Routing script tutorial:Mini Development Guide

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(Added value "video" for "information_transfer_capability")
(Adding sip uri parameters for to, from, private and contact headers)
Line 958: Line 958:
 
Encryption          Reject-Contact        Service-Route             
 
Encryption          Reject-Contact        Service-Route             
 
                                             Session-Expires
 
                                             Session-Expires
 +
</pre>
 +
 +
=== sip uri parameters  ===
 +
Access to parameters of several SIP headers (To, From, P-Asserted-Identity, Remote-Party-ID, Contact).
 +
 +
<pre>
 +
call[ :calling_parameters ]         
 +
call[ :called_parameters ]
 +
call[ :private_address_parameters ]  (for P-Asserted-Identity or Remote-Party-ID)
 +
call[ :contact_parameters ]
 +
</pre>
 +
 +
These parameters (if present) contains a hash with 3 keys: user_param, uri_param and header_param.
 +
If a param has already a specific field: oli, isub, cpc, transport. It won't be present in the new generic structure.
 +
By default, all parameters will be forwarded in a SIP to SIP call flow.
 +
 +
Using this code in routing script:
 +
<pre>
 +
puts "calling_param        = #{ call[ :calling_parameters].inspect}"
 +
</pre>
 +
 +
Example, From header versus the call[ :calling_parameters] value:
 +
<pre>
 +
From:"test"<sip:4440000;user_param1=1;user_param2@10.3.10.217;uri_param3=3>;header_param4=4;tag=6F97303034343030002A2484
 +
calling_param        = {"user_param"=>"user_param1=1;user_param2", "uri_param"=>"uri_param3=3", "header_param"=>"header_param4=4"}
 +
</pre>
 +
 +
<pre>
 +
From:"test"<sip:4440000@10.3.10.217;user=phone>;tag=6F97303034343030002A2484
 +
calling_param        = {"uri_param"=>"user=phone"}
 +
</pre>
 +
 +
Example to insert outgoing parameters and overwrite received values.
 +
<pre>
 +
call [:calling_parameters] = {"user_param"=>"user_param7=7;user_param8", "uri_param"=>"uri_param9=9", "header_param"=>"header_paramA=A"}
 +
</pre>
 +
 +
Example to add user=phone and keep all other uri parameters.
 +
<pre>
 +
    if call [:calling_parameters]
 +
      if call [:calling_parameters]["uri_param"]
 +
        call[:calling_parameters]["uri_param"] << ";user=phone"
 +
      else
 +
        call [:calling_parameters]["uri_param"] = "user=phone"
 +
      end
 +
    else
 +
      call [:calling_parameters] = {"uri_param"=>"user=phone"}
 +
    end
 
</pre>
 
</pre>
  

Revision as of 15:55, 13 October 2016

Contents

Call object

Get

This function is used to get the call parameters. The possible parameters are described in the section "Call parameters" 

 called_number = caf_call.get :called

List_params

This function is used to retrieve the list of supported call parameters. For example to extract all the possible call params from the the call object and put it in hash. 

 caf_call.list_params.each {|param| call[param] = caf_call.get param }

Accept

This function is used to accept a call. It actually creates one outgoing route that gateway application will use to bridge the incoming call leg. If more than one outgoing route is "accepted", gateway will try them one by one in the same order they were accepted. If an outgoing call leg fails (according to 'route retry' parameters), the next route in line will be used.

This method takes 2 arguments, the call parameters (hash) and the route parameters (hash). Note that calling this method does NOT stop the flow of the script.

Apply route remapping rules 

 caf_call.accept out_call, route

Refuse

This function is used to set the reason code for the incoming call leg refusal. However, this function does NOT stop the flow of the script. 

 caf_call.refuse :reason => :temporary_failure

To immediately refuse the incoming call leg and stop processing the script, the script must raise an exception. Exiting the script by raising the exception overwrites any reason cause previously stored using refuse(). 

 raise RoutingException, :no_route

The supported refusal cause values for both refuse() and raise() are described in the section "Reason values".

Script parameters protocol mapping

The following call parameters are available in the call object. For example 

 called_number = call[:called]
Script parameter name ISDN
 		R2 CAS
 SS7
 		SIP
 		Comment
 		Toolpack version
leg_id
 N/A
 N/A
 N/A
 N/A
 Leg ID
session_id
 N/A
 N/A
 N/A
 N/A
 Session ID
calling
 Q931: 'Calling party number' IE - Number digits
ANI (Group B)
 Q763: 'Calling party number' IE - address signals (*)
 SIP:From - user-info
 * In ANSI SS7 LNP networks, the IE 'generic address parameter' is used (when present) instead.
calling_noa
 Q931: 'Calling party number' IE - Type of number
 N/A
 Q763: 'Calling party number' IE - nature of address indicator (*)
 N/A
 * In ANSI SS7 LNP networks, the IE 'generic address parameter' is used (when present) instead
calling_npi
Q931: 'Calling party number' IE - Numbering plan identification
 N/A
 Q763: 'Calling party number' IE - numbering plan indicator (*)
 N/A
 * In ANSI SS7 LNP networks, the IE 'generic address parameter' is used (when present) instead
calling_display
Q931: 'Display' IE - Display information

Q931: 'Facility CNAM' IE when presentation is allowed for DMS/NI2 variants

 N/A
 Q763

ITU97: 'Display information' IE - display information ANSI95: 'Generic name' IE - display information

SIP:From - display-name

calling_display_type
 Q931: 'Display' IE - Display information (present and/or first byte)
 N/A
 Q763: 'Display information' IE - present or not
 N/A

calling_presentation
Q931: 'Calling party number' IE - Presentation indicator
 N/A
 Q763: 'Calling party number' IE - address presentation restricted indicator

SIP:From - display-name (displays 'anonymous' or not)

SIP:Remote-party-id - privacy

calling_screening
 Q931: 'Calling party number' IE - Screening indicator
 N/A
 Q763: 'Calling party number' IE - screening
 SIP:Remote-party-id - screen

calling_category
 N/A
 Call party category (Group A)
 Q763: 'Calling party's category' IE - calling party's category

SIP:From - cpc

SIP:P-asserted-identity - cpc

calling_subscriber

(Generic Number / NDS)

 Q931: 2nd 'Calling party number' IE - Number digits
N/A
 Q763: Generic number IE with type 'additional calling party number' - Number digits

SIP:P-asserted-identity - userinfo

SIP:Remote-party-id - user-info

 Requires option 'support 2 calling number IE' in the profile. This variable has priority over 'private_address' in the outgoing direction.
calling_subscriber_noa
 Q931: 2nd 'Calling party number' IE - Type of number
 N/A
 Q763: Generic number IE with type 'additional calling party number' - nature of address indicator
 SIP:P-asserted-identity - userinfo

SIP:Remote-party-id - user-info

calling_subscriber_npi
Q931: 2nd 'Calling party number' IE - Numbering plan identification
 N/A
 Q763: Generic number IE with type 'additional calling party number' - numbering plan indicator
 SIP:P-asserted-identity - userinfo

SIP:Remote-party-id - user-info



calling_subscriber_presentation
 Q931: 2nd 'Calling party number' IE - Presentation indicator
 N/A
 Q763: Generic number IE with type 'additional calling party number' - presentation restricted indicator
 SIP:P-asserted-identity - userinfo

SIP:Remote-party-id - user-info

calling_subscriber_screening
Q931: 2nd 'Calling party number' IE - Screening indicator
 N/A
 Q763: Generic number IE with type 'additional calling party number' - screening
 SIP:P-asserted-identity - userinfo

SIP:Remote-party-id - user-info

private_display
 Q931: 'Facility CNAM' IE when presentation is restricted for DMS/NI2 variants
 N/A
 N/A

SIP:P-asserted-identity - display-name

SIP:Remote-party-id - display-name

private_display_type
N/A
 N/A
 N/A
 N/A
 Indicate presence or not of the private calling information
private_address
 N/A
 N/A
 N/A

SIP:P-asserted-identity - userinfo

SIP:Remote-party-id - user-info

called
Q931: 'Called party number' IE - Number digits
DNIS (Group A)
 Q763: 'Called party number' IE - address signals
 SIP:To - user-info and host

called_noa
Q931: 'Called party number' IE - Type of number
 N/A
 Q763: 'Called party number' IE - nature of address indicator
 N/A

called_npi
Q931: 'Called party number' IE - Numbering plan identification
 N/A
 Q763: 'Called party number' IE - numbering plan indicator
 N/A

charge_number
 N/A
 N/A
 ANSI: 'Charge number' IE - address signals
 N/A

charge_number_noa
N/A
 N/A
 ANSI: 'Charge number' IE - nature of address indicator
 N/A

charge_number_npi
 N/A
 N/A
 ANSI: 'Charge number' IE - numbering plan indicator
 N/A

redirecting_number_forward_enabled
N/A
 N/A
 N/A
 N/A
 Overwrite default redirecting number and original called number forwarding behavior from incoming to outgoing leg

redirecting_number
Q931: 'Redirecting number' 1st IE - Number digits
N/A
 Q763: 'Redirecting number' IE - address signals
 SIP:Diversion (2nd header) - display-name

redirecting_number_noa
Q931: 'Redirecting number' 1st IE - Type of number
 N/A
 Q763: 'Redirecting number' IE - nature of address indicator
 N/A

redirecting_number_npi
Q931: 'Redirecting number' 1st IE - Numbering plan identification
 N/A
 Q763: 'Redirecting number' IE - numbering plan indicator
 N/A

redirecting_number_presentation
Q931: 'Redirecting number' 1st IE - Presentation indicator
N/A
 Q763: 'Redirecting number' IE - address presentation restricted indicator
 SIP:Diversion (2nd header) - diversion-privacy

redirecting_number_indicator
N/A
 N/A
 Q763: 'Redirection information' IE - redirecting indicator
 N/A

redirecting_number_reason
Q931: 'Redirecting number' 1st IE - Reason for redirection
 N/A
 Q763: 'Redirection information' IE - redirecting reason
 SIP:Diversion (2nd header) - diversion-reason

redirecting_number_counter
N/A
 N/A
 Q763: 'Redirection information' IE - redirection counter
 SIP:Diversion (2nd header) - diversion-counter

original_called_number

(OCN)

Q931: 'Redirecting number' 2nd IE - Number digits
N/A
 Q763: 'Redirection number' IE - address signals
 SIP:Diversion  (1st header) - display-name

original_called_number_noa
Q931: 'Redirecting number' 2nd IE - Type of number
 N/A
 Q763: 'Redirection number' IE - nature of address indicator
 N/A

original_called_number_npi
Q931: 'Redirecting number' 2nd IE - Numbering plan identification
 N/A
 Q763: 'Redirection number' IE - numbering plan indicator
 N/A

original_called_number_presentation
Q931: 'Redirecting number' 2nd IE - Presentation indicator
N/A
 Q763: 'Redirection number' IE - address presentation restricted indicator
 SIP:Diversion (1st header) - diversion-privacy

original_called_number_reason
Q931: 'Redirecting number' 2nd IE - Reason for redirection
 N/A
 Q763: 'Redirection information' IE - original redirection reason
 SIP:Diversion (1st header) - diversion-reason

original_called_number_counter
N/A
 N/A
 N/A
 SIP:Diversion (1st header) - diversion-counter

ported_number
N/A
 N/A
 Q763: 'Called party number' IE - address signals
 SIP:RequestURI - rn
 Only valid if SIP/SS7 supports LNP
ported_number_noa
N/A
 N/A
 Q763: 'Called party number' IE - nature of address indicator
 N/A
 Only valid if SIP/SS7 supports LNP
ported_number_npi
N/A
 N/A
 Q763: 'Called party number' IE - numbering plan indicator
 N/A
 Only valid if SIP/SS7 supports LNP
oli

(Originating line information)

5ESS Codeset 6 OLI - Value
 N/A
 ANSI: 'Originating line information' IE - OLI

SIP:From - oli

SIP:P-asserted-identity - oli

request_uri
N/A
 N/A
 N/A
 Complete Request URI string

request_uri_forward_enabled
N/A
 N/A
 N/A
 N/A
 Overwrite default URI forwarding behavior from incoming to outgoing leg
sip_header
 N/A
 N/A
 N/A
 Any header
 Requires option 'Forward custom headers' in Profiles->SIP
2.7.63
nap

(Network Access Point)

N/A
 N/A
 N/A
 N/A
 Incoming leg NAP name (read-only)
type_of_network_identification
 Q931: 'Transit network selection' IE - Type of network identification
N/A
 Q763: 'Transit network selection' IE - Type of network identification
N/A
2.7
network_identification
 Q931: 'Transit network selection' IE - Network identification
N/A
 Q763: 'Transit network selection' IE - Network identification
SIP: Request-Line - cic
2.7
network_identification_plan
 Q931: 'Transit network selection' IE - Network identification plan
N/A
 Q763: 'Transit network selection' IE - Network identification plan
N/A
2.7
location_number_forward_enabled
N/A
 N/A
 N/A
 N/A
 Overwrite default location number forwarding behavior from incoming to outgoing leg
2.7
location_number
N/A
 N/A
 Q763: 'Location number' IE - address signals
 N/A
2.7
location_number_noa
N/A
 N/A
 Q763: 'Location number' IE - nature of address indicator
 N/A
2.7
location_number_npi
N/A
 N/A
 Q763: 'Location number' IE - numbering plan indicator
 N/A
2.7
location_number_presentation
 N/A
 N/A
 Q763: 'Location number' IE - presentation restricted indicator
 N/A
2.7
location_number_screening
N/A
 N/A
 Q763: 'Location number' IE - screening
 N/A
2.7
mlpp_forward_enabled
N/A
 N/A
 N/A
 N/A
 A script needs to set this to true if it wants to overwrite MLPP information in the outgoing leg. Otherwise, profile relay 'outgoing mode' applies automatically.
 2.7
mlpp_look_for_busy
N/A
 N/A
 Q763: 'MLPP precedence' IE - look ahead for busy
 N/A
2.7
mlpp_precedence_level
N/A
 N/A
 Q763: 'MLPP precedence' IE - precedence level
 SIP:Resource-Priority - q735
2.7
mlpp_network_identity
N/A
 N/A
 Q763: 'MLPP precedence' IE - network identity
 N/A
2.7
mlpp_service_domain
 N/A
 N/A
 Q763: 'MLPP precedence' IE - MLPP service domain
 N/A
2.7
called_isub
Q931: 'Called party subaddress' IE - subaddress information
 N/A
 Q763: 'Access transport' IE
 SIP:To - isub parameter
2.7
called_isub_type
 Q931: 'Called party subaddress' IE - type of subaddress
 N/A
 Q763: 'Access transport' IE
 SIP:To - isub-encoding parameter
2.7
calling_isub
Q931: 'Calling party subaddress' IE - subaddress information
 N/A
 Q763: 'Access transport' IE
 SIP:From - isub
2.7
calling_isub_type
 Q931: 'Callinf party subaddress' IE - type of subaddress
 N/A
 Q763: 'Access transport' IE
 SIP:From - isub-encoding
2.7
ss7_fci_default
N/A
 N/A
 Default forward call indicator (FCI) value.
 N/A
 Toolpack will overwrite FCI bits A, D, F, I and M with appropriate values according to call conditions
 2.7
ss7_fci_force_mask
N/A
 N/A
 Mask to select bits from ss7_fci_default that must be forced.
 N/A
 Bits from ss7_fci_default which corresponding bit in ss7_fci_force_mask is set will be forced, and no more controlled by Toolpack
 2.7
ss7_bci_default
N/A
 N/A
 Default backward call indicator (BCI) value.
 N/A
 Toolpack will overwrite BCI bits AB, I, K, M and N with appropriate values according to call conditions
 2.7
ss7_bci_force_mask
N/A
 N/A
 Mask to select bits from ss7_bci_default that must be forced.
 N/A
 Bits from ss7_bci_default which corresponding bit in ss7_bci_force_mask is set will be forced, and no more controlled by Toolpack
 2.7
tdm_ls_name

(Line Service or T1/E1 trunk)

Incoming leg line service name
 Incoming leg line service name
 Incoming leg line service name
 N/A
 (read-only)
 2.7
tdm_timeslot_nb
 Incoming leg timeslot number
 Incoming leg timeslot number
 Incoming leg timeslot number
 N/A
 (read-only)
 2.7
rtp_local_addr
 N/A
 N/A
 N/A
 Incoming leg local SDP IP address
 (read-only)
 2.7
rtp_local_port
 N/A
 N/A
 N/A
 Incoming leg local SDP IP port
 (read-only)
 2.7
rtp_remote_addr
 N/A
 N/A
 N/A
 Incoming leg remote SDP IP address
 (read-only)
 2.7
rtp_remote_port
 N/A
 N/A
 N/A
 Incoming leg remote SDP IP port
 (read-only)
 2.7
ss7_cot_enabled
N/A
 N/A
 Requests SS7 in-call continuity test for this outgoing SS7 call
 N/A
 Toolpack will request continuity test on the timeslot before making the outgoing call. If COT fails, the call will be dropped (then another route may be attempted)
 2.8
reverse_charging_indication
 Incoming leg Reverse charging indication IE present
 N/A
 N/A
 N/A
 If set in routing script, will add Reverse charging indication IE in outgoing leg (also use reverse_charging_indication_forward_enabled)
 2.8.12
reverse_charging_indication_forward_enabled
 N/A
 N/A
 N/A
 N/A
 Enable forwarding of reverse charging indication from incoming to outgoing leg
 2.8.12
sip_local_addr
 N/A
 N/A
 N/A
 Incoming leg local SIP IP address
 (read-only)
 2.8.13
sip_local_port
 N/A
 N/A
 N/A
 Incoming leg local SIP UDP port
 (read-only)
 2.8.13
sip_remote_addr
 N/A
 N/A
 N/A
 Incoming leg remote SIP IP address
 (read-only)
 2.8.13
sip_remote_port
 N/A
 N/A
 N/A
 Incoming leg remote SIP UDP port
 (read-only)
 2.8.13


Noa values

  • unknown_number (0x2)
  • international_number (0x4)
  • national_number (0x3)
  • subscriber_number (0x1)
  • network_specific (0x5)
  • network_routing_national_format (0x7)
  • network_routing_international_format (0x8)
  • abbreviated_number (0x6)
  • subscriber_number_operator_requested (0x71)
  • national_number_operator_requested (0x72)
  • international_number_operator_requested (0x73)
  • no_number_present_operator_requested (0x74)
  • no_number_present_cut_through_call_to_carrier (0x75)
  • test_line_test_code (0x77)
  • non_unique_subscriber_number (0x71)
  • non_unique_national_number (0x73)
  • non_unique_international_number (0x74)
  • call_950_numbe (0x76)
  • special_number (0x73)
  • national_number_with_transit_network_selection (0x74)
  • international_number_with_transit_network_selection (0x75)

Those values will be remapped to the protocol specific NOA value. To provide protocol specific value:

  • call_params[:called_noa] = 0x70

or

  • call_params[:called_noa] = 112

Npi values

  • unknown_number
  • isdn
  • telephony
  • private
  • data
  • telex
  • national

Calling Display Type values

  • unspecified => Type is unspecified.
  • calling_party_name => Type is 0xB1.

Those values will be remapped to the protocol specific Display Information Type value. To provide protocol specific value:

  • call_params[:calling_display_type] = 0xB1

or

  • call_params[:calling_display_type] = 177

Calling Display value

  • call_params[:calling_display] = "Roger Fluffy"

Presentation values for Calling number, Calling Subscriber (Generic Number), Redirecting Number, Original Called Number (OCN) and Location Number

  • unspecified
  • not_available (0x2)
  • allowed (0x0)
  • restricted (0x1)
  • addr_restricted
  • name_restricted

Calling Party Category

values for calling_category

  • unspecified (0xa)
  • unknown (0x0)
  • operator_french (0x1)
  • operator_english (0x2)
  • operator_german (0x3)
  • operator_russian (0x4)
  • operator_spanish (0x5)
  • subscriber (0xa)
  • subscriber_with_priority (0xb)
  • data (0xc)
  • test (0xd)
  • payphone (0xf)

Screening values for Calling number, Calling Subscriber (Generic Number), and Location Number

  • unspecified
  • no (0x0)
  • pass (0x1)
  • fail (0x2)
  • network_provided (0x3)

Redirecting indicator values

SS7:

  • no_redirection
  • call_rerouted
  • call_rerouted_all_restricted
  • call_diverted
  • call_diverted_all_restricted
  • call_rerouted_restricted
  • call_diverted_restricted
  • spare

Redirecting number, Original Called Number and Diversion Reason

ISDN:

  • unknown
  • busy
  • no_reply
  • deflection
  • dte_out_of_order
  • forwarding_by_called_dte
  • unconditional

SS7:

  • unknown
  • busy (SIP: user-busy)
  • no_reply (SIP: no-answer)
  • unconditional
  • deflection
  • deflection_immediate
  • mobile_not_reachable

OLI (originating line information) values

The OLI parameter is a string that represents an integer value from 0 to 255.

Information Transfer Capability values

information_transfer_capability:

  • digital
  • restricted_digital
  • digital_with_tones
  • speech
  • 3_1_khz_audio
  • video

redirecting_number_forward_enabled values

Controls forwarding or discarding of redirecting number (SIP: diversion header) to outgoing call leg.

Values for this parameter are "0", "1", "false" or "true.

  • 0/false: Redirecting number (and original called number) is not forwarded to outgoing call leg
  • 1/true: Redirecting number (and original called number) is forwarded to outgoing call leg

The value for this parameter at input of routing script depends on the "Forward redirecting number" parameter in the "Advanced" section of the Gateway configuration page of the Web Portal. The script may change this value to override the Gateway configuration.

Note: To "insert" a new redirecting number value on the outgoing leg, redirecting_number_forward_enabled must also be set to true.

request_uri

Enables access to the Request-Line URI.

For example, if the Request-Line is: 

Request-Line: INVITE sip:4175162082@172.22.45.13:5060;user=phone;transport=udp SIP/2.0

Then the retrieved request_uri will be "sip:4175162082@172.22.45.13:5060;user=phone;transport=udp SIP/2.0".

In the routing scripts, to retrieve only the called number, this script can be used:

    if call_params[:request_uri] && call_params[:request_uri] =~ /sip:(.*)@.*/
       call_params[:called] = $1
    end

request_uri_forward_enabled values

Controls forwarding or discarding of request uri to outgoing call leg.The request uri is the information in the "Request-Line:" of the SIP INVITE message.

Values for this parameter are "0", "1", "false" or "true.

  • 0/false: Request uri is not forwarded to outgoing call leg
  • 1/true: Request uri is forwarded to outgoing call leg

The value for this parameter at input of routing script is always false.

sip_header values

Contains a hash table of custom sip headers from the inbound call leg. Any custom sip header can be added to an outgoing call leg:

call[ :sip_header ] = {"P-my-custom-header"=>"value1", "P-my-custom-header2"=>"value2", "P-my-custom-header3"=>"value3"}

p_access_network_info = call[ :sip_header ][ 'P-Access-Network-Info' ] = "3GPP-E-UTRAN-FDD; utran-cell-id-3gpp=XYZ"

List of sip headers that will not appear in call[:sip_header] since they are already processed by the SIP stack: 

Accept               Error-Info             Remote-Party-ID      
Accept-Contact       Event                  Replaces                        
Accept-Encoding      Expires                Reply-To               
Accept-Language      From                   Request-Disposition    
Alert-Info           In-Reply-To            Subject          
Allow                Max-Forwards           Subscription-State  
Allow-Events         MIME-version           Supported           
Also                 Min-Expires            Timestamp           
Anonymity            Min-SE                 To             
Authorization        Organization           Unsupported  
Authentication-Info  Path                   User-Agent  
Call-ID              Priority               Via  
Call-Info            Privacy                Warning  
Contact              Proxy-Authenticate     WWW-Authenticate  
Content-Disposition  Proxy-Authorization    Require  
Content-Encoding     Proxy-Require          Response-Key  
Content-Language     P-Media-Authorization  Retry-After  
Content-Length       P-Preferred-Identity   RPID-Privacy  
Content-Type         P-Asserted-Identity    Route  
CSeq                 RAck                   RSeq  
RAck                 Reason                 Security-Client  
Reason               Record-Route           Security-Server  
Date                 Refer-To               Security-Verify
Diversion            Referred-By            Server
Encryption           Reject-Contact         Service-Route             
                                            Session-Expires

sip uri parameters

Access to parameters of several SIP headers (To, From, P-Asserted-Identity, Remote-Party-ID, Contact). 

call[ :calling_parameters ]           
call[ :called_parameters ]
call[ :private_address_parameters ]   (for P-Asserted-Identity or Remote-Party-ID)
call[ :contact_parameters ]

These parameters (if present) contains a hash with 3 keys: user_param, uri_param and header_param. If a param has already a specific field: oli, isub, cpc, transport. It won't be present in the new generic structure. By default, all parameters will be forwarded in a SIP to SIP call flow.

Using this code in routing script: 

puts "calling_param        = #{ call[ :calling_parameters].inspect}"

Example, From header versus the call[ :calling_parameters] value: 

From:"test"<sip:4440000;user_param1=1;user_param2@10.3.10.217;uri_param3=3>;header_param4=4;tag=6F97303034343030002A2484
calling_param        = {"user_param"=>"user_param1=1;user_param2", "uri_param"=>"uri_param3=3", "header_param"=>"header_param4=4"}

From:"test"<sip:4440000@10.3.10.217;user=phone>;tag=6F97303034343030002A2484
calling_param        = {"uri_param"=>"user=phone"}

Example to insert outgoing parameters and overwrite received values. 

call [:calling_parameters] = {"user_param"=>"user_param7=7;user_param8", "uri_param"=>"uri_param9=9", "header_param"=>"header_paramA=A"}

Example to add user=phone and keep all other uri parameters. 

    if call [:calling_parameters]
      if call [:calling_parameters]["uri_param"]
        call[:calling_parameters]["uri_param"] << ";user=phone"
      else
        call [:calling_parameters]["uri_param"] = "user=phone"
      end
    else
      call [:calling_parameters] = {"uri_param"=>"user=phone"}
    end

MLPP Precedence values

mlpp_look_for_busy:

  • allowed
  • path_reserved
  • not_allowed


mlpp_precedence_level:

  • flash_override
  • flash
  • immediate
  • priority
  • routine


mlpp_network_identity:

3 digits value from 0 to 999


mlpp_service_domain:

24 bits value from 0 to 16777215

ISUB subaddress information values

called_isub_type: calling_isub_type:

  • nsap
  • nsap_ia5
  • nsap_bcd
  • user


called_isub: calling_isub:

Digits for the subaddress information.

Network Identification Plan

network_identification_plan:

  • Unknown (value 0)
  • cic (3 digits carrier identification code plus circuit code, value 1, SS7 or ISDN)
  • user (User, value 2, ISDN only)
  • cic4 (4 digits carrier identification code plus circuit code, value 2, SS7 only)
  • dnic (public Data Network ID, value 3, SS7 only)
  • mnic (public land mobile network, value 6, SS7 only)

Route parameters

All route may have these parameters:

  • calling
  • called
  • nap
  • remapped_calling
  • remapped_called
  • remapped_nap
  • remapped_destination_leg_profile (called remapped_profile prior to Toolpack 2.9)
  • remapped_source_leg_profile (called remapped_incoming_profile prior to Toolpack 2.9)

Example: 

 route[:remapped_nap]

Additionally it is possible to add dynamic route attributes in the web portal. These can be referenced by their name.

Playing prompts announcements or tones

New feature in release 2.6, all bridges may have these parameters. These can be used to play IVR prompts (audio files) in different states of the call flow.

  • announcement_tone (played before outgoing call is routed)
  • ring_tone (played after when waiting for outgoing call to answer)
  • busy_tone (played if outgoing call failed)
  • disconnect_tone (played after the call has reached it's maximum duration)


The following sections show how to access those different parameters from inside a filter. The 'params' hash is the sole argument passed to a filter function and contains call, nap, route and bridge parameters. This hash is created by the base_routing.rb scripts to regroup all relevant information just before calling the filter functions.

Tone string format

All tone strings (:announcement_tone, :busy_tone, :ring_tone, :disconnect_tone) inside bridge parameters are using this format. 

"file1.wav:repeat:start_off:end_off,file2.wav:repeat:start_off:end_off,file3.wav:repeat:start_off:end_off"

Optional parameters:

  • repeat: number of times to play the file (0 and 1 have the same result)
  • start_off: Start offset in milliseconds
  • end_off: End offset in milliseconds

Http and other path formats are described here: Path format

Example 1

The following example will play file1.wav once, and then play file2.wav in loop: 

 "file1.wav,file2.wav:-1"

Example 2

The following example will play file1.wav from start offset of 1 second to end offset of 3 seconds, then twice file2.wav from second 5 to second 10. 

 "file1.wav:0:1000:3000,file2.wav:2:5000:10000"

Example 3

The following example will play file1.wav once, ending at offset of 30 seconds. 

 "file1.wav:0:0:30000"

announcement_tone 

 params[:bridge][:announcement_tone] = "announcement.wav"

Audio file played on the incoming call before any outgoing call is placed. The outgoing call occurs when the file finished playing.

announcement_tone options

announcement_tone_answer 
 params[:bridge][:announcement_tone_answer] = "yes"

Forces an answer of the call before playing the announcement. Default if argument not provided is "no", in which case call is only alerted with in-band media.

announcement_code_detect

This option allows to enable tone detection during the announcement play.

Collected digits can be inserted into the CDR logs (radius attribute "Telcob-CollectedDigits", or text CDR variable @{CollectedDigits}).

Collected digits can also be sent back to routing script, which is called again with the same call attributes, except that the called number is replaced by the collected digits.

Code detect has multiple options, as shown in the following code: 

 code_detect = {
   :type                   => :DTMF,   # :DTMF or :MFR1 tone detection.
                                       # Default is MFR1.
   :prefix                 => "",      # Prefix (digits) that is removed from collected digits.
                                       # Default is empty.
   :suffix                 => "",      # Suffix (digits) that is removed from collected digits
                                       # and causes routing script to be immediately called.
                                       # Default is empty.
   :suffix_removal         => false,   # Controls the removal of the suffix from the collected digit string that's reported to routing script.
                                       # Default is false
   :timeout                => 0,       # Inter-digit timeout (ms) after which collected digits are passed to the routing script.
                                       # Use 0 for "no timeout".
                                       # Default is 1000ms
   :barge_in_interruption  => true,    # When enabled, playing announcement is stopped as soon as first digit is collected.
                                       # Default is true.
   :proceed_on_play_done   => false,   # When true:  Outgoing call is made after announcement finishes playing.
                                       #             Routing script is not called again.
                                       # When false: Outgoing call is never made.
                                       #             Digits are collected until timeout or suffix match,
                                       #             then routing script is called again.
                                       # Default is false.
   :cas_on_hook            => false,   # Specific for CAS-R1 calls. Makes CAS bits switch to "on-hook" when announcement finished playing
                                       # (but the call is not "terminated" from Toolpack point of view)
                                       # Default is false.
   :cas_on_hook_delay      => 0,       # Duration of cas bits "on-hook" state.
                                       # Only effective if cas_on_hook is set to true.
                                       # Value of 0 stands for "infinite delay".
                                       # Default is 0.
   :repeat_delay           => 0,       # Delay between repetition of the announcement. The announcement will repeat
                                       # itself every "repeat_delay" until a code is detected (suffix match or timetout).
                                       # Value of 0 stands for "infinite delay" (no repeating).
                                       # Default is 0.
 }

Example 1: Collect DTMF digits, and call routing script again with collected digits upon timeout or suffix match. 

 code_detect = { :type => :DTMF, :suffix => "#", :timeout => 5000 }
 params[:bridge][:announcement_code_detect] = code_detect

Example 2: Collect digits during the announcement (for CDR logs), then proceed (make outgoing call) after announcement finishes playing 

 code_detect = { :type => :DTMF, :timeout => 0, :barge_in_interruption => false, :proceed_on_play_done => true }
 params[:bridge][:announcement_code_detect] = code_detect

Controlling what happens after announcement

The routing script can control what happens with the call after the announcement finishes playing:

  • An outgoing call is made
  • Incoming call is hung-up
  • Do nothing (wait for the incoming call to hang-up)
An outgoing call is made

This happens when the script has returned matching routes (and did not raise RoutingException)

Incoming call is hung-up

This happens when the script returns no routes (in which case base_routing will raises RoutingException with cause :no_route).

It also happens when the script explicitly raises RoutingException.

The incoming call will be terminated with the specified cause. For example 

   raise RoutingException, :temporary_failure

(See "Reason values" section in this page for list of available causes)

Do nothing (wait for the incoming call to hang-up)

If a filter raises RoutingException with code :ok, then the incoming call will not be terminated at the end of the announcement play. Announcement digit collection will remain active if appropriate. For example: 

   raise RoutingException, :ok



ring_tone 

 params[:bridge][:ring_tone] = "ringing.wav"

Audio file played on the incoming call while waiting for the outgoing call to be answered.

Ring tone playback can also be configured in the Web Portal, from the incoming call's profile (under "Tones and Call Progress Options").

Routing script has precedence over profile (a routing script that fills params[:bridge][:ring_tone] will override the profile's ring tone behavior).

ring_tone options

ring_tone_state 
 params[:bridge][:ring_tone_state] = :alerted

Call state from which ring tone is being played. Available values are:

  • immediately: Ring tone starts playing immediately on the incoming leg
  • accepted: Ring tone starts playing as soon as outgoing call is accepted
  • callprogress: Ring tone starts playing as soon as "call progress" is received on the outgoing call
  • alerted (default): Ring tone starts playing only once outgoing call is alerted (but won't play if alert indicates early media from outgoing call)

This option also apply when params[:bridge][:ring_tone] is not used, because it also apply to ring tone playback configured in the Web Portal, from the incoming call's profile.

busy_tone 

 Toolpack 2.8 and above:
   params[:bridge][:busy_tone] = "no_route.wav"

 Note: Obsolete name (toolpack 2.7.153 and earlier, but still supported in recent releases):
   params[:bridge][:call_progress_tone] = "no_route.wav"

Audio file played on the incoming call when outgoing call fails (never answered).

Note that announcement_tone, if used, is played before the outgoing call attempt is made, and thus before the busy_tone.

Busy tone playback can also be configured in the Web Portal, from the incoming call's profile (under "Tones and Call Progress Options").

Routing script has precedence over profile (a routing script that fills params[:bridge][:busy_tone] will override the profile's busy tone behavior).

Special value "none" can be used by routing script to force playing nothing (as empty string would default to profile's behavior)

busy_tone options

busy_tone_answer 
 params[:bridge][:busy_tone_answer] = "yes"

Forces an answer of the call before playing the busy tone. Default if argument not provided is "no", in which case call is only alerted with in-band media.

disconnect_tone 

 params[:bridge][:disconnect_tone] = "max_duration.wav"

Audio file played on the incoming call when call duration (:max_call_duration) is reached. Then the leg will be terminated with specified reason (:call_duration_reason).

disconnect_tone options

max_call_duration 
 params[:bridge][:max_call_duration] = "60000"

Maximum call duration in millisecond. This timer is started when entering answer state.

call_duration_reason 
 params[:bridge][:call_duration_reason] = :resource_unavailable

Drop both legs with this reason when call duration (:max_call_duration) is reached.

Managing audio prompts through Web Portal

Audio prompts can be uploaded or deleted from the TMedia unit through the Web Portal: Managing audio prompts

Prompts management must be done using the Web Portal of the primary server (in systems with redundant TMedia units or redundant host servers). The file will automatically get replicated to the secondary server.

Managing audio prompts manually

Any file on the TMedia host file system can be played. This means it's possible to manage prompts through ssh/scp.

The default (replicated) prompts folder

By default, when playing a prompt, Toolpack will look in the default prompts folder: 

/lib/tb/toolpack/pkg/prompts

The root of this "prompts" directory is automatically replicated to secondary unit of redundant setups (1+1, N+1, redundant hosts). Sub-folders won't be replicated.

Any prompt play request without explicit file path will map to this folder. For example: 

 params[:bridge][:busy_tone] = "no_route.wav"

This will correspond to file /lib/tb/toolpack/pkg/prompts/no_route.wav

Relative file paths

Any file path that begins with "file://" is considered relative to the tbstreamserver application's working directory: 

/lib/tb/toolpack/setup/12358/2.8/apps/tbstreamserver/

(Where "2.8" may be replaced by the current major version of your system)

For example: 

 params[:bridge][:busy_tone] = "file://my_folder/no_route.wav"

This will correspond to file /lib/tb/toolpack/setup/12358/2.8/apps/tbstreamserver/my_folder/no_route.wav

Absolute file paths

Absolute paths can also be provided. For example: 

 params[:bridge][:busy_tone] = "file:///root/my_folder/no_route.wav"

This will correspond to file /root/my_folder/no_route.wav

Recording call legs

Introduced in release 2.6.44, it's now possible to use routing scripts to ask for recording incoming and/or outgoing call legs.

See example filter script "call_recording" (created by default in Web Portal routing scripts starting with 2.6.44) for an example.

Recording the incoming call leg

To record the incoming call leg, the routing script (in a "after filter" for example) has to set the following parameter: 

 bridge[ :record_incoming ]  = ""

Recording the outgoing call leg

To record the outgoing call leg, the routing script (in a "after filter" for example) has to set the following parameter, per route (the decision to record or not, or the file name to record to, can be set per matching route): 

 # Need to clone the routes in order to have the right to modify them
 routes = clone_routes params[:routes]
 routes.each do |route|
   route[ :record_outgoing ]  = ""
 end
 # Store modified routes back to the parameters for this outgoing call
 params[:routes] = routes

Record the outgoing call leg within incoming leg's recorded file (mixing) 

 [...]
   route[ :record_outgoing ]  = "@{MixWithIncoming}"
 [...]

Choosing file path to record to

The value assigned to ":record_incoming" or ":record_outgoing" is the path to record the file to.

The paths can be absolute, or relative. When relative, they are relative to the "tbstreamserver" application working directory, for example: 

 /lib/tb/toolpack/setup/12358/2.7/apps/tbstreamserver/
  • Empty file name will default to a name that contains various information about the call:
    • LinkId: Id common between all legs of this call bridge
    • LegId: Unique Id for this leg
    • Nap: Current NAP name this call leg is from
    • Direction: "IN" or "OUT" (depends if call leg is incoming or outgoing leg)
    • Calling: The calling number of this call leg
    • Called: The called number of this call leg
    • Protocol: The signaling protocol of this call leg (SS7, ISDN, CAS, SIP)
    • Media info: Codec + IP/Port for SIP calls, Trunk/Timeslot for TDM calls
  • To record outgoing call leg in the same audio file as incoming call leg (mixing), use the following:
    • @{MixWithIncoming}: Record outgoing legs in same file as incoming legs
  • Variables can be used to insert in the recording path information that's not already available from routing scripts:
    • @{CURRENT_PKG}: Version of current package
      • Example: 2.6.45
    • @{DATE format}: Prints the date, where 'format' is expressed as described for the 'strftime' function
      • Example: @{DATE %Y-%m-%d} => 2013-01-28
    • @{DefaultName}: Replaced by the default file name for recording, which contains:
      • LinkId: Id common between all legs of this call bridge
      • LegId: Unique Id for this leg
      • Nap: Current NAP name this call leg is from
      • Direction: "IN" or "OUT" (depends if call leg is incoming or outgoing leg)
      • Calling: Calling number
      • Called: Called number
      • Protocol: Protocol type of this call (SS7, ISDN, CASR2, SIP)
      • Media info: Codec + IP/Port for SIP calls, Trunk/Timeslot for TDM calls
      • Example: "73EBA698-F3D67B4B-NAP_SS7-IN-5550000-5550001-SS7-TRUNK_BELL_11-24.wav"
      • Example: "73EBA698-73EBA698-NAP_SIP-OUT-5550000-5550001-SIP-G723-10.3.10.101-1050.wav"
    • @{DefaultPath}: Default recording folder and file name: "@{RECORD_PATH}/@{DATE %Y-%m-%d}/@{DefaultName}"
      • Example: "/lib/tb/toolpack/setup/12358/recorded_calls/73EBA698-F3D67B4B-NAP_SS7-IN-5550000-5550001-SS7-TRUNK_BELL_11-24.wav"
      • Example: "/lib/tb/toolpack/setup/12358/recorded_calls/73EBA698-73EBA698-NAP_SIP-OUT-5550000-5550001-SIP-G723-10.3.10.101-1050.wav"
    • @{Direction}: Direction of current leg (IN our OUT)
      • Example: IN
    • @{LegId}: Current LegId (Unique Id for this leg)
      • Example: F3D67B4B
    • @{LinkId}: Current LinkId (Id common between all legs of this call bridge)
      • Example: 73EBA698
    • @{PKG_HOME}: Path where packages are stored.
      • Note: It's not recomended to use that path on redundant systems, package file replication may cause confusion in recorded files.
      • Example: /lib/tb/toolpack/pkg
    • @{PROMPT_PATH}: Default path where audio prompts are stored
      • Note: It's not recomended to use that path on redundant systems, package file replication may cause confusion in recorded files.
      • Example: /lib/tb/toolpack/pkg/prompts
    • @{Protocol}: Protocol of current leg
      • Example: SS7
    • @{RECORD_PATH}: Default recording folder: "@{TB_SETUP_HOME}/recorded_calls/"
    • @{TBX_GW_PORT}: Current "System Id" (also called "Gateway Port")
      • Example: 12358
    • And all variables listed here: Building play or record file path

Controlling UUI (user-to-user information) relay

UUI (user-to-user information) can be present in different messages received by either call leg during a call. For example, information can be carried during the initial invite, other information can be carried when the call is alerted, answered, or terminated.

Routing scripts can control if the UUI received from one leg through the call will be forwarded or not to the other call leg:

  • uui_forward_enabled

Routing scripts can also read and modify the UUI received with the incoming call leg, before it gets forwarded upon creation of the outgoing call leg:

  • uui

UUI (user-to-user indication) values

Byte array represented as ruby String. Use bridge=params[:bridge], then bridge[:uui] to access the data.

To access the bytes in Ruby, use ruby String operator []. For example: bridge[:uui][0] will return the binary value of the first UUI byte.

Function each_byte can also be useful to iterate through all bytes of the UUI.

uui_forward_enabled values

Controls forwarding or discarding of UUI to outgoing call leg.

Values for this parameter are "0", "1", "false" or "true.

  • 0/false: UUI is not forwarded between call legs
  • 1/true: UUI is forwarded between call legs

The value for this parameter at input of routing script depends on the "Forward UUI" parameter in the "Advanced" section of the Gateway configuration page of the Web Portal. The script may change this value to override the Gateway configuration.

Authorization

Starting with release 2.7, it is possible to issue RADIUS authorization requests from routing scripts. To do so, the params[:authorization] object must be filled with the required RADIUS attributes and an exception must be raised with reason :authorization_required.

When the authorization is completed, the routing script is called again with the result. The params[:authorization] object will be filled with the RADIUS attributes from the response. The params[:authorization][:result] field will also contain a string indicating the result of the authorization:

  • accept: The authorization was successful.
  • reject: The authorization was refused.
  • challenge: The authorization was challenged.
  • timeout: The authorization was not answered.

Call diversion options

It's possible to control the call flow when a call diversion information is received in the alerting state.

Two fields are available: bridge[ :diversion ] and bridge[ :diversion_reason ]

The internal release cause TOOLPACK_DIVERT_NOT_ALLOWED is used by gateway application to terminate both legs. 

 bridge[ :diversion ] = :allowed

The alert message will not be analyzed and the call will be progressed. Default behavior. 

 bridge[ :diversion ] = :not_allowed

If the alert message indicates that the call is diverted, the call will be released no matter the In-band information to allow early media. 

 bridge[ :diversion ] = :not_allowed_w_early_media

The call will be released If the alert message indicates that the call is diverted with in-band information to allow early media. 

 bridge[ :diversion_reason ] = "*"

If the diversion is not allowed, the gateway will drop the call for any redirecting reason. 

 bridge[ :diversion_reason ] = []
 bridge[ :diversion_reason ] << :busy
 bridge[ :diversion_reason ] << :unconditional

If the diversion is not allowed, the redirecting reason will be analyzed and the call will only be dropped for the configured cases.

See section Redirecting number reason values.


Call transfer requests

Toolpack allows to relay Call transfer requests from one leg to the other, or to process them locally (making another outgoing call to replace the call that requested the call transfer).

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)
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 field are 'read-only'. They won't be included in the outgoing call, as they represent the content of the call transfer request, not the outgoing call to make.

To insert/modify attributes of the outgoing call, the parameters from params[ :call ] must be edited instead.

Redirection

In release 2.8 and above, redirection contacts are obtained from the routing engine in the following format: 

 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:

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

All the status fields of the NAPs are provided for use by the routing scripts. 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
 --------------------------------------------------------------------------------------------
 "signaling_type"                      Signaling type (SS7, ISDN, CASR2, SIP)
 "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.
 "asr_statistics_struct"               Detailed Answer-Seizure Rate Statistics.
 "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.
 "availability_detection_struct"       Detailed availibility detection Statistics
 "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
 "registration_struct"                 Detailed registration Statistics
 "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


If the nap status is part of a substructure, it's name in the routing scripts must be composed of the structure name appended by an underscore and the field name.

For example the name to use for the global ASR percentage is: 

 asr_statistics_struct_global_asr_percent


It is also possible to add dynamic nap attributes in the web portal. These can be referenced by their name.

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 passed to the routing script. This is equivalent to the incoming call parameters.


@nap_list

A list of 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"
    }
 }



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