call transfer).
+[[Routing Script Tutorial: Call Transfer Requests|Learn how to route call transfer requests.]]
-If the chosen [[Call transfer]] mode is to process requests locally, upon reception of a call transfer request (SIP REFER or ISDN Facility), routing script will be called once again, to select the routes for the new outgoing call (call transfer target).
-=== How to route call transfer request ===
-Routing of a call transfer request is done exactly like routing of a normal incoming call.
-The routing script generally does not need any modification to support that.
-In some cases, the routing script may want to use information related to the transfer request to perform routing, or to insert information in the outgoing call leg.
-Additional information is provided to the routing script, allowing routing decisions using information from the call transfer request (SIP REFER or ISDN Facility).
-See below...
-=== params[ :call ] content during transfer request ===
-When processing a call transfer request, the params[ :call ] hash contains the information from the inbound call (same as was passed to the routing script upon arrival of the inbound call)
- call = params[ :call ]          -> Information from original inbound call, with exception of call[ :called ]
-One exception (convenient because it allows a unmodified routing script to process call transfer request the same way as any other routing request):
- call[ :called ]                 -> Replaced by the called number from the call transfer request (also called "redirection number")
-Complementary information:
- call[ :original_called_number ] -> Contains the called number that was initially received from the incoming call, prior to call transfer request
- call[ :redirecting_number ]     -> Number of the call from which the call transfer request was received (generally equals to original_called_number)
-These fields will also be included in the outgoing call made after routing:
-* original called number and redirecting number are existing fields on SS7 and ISDN calls
-* SIP "diversion" header is used for SIP calls
-=== params[ :transfer ] content ===
-(this if valid only for release 2.7.102 and above)<br>
-When processing a call transfer request, information from the call transfer request message (SIP REFER, ISDN Facility) is provided in params[ :transfer ]:
-  transfer = params[ :transfer ]
-The following field is always present:
-  transfer[ :original_nap ]      -> Contains the NAP of the first call from which a call transfer request was received
-  transfer[ :redirecting_nap ]   -> Contains the NAP of the call from which the current call transfer request was received
-                                    (same as :original_nap for the first call transfer, different for subsequent transfers)
-Examples of other fields that may be present, when appropriate:
-  transfer[ :uui ]               -> The UUI (user-to-user information) found in the call transfer request
-  transfer[ :sip_header ]        -> Contains custom SIP headers from the call transfer request
-  transfer[ :request_uri ]       -> Contains the SIP Request URI
-These fields are 'read-only'. They will not be included in the outgoing call, as they represent the contents of the call transfer request, and not the outgoing call to be made.
-To insert/modify attributes of the outgoing call, the parameters from params[ :call ] must be edited instead.
 == Redirection ==