TBwiki - User contributions [en]

Toolpack:RADIUS CDR attributes C

2013-10-29T19:49:02Z

Alexandre Tremblay: Added call-id, removed TelcoB-SipCallId

{{DISPLAYTITLE:Toolpack:RADIUS CDR attributes}}
=== '''''Applies to version(s): v2.7''''' ===
This page contains the list of RADIUS attributes that are sent to a RADIUS accounting server.

From RFC 2865 and RFC 2866 (Accounting) :
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack CDR parameter
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 1 
| User-Name
| align="center" | string 
| IncomingNAP
| Incoming NAP name
|-
| align="center" | 4 
| NAS-IP-Address
| align="center" | IP address 
| -
| IP address of the TMedia generating the CDR record
|-
| align="center" | 30 
| Called-Sation-Id
| align="center" | string 
| Called Number
| Called party number
|-
| align="center" | 31 
| Calling-Station-Id
| align="center" | string 
| Calling Number
| Calling party number
|-
| align="center" | 32 
| NAS-Identifier
| align="center" | string 
| Application Name
| Application name of the CDR provider
|-
| align="center" | 40 
| Acct-Status-Type
| align="center" | integer 
| -
| Start or Stop
|-
| align="center" | 44 
| Acct-Session-Id
| align="center" | integer 
| Leg Id
| Call Leg Identifier
|-
| align="center" | 46 
| Acct-Session-Time
| align="center" | integer 
| Session duration
| Duration of the call, in seconds. Corresponds to h323-disconnect-time minus h323-connect-time.
|}

Attribute ID 26, Vendor ID: 9 (Cisco)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 2 
| Cisco-NAS-Port
| align="center" | string 
| NAP name
| Network Access Point name for the call leg
|-
| align="center" | 24 
| h323-conf-id
| align="center" | string 
| Unique Id
| Unique call identifier for the two initial legs (incoming and outgoing) - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 25 
| h323-setup-time
| align="center" | string 
| Start Time
| Represent the call leg setup time - Coordinated Universal Time (UTC)
|-
| align="center" | 26 
| h323-call-origin
| align="center" | string 
| Originator Name
| "originate" for an outgoing leg - "answer" for an incoming leg
|-
| align="center" | 27 
| h323-call-type
| align="center" | string 
| Protocol Type
| If protocol is SIP the value is "VOIP", otherwise it is "Telephony"
|-
| align="center" | 28 
| h323-connect-time
| align="center" | string 
| Connected Time
| Represent the call leg answer time (connect time) - Coordinated Universal Time (UTC)
|-
| align="center" | 29 
| h323-disconnect-time
| align="center" | string 
| EndTime
| Represent the call leg disconnect time - Coordinated Universal Time (UTC)
|-
| align="center" | 30 
| h323-disconnect-cause
| align="center" | string 
| Termination Reason
| Q.931 disconnect (1 to 160) cause, TB Toolpack system cause (200 to 300) and SIP cause (400 to 600)
|-
| align="center" | 35 
| h323-incoming-conf-id
| align="center" | string 
| Unique Id
| Contains the original h323-conf-id in case of call transfer for subsequent outgoing legs - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 115
| release-source
| align="center" | string 
| Termination Source
| "localLeg" if this leg terminate the call or "connectedLeg" if its the connected leg - We use a Cisco string field with our own value definition
|-
| align="center" | 141 
| call-id
| align="center" | String 
| SipCallId
| Content of the "call-id" SIP header
|}

Attribute ID 26, Vendor ID: 21776 (TelcoBridges)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 9 
| Telcob-ChargeIndicator
| align="center" | String 
| ChargeIndicator
| Represent the charge indicator value
|-
| align="center" | 10 
| Telcob-Protocol
| align="center" | String 
| Protocol
| Protocol used for this LegID. Ex. SIP, ISDN, etc.
|-
| align="center" | 11 
| Telcob-Codec
| align="center" | String 
| Codec
| Codec used for this LegID
|-
| align="center" | 12 
| Telcob-RemoteMediaIP
| align="center" | IP address 
| RemoteIP
| RemoteIP used for the media. This is only for VOIP.
|-
| align="center" | 13 
| Telcob-RemoteMediaPort
| align="center" | Integer 
| RemotePort
| RTP Port of the remote peer. This is only for VOIP.
|-
| align="center" | 14 
| Telcob-TrunkName
| align="center" | String 
| TrunkName
| Name of the trunk (ex. "LS004400E1_0_00"). 
This attribute is only for non-VOIP.
|-
| align="center" | 15 
| Telcob-TimeslotNumber
| align="center" | Integer 
| TimeslotNumber
| This is the Timeslot Number. 
This attribute is only for non-VOIP.
|-
| align="center" | 16 
| Telcob-MediaInfo
| align="center" | String 
| MediaInfo
| String that represents the MediaInfo (ex. "PCMU@10.3.10.124:20054" for SIP and "LS_T1_BITS_00:1" for ISDN)
|-
| align="center" | 17 
| Telcob-StartTime
| align="center" | String 
| StartTime
| String that represents the StartTime of the call.
|-
| align="center" | 18 
| Telcob-ConnectedTime
| align="center" | String 
| ConnectedTime
| String that represents the time when the call was connected
|-
| align="center" | 19 
| Telcob-EndTime
| align="center" | String 
| EndTime
| String that represents when the call was completed.
|-
| align="center" | 20 
| Telcob-TerminationCause
| align="center" | Integer 
| TerminationCause
| Integer that represents the Termination Cause
|-
| align="center" | 21 
| Telcob-Other-Leg-Id
| align="center" | Integer 
| Other-Leg-Id
| Integer that represents the other LegID bridged for this call
|-
| align="center" | 22 
| Telcob-TerminationCauseString
| align="center" | String 
| TerminationCauseString
| String that represents the TerminationCause, but in a String format.
|-
| align="center" | 23 
| Telcob-TerminationSource
| align="center" | String 
| TerminationSource
| String that represents the Termination Source
|-
| align="center" | 24 
| Telcob-LocalSipIP*
| align="center" | IPAddr 
| LocalSipIP
| IP Address that represents the Local IP used for SIP.
|-
| align="center" | 25 
| Telcob-LocalSipPort*
| align="center" | Integer 
| LocalSipPort
| Integer that represents the Local port used for SIP.
|-
| align="center" | 26 
| Telcob-LocalMediaIP
| align="center" | IPAddr 
| LocalMediaIP
| IP Address that represents the Local IP used for the media.
|-
| align="center" | 27 
| Telcob-LocalMediaPort
| align="center" | Integer 
| LocalMediaPort
| Integer that represents the Local port used for the media.
|-
| align="center" | 28 
| Telcob-LocalMediaInfo
| align="center" | String 
| LocalMediaInfo
| String that represents the Local Media Info.(ex. "PCMU@10.3.10.124:20054")
Only for VOIP calls.
|-
| align="center" | 29 
| Telcob-RemoteMediaInfo
| align="center" | String 
| RemoteMediaInfo
| String that represents the Remote Media Info.(ex. "PCMU@10.3.10.124:20054")
This is the same as Telcob-MediaInfo. Only for VOIP calls.
|-
| align="center" | 30 
| Telcob-Alert-Time
| align="center" | String 
| AlertTime
| Time where the call has started ringing.
|-
| align="center" | 31 
| Telcob-Redirecting-Number
| align="center" | String 
| RedirectingNumber
| Redirecting number
|-
| align="center" | 32 
| Telcob-Original-Called-Number
| align="center" | String 
| OriginalCalledNumber
| Original called number
|-
| align="center" | 33 
| Telcob-Calling-Presentation
| align="center" | String 
| CallingPresentation
| Calling presentation: "Unspecified", "NotAvailable", "Allowed", "Restricted", "AddressRestricted" or "NameRestricted"
|-
| align="center" | 34 
| Telcob-CallingSubscriberNumber
| align="center" | String 
| CallingSubscriberNumber
| Second calling number (ISDN) or Generic number of type additional calling party number (SS7)
|}
* = Not implemented.

Routing script tutorial:Mini Development Guide

2013-10-15T14:39:07Z

Alexandre Tremblay: Added leg_id and session_id

== Call object ==

=== Get ===

Those function are 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:

{| cellspacing="1" cellpadding="1" border="1" style="width: 921px; height: 805px;"
|-
! scope="col" | '''Script parameter name'''
! scope="col" | '''ISDN '''
! scope="col" | '''R2 CAS''' 
! scope="col" | '''SS7 '''
! scope="col" | '''SIP '''
! scope="col" | '''Comment '''
! scope="col" | '''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 
| N/A 
| Q763: 'Display information' 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 
| 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 
| N/A 
| 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_screening 
| Q931: 'Redirecting number' 1st IE - Screening indicator 
| N/A 
| Q763: 'Redirecting number' IE - screening (national) 
| SIP:Diversion (2nd header) - diversion-screen 
| 
| 
|-
| 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 
| 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_screening 
| Q931: 'Redirecting number' 2nd IE - Screening indicator 
| N/A 
| Q763: 'Redirection number' IE - screening (national) 
| SIP:Diversion (1st header) - diversion-screen 
| 
| 
|-
| 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 
| 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 
| Outgoing leg (write) only 
| 
|-
| nap 
| 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 
|}

 

'''Notice:''' All values are documented in the '''noa_npi_remap.rb''' script and may change between major release.

=== Noa values ===

*<tt>unknown_number</tt>
*<tt>international_number</tt>
*<tt>national_number</tt>
*<tt>subscriber_number</tt>
*<tt>network_specific</tt>
*<tt>network_routing_national_format</tt>
*<tt>network_routing_international_format</tt>
*<tt>abbreviated_number</tt>
*<tt>subscriber_number_operator_requested</tt>
*<tt>national_number_operator_requested</tt>
*<tt>international_number_operator_requested</tt>
*<tt>no_number_present_operator_requested</tt>
*<tt>no_number_present_cut_through_call_to_carrier</tt>
*<tt>test_line_test_code</tt>
*<tt>non_unique_subscriber_number</tt>
*<tt>non_unique_national_number</tt>
*<tt>non_unique_international_number</tt>
*<tt>call_950_number</tt>

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

*<tt>call_params[:called_noa] = 0x70</tt>

or

*<tt>call_params[:called_noa] = 112</tt>

=== Npi values ===

*<tt>unknown_number</tt>
*<tt>isdn</tt>
*<tt>telephony</tt>
*<tt>private</tt>
*<tt>data</tt>
*<tt>telex</tt>
*<tt>national</tt>

=== Calling Display Type values ===

*<tt>unspecified</tt> => Type is unspecified.
*<tt>calling_party_name</tt> => Type is 0xB1.

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

*<tt>call_params[:calling_display_type] = 0xB1</tt>

or

*<tt>call_params[:calling_display_type] = 177</tt>

=== Calling Display value ===

*<tt>call_params[:calling_display] = "Roger Fluffy"</tt>

=== Presentation values ===

*<tt>unspecified</tt>
*<tt>not_available</tt>
*<tt>allowed</tt>
*<tt>restricted</tt>
*<tt>addr_restricted</tt>
*<tt>name_restricted</tt>

=== Calling Party Category ===
values for calling_category
*<tt>unspecified</tt> <tt>(0xa)</tt>
*<tt>unknown</tt> <tt>(0x0)</tt>
*<tt>operator_french</tt> <tt>(0x1)</tt>
*<tt>operator_english</tt> <tt>(0x2)</tt>
*<tt>operator_german</tt> <tt>(0x3)</tt>
*<tt>operator_russian</tt> <tt>(0x4)</tt>
*<tt>operator_spanish</tt> <tt>(0x5)</tt>
*<tt>subscriber</tt> <tt>(0xa)</tt>
*<tt>subscriber_with_priority</tt> <tt>(0xb)</tt>
*<tt>data</tt> <tt>(0xc)</tt>
*<tt>test</tt> <tt>(0xd)</tt>
*<tt>payphone</tt> <tt>(0xf)</tt>

=== Screening values ===

*<tt>unspecified</tt>
*<tt>no</tt>
*<tt>pass</tt>
*<tt>fail</tt>
*<tt>network_provided</tt>

=== Redirecting indicator values ===

SS7:

*<tt>no_redirection</tt>
*<tt>call_rerouted</tt>
*<tt>call_rerouted_all_restricted</tt>
*<tt>call_diverted</tt>
*<tt>call_diverted_all_restricted</tt>
*<tt>call_rerouted_restricted</tt>
*<tt>call_diverted_restricted</tt>
*<tt>spare</tt>

=== Redirecting number reason values ===

ISDN:

*<tt>unknown</tt>
*<tt>busy</tt>
*<tt>no_reply</tt>
*<tt>deflection</tt>
*<tt>dte_out_of_order</tt>
*<tt>forwarding_by_called_dte</tt>
*<tt>unconditional</tt>

SS7:

*<tt>unknown</tt>
*<tt>busy</tt>
*<tt>no_reply</tt>
*<tt>unconditional</tt>
*<tt>deflection</tt>
*<tt>deflection_immediate</tt>
*<tt>mobile_not_reachable</tt>

=== OLI (originating line information) values ===

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

 

=== 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.

 

=== request_uri ===

Enables access to the Request-Line URI. 

For example, if the Request-Line is:
<pre>Request-Line: INVITE sip:4175162082@172.22.45.13:5060;user=phone;transport=udp SIP/2.0</pre>
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: 
<pre> if call_params[:request_uri] && call_params[:request_uri] =~ /sip:(.*)@.*/
call_params[:called] = $1
end
</pre>
=== 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 ===
Any custom sip header can be added to an outgoing call leg: 
<pre>call[ :sip_header ] = ["P-my-custom-header:any value"]</pre>

Many can be added at one time: 
<pre>call[ :sip_header ] = ["P-my-custom-header:value1", "P-my-custom-header2:value2", "P-my-custom-header3:value3"]</pre>

Note: It is not possible to read custom sip headers from inbound call legs. 

== Route parameters ==

All route may have these parameters:

*calling
*called
*nap
*remapped_calling
*remapped_called
*remapped_nap
*remapped_profile

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 or wav files in different states of the call flow, like playing an announcement when the caller dials in. You can also manage the User to User information element.

*announcement_tone
*call_progress_tone
*ring_tone
*max_call_duration
*call_duration_reason
*disconnect_tone
*uui
*uui_forward_enabled

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, :call_progress_tone, :ring_tone, :disconnect_tone) inside bridge parameters are using this format.
<pre>"file1.wav:repeat:start_off:end_off,file2.wav:repeat:start_off:end_off,file3.wav:repeat:start_off:end_off"
</pre>
optional repeat parameter: number of times to play the file (0 and 1 have the same result) optional start_off parameter : Start offset in milliseconds optional end_off parameter: End offset in milliseconds Example: "file1.wav,file2.wav:-1" will play file1.wav one time and then play file2.wav in loop. Example: "file1.wav:0:1000:3000,file2.wav:2:5000:10000" will play file1.wav from second 1 to second 3 then file2.wav from second 5 to second 10 two times. Example: "file1.wav:0:0:30000" will play file1.wav one time for a max duration of 30 seconds.

=== announcement_tone ===

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

Prompt played on the incoming leg with the stream_server before any outgoing call is placed. The outgoing call occurs when the wav file is completed or when the "profile:Busy Tone max duration" is reached.

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

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

=== call_progress_tone ===

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

Prompt played after "announcement_tone" when reason is not "ok" or when outgoing call disconnect before answer state. "none" value is used to bypass "profile:Generate Busy (Congestion) Tone" and to play no audio prompt.

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

Answer the call before playing the call progress tone. Default if argument not provided is "no", in which case call is only alerted with in-band media.

=== ring_tone ===

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

Prompt played between alerting state and answer state. Bypass any other ring back tone (RBT) configured in profile.

=== max_call_duration ===

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

Maximum call duration in millisecond for the current bridge. 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.

=== disconnect_tone ===

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

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

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

Answer the call before playing the disconnect tone. Default if argument not provided is "no", in which case call is only alerted with in-band media.

=== 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 to outgoing call leg 1/true: UUI is forwarded to outgoing call leg 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.

=== Managing audio prompts through Web Portal ===
Audio prompts can be uploaded or deleted from the TMedia unit through the Web Portal:
[[Toolpack:Configuring_Audio_Prompts_A|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.

Provided paths are always relative to the "current working directory" of the tbstreamserver application:
/lib/tb/toolpack/setup/12358/2.7/apps/tbstreamserver/
(Where "2.7" may be replaced by the current major version of your system)

Absolute paths can also be provided.

== 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:
** @{DefaultName}: Same file name (without parent folder path) as the one used when empty string is provided
*** 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}: Same path as the one used when empty string is provided
*** Example: "recorded_calls/73EBA698-F3D67B4B-NAP_SS7-IN-5550000-5550001-SS7-TRUNK_BELL_11-24.wav"
*** Example: "recorded_calls/73EBA698-73EBA698-NAP_SIP-OUT-5550000-5550001-SIP-G723-10.3.10.101-1050.wav"
** @{LinkId}: Current LinkId (Id common between all legs of this call bridge)
*** Example: "73EBA698"
** @{LegId}: Current LegId (Unique Id for this leg)
*** Example: "F3D67B4B"
** @{Protocol}: Protocol type of current leg
*** Example: "SS7"
** @{Direction}: Direction of current leg (IN our OUT)
*** Example: "IN"
** @{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"
** @{CURRENT_PKG}: Version of current package
*** Example: "2.6.45"
** @{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"
** @{TBX_GW_PORT}: Current "System Id" (also called "Gateway Port")
*** Example: "12358"

== 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 [[#Refuse|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 [[Routing_script_tutorial:Mini_Development_Guide#Redirecting_number_reason_values|Redirecting number reason values]].

== Reason values ==

Check here for Termination Reason Cause codes: 

[[Termination_cause_codes|Termination Reason Cause codes]]
 

== 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.

#define NAP_STATS_FIELDS \
/* Field, szName, szDescription, szOptions */ \
( SIGNALING_TYPE, "signaling_type", "Signaling type.", "" ) \
( INCOMING_CALL_CNT, "inst_incoming_call_cnt", "Instantaneous Count of incoming calls.", "" ) \
( OUTGOING_CALL_CNT, "inst_outgoing_call_cnt", "Instantaneous Count of outgoing calls.", "" ) \
( AVAILABLE_CNT, "available_cnt", "Number of available circuits or channels.", "" ) \
( UNAVAILABLE_CNT, "unavailable_cnt", "Number of unavailable circuits or channels.", "" ) \
( AVAILABILITY_PCT, "availability_percent", "Percentage of available circuits or channels.", "" ) \
( USAGE_PCT, "usage_percent", "Percentage of used circuits or channels.", "" ) \
( TOTAL_INCOMING_CALL_CNT, "total_incoming_call_cnt", "Total Count of incoming calls.", "" ) \
( ASR_STRUCT, "asr_statistics_struct", "Detailed Answer-Seizure Rate Statistics.", "" ) \
( GLOBAL_ASR_PCT, "global_asr_percent", "Global calculated ASR percentage.", "" ) \
( TOTAL_OUTGOING_CALL_CNT, "total_outgoing_call_cnt", "Total Count of outgoing calls.", "" ) \
( LAST_24H_ASR_PCT, "last_24h_asr_percent", "Last 24 hours calculated ASR percentage.", "" ) \
( LAST_24H_OUTGOING_CALL_CNT, "last_24h_outgoing_call_cnt", "Last 24 hours outgoing calls.", "" ) \
( HOUR_ASR_PCT, "current_hour_asr_percent", "Current hour calculated ASR percentage.", "" ) \
( HOUR_OUTGOING_CALL_CNT, "current_hour_outgoing_call_cnt", "Current hour outgoing calls.", "" ) \
( LAST_HOUR_ASR_PCT, "last_hour_asr_percent", "Last hour calculated ASR percentage.", "" ) \
( LAST_HOUR_OUTGOING_CALL_CNT, "last_hour_outgoing_call_cnt", "Last hour outgoing calls.", "" ) \
( AVAILABILITY_DETECTION_STRUCT, "availability_detection_struct", "Detailed availibility detection Statistics", "" ) \
( POLL_REMOTE_PROXY, "poll_remote_proxy", "Remote proxy polling enabled", "" ) \
( IS_AVAILABLE, "is_available", "Remote proxy actually available or not", "" ) \
( TIME_SINCE_POLLING, "time_since_polling", "Time since the last availibility polling", "" ) \
( TIME_AVAILABLE_SECONDS, "time_available_seconds", "Number of seconds since the NAP is available", "x" ) \
( TIME_UNAVAILABLE_SECONDS, "time_unavailable_seconds", "Number of seconds since the NAP is unavailable", "x" ) \
( REGISTRATION_STRUCT, "registration_struct", "Detailed registration Statistics", "" ) \
( REGISTER_TO_PROXY, "register_to_proxy", "Register to proxy enabled", "" ) \
( IS_REGISTERED, "registered", "Actually registered or not", "" ) \
( TIME_SINCE_REFRESH, "time_since_refresh", "Time since the last refresh", "" ) \
( TIME_REGISTERED_SECONDS, "time_registered_seconds", "Number of seconds since the NAP is registered", "x" ) \
( TIME_NOT_REGISTERED_SECONDS, "time_not_registered_seconds", "Number of seconds since the NAP is not registered", "x" ) \
/*!< Nap Status Fields */

 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.

== 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.

Example: @params = {:bridge => {:announcement_tone, "announcement.wav"}}

 

----

 Back to [[Routing script tutorial|Routing Script Tutorial]].

[[category:Needs revising]]

Toolpack:RADIUS CDR attributes C

2013-09-17T20:31:45Z

Alexandre Tremblay: Fixed attributes

{{DISPLAYTITLE:Toolpack:RADIUS CDR attributes}}
=== '''''Applies to version(s): v2.7''''' ===
This page contains the list of RADIUS attributes that are sent to a RADIUS accounting server.

From RFC 2865 and RFC 2866 (Accounting) :
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack CDR parameter
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 1 
| User-Name
| align="center" | string 
| IncomingNAP
| Incoming NAP name
|-
| align="center" | 4 
| NAS-IP-Address
| align="center" | IP address 
| -
| IP address of the TMedia generating the CDR record
|-
| align="center" | 30 
| Called-Sation-Id
| align="center" | string 
| Called Number
| Called party number
|-
| align="center" | 31 
| Calling-Station-Id
| align="center" | string 
| Calling Number
| Calling party number
|-
| align="center" | 32 
| NAS-Identifier
| align="center" | string 
| Application Name
| Application name of the CDR provider
|-
| align="center" | 40 
| Acct-Status-Type
| align="center" | integer 
| -
| Start or Stop
|-
| align="center" | 44 
| Acct-Session-Id
| align="center" | integer 
| Leg Id
| Call Leg Identifier
|-
| align="center" | 46 
| Acct-Session-Time
| align="center" | integer 
| Session duration
| Duration of the call, in seconds. Corresponds to h323-disconnect-time minus h323-connect-time.
|}

Attribute ID 26, Vendor ID: 9 (Cisco)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 2 
| Cisco-NAS-Port
| align="center" | string 
| NAP name
| Network Access Point name for the call leg
|-
| align="center" | 24 
| h323-conf-id
| align="center" | string 
| Unique Id
| Unique call identifier for the two initial legs (incoming and outgoing) - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 25 
| h323-setup-time
| align="center" | string 
| Start Time
| Represent the call leg setup time - Coordinated Universal Time (UTC)
|-
| align="center" | 26 
| h323-call-origin
| align="center" | string 
| Originator Name
| "originate" for an outgoing leg - "answer" for an incoming leg
|-
| align="center" | 27 
| h323-call-type
| align="center" | string 
| Protocol Type
| If protocol is SIP the value is "VOIP", otherwise it is "Telephony"
|-
| align="center" | 28 
| h323-connect-time
| align="center" | string 
| Connected Time
| Represent the call leg answer time (connect time) - Coordinated Universal Time (UTC)
|-
| align="center" | 29 
| h323-disconnect-time
| align="center" | string 
| EndTime
| Represent the call leg disconnect time - Coordinated Universal Time (UTC)
|-
| align="center" | 30 
| h323-disconnect-cause
| align="center" | string 
| Termination Reason
| Q.931 disconnect (1 to 160) cause, TB Toolpack system cause (200 to 300) and SIP cause (400 to 600)
|-
| align="center" | 35 
| h323-incoming-conf-id
| align="center" | string 
| Unique Id
| Contains the original h323-conf-id in case of call transfer for subsequent outgoing legs - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 115
| release-source
| align="center" | string 
| Termination Source
| "localLeg" if this leg terminate the call or "connectedLeg" if its the connected leg - We use a Cisco string field with our own value definition
|-
|}

Attribute ID 26, Vendor ID: 21776 (TelcoBridges)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 9 
| Telcob-ChargeIndicator
| align="center" | String 
| ChargeIndicator
| Represent the charge indicator value
|-
| align="center" | 10 
| Telcob-Protocol
| align="center" | String 
| Protocol
| Protocol used for this LegID. Ex. SIP, ISDN, etc.
|-
| align="center" | 11 
| Telcob-Codec
| align="center" | String 
| Codec
| Codec used for this LegID
|-
| align="center" | 12 
| Telcob-RemoteMediaIP
| align="center" | IP address 
| RemoteIP
| RemoteIP used for the media. This is only for VOIP.
|-
| align="center" | 13 
| Telcob-RemoteMediaPort
| align="center" | Integer 
| RemotePort
| RTP Port of the remote peer. This is only for VOIP.
|-
| align="center" | 14 
| Telcob-TrunkName
| align="center" | String 
| TrunkName
| Name of the trunk (ex. "LS004400E1_0_00"). 
This attribute is only for non-VOIP.
|-
| align="center" | 15 
| Telcob-TimeslotNumber
| align="center" | Integer 
| TimeslotNumber
| This is the Timeslot Number. 
This attribute is only for non-VOIP.
|-
| align="center" | 16 
| Telcob-MediaInfo
| align="center" | String 
| MediaInfo
| String that represents the MediaInfo (ex. "PCMU@10.3.10.124:20054" for SIP and "LS_T1_BITS_00:1" for ISDN)
|-
| align="center" | 17 
| Telcob-StartTime
| align="center" | String 
| StartTime
| String that represents the StartTime of the call.
|-
| align="center" | 18 
| Telcob-ConnectedTime
| align="center" | String 
| ConnectedTime
| String that represents the time when the call was connected
|-
| align="center" | 19 
| Telcob-EndTime
| align="center" | String 
| EndTime
| String that represents when the call was completed.
|-
| align="center" | 20 
| Telcob-TerminationCause
| align="center" | Integer 
| TerminationCause
| Integer that represents the Termination Cause
|-
| align="center" | 21 
| Telcob-Other-Leg-Id
| align="center" | Integer 
| Other-Leg-Id
| Integer that represents the other LegID bridged for this call
|-
| align="center" | 22 
| Telcob-TerminationCauseString
| align="center" | String 
| TerminationCauseString
| String that represents the TerminationCause, but in a String format.
|-
| align="center" | 23 
| Telcob-TerminationSource
| align="center" | String 
| TerminationSource
| String that represents the Termination Source
|-
| align="center" | 24 
| Telcob-LocalSipIP*
| align="center" | IPAddr 
| LocalSipIP
| IP Address that represents the Local IP used for SIP.
|-
| align="center" | 25 
| Telcob-LocalSipPort*
| align="center" | Integer 
| LocalSipPort
| Integer that represents the Local port used for SIP.
|-
| align="center" | 26 
| Telcob-LocalMediaIP
| align="center" | IPAddr 
| LocalMediaIP
| IP Address that represents the Local IP used for the media.
|-
| align="center" | 27 
| Telcob-LocalMediaPort
| align="center" | Integer 
| LocalMediaPort
| Integer that represents the Local port used for the media.
|-
| align="center" | 28 
| Telcob-LocalMediaInfo
| align="center" | String 
| LocalMediaInfo
| String that represents the Local Media Info.(ex. "PCMU@10.3.10.124:20054")
Only for VOIP calls.
|-
| align="center" | 29 
| Telcob-RemoteMediaInfo
| align="center" | String 
| RemoteMediaInfo
| String that represents the Remote Media Info.(ex. "PCMU@10.3.10.124:20054")
This is the same as Telcob-MediaInfo. Only for VOIP calls.
|-
| align="center" | 30 
| Telcob-Alert-Time
| align="center" | String 
| AlertTime
| Time where the call has started ringing.
|-
| align="center" | 31 
| Telcob-Redirecting-Number
| align="center" | String 
| RedirectingNumber
| Redirecting number
|-
| align="center" | 32 
| Telcob-Original-Called-Number
| align="center" | String 
| OriginalCalledNumber
| Original called number
|-
| align="center" | 33 
| Telcob-Calling-Presentation
| align="center" | String 
| CallingPresentation
| Calling presentation: "Unspecified", "NotAvailable", "Allowed", "Restricted", "AddressRestricted" or "NameRestricted"
|-
| align="center" | 34 
| Telcob-CallingSubscriberNumber
| align="center" | String 
| CallingSubscriberNumber
| Second calling number (ISDN) or Generic number of type additional calling party number (SS7)
|}
* = Not implemented.

Toolpack:Enabling VoIP Bonding A

2013-07-08T19:14:13Z

Alexandre Tremblay: Fixed layout

=== '''''Applies to version(s): v2.7.''''' ===
{{DISPLAYTITLE:Enabling VoIP Bonding}}

1- Access the telecom baseboard of the Tmedia unit:
:a. For TMG800 and TMG3200, access the unit using SSH, then connect to the telecom baseboard using telnet.
telnet 172.31.1.1
:b. For other platforms, such as TMP800, TMP3200 and TMG7800, connect to the telecom baseboard directly using telnet and the eth0 IP address.
telnet 10.3.2.21

2- In the baseboard prompt, use the set_voip0bonding shell command to enable VoIP Bonding.
set_voip0bonding <0|1> [IP_To_Ping] [Ping_Timeout] [Retries]
:Where:
::<code><0|1></code> selects if the feature is enabled (1 to enable). When the feature is enabled, the Ethernet PHY states are polled periodically to discover which port is available.
::<code>[IP_To_Ping]</code> is the IP address to send ICMP echo requests to verify if a logical path is available. When no response is received within the requested timeout after the specified number of retries, the port is considered temporarly unusable and the other VoIP port is used instead.
::<code>[Ping_Timeout]</code> is the ICMP echo request timeout.
::<code>[Retries]</code> is the number of ICMP echo request retries to send before doing a switchover.
:Examples:
::<code>set_voip0bonding 1</code>
::<code>set_voip0bonding 1 10.3.10.187 1000 1</code>
::<code>set_voip0bonding 1 123.45.67.89 2500 3</code>

3- Reboot the unit (note: this causes downtime).
reboot

4- (optional) After the reboot, access the telecom baseboard again and verify that the VoIP Bonding was successfully enabled:
print_bonding

Toolpack:Enabling VoIP Bonding A

2013-07-08T17:32:55Z

Alexandre Tremblay: Changed instructions for connecting to the telecom baseboard

=== '''''Applies to version(s): v2.7.''''' ===
{{DISPLAYTITLE:Enabling VoIP Bonding}}

1- Access the telecom baseboard of the Tmedia unit:
:a. For TMG800 and TMG3200, access the unit using SSH, then connect to the telecom baseboard using telnet.
telnet 172.31.1.1
:b. For other platforms, such as TMP800, TMP3200 and TMG7800, connect to the telecom baseboard directly using telnet and the eth0 IP address.
telnet 10.3.2.21

2- In the baseboard prompt, use the set_voip0bonding shell command to enable VoIP Bonding.
set_voip0bonding <0|1> [IP_To_Ping] [Ping_Timeout] [Retries]
Where:
<0|1> selects if the feature is enabled (1 to enable). When the feature is enabled, the Ethernet PHY states are polled periodically to discover which port is available.
[IP_To_Ping] is the IP address to send ICMP echo requests to verify if a logical path is available. When no response is received within the requested timeout after the specified number of retries, the port is considered temporarly unusable and the other VoIP port is used instead.
[Ping_Timeout] is the ICMP echo request timeout.
[Retries] is the number of ICMP echo request retries to send before doing a switchover.
Examples:
set_voip0bonding 1
set_voip0bonding 1 10.3.10.187 1000 1
set_voip0bonding 1 123.45.67.89 2500 3

3- Reboot the unit (note: this causes downtime).
reboot

4- (optional) After the reboot, access the telecom baseboard again and verify that the VoIP Bonding was successfully enabled:
print_bonding

Tctrl Version 2.7

2013-07-08T17:07:31Z

Alexandre Tremblay: Added link to VoIP Bonding

== Features ==

The [[TMG-CONTROL]] provides the following [[:category:Tmedia Features|features]].

 

{| cellpadding="0" border="0" style="width: 820px; height: 383px;" class="wikitable"
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(200, 200, 200);" | Features
! width="150" style="background: none repeat scroll 0% 0% rgb(200, 200, 200);" | Enhancement'''1'''
! width="150" style="background: none repeat scroll 0% 0% rgb(200, 200, 200);" | Minor Release
|-
! width="800" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" colspan="3" | TMG Product
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[N+1]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | -
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[Symmetric_NAT_Traversal|NAT Traversal]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 5656
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[Lawful_intercept|Lawful Intercept]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 12894
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[RADIUS AAA]] (authorization and authentication support)
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 7984
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | Outgoing COT testing
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 8710
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[VoIP_Bonding|VoIP Bonding]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 14595
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.30
|-
! width="800" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" colspan="3" | Fax
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[SPIROU|SS7 SPIROU variant]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 6337
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[NSE]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 12665
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="800" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" colspan="3" | OAM&P
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[Web Profile]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | -
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[Test_Call|Test Call]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | -
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[Target_Recording|Target Recording]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | -
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[System_Statistics|System Statistics]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | -
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[Users|User authentication and permissions]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | -
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[Audit_Log|Audit Logs]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 10838
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.26
|-
! width="800" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" colspan="3" | Development Platform
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[CAF:_Working_With_Caf_Call_Legs|Mixing API]] (for conferencing and other functions)
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 4850
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.12
|-
! width="800" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" colspan="3" | New Products
|-
! width="500" style="background: none repeat scroll 0% 0% rgb(255, 255, 255); text-align:left;" | [[TM3000]]
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | -
! width="150" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | 2.7.15
|}

[1] Customer's RFC (Request For Change) tracking number.

Toolpack:Enabling VoIP Bonding A

2013-07-08T17:05:00Z

Alexandre Tremblay:

=== '''''Applies to version(s): v2.7.''''' ===
{{DISPLAYTITLE:Enabling VoIP Bonding}}

1- Access the Tmedia unit using SSH.

2- Access the telecom baseboard from the Tmedia using telnet.
telnet 172.31.1.1

3- In the baseboard prompt, use the set_voip0bonding shell command to enable VoIP Bonding.
set_voip0bonding <0|1> [IP_To_Ping] [Ping_Timeout] [Retries]
Where:
<0|1> selects if the feature is enabled (1 to enable). When the feature is enabled, the Ethernet PHY states are polled periodically to discover which port is available.
[IP_To_Ping] is the IP address to send ICMP echo requests to verify if a logical path is available. When no response is received within the requested timeout after the specified number of retries, the port is considered temporarly unusable and the other VoIP port is used instead.
[Ping_Timeout] is the ICMP echo request timeout.
[Retries] is the number of ICMP echo request retries to send before doing a switchover.
Examples:
set_voip0bonding 1
set_voip0bonding 1 10.3.10.187 1000 1
set_voip0bonding 1 123.45.67.89 2500 3

4- Reboot the unit (note: this causes downtime).
reboot

5- (optional) After the reboot, access the telecom baseboard again and verify that the VoIP Bonding was successfully enabled:
print_bonding

Toolpack:Enabling VoIP Bonding A

2013-07-08T16:59:53Z

Alexandre Tremblay: Added step to verify changes

VoIP Bonding

2013-07-08T16:55:35Z

Alexandre Tremblay: Changed wording for voip0 usage

VoIP Bonding refers to the capability of Tmedia units to use the two available VoIP Ethernet ports as a bonding pair. When the active port becomes unusable (cable disconnected, limited connectivity), the other port is used to transport Ethernet traffic.

Starting with Toolpack release 2.7.30, VoIP Bonding is available as a feature for Tmedia units. Detection of offline ports is currently achieved through two methods:
# Ethernet PHY polling to quickly discover if a cable is connected or not to the unit;
# Sending IPv4 ping requests to a known host to ensure that a logical IP path is available to the unit.

The voip0 (first VoIP port) MAC and IP addresses are used on both VoIP ports of the Tmedia unit when VoIP bonding is activated. However, in that mode, the voip1 port MAC and IP address are not usable (i.e. protocol stacks assigned to voip1 will not work and voip1 will not respond to ICMP requests). Therefore, to bind services and stacks (such as SIP, SIGTRAN and port ranges), the voip0 port must be assigned to them.

The VoIP Bonding feature also makes the Tmedia units report the bonding state as the voip0 port state. For instance, the IPv4 statistics reported by the Web Portal for voip0 are actually for the bonding pair. The PHY state is also for the bonding pair (so it would only be down if both voip ports are down). To know the true state of the physical voip0 and voip1 ports, the SNMP protocol must be used.

The feature has no effect on Virtual IP Interfaces configured in the Web Portal. Virtual IP Interfaces configured on either voip0 or voip1 physical ports still work exactly the same whether VoIP Bonding is activated.

== Configuration ==

*[[Web_Portal_Tutorial_Guide_v2.7#VoIP_Bonding|Toolpack 2.7: VoIP Bonding]]

Toolpack:Enabling VoIP Bonding A

2013-07-08T16:36:36Z

Alexandre Tremblay: Added page

Ip Teaming/Ip Bonding

2013-07-08T16:36:10Z

Alexandre Tremblay: Changed page for redirect to VoIP Bonding

#REDIRECT [[VoIP Bonding]]

VoIP Bonding

2013-07-08T15:31:38Z

Alexandre Tremblay: Added link to Web Portal Tutorial Guide

VoIP Bonding refers to the capability of Tmedia units to use the two available VoIP Ethernet ports as a bonding pair. When the active port becomes unusable (cable disconnected, limited connectivity), the other port is used to transport Ethernet traffic.

Starting with Toolpack release 2.7.30, VoIP Bonding is available as a feature for Tmedia units. Detection of offline ports is currently achieved through two methods:
# Ethernet PHY polling to quickly discover if a cable is connected or not to the unit;
# Sending IPv4 ping requests to a known host to ensure that a logical IP path is available to the unit.

The voip0 (first VoIP port) MAC and IP addresses are used on both VoIP ports of the Tmedia unit when VoIP bonding is activated. However, in that mode, the voip1 port MAC and IP address are not usable (i.e. protocol stacks assigned to voip1 will not work and voip1 will not respond to ICMP requests).

The VoIP Bonding feature also makes the Tmedia units report the bonding state as the voip0 port state. For instance, the IPv4 statistics reported by the Web Portal for voip0 are actually for the bonding pair. The PHY state is also for the bonding pair (so it would only be down if both voip ports are down). To know the true state of the physical voip0 and voip1 ports, the SNMP protocol must be used.

The feature has no effect on Virtual IP Interfaces configured in the Web Portal. Virtual IP Interfaces configured on either voip0 or voip1 physical ports still work exactly the same whether VoIP Bonding is activated.

== Configuration ==

*[[Web_Portal_Tutorial_Guide_v2.7#VoIP_Bonding|Toolpack 2.7: VoIP Bonding]]

Web Portal Tutorial Guide v2.7

2013-07-08T15:29:44Z

Alexandre Tremblay: Added VoIP Bonding

=Introduction=
This guide is designed to provide users with a scenario-based approach to installing [[Tmedia]] and [[Tdev]] systems, using the Web Portal configuration tool. This means that users must first go through the steps that are general to all installations, and subsequently follow the instructions that are specific to their particular application scenario (this corresponds to the type of service, and hardware system being set up).

==Prerequisites==
This document assumes that the [[Tmedia]] or [[Tdev]] devices are installed as described in their Quick Reference Guides, and in the case of the Tmedia system communication has been established with the control network.

(Note: Tmedia devices are shipped with the Tmedia Web Portal pre-installed, while Tdev devices require that Toolpack be installed as described in the [[Tmedia_Installation#Tdev|Tdev installation guides]].)

Furthermore, in the case of a Tdev device and the initial configuration of the [[Toolpack]] application server, the following will have been provided:
*The Tdev device to which the Web Portal will first connect.
*The Tdev device will have been pre-configured with its physical TDM interface.
*The serial number of the Tdev device will have been entered into the configuration file.
*The application software for the Tdev device will have been pre-installed on the Toolpack application server, as described in the [[Tmedia_Installation#Tdev|Tdev installation guides]].

This document also assumes familiarity with topics, such as [[ISDN]] signaling, [[SIP]] signaling, [[SS7]], [[CAS|CAS R2]] signaling, and [[SIGTRAN]] signaling.

==Key Concepts==
*'''System:''' A system is defined as the complete solution that is designed for a network. If a system solution is comprised of 4 TMP6400’s and a TMS1600, then the combination of these TelcoBridges devices is termed a system.
*'''Hardware:''' Each TelcoBridges devices, whether it is a TMP6400, TMG3200, or TMS1600 is referred to as a hardware device or hardware adapter.
*'''Line Interface:''' The physical TDM module installed on the TelcoBridges device is referred to as a line interface.

== General Web Portal Applications ==

'''''IMPORTANT!''''' The Topics covered in the following table are common to all TMedia installations. Therefore, this section should be consulted before moving on to any other.

'''Topics covered here:'''

=System=

==Access and User Management==
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" rowspan="2" | Web Portal Access
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" colspan="3"| User Management
|-

! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Add User
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Modify User Permission
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Delete User
|-
| valign="top" |
*[[Toolpack:Connecting to the Web Server and logging on to the web portal A|Connect to the Web Portal]]
*[[Toolpack:Logging Off B|Logging Off]]

|

*[[Toolpack:Creating User Groups A|Creating User Groups]]
*[[Toolpack:Creating Users B|Creating Users]]

| valign="top" |

*[[Modify User Permission]]

| valign="top" |

*[[Delete User]]

|}

==Database Backup==
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Create Database Backup
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Restore Database
|-
| valign="top" |
*[[Toolpack:Database Backup B|Carrying out a First Database Backup]]
*[[Toolpack:Downloading a Database Backup B|Downloading a Database Backup]]
| valign="top" |
*[[Toolpack:Uploading a Database Backup B|Uploading a Database Backup]]
*[[Toolpack:Restoring a Database Backup B|Restoring a Database Backup]]
|}

==Configuration==
{| cellpadding="5" border="1" class="wikitable"
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Activating a Configuration
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Web Portal Profile
|-
| valign="top" |
*[[Toolpack:Activating_the_Configuration B|Activating the Configuration]]
| valign="top" |
*[[Toolpack:Web_Portal_Profile A|Web Portal Profile]]
|}

<div id="Minor Package Upgrade">
==Minor Package Upgrade==
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Installing a Package
|-
| valign="top" |
*[[Toolpack:Downloading_a_Release_A|Retrieving a software release]]
*[[Toolpack:Uploading_a_Release_A|Uploading a software release]]
*[[Toolpack:Activating_a_Release_A|Activating a software release]]
|}
</div>

==Statistics==
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Configuration
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Profile
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Graphs
|-
| valign="top" |
* [[Toolpack:Configuring_Statistics_A|Configuring Statistics Polling]]
| valign="top" |
* [[Toolpack:Creating_Statistics_Profile_A|Creating Statistics Profiles]]
* [[Toolpack:Enabling_Statistics_Profile_A|Enabling Statistics Profiles]]
| valign="top" |
* [[Toolpack:Generating_Graphs_A|Generating Graphs]]
|}

==Audit Logs==
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Audit Logs
|-
| valign="top" |
*[[Toolpack:Accessing_Audit_Logs_A|Working with Audit Logs]]

|}

==VoIP Bonding==
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | VoIP Bonding
|-
| valign="top" |
*[[Toolpack:Enabling_VoIP_Bonding_A|Enabling VoIP Bonding]]
|}

=Hardware=

==New Hardware==
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Add telecom hardware
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Add TMS hardware
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Add 1+1 group
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Add N+1 group

|-
| valign="top" |
*[[Toolpack:Adding_an_Adapter_C|Add telecom hardware]]
| valign="top" |
*[[Toolpack:Adding_an_TMS_B|Add TMS hardware]]
| valign="top" |
*[[Toolpack:Creating_a_1+1_Group_A|Add 1+1 group]]
| valign="top" |
*[[Toolpack:Creating_a_N+1_Group_A|Add N+1 group]]
|}

==TDM Network==
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | (Step 2) System Clock

|-
| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create Line Interfaces]]
*[[Toolpack:Creating_a_Line_Service B|Create Line Services]]
| valign="top" |
*[[Toolpack:Defining_a_Clocking_Source_B|System Clock]]

|}

==IP Network==
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) IP Interfaces
! width="200" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Create an IP port range

|-
| valign="top" |
*[[Toolpack:Configuring_VoIP_Interfaces B|Configuring VoIP Interfaces]]
| valign="top" |
*[[Toolpack:Creating_an_IP_Port_Range A|Create an IP port range]]

|}

=Protocol Stacks=

==ISDN NFAS==

{| cellpadding="5" border="1" class="wikitable"
|-

! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces

! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Configure ISDN-PRI Signaling

! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Modify NAP profile

! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Configure Network Access Points (NAPs)
|-

| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create Line interfaces]]
*[[Toolpack:Creating_a_Line_Service B|Create Line Services]]

| valign="top" |
*[[Toolpack:Configuring_an_ISDN_Stack C|Configure an ISDN stack]]
OR 
*[[Toolpack:Configuraing_NFAS A|Configure NFAS]]

| valign="top" |
*[[Toolpack:Modify_NAP_profile B|Modify profile]]

| valign="top" |
*[[Toolpack:Allocating_an_ISDN_Network_Access_Point_(NAP) C|Allocate an ISDN NAP]]

|}

===''ISDN Advanced Features''===

<div class="mw-collapsible mw-collapsed" data-collapsetext="ISDN NFAS Overlap" data-expandtext="ISDN NFAS Overlap" style="width: 400px;">

{| cellpadding="5" border="1" class="wikitable"
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |ISDN Overlap Sending
|-
| valign="top" |
*[[Toolpack:Configuring_Overlap A|''ISDN NFAS Overlap Parameters'']]
|}

</div>
 

==CASR2==

{| cellpadding="5" border="1" class="wikitable"
|-

! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Configuring a CAS R2 stack
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Modify NAP profile
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Configure Network Access Points (NAPs)

|-
| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create Line Interfaces]]
*[[Toolpack:Creating_a_Line_Service B|Create Line services]]
| valign="top" |
*[[Toolpack:Copy variant script B|Copy variant script]]
*[[Toolpack:Create CAS R2 stack B|Create CAS R2 stack]]
*[[Toolpack:Setting CAS R2 BACK DIGITS variable B|Set backward digits string]]
| valign="top" |
*[[Toolpack:Modify_NAP_profile B|Modify profile]]
| valign="top" |
*[[Toolpack:Creating_a_CAS_R2_NAP B|Create a CASR2 NAP]]

|}

==''SIP''==

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Configure IP port range
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Configure SIP Signaling
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Configure Codecs and Tone Detection
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Modify NAP profile
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 5) Configure Network Access Points (NAPs)
|-
| valign="top" |
*[[Toolpack:Creating_an_IP_Port_Range A|Create an IP port range]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Stack A|Create a SIP stack]]
*[[Toolpack:Creating_a_SIP_Transport_Server A|Create a SIP transport server]]
*[[Toolpack:Creating_a_Service_Access_Point_(SAP) A|Create a service access point (SAP)]]

| valign="top" |
*[[Toolpack:Accessing_Codec_and_Tone_Detection_Profiles_and_Validating_Settings A|Access profiles and validate settings]]

| valign="top" |
*[[Toolpack:Modify_NAP_profile B|Modify profile]]

| valign="top" |
*[[Toolpack:Allocating_a_SIP_Network_Access_Point_(NAP) A|Allocate a SIP NAP]]

|}

===''SIP Advanced Features''===
<div class="mw-collapsible mw-collapsed" data-collapsetext="SIP-I SIP-T" data-expandtext="SIP-I SIP-T" style="width: 400px;">
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | SIP-I SIP-T
|-
| valign="top" |
*[[Enabling_SIP_I_SIP_T_A|Enabling SIP-I SIP-T]]
|}
</div>
<div class="mw-collapsible mw-collapsed" data-collapsetext="SIP Spiral" data-expandtext="SIP Spiral" style="width: 400px;">
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | SIP Spiral
|-
| valign="top" |
*[[Enabling_SIP_Spiral_A|Enabling SIP Spiral]]
|}

</div>

<div class="mw-collapsible mw-collapsed" data-collapsetext="NAT traversal" data-expandtext="NAT traversal" style="width: 400px;">
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | Remote NAT Traversal
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | Local NAT Traversal
|-
| valign="top" |
*[[Enabling_Remote_NAT_Traversal_A|Enabling Remote NAT Traversal]]

| valign="top" |
*[[Enabling_Local_NAT_Traversal_A|Enabling Local NAT Traversal]]
|}
</div>

==SS7 ISUP==

===''MTP2-MTP3-ISUP''===

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Configuring the MTP2 Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Configuring the MTP3 Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Configuring the ISUP Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 5) Modify NAP profile
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 6) Configure Network Access Points (NAPs)

|-
| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create a new line interface]]
*[[Toolpack:Creating_a_Line_Service B|Create a new line service]]

| valign="top" |
*[[Toolpack:Creating_an_MTP2_Configuration A|Create an MTP2 Configuration]]
*[[Toolpack:Creating_MTP2_Links A|Create an MTP2 Link]]

| valign="top" |
*[[Toolpack:Creating_an_MTP3_Configuration B|Create an MTP3 Configuration]]
*[[Toolpack:Creating_an_MTP3_Network A|Create an MTP3 Network]]
**[[Toolpack:Creating_SS7_Point_Codes A|Create an MTP3 Point Code]]
**[[Toolpack:Creating_an_MTP3_Linkset A|Create an MTP3 Linkset]]
***[[Toolpack:Creating_MTP3_Links A|Create MTP3 Links]]
**[[Toolpack:Creating_an_MTP3_Route A|Create an MTP3 Route]]

| valign="top" |
*[[Toolpack:Creating_an_ISUP_Stack A|Create an ISUP Stack]]
*[[Toolpack:Creating_an_ISUP_Network A|Create an ISUP Network]]
*[[Toolpack:Creating_an_ISUP_User_Part A|Create an ISUP User Part]]
*[[Toolpack:Creating_an_ISUP_Interface A|Create an ISUP Interface]]
*[[Toolpack:Creating ISUP CIC Groups A|Create ISUP CIC Groups]]

| valign="top" |
*[[Toolpack:Modify_NAP_profile B|Modify profile]]

| valign="top" |
*[[Toolpack:Creating_a_NAP_(SIGTRAN)_A|Create an ISUP NAP]]

|}

===''M2PA-MTP3-ISU''P===

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Configuring the M2PA Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Configuring the MTP3 Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Configuring the ISUP Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 5) Modify NAP profile
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 6) Configure Network Access Points (NAPs)

|-
| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create a new line interface]]
*[[Toolpack:Creating_a_Line_Service B|Create a new line service]]

| valign="top" |
*[[Toolpack:Creating the SCTP Configuration A|Edit the SCTP Configuration]]
*[[Toolpack:Creating the M2PA Configuration A|Create an M2PA Configuration]]
**[[Toolpack:Creating an M2PA SAP A|Create an M2PA SAP]]
**[[Toolpack:Adding M2PA Links A|Add M2PA Links]]

| valign="top" |
*[[Toolpack:Creating_an_MTP3_Configuration B|Create an MTP3 Configuration]]
*[[Toolpack:Creating_an_MTP3_Network A|Create an MTP3 Network]]
**[[Toolpack:Creating_SS7_Point_Codes A|Create an MTP3 Point Code]]
**[[Toolpack:Creating_an_MTP3_Linkset A|Create an MTP3 Linkset]]
***[[Toolpack:Creating_MTP3_Links A|Create MTP3 Links]]
**[[Toolpack:Creating_an_MTP3_Route A|Create an MTP3 Route]]

| valign="top" |
*[[Toolpack:Creating_an_ISUP_Stack A|Create an ISUP Stack]]
*[[Toolpack:Creating_an_ISUP_Network A|Create an ISUP Network]]
*[[Toolpack:Creating_an_ISUP_User_Part A|Create an ISUP User Part]]
*[[Toolpack:Creating_an_ISUP_Interface A|Create an ISUP Interface]]
*[[Toolpack:Creating ISUP CIC Groups A|Create ISUP CIC Groups]]

| valign="top" |
*[[Toolpack:Modify_NAP_profile B|Modify profile]]

| valign="top" |
*[[Toolpack:Creating_a_NAP_(SIGTRAN)_A|Create an ISUP NAP]]

|}

===''M3UA-ISUP''===

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Configuring the M3UA Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Configuring the ISUP Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Modify NAP profile
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 5) Configure Network Access Points (NAPs)

|-

| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create a new line interface]]
*[[Toolpack:Creating_a_Line_Service B|Create a new line service]]

| valign="top" |
*[[Toolpack:Creating the SCTP Configuration A|Edit the SCTP Configuration]]
*[[Toolpack:Creating SS7 Point Codes A|Create SS7 Point Codes]]
*[[Toolpack:Creating an M3UA Configuration (IPSP) A|Create an M3UA Configuration]]
**[[Toolpack:Creating an M3UA SAP A|Create an M3UA SAP]]
**[[Toolpack:Creating an M3UA Network A|Create an M3UA Network]]
**[[Toolpack:Creating an M3UA User Part A|Create an M3UA User Part]]
**[[Toolpack:Creating an M3UA Peer Signalling Process (IPSP) A|Create an M3UA Peer Signalling Process]]
**[[Toolpack:Creating an M3UA Peer Server A|Create an M3UA Peer Server]]
**[[Toolpack:Creating an M3UA Route A|Create an M3UA Route]]

| valign="top" |
*[[Toolpack:Creating_an_ISUP_Stack A|Create an ISUP Stack]]
*[[Toolpack:Creating_an_ISUP_Network A|Create an ISUP Network]]
*[[Toolpack:Creating_an_ISUP_User_Part A|Create an ISUP User Part]]
*[[Toolpack:Creating_an_ISUP_Interface A|Create an ISUP Interface]]
*[[Toolpack:Creating ISUP CIC Groups A|Create ISUP CIC Groups]]

| valign="top" |
*[[Toolpack:Modify_NAP_profile B|Modify profile]]

| valign="top" |
*[[Toolpack:Creating_a_NAP_(SIGTRAN)_A|Create an ISUP NAP]]

|}

===SS7 Advanced Features===

<div class="mw-collapsible mw-collapsed" data-collapsetext="SS7 SPIROU configuration" data-expandtext="SS7 SPIROU configuration" style="width: 400px;">
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | Configuring SPIROU
|-
| valign="top" |
*[[Toolpack:Modifying_Profile_for_SPIROU A|Modifying Profile]]
*[[Toolpack:Modifying_Gateway_Configurations_for_SPIROU A|Modifying Gateway Configurations]]
*[[Toolpack:Modifying_ISUP_Stack_for_SPIROU A|Modifying ISUP Stack]]
*[[Toolpack:Modifying_ISUP_Userpart_for_SPIROU A|Modifying ISUP Userpart]]
*[[Toolpack:Optional_configurations_for_SPIROU A|Optional modifications]]
|}

</div>

==''SCCP-TCAP''==

===''MTP2-MTP3-SCCP-TCAP''===

{| cellpadding="5" border="1" class="wikitable"

! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Configuring the MTP2 Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Configuring the MTP3 Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Configuring the SCCP Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 5) Configuring the TCAP Layer (optional)

|-

| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create a new line interface]]
*[[Toolpack:Creating_a_Line_Service B|Create a new line service]]

| valign="top" |
*[[Toolpack:Creating_an_MTP2_Configuration A|Create an MTP2 Configuration]]
*[[Toolpack:Creating_MTP2_Links A|Create an MTP2 Link]]

| valign="top" |
*[[Toolpack:Creating_an_MTP3_Configuration B|Create an MTP3 Configuration]]
*[[Toolpack:Creating_an_MTP3_Network A|Create an MTP3 Network]]
**[[Toolpack:Creating_SS7_Point_Codes A|Create an MTP3 Point Code]]
**[[Toolpack:Creating_an_MTP3_Linkset A|Create an MTP3 Linkset]]
***[[Toolpack:Creating_MTP3_Links A|Create MTP3 Links]]
**[[Toolpack:Creating_an_MTP3_Route A|Create an MTP3 Route]]

| valign="top" |
*[[Toolpack:Creating_an_SCCP_Stack A| Create an SCCP Stack]]
*[[Toolpack:Creating_an_SCCP_Network A|Create an SCCP Network]]
**[[Toolpack:Creating_an_SCCP_Userpart A|Create an SCCP Userpart]]
**[[Toolpack:Creating_an_SCCP_LSAP A|Create an SCCP LSAP]]
***[[Toolpack:Creating_an_SCCP_Route A|Create an SCCP Route]]
****[[Toolpack:Creating_an_SCCP_SSN A|Create an SCCP SSN]]
*[[Toolpack:Creating_an_SCCP_GTT_Association A|Create an SCCP GTT Association]]
**[[Toolpack:Creating_an_SCCP_GTT_Address_Map A|Create an SCCP GTT Address Map]]

| valign="top" |
*[[Toolpack:Creating_a_TCAP_Stack A|Create a TCAP Stack]]
**[[Toolpack:Creating_a_TCAP_Userpart A| Create a TCAP Userpart]]
*[[Toolpack:Activating_the_Configuration A|Activating the Configuration]]

|}

===''M2PA-MTP3-SCCP-TCAP''===

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Configuring the M2PA Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Configuring the MTP3 Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Configuring the SCCP Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 5) Configuring the TCAP Layer (optional)

|-
| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create a new line interface]]
*[[Toolpack:Creating_a_Line_Service B|Create a new line service]]

| valign="top" |
*[[Toolpack:Creating the SCTP Configuration A|Edit the SCTP Configuration]]
*[[Toolpack:Creating the M2PA Configuration A|Create an M2PA Configuration]]
**[[Toolpack:Creating an M2PA SAP A|Create an M2PA SAP]]
**[[Toolpack:Adding M2PA Links A|Add M2PA Links]]

| valign="top" |
*[[Toolpack:Creating_an_MTP3_Configuration B|Create an MTP3 Configuration]]
*[[Toolpack:Creating_an_MTP3_Network A|Create an MTP3 Network]]
**[[Toolpack:Creating_SS7_Point_Codes A|Create an MTP3 Point Code]]
**[[Toolpack:Creating_an_MTP3_Linkset A|Create an MTP3 Linkset]]
***[[Toolpack:Creating_MTP3_Links A|Create MTP3 Links]]
**[[Toolpack:Creating_an_MTP3_Route A|Create an MTP3 Route]]

| valign="top" |
*[[Toolpack:Creating_an_SCCP_Stack A| Create an SCCP Stack]]
*[[Toolpack:Creating_an_SCCP_Network A|Create an SCCP Network]]
**[[Toolpack:Creating_an_SCCP_Userpart A|Create an SCCP Userpart]]
**[[Toolpack:Creating_an_SCCP_LSAP A|Create an SCCP LSAP]]
***[[Toolpack:Creating_an_SCCP_Route A|Create an SCCP Route]]
****[[Toolpack:Creating_an_SCCP_SSN A|Create an SCCP SSN]]
*[[Toolpack:Creating_an_SCCP_GTT_Association A|Create an SCCP GTT Association]]
**[[Toolpack:Creating_an_SCCP_GTT_Address_Map A|Create an SCCP GTT Address Map]]

| valign="top" |
*[[Toolpack:Creating_a_TCAP_Stack A|Create a TCAP Stack]]
**[[Toolpack:Creating_a_TCAP_Userpart A| Create a TCAP Userpart]]
*[[Toolpack:Activating_the_Configuration A|Activating the Configuration]]

|}

===''M3UA-SCCP-TCAP''===

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Configuring the M3UA Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Configuring the SCCP Layer
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Configuring the TCAP Layer (optional)

|-

| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create a new line interface]]
*[[Toolpack:Creating_a_Line_Service B|Create a new line service]]

| valign="top" |
*[[Toolpack:Creating the SCTP Configuration A|Edit the SCTP Configuration]]
*[[Toolpack:Creating SS7 Point Codes A|Create SS7 Point Codes]]
*[[Toolpack:Creating an M3UA Configuration (IPSP) A|Create an M3UA Configuration]]
**[[Toolpack:Creating an M3UA SAP A|Create an M3UA SAP]]
**[[Toolpack:Creating an M3UA Network A|Create an M3UA Network]]
**[[Toolpack:Creating an M3UA User Part A|Create an M3UA User Part]]
**[[Toolpack:Creating an M3UA Peer Signalling Process (IPSP) A|Create an M3UA Peer Signalling Process]]
**[[Toolpack:Creating an M3UA Peer Server A|Create an M3UA Peer Server]]
**[[Toolpack:Creating an M3UA Route A|Create an M3UA Route]]

| valign="top" |
*[[Toolpack:Creating_an_SCCP_Stack A| Create an SCCP Stack]]
*[[Toolpack:Creating_an_SCCP_Network A|Create an SCCP Network]]
**[[Toolpack:Creating_an_SCCP_Userpart A|Create an SCCP Userpart]]
**[[Toolpack:Creating_an_SCCP_LSAP A|Create an SCCP LSAP]]
***[[Toolpack:Creating_an_SCCP_Route A|Create an SCCP Route]]
****[[Toolpack:Creating_an_SCCP_SSN A|Create an SCCP SSN]]
*[[Toolpack:Creating_an_SCCP_GTT_Association A|Create an SCCP GTT Association]]
**[[Toolpack:Creating_an_SCCP_GTT_Address_Map A|Create an SCCP GTT Address Map]]

| valign="top" |
*[[Toolpack:Creating_a_TCAP_Stack A|Create a TCAP Stack]]
**[[Toolpack:Creating_a_TCAP_Userpart A| Create a TCAP Userpart]]
*[[Toolpack:Activating_the_Configuration A|Activating the Configuration]]

|}

==''H.248 MEGACO''==

If the Tmedia is used to forward signalling messages to the SSW, you will require to configure M2UA, M3UA or IUA before configuring H.248.

If the Tmedia is used to connect media only, then you can go straight to [[#H.248|H.248 Configuration]]

===''M2UA''===

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Create Line Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Create MTP2 links
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Create M2UA configuration

|-

| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create line interfaces]]
*[[Toolpack:Creating_a_Line_Service B|Create line services]]

| valign="top" |

*[[Toolpack:Creating an MTP2 Configuration A|Create an MTP2 Configuration]]
**[[Toolpack:Creating MTP2 Links A|Create MTP2 Links]]

| valign="top" |

*[[Toolpack:Creating the SCTP Configuration A|Edit the SCTP Configuration]]
*[[Toolpack:Creating the M2UA Configuration A|Create an M2UA Configuration]]
**[[Toolpack:Creating an M2UA SAP A|Create an M2UA SAP]]
**[[Toolpack:Creating an M2UA Cluster A|Create an M2UA Cluster]]
**[[Toolpack:Creating M2UA Links A|Create M2UA Links]]
**[[Toolpack:Creating M2UA Peers A|Create M2UA Peers]]

|}

===''M3UA''===

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Create Line Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Create MTP2 links
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Create MTP3 configuration
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Create M3UA configuration
|-

| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create line interfaces]]
*[[Toolpack:Creating_a_Line_Service B|Create line services]]

| valign="top" |
*[[Toolpack:Creating an MTP2 Configuration A|Create an MTP2 Configuration]]
*[[Toolpack:Creating MTP2 Links A|Create MTP2 Links]]

| valign="top" |
*[[Toolpack:Creating SS7 Point Codes A|Create SS7 Point Codes]]
*[[Toolpack:Creating an MTP3 Configuration B|Create an MTP3 Configuration]]
**[[Toolpack:Creating an MTP3 Network A|Create an MTP3 Network]]
**[[Toolpack:Creating an MTP3 Linkset A|Create an MTP3 Linkset]]
**[[Toolpack:Creating MTP3 Links (M3UA) A|Create MTP3 Links]]
**[[Toolpack:Creating an MTP3 Route A|Create an MTP3 Route]]

| valign="top" |
*[[Toolpack:Creating the SCTP Configuration A|Edit the SCTP Configuration]]
*[[Toolpack:Creating an M3UA Configuration (SGP) A|Create an M3UA Configuration]]
**[[Toolpack:Creating an M3UA SAP A|Create an M3UA SAP]]
**[[Toolpack:Creating an M3UA Network A|Create an M3UA Network]]
**[[Toolpack:Creating an M3UA User Part A|Create an M3UA User Part]]
**[[Toolpack:Creating an M3UA Peer Signalling Process (SGP) A|Create an M3UA Peer Signalling Process]]
**[[Toolpack:Creating an M3UA Peer Server A|Create an M3UA Peer Server]]
**[[Toolpack:Creating an M3UA Route (SGP) A|Create an M3UA Route]] 

|}

===''IUA''===

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Create Line Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Create M2UA Configuration

|-

| valign="top" |
*[[Toolpack:Creating a Line Interface B|Create Line Interfaces]]
*[[Toolpack:Creating a Line Service B|Create Line services]]

| valign="top" |
*[[Toolpack:Creating the SCTP Configuration A|Edit the SCTP Configuration]]
*[[Toolpack:Creating the IUA Configuration A|Create an IUA Configuration]]
**[[Toolpack:Creating an IUA SAP A|Create an IUA SAP]]
**[[Toolpack:Creating an IUA Cluster A|Create an IUA Cluster]]
**[[Toolpack:Creating IUA Links A|Create IUA Links]]
**[[Toolpack:Creating IUA Peers A|Create IUA Peers]]

|}

===''H.248''===

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 1) Allocate Physical Interfaces
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 2) Create IP Port Ranges
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 3) Modify NAP profile
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 4) Configure Network Access Points (NAPs)

|-

| valign="top" |
*[[Toolpack:Creating_a_Line_Interface B|Create line interfaces]]
*[[Toolpack:Creating_a_Line_Service B|Create line services]]

| valign="top" |
*[[Toolpack:Creating_an_IP_Port_Range A|Create IP port ranges]]
| valign="top" |
*[[Toolpack:Modify_NAP_profile A|Modify profile]]

| valign="top" |
*[[Toolpack:Allocating_a_VoIP_Network_Access_Point_(NAP) A|Allocate a VoIP NAP]]
*[[Toolpack:Allocating_a_TDM_Network_Access_Point_(NAP) A|Allocate a TDM NAP]]
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 5) Configure Media Gateway for H.248 mode
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 6) Add a New Media Gateway Controller
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (Step 7) Associate NAPs Resources to H.248 Configuration

|-

| valign="top" |
*[[Toolpack:Configure the Media_Gateway_for_H248_Mode A|Configure the Media Gateway for H.248 mode]]
*[[Toolpack:Creating_an_H.248_Configuration A|Create a new H.248 configuration]]
| valign="top" |
*[[Toolpack:Adding_a_Media_Gateway_Controller_(MGC) A|Add a new media gateway controller (MGC)]]
| valign="top" |
*[[Toolpack:Associating_NAPs_with_an_H.248_Configuration A|Associate the NAPs with the H.248 configuration]]
*[[Toolpack:Selecting_Timeslots_for_TDM_Interfaces A|Select timeslots for the TDM interfaces]]

|}

<div id="CDR">

=''CDR (Call Detail Record)''=
{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Text CDR
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |RADIUS
|-

| valign="top" |
*[[Text_cdr|Text CDR]]

| valign="top" |
*[[Toolpack:Enabling RADIUS A|Enabling RADIUS]]
*[[Toolpack:Configuring_RADIUS_client A|Configuring RADIUS client]]
*[[Toolpack:Adding_RADIUS_server A|Adding RADIUS server(s)]]

|}
</div>

=''Fax''=

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |FAX passthrough
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |FAX T.38
|-
| valign="top" |
*[[Configure_Fax_Relay_A|Configure Fax Relay]]
*[[Configure_Fax_Passthrough_A|Configure Fax Passthrough]]
| valign="top" |
*[[Configure_Fax_Relay_A|Configure Fax Relay]]
*[[Configure_Fax_T38_A|Configure Fax T38]]
|}

=Routes=

==''Static Route''==
{| cellpadding="5" border="1" class="wikitable"
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Add routes
|-
| valign="top" |
*[[Toolpack:Creating_a_First_Call_Route A|Create a first call route]]
|}

==''Routing Script''==
TelcoBridges provides a wide variety of routing script examples which customers can apply directly in their web configuration according to their needs by following just a few simple steps. Please select one of the following examples according to your needs. You may modify it when necessary.

'''*Flexible NOA:''' Remap the Calling/Called NOA flexibly based on the routes. It should cover most of the NOA requests since the routes include the calling/called remapping as well.

'''*Prefix NOA:''' Remapp Calling/Called NOA according to the prefix of the calling/called number, manipulate calling/called number. For calling number, if it is a privacy call, use the number in P-Asserted-Identity line or Remote-Party-ID line as the calling number to remap the NOA and do the manipulation.

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Flexible NOA
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Prefix NOA
|-
| valign="top" |

*[[Toolpack:Create_New_Routing_Script_B|Create new routing script]]
*[[Import_Customized_Routing_Script_A|Import customized routing script]]
*[[Add_NOA_Columns_in_Routes_A|Add NOA columns in routes]]
*[[Enable_Routing_Script_A|Enable routing script]]
*[[Activate_Configuration_A|Activate configuration]]

| valign="top" |

*[[Toolpack:Create_New_Routing_Script_B|Create new routing script]]
*[[Import_Customized_Routing_Script_A|Import customized routing script]]
*[[Enable_Routing_Script_A|Enable routing script]]
*[[Activate_Configuration_A|Activate configuration]]

|}

<div class="mw-collapsible mw-collapsed" data-collapsetext="Configuring Audio Prompts" data-expandtext="Audio Prompts" style="width: 400px;">

{| cellpadding="5" border="1" class="wikitable"
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Audio Prompts
|-
| valign="top" |
*[[Toolpack:Configuring_Audio_Prompts A|Audio Prompts]]
|}

</div>
 

==''Label Routing''==

This section assumes:
* '''The routing script is already implemented''' (you just need to add the label routing functionality)
* '''The routeset definition and digimap files are already created'''

{|
|STYLE="vertical-align: top"|
{| cellpadding="5" border="1" class="wikitable"
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Add label routing functionality
|-
| valign="top" |
*[[Adding_Label_Routing_to_any_Routing_Script|Add Label Routing to script]]
*[[Create_Databse_Files A|Create Database Files]]
*[[Assign_Routing_Script_DB_Files_to_Gateway_Application A|Assign Routing/Script Files to the Gateway Application]]
*[[Add_Digitmap_Files A|Add the Digit Map Files to the System]]
*[[Add_Routeset_Definition_Files A|Add the Routset Definition Files to the System]]
*[[Assign_Definition_Digitmap_Files_per_NAP A|Assign Defintion/Digitmap Files on a per NAP Basis]]
*[[Generate_Dynamic_Routes A|Generate Dynamic Routes]]
*[[Activate_Configuration_A|Activate configuration]]
|}
|STYLE="vertical-align: top"|
{| cellpadding="5" border="1" class="wikitable"
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Update definition/digitmap files
|-
| valign="top" |
*[[Update_Digitmap_Files A|Update the Digitmap Files]]
*[[Update_Routeset_Definition_Files A|Update the Routeset Definition Files]]
*[[Generate_Dynamic_Routes A|Generate Dynamic Routes]]
*[[Activate_Configuration_A|Activate configuration]]
|}
|}

==RADIUS Authorization and Authentication==
{|
|STYLE="vertical-align: top"|
{| cellpadding="5" border="1" class="wikitable"
! width="240" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |RADIUS
|-
| valign="top" |
*[[Toolpack:Configuring_RADIUS_client_B|Configure RADIUS client]]
*[[Toolpack:Adding_RADIUS_server_B|Add RADIUS authorization server]]
*[[Toolpack:Configuring_RADIUS_authorization_A|Configure RADIUS authorization]]
|}
|STYLE="vertical-align: top"|
{| cellpadding="5" border="1" class="wikitable"
! width="300" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |RADIUS (with custom dictionary)
|-
| valign="top" |
*[[Toolpack:Import_RADIUS_Custom_Dictionary_A|Import RADIUS Custom Dictionary]]
*[[Toolpack:Configuring_RADIUS_client_B|Configure RADIUS Client]]
*[[Toolpack:Adding_RADIUS_server_B|Add RADIUS Authorization Server]]
*[[Toolpack:Create_New_Routing_Script_B|Create New Routing Script]]
*[[Toolpack:Configuring_RADIUS_authorization_A|Configure RADIUS Authorization]]
*[[Toolpack:Enable_Routing_Script_B|Enable Routing Script]]
|}
|}

<div id="Lawful_Intercept">

=Lawful Interception=
{| cellpadding="5" border="1" class="wikitable"
! width="300" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Lawful Interception
|-
| valign="top" |
*[[Toolpack:Verifying_lawful_interception_A|Verify Lawful Interception Prerequisite]]
*[[Toolpack:Importing_lawful_interception_A|Import a Lawful Interception CSV File]]
*[[Toolpack:Create_New_Routing_Script_B|Create New Routing Script]]
*[[Toolpack:Enable_Lawful_Interception_in_Routing_Script_A|Enable Lawful Interception in Routing script]]
*[[Toolpack:Enable_Routing_Script_B|Enable Routing Script]]
*[[Toolpack:Configuring_lawful_interception_A|Configure Lawful Agencies]]
|}
</div>

=Troubleshooting=

==''Call Trace''==
{| cellpadding="5" border="1" class="wikitable"
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Call trace
|-
| valign="top" |
*[[Toolpack:Configuring_Call_Trace A|Call trace]]
|}

==''Test Call''==
{| cellpadding="5" border="1" class="wikitable"
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" |Test Call
|-
| valign="top" |
*[[Creating_A_Test_Call_A|Creating a Test Call]]
|}

==''Target Recording''==
{| cellpadding="5" border="1" class="wikitable"
! width="300" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Activating a Configuration
! width="300" style="background: none repeat scroll 0% 0% rgb(239, 239, 239); -moz-background-inline-policy: continuous;" | Web Portal Profile
|-
| valign="top" |
*[[Enabling_Recordings_A|Enabling the Recording of Calls]]
| valign="top" |
* [[Accessing_Recording|Accessing the Recording]]
|}

=''Status Menus''=

The Web Portal configuration tool will guide you through a number of status menus, each of which indicate the current state of a certain hardware service (assuming said service has been configured).
The following table contains the general conventions of the Web Portal's display '''(A)''', as well as a variety of specific status menus '''(B)'''.

{| cellpadding="5" border="1" class="wikitable"
|-
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (A) Menus Overview
! width="200" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | (B) Individual Menu Descriptions

|-
| valign="top" |

*[[Toolpack:Tmedia_Status_Menus A|Tmedia Status Menus]]
**[[Toolpack:Status_Menus:Navigation A|Navigation]]
**[[Toolpack:Status_Menus:Screen_Conventions A|Status Screen Conventions]]

| valign="top" |

*[[Toolpack:Status_Menus:System B|System]]
*[[Toolpack:Status_Menus:Adapters_and_IP_Interfaces B|Adapters and IP Interfaces]]
*[[Toolpack:Status_Menus:1+1|1+1]]
*[[Toolpack:Status_Menus:Hosts A|Hosts]]
*[[Toolpack:Status_Menus:Applications A|Applications]]
*[[Toolpack:Status_Menus:TDM_Lines A|TDM Lines]]
**[[Toolpack:Status_Menus:Line Interfaces A|Line Interfaces]]
*[[Toolpack:Status_Menus:SCTP A|SCTP]]
*[[Toolpack:Status_Menus:SS7_MTP2 A|SS7 MTP2]]
*[[Toolpack:Status_Menus:SS7_M2UA A|SS7 M2UA]]
*[[Toolpack:Status_Menus:SS7_IUA A|SS7 IUA]]
*[[Toolpack:Status_Menus:SS7_M2PA A|SS7 M2PA]]
*[[Toolpack:Status_Menus:SS7_M3UA A|SS7 M3UA]]
*[[Toolpack:Status_Menus:SS7_MTP3 A|SS7 MTP3]]
*[[Toolpack:Status_Menus:SS7_ISUP A|SS7 ISUP]]
*[[Toolpack:Status_Menus:SS7_SCCP A|SS7 SCCP]]
*[[Toolpack:Status_Menus:SS7_TCAP A|SS7 TCAP]]
*[[Toolpack:Status_Menus:ISDN A|ISDN]]
**[[Toolpack:Status_Menus:NFAS A|NFAS]]
*[[Toolpack:Status_Menus:CAS R2 A|CAS R2]]
*[[Toolpack:Status_Menus:SIP A|SIP]]
*[[Toolpack:Status_Menus:Clock B|Clock]]
**[[Toolpack:Status_Menus:Clock Status List A|Clock Status List]]
*[[Toolpack:Status_Menus:TMS A|TMS]]
*[[Toolpack:Status_Menus:NAP A|NAP]]
**[[Toolpack:Status_Menus:NAP Status List A|NAP Status List]]
*[[Toolpack:Status_Menus:H.248 A|H.248]]
*[[Toolpack:Status_Menus:RADIUS A|RADIUS]]

|}

[[category:Toolpack]]
[[category:Coming Soon]]

VoIP Bonding

2013-07-08T15:25:13Z

Alexandre Tremblay: First draft

VoIP Bonding refers to the capability of Tmedia units to use the two available VoIP Ethernet ports as a bonding pair. When the active port becomes unusable (cable disconnected, limited connectivity), the other port is used to transport Ethernet traffic.

Starting with Toolpack release 2.7.30, VoIP Bonding is available as a feature for Tmedia units. Detection of offline ports is currently achieved through two methods:
# Ethernet PHY polling to quickly discover if a cable is connected or not to the unit;
# Sending IPv4 ping requests to a known host to ensure that a logical IP path is available to the unit.

The voip0 (first VoIP port) MAC and IP addresses are used on both VoIP ports of the Tmedia unit when VoIP bonding is activated. However, in that mode, the voip1 port MAC and IP address are not usable (i.e. protocol stacks assigned to voip1 will not work and voip1 will not respond to ICMP requests).

The VoIP Bonding feature also makes the Tmedia units report the bonding state as the voip0 port state. For instance, the IPv4 statistics reported by the Web Portal for voip0 are actually for the bonding pair. The PHY state is also for the bonding pair (so it would only be down if both voip ports are down). To know the true state of the physical voip0 and voip1 ports, the SNMP protocol must be used.

The feature has no effect on Virtual IP Interfaces configured in the Web Portal. Virtual IP Interfaces configured on either voip0 or voip1 physical ports still work exactly the same whether VoIP Bonding is activated.

CAS R2 scripting

2013-06-10T20:09:06Z

Alexandre Tremblay:

This page contains information about TelcoBridges CAS R2 variant scripting.

Refer to the default script (itu.lua) in the Toolpack Web Portal for a complete working variant script example.

In this page, using the term "application" may refer to Toolpack or any other application using the basic TelcoBridges CAS R2 stack API.

= Overview =
[[CAS|CAS R2]] signaling is done in two parts: line signaling and register signaling. In CAS R2 for [[E1]] trunks, [[Line signaling|line signaling]] uses a single [[timeslot]] to communicate call states (such as off-hook, answering) for the remaining timeslots. [[Register signaling]] is the exchange of MFR2 tones to convey information such as the calling number, called number and calling party category. TelcoBridges CAS R2 scripts allow complete customization of the register signaling process, while having some characteristics of the line signaling process customizable. Scripts also contain information relative to the conversion of register signaling data - group B signals and categories - into a format that the application can interpret. The scripting language is [[Lua]].

= Mandatory elements =
In order for CAS R2 variant scripts to work, some variables, tables and events must be defined from within scripts. All of the following elements are required; if one of these elements is not present, it will not be possible to save the script in the Toolpack Web Portal or load it on the adapter.

=== VARIANT_ID ===
The VARIANT_ID variable must be filled to distinguish variants from another. More precisely, it is used when transporting [[#Group B conversion information|group B]] and [[#Category conversion information|category]] information with calls in order to possibly convert information meaning across variants.

Default CAS R2 TelcoBridges scripts have variant IDs starting from 1. To avoid future conflicts, user scripts variant IDs should be prefixed with a sufficiently high value. For example, a modified ITU variant could have a ID equal to 0x0001001. Scripts may only share the same variant ID as other scripts if and only if the category and group B meanings are exactly identical.

== Group B conversion information ==
Group B conversion information in scripts is used to convert group B information between CAS R2 variants and between other signaling types. The following elements must be filled so that these conversions can be completed successfully.

=== GROUPB_DIGIT_TO_MEANING ===
Each GROUPB_DIGIT_TO_MEANING table element must contain two values: a digit and a meaning. The digit must be a valid digit value (1 to 15) and the meaning must be a valid meaning (see [[#Group B meanings|Group B meanings]]). Each digit must be present in the table.

This table is used to convert group B digits to meaning in the context of an outgoing call. For a given digit, the specified meaning will be propagated to the application.

=== GROUPB_MEANING_TO_DIGIT ===
Each GROUPB_MEANING_TO_DIGIT table element must contain two values: a meaning and a digit. Meanings must not appear more than once in the table.

This table is used to convert meanings to digits for an incoming call. This is mostly used for calls where the other leg belongs to another signaling type, although it will be used if the other leg's signaling type is CAS R2 but the variant is different and there was no match in the [[#GROUPB_OTHER_DIGITS|GROUPB_OTHER_DIGITS]] table.

If a meaning is not found in this table, the meaning corresponding to [[#GROUPB_DEFAULT_DIGIT|GROUPB_DEFAULT_DIGIT]] will be used.

=== GROUPB_DEFAULT_DIGIT ===
This variable must be filled with a valid digit value. It will be used when lookups in the [[#GROUPB_MEANING_TO_DIGIT|GROUPB_MEANING_TO_DIGIT]] table fail.

=== GROUPB_OTHER_DIGITS ===
Each GROUPB_OTHER_DIGITS table element must contain three values: a variant ID and two valid digit values. This table may be empty but cannot be omitted.

This table is used to convert a variant ID and a digit combination to a digit for the current variant in an outgoing call scenario.

== Category conversion information ==
Category conversion information in scripts is used to convert category information between CAS R2 variants and between other signaling types. The following elements must be filled so that these conversions can be completed successfully.

The pattern used to handle categories is identical to the one used for [[#Group B conversion information|group B conversions]].

=== CATEGORY_DIGIT_TO_MEANING ===
Each CATEGORY_DIGIT_TO_MEANING table element must contain two values: a digit and a meaning. The digit must be a valid digit value (1 to 15) and the meaning must be a valid meaning (see [[#Category meanings|Category meanings]]). Each digit must be present in the table.

This table is used to convert category digits to meaning in the context of an outgoing call. For a given digit, the specified meaning will be propagated to the application.

=== CATEGORY_MEANING_TO_DIGIT ===
Each CATEGORY_MEANING_TO_DIGIT table element must contain two values: a meaning and a digit. Meanings must not appear more than once in the table.

This table is used to convert meanings to digits for an incoming call. This is mostly used for calls where the other leg belongs to another signaling type, although it will be used if the other leg's signaling type is CAS R2 but the variant is different and there was no match in the [[#CATEGORY_OTHER_DIGITS|CATEGORY_OTHER_DIGITS]] table.

If a meaning is not found in this table, the meaning corresponding to [[#CATEGORY_DEFAULT_DIGIT|CATEGORY_DEFAULT_DIGIT]] will be used.

=== CATEGORY_DEFAULT_DIGIT ===
This variable must be filled with a valid digit value. It will be used when lookups in the [[#CATEGORY_MEANING_TO_DIGIT|CATEGORY_MEANING_TO_DIGIT]] table fail.

=== CATEGORY_OTHER_DIGITS ===
Each CATEGORY_OTHER_DIGITS table element must contain three values: a variant ID and two valid digit values. This table may be empty but cannot be omitted.

This table is used to convert a variant ID and a digit combination to a digit for the current variant in an outgoing call scenario.

== Event handlers ==
Four register signaling event handlers must be defined in the script.

=== Call context table ===
Each event handler will have a table passed as a parameter. This table is the call context table, and contains critical information about the state of the call. Once calls are created (through Bwd_HandleNewCall and Fwd_HandleNewCall), the script must initialize the table with its own state information. Also, for incoming calls, as they go on, the script must fill the table with the collected signaling information. For outgoing calls, the table already contains this signaling information. The signaling information fields contained in the table are as follows:

{| class="wikitable" border="1" cellspacing="0"
! align="left" | Table element
! align="left" | Description
|-
| DestinationNumber|| String of digits representing the called (destination) number.
|-
| OriginNumber|| String of digits representing the calling (origin) number.
|-
| Category|| Digit representing the calling party category. See [[#Category conversion information|Category conversion information]].
|}

=== Return values ===
Each event handlers must return two values. The first value is the action to take immediately or shortly following the call of the event. It must be set to one of the four following values:

{| class="wikitable" border="1" cellspacing="0"
! align="left" | First return value
! align="left" | Description
|-
| WAIT_SIGNAL || Generally emits the digit provided as the second return value, and waits for the response or the next compelled sequence.
|-
| END_SIGNALING || Ends the call with the provided digit.
|-
| VALIDATE_CALL || (Incoming only) Sends the cumulated signaling information to the application and ends the signaling.
|-
| PULSE_SIGNAL || (Incoming only) Emits a pulsed signal.
|}

The second return value is the digit that accompanies the action, or '''NO_DIGIT''' if no digit must be emitted. Its meaning depends on the first return value.

=== Backward event handlers ===
The following event functions must be defined so that the CAS R2 stack can proceed with register signaling for incoming calls.

==== Bwd_HandleNewCall ====
The Bwd_HandleNewCall event is called whenever an incoming call is discovered on a timeslot. The purpose of this event is for the script to initialize local variables that will be used throughout the tone exchanges.

The event will have the call context table passed as a parameter. This table must be filled with the call state information required by the script for a new incoming call.

The event must return two values as described in the following table:
{| class="wikitable" border="1" cellspacing="0"
! align="left" | First return value
! align="left" | Description
|-
| WAIT_SIGNAL || Waits for the first compelled sequence to be initiated. The second return value must be '''NO_DIGIT'''.
|-
| END_SIGNALING || Ends the call immediately. The second return value must be '''NO_DIGIT'''.
|}

==== Bwd_HandleFwdSignal ====
The Bwd_HandleFwdSignal is called whenever a tone signal has been received, which is most likely the beginning of a compelled sequence.

The event will receive two parameters: the call context table (which was also passed to Bwd_HandleNewCall) and the received tone signal. The event must use the received tone signal to fill the DestinationNumber, OriginNumber and Category fields of the call context table.

The event must return two values as described in the following table:
{| class="wikitable" border="1" cellspacing="0"
! align="left" | First return value
! align="left" | Description
|-
| WAIT_SIGNAL || Waits for the next compelled sequence to be initiated. The second return value must be a valid digit as it is emitted as a response to the current compelled sequence.
|-
| END_SIGNALING || Ends the call immediately. The second return value can either be '''NO_DIGIT''' or a valid digit. If the digit is valid and has an acceptance meaning (see the [[#Group B conversion information|Group B conversion information section]]), the call will be accepted without explicit validation by the application. Otherwise, the call will be refused. In both cases, the digit will be emitted.
|-
| VALIDATE_CALL || Sends the cumulated signaling information to the application. The application will then provide a group B meaning (or digit) that will be sent to the outgoing end.
|-
| PULSE_SIGNAL || Instead of waiting for the next compelled sequence, the event handler will be called again with '''NO_DIGIT''' as a second parameter. The second return value must be a valid digit since it will be sent to complete the current compelled sequence.
|}

=== Forward event handlers ===
The following event functions must be defined so that the CAS R2 stack can proceed with register signaling for outgoing calls.

==== Fwd_HandleNewCall ====
The Fwd_HandleNewCall event is called whenever an outgoing call is initiated on a timeslot. The purpose of this event is for the script to initialize local variables that will be used throughout the tone exchanges.

The event will have the call context table passed as a parameter. This table must be filled with the call state information required by the script for a new incoming call.

The event must return two values as described in the following table:
{| class="wikitable" border="1" cellspacing="0"
! align="left" | First return value
! align="left" | Description
|-
| WAIT_SIGNAL || Waits for the response to the compelled sequence to be received. The second return value can be '''NO_DIGIT''' if a pulsed signal is expected. Otherwise, it must be a valid digit that will be used to initiate the first compelled sequence.
|-
| END_SIGNALING || Ends the call immediately. The second return value must be '''NO_DIGIT'''.
|}

==== Fwd_HandleBwdSignal ====
The Fwd_HandleBwdSignal event is called whenever a tone signal has been received as a response to the current compelled sequence, which is most likely the beginning of a compelled sequence.

The event will receive two parameters: the call context table (which was also passed to Fwd_HandleNewCall) and the received tone signal. The event must use the available DestinationNumber, OriginNumber and Category fields from the call context table to communicate information to the incoming end.

The event must return two values as described in the following table:
{| class="wikitable" border="1" cellspacing="0"
! align="left" | First return value
! align="left" | Description
|-
| WAIT_SIGNAL || Waits for the response to the compelled sequence to be received. The second return value can be '''NO_DIGIT''' if a pulsed signal is expected. Otherwise, it must be a valid digit that will be used to initiate the first compelled sequence.
|-
| END_SIGNALING || Ends the call immediately. The second value must be '''NO_DIGIT'''.
|}

= Optional elements =
The following elements may or may not be defined into a CAS R2 script.

=== SCRIPT_FILE_NAME ===
When affecting a value to this variable, the Toolpack Web Portal will automatically replace the value to the script name followed by the script revision. If the following piece of code is added appended at the end of the script (this is done for default scripts), it will print a trace when loading a CAS R2 stack which can be recovered using an application such as [[Toolpack Application:tblogtrace|tblogtrace]].

=== VARIABLES ===
The VARIABLES table is used to declare variables which can be altered through the Toolpack Web Portal. Typically, variables declared here contain parameters that affect particular trunks. Each row of this table must contain three elements: the variable name, the variable's default value and the variable's description.

= Good practices =
* Use the '''local''' keyword for declaring variables as much as possible. Non-local variables will stay alive outside the scope of event handlers, which in turn will take more CPU time in the form of garbage collection.

= Tone generation module selection =

By default the VoIP DSP devices are used to generate the MFCR2 tones. If the unit has only IVR DSP devices, or for other reasons, this can be changed by adding these lines to the scripts: 
<pre>BWD_VP_GROUP = 0
FWD_VP_GROUP = 0
</pre>
= Script keyword reference =

== Stack variables ==

Here is the list of stack variables. These variables are declared before the VARIABLES table is parsed in a script. A single script affectation will modify the value one of these variables. It is also possible to edit these variables in the Toolpack Web Portal.

{| cellspacing="0" border="1" class="wikitable"
|-
! align="left" | Variable name
! align="left" | Default value
! align="left" | Description
|-
| TIMER_BWD_TONE
| 15000
| Maximum time before a call is closed because the forward end did not emit a tone.
|-
| TIMER_BWD_PULSE
| 250
| Length of a pulsed digit.
|-
| TIMER_BWD_PULSE_GUARD
| 100
| Time between the end of a compelled sequence and the beginning of a pulsed digit.
|-
| TIMER_BWD_PULSE_DETECTION
| 3000
| Time elapsed after a compelled sequence that triggers a pulsed digit.
|-
| TIMER_BWD_ALERT
| 14000
| Maximum wait time for an ALERT API message.
|-
| TIMER_BWD_ANSWER
| 300000
| Maximum wait time for an ANSWER API message.
|-
| TIMER_BWD_FIRST_METERING_INTERVAL
| 150
| Maximum time for which the CAS bits should stay in the ANSWER state before changing to METERING.
|-
| TIMER_BWD_METERING_INTERVAL
| 60000
| Maximum time for which the CAS bits should stay in the ANSWER state before changing to METERING.
|-
| TIMER_BWD_METERING_PULSE
| 150
| Maximum time for which the CAS bits should stay in the METERING state before changing to ANSWER.
|-
| TIMER_BWD_TERMINATE
| 3000
| Maximum wait time for an TERMINATE API message.
|-
| TIMER_BWD_WAIT_CLEAR_FORWARD
| 2000
| Maximum wait time for CLEAR_FORWARD CAS bits change when closing a call.
|-
| TIMER_BWD_HOLD_ANSWER
| 100
| Maximum time for which the CAS bits should stay in the ANSWER state before changing to CLEAR_BACKWARD.
|-
| TIMER_BWD_HOLD_CLEAR_BACKWARD
| 200
| Maximum time for which the CAS bits should stay in the CLEAR_BACKWARD state.
|-
| TIMER_BWD_HOLD_FORCED_RELEASE
| 200
| Maximum time for which the CAS bits should stay in the FORCED_RELEASE state.
|-
| TIMER_BWD_HOLD_IDLE
| 100
| Maximum time for which the CAS bits should stay in the IDLE state before starting a new call on the same timeslot.
|-
| TIMER_FWD_TONE
| 15000
| Maximum time before a call is closed because the forward end did not emit a tone.
|-
| TIMER_FWD_TONE_GUARD
| 0
| Time between the end of a compelled sequence and the beginning of the next one.
|-
| TIMER_FWD_SEIZURE_ACK_BIT
| 200
| Maximum time for which the forward end should wait for a CAS bits SEIZURE_ACKNOWLEDGE state.
|-
| TIMER_FWD_ANSWER_BIT
| 300000
| Maximum time for which the forward end should wait for a CAS bits ANSWER state.
|-
| TIMER_FWD_METERING_PULSE
| 0
| Maximum wait time for an ANSWER CAS bit state to be triggered before a call is terminated.
|-
| TIMER_FWD_TERMINATE
| 3000
| Maximum wait time for an TERMINATE API message.
|-
| TIMER_FWD_HOLD_SEIZURE
| 100
| Maximum time for which the SEIZURE CAS bits state should be held in case of a collision.
|-
| TIMER_FWD_HOLD_IDLE
| 100
| Maximum time for which the CAS bits should stay in the IDLE state before starting a new call on the same timeslot.
|-
| TIMER_CAS_CHANGE
| 20
| Time required for a CAS bits state change to take effect.
|-
| ABCD_PATTERN_IDLE
| 0x1001
| CAS R2 bit pattern for the IDLE state.
|-
| ABCD_PATTERN_BLOCKED
| 0x1101
| CAS R2 bit pattern for the BLOCKED state.
|-
| ABCD_PATTERN_FWD_SEIZURE
| 0x0001
| CAS R2 bit pattern for the forward SEIZURE state.
|-
| ABCD_PATTERN_FWD_CLEAR_FWD
| 0x1001
| CAS R2 bit pattern for the forward CLEAR state.
|-
| ABCD_PATTERN_BWD_SEIZURE_ACK
| 0x1101
| CAS R2 bit pattern for the backward SEIZURE ACKNOWLEDGE state.
|-
| ABCD_PATTERN_BWD_ANSWERED
| 0x0101
| CAS R2 bit pattern for the backward ANSWERED state.
|-
| ABCD_PATTERN_BWD_CLEAR_BWD
| 0x1101
| CAS R2 bit pattern for the backward CLEAR state.
|-
| ABCD_PATTERN_BWD_FORCED_RELEASE
| 0x0001
| CAS R2 bit pattern for the backward FORCED RELEASE state.
|-
| ABCD_PATTERN_MATCH_MASK
| 0x1100
| CAS R2 bit match mask pattern.
|-
| USE_FORCED_RELEASE
| 0
| Flag indicating if the CAS R2 line signaling state machine should use the forced release method.
|-
| BWD_METERING_PULSE_COUNT
| 0
| Number of metering pulses that the CAS R2 line signaling state machine should emit.
|}

== Group B meanings ==

Here is the list of all defined group B meanings:

ACCEPT_CHARGE
ACCEPT_NO_CHARGE
REJECT_UNSPECIFIED
REJECT_UNALLOCATED_NUMBER
REJECT_NO_ROUTE_TO_NETWORK
REJECT_NO_ROUTE_TO_DESTINATION
REJECT_USER_BUSY
REJECT_SUBSCRIBER_ABSENT
REJECT_CALL_REJECTED
REJECT_NUMBER_CHANGED
REJECT_EXCHANGE_ROUTING_ERROR
REJECT_DESTINATION_OUT_OF_ORDER
REJECT_ADDRESS_INCOMPLETE
REJECT_NO_CIRCUIT_AVAILABLE
REJECT_NETWORK_OUT_OF_ORDER
REJECT_TEMPORARY_FAILURE
REJECT_SWITCHING_EQUIP_CONGESTION
REJECT_REQUESTED_CIRCUIT_NOT_AVAILABLE
REJECT_RESOURCE_UNAVAILABLE
REJECT_SERVICE_NOT_AVAILABLE
REJECT_CALL_IDENTITY_DOES_NOT_EXIST
REJECT_CALL_IDENTITY_IN_USE
REJECT_PROTOCOL_ERROR
REJECT_INTERWORKING_UNSPECIFIED

== Category meanings ==

Here is the list of all defined category meanings:

CATEGORY_UNKNOWN
CATEGORY_OPERATOR_FRENCH
CATEGORY_OPERATOR_ENGLISH
CATEGORY_OPERATOR_GERMAN
CATEGORY_OPERATOR_RUSSIAN
CATEGORY_OPERATOR_SPANISH
CATEGORY_SUBSCRIBER
CATEGORY_SUBSCRIBER_WITH_PRIORITY
CATEGORY_DATA
CATEGORY_TEST
CATEGORY_PAYPHONE

= References =

* [http://www.lua.org Lua Programming Language official website]
* [http://en.wikipedia.org/wiki/Line_signaling Line signaling Wikipedia article]
* [http://en.wikipedia.org/wiki/Register_signaling Register signaling Wikipedia article]

[[category:Needs revising]]

Routing script tutorial:Mini Development Guide

2013-06-07T20:37:16Z

Alexandre Tremblay:

== Call object ==

=== Get ===

Those function are 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:

{| cellspacing="1" cellpadding="1" border="1" style="width: 921px; height: 805px;"
|-
! scope="col" | '''Script parameter name'''
! scope="col" | '''ISDN '''
! scope="col" | '''R2 CAS''' 
! scope="col" | '''SS7 '''
! scope="col" | '''SIP '''
! scope="col" | '''Comment '''
! scope="col" | '''Toolpack version '''
|-
| 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 
| N/A 
| Q763: 'Display information' 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 
| 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 
| N/A 
| 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_screening 
| Q931: 'Redirecting number' 1st IE - Screening indicator 
| N/A 
| Q763: 'Redirecting number' IE - screening (national) 
| SIP:Diversion (2nd header) - diversion-screen 
| 
| 
|-
| 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 
| 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_screening 
| Q931: 'Redirecting number' 2nd IE - Screening indicator 
| N/A 
| Q763: 'Redirection number' IE - screening (national) 
| SIP:Diversion (1st header) - diversion-screen 
| 
| 
|-
| 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 
| 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 
| Outgoing leg (write) only 
| 
|-
| nap 
| 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 
|}

 

'''Notice:''' All values are documented in the '''noa_npi_remap.rb''' script and may change between major release.

=== Noa values ===

*<tt>unknown_number</tt>
*<tt>international_number</tt>
*<tt>national_number</tt>
*<tt>subscriber_number</tt>
*<tt>network_specific</tt>
*<tt>network_routing_national_format</tt>
*<tt>network_routing_international_format</tt>
*<tt>abbreviated_number</tt>
*<tt>subscriber_number_operator_requested</tt>
*<tt>national_number_operator_requested</tt>
*<tt>international_number_operator_requested</tt>
*<tt>no_number_present_operator_requested</tt>
*<tt>no_number_present_cut_through_call_to_carrier</tt>
*<tt>test_line_test_code</tt>
*<tt>non_unique_subscriber_number</tt>
*<tt>non_unique_national_number</tt>
*<tt>non_unique_international_number</tt>
*<tt>call_950_number</tt>

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

*<tt>call_params[:called_noa] = 0x70</tt>

or

*<tt>call_params[:called_noa] = 112</tt>

=== Npi values ===

*<tt>unknown_number</tt>
*<tt>isdn</tt>
*<tt>telephony</tt>
*<tt>private</tt>
*<tt>data</tt>
*<tt>telex</tt>
*<tt>national</tt>

=== Calling Display Type values ===

*<tt>unspecified</tt> => Type is unspecified.
*<tt>calling_party_name</tt> => Type is 0xB1.

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

*<tt>call_params[:calling_display_type] = 0xB1</tt>

or

*<tt>call_params[:calling_display_type] = 177</tt>

=== Calling Display value ===

*<tt>call_params[:calling_display] = "Roger Fluffy"</tt>

=== Presentation values ===

*<tt>unspecified</tt>
*<tt>not_available</tt>
*<tt>allowed</tt>
*<tt>restricted</tt>
*<tt>addr_restricted</tt>
*<tt>name_restricted</tt>

=== Calling Party Category ===
values for calling_category
*<tt>unspecified</tt> <tt>(0xa)</tt>
*<tt>unknown</tt> <tt>(0x0)</tt>
*<tt>operator_french</tt> <tt>(0x1)</tt>
*<tt>operator_english</tt> <tt>(0x2)</tt>
*<tt>operator_german</tt> <tt>(0x3)</tt>
*<tt>operator_russian</tt> <tt>(0x4)</tt>
*<tt>operator_spanish</tt> <tt>(0x5)</tt>
*<tt>subscriber</tt> <tt>(0xa)</tt>
*<tt>subscriber_with_priority</tt> <tt>(0xb)</tt>
*<tt>data</tt> <tt>(0xc)</tt>
*<tt>test</tt> <tt>(0xd)</tt>
*<tt>payphone</tt> <tt>(0xf)</tt>

=== Screening values ===

*<tt>unspecified</tt>
*<tt>no</tt>
*<tt>pass</tt>
*<tt>fail</tt>
*<tt>network_provided</tt>

=== Redirecting indicator values ===

SS7:

*<tt>no_redirection</tt>
*<tt>call_rerouted</tt>
*<tt>call_rerouted_all_restricted</tt>
*<tt>call_diverted</tt>
*<tt>call_diverted_all_restricted</tt>
*<tt>call_rerouted_restricted</tt>
*<tt>call_diverted_restricted</tt>
*<tt>spare</tt>

=== Redirecting number reason values ===

ISDN:

*<tt>unknown</tt>
*<tt>busy</tt>
*<tt>no_reply</tt>
*<tt>deflection</tt>
*<tt>dte_out_of_order</tt>
*<tt>forwarding_by_called_dte</tt>
*<tt>unconditional</tt>

SS7:

*<tt>unknown</tt>
*<tt>busy</tt>
*<tt>no_reply</tt>
*<tt>unconditional</tt>
*<tt>deflection</tt>
*<tt>deflection_immediate</tt>
*<tt>mobile_not_reachable</tt>

=== OLI (originating line information) values ===

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

 

=== 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.

 

=== request_uri ===

Enables access to the Request-Line URI. 

For example, if the Request-Line is:
<pre>Request-Line: INVITE sip:4175162082@172.22.45.13:5060;user=phone;transport=udp SIP/2.0</pre>
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: 
<pre> if call_params[:request_uri] && call_params[:request_uri] =~ /sip:(.*)@.*/
call_params[:called] = $1
end
</pre>
=== 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 ===
Any custom sip header can be added to an outgoing call leg: 
<pre>call[ :sip_header ] = ["P-my-custom-header:any value"]</pre>

Many can be added at one time: 
<pre>call[ :sip_header ] = ["P-my-custom-header:value1", "P-my-custom-header2:value2", "P-my-custom-header3:value3"]</pre>

Note: It is not possible to read custom sip headers from inbound call legs. 

== Route parameters ==

All route may have these parameters:

*calling
*called
*nap
*remapped_calling
*remapped_called
*remapped_nap
*remapped_profile

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 or wav files in different states of the call flow, like playing an announcement when the caller dials in. You can also manage the User to User information element.

*announcement_tone
*call_progress_tone
*ring_tone
*max_call_duration
*call_duration_reason
*disconnect_tone
*uui
*uui_forward_enabled

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, :call_progress_tone, :ring_tone, :disconnect_tone) inside bridge parameters are using this format.
<pre>"file1.wav:repeat:start_off:end_off,file2.wav:repeat:start_off:end_off,file3.wav:repeat:start_off:end_off"
</pre>
optional repeat parameter: number of times to play the file (0 and 1 have the same result) optional start_off parameter : Start offset in milliseconds optional end_off parameter: End offset in milliseconds Example: "file1.wav,file2.wav:-1" will play file1.wav one time and then play file2.wav in loop. Example: "file1.wav:0:1000:3000,file2.wav:2:5000:10000" will play file1.wav from second 1 to second 3 then file2.wav from second 5 to second 10 two times. Example: "file1.wav:0:0:30000" will play file1.wav one time for a max duration of 30 seconds.

=== announcement_tone ===

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

Prompt played on the incoming leg with the stream_server before any outgoing call is placed. The outgoing call occurs when the wav file is completed or when the "profile:Busy Tone max duration" is reached.

=== call_progress_tone ===

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

Prompt played after "announcement_tone" when reason is not "ok" or when outgoing call disconnect before answer state. "none" value is used to bypass "profile:Generate Busy (Congestion) Tone" and to play no audio prompt.

=== ring_tone ===

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

Prompt played between alerting state and answer state. Bypass any other ring back tone (RBT) configured in profile.

=== max_call_duration ===

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

Maximum call duration in millisecond for the current bridge. 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.

=== disconnect_tone ===

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

Prompt played (on the incoming leg only) when call duration (:max_call_duration) is reached before terminating the leg with call duration reason (:call_duration_reason).

=== 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 to outgoing call leg 1/true: UUI is forwarded to outgoing call leg 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.

=== Adding audio prompts ===

[[Web_Portal_Tutorial_Guide_v2.6#Routing_script|Adding Audio prompts in v2.6]]

== 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:
** @{DefaultName}: Same file name (without parent folder path) as the one used when empty string is provided
*** 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}: Same path as the one used when empty string is provided
*** Example: "recorded_calls/73EBA698-F3D67B4B-NAP_SS7-IN-5550000-5550001-SS7-TRUNK_BELL_11-24.wav"
*** Example: "recorded_calls/73EBA698-73EBA698-NAP_SIP-OUT-5550000-5550001-SIP-G723-10.3.10.101-1050.wav"
** @{LinkId}: Current LinkId (Id common between all legs of this call bridge)
*** Example: "73EBA698"
** @{LegId}: Current LegId (Unique Id for this leg)
*** Example: "F3D67B4B"
** @{Protocol}: Protocol type of current leg
*** Example: "SS7"
** @{Direction}: Direction of current leg (IN our OUT)
*** Example: "IN"
** @{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"
** @{CURRENT_PKG}: Version of current package
*** Example: "2.6.45"
** @{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"
** @{TBX_GW_PORT}: Current "System Id" (also called "Gateway Port")
*** Example: "12358"

== 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 [[#Refuse|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 [[Routing_script_tutorial:Mini_Development_Guide#Redirecting_number_reason_values|Redirecting number reason values]].

== Reason values ==

Check here for Termination Reason Cause codes: 

[[Termination_cause_codes|Termination Reason Cause codes]]
 

== 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.

#define NAP_STATS_FIELDS \
/* Field, szName, szDescription, szOptions */ \
( SIGNALING_TYPE, "signaling_type", "Signaling type.", "" ) \
( INCOMING_CALL_CNT, "inst_incoming_call_cnt", "Instantaneous Count of incoming calls.", "" ) \
( OUTGOING_CALL_CNT, "inst_outgoing_call_cnt", "Instantaneous Count of outgoing calls.", "" ) \
( AVAILABLE_CNT, "available_cnt", "Number of available circuits or channels.", "" ) \
( UNAVAILABLE_CNT, "unavailable_cnt", "Number of unavailable circuits or channels.", "" ) \
( AVAILABILITY_PCT, "availability_percent", "Percentage of available circuits or channels.", "" ) \
( USAGE_PCT, "usage_percent", "Percentage of used circuits or channels.", "" ) \
( TOTAL_INCOMING_CALL_CNT, "total_incoming_call_cnt", "Total Count of incoming calls.", "" ) \
( ASR_STRUCT, "asr_statistics_struct", "Detailed Answer-Seizure Rate Statistics.", "" ) \
( GLOBAL_ASR_PCT, "global_asr_percent", "Global calculated ASR percentage.", "" ) \
( TOTAL_OUTGOING_CALL_CNT, "total_outgoing_call_cnt", "Total Count of outgoing calls.", "" ) \
( LAST_24H_ASR_PCT, "last_24h_asr_percent", "Last 24 hours calculated ASR percentage.", "" ) \
( LAST_24H_OUTGOING_CALL_CNT, "last_24h_outgoing_call_cnt", "Last 24 hours outgoing calls.", "" ) \
( HOUR_ASR_PCT, "current_hour_asr_percent", "Current hour calculated ASR percentage.", "" ) \
( HOUR_OUTGOING_CALL_CNT, "current_hour_outgoing_call_cnt", "Current hour outgoing calls.", "" ) \
( LAST_HOUR_ASR_PCT, "last_hour_asr_percent", "Last hour calculated ASR percentage.", "" ) \
( LAST_HOUR_OUTGOING_CALL_CNT, "last_hour_outgoing_call_cnt", "Last hour outgoing calls.", "" ) \
( AVAILABILITY_DETECTION_STRUCT, "availability_detection_struct", "Detailed availibility detection Statistics", "" ) \
( POLL_REMOTE_PROXY, "poll_remote_proxy", "Remote proxy polling enabled", "" ) \
( IS_AVAILABLE, "is_available", "Remote proxy actually available or not", "" ) \
( TIME_SINCE_POLLING, "time_since_polling", "Time since the last availibility polling", "" ) \
( TIME_AVAILABLE_SECONDS, "time_available_seconds", "Number of seconds since the NAP is available", "x" ) \
( TIME_UNAVAILABLE_SECONDS, "time_unavailable_seconds", "Number of seconds since the NAP is unavailable", "x" ) \
( REGISTRATION_STRUCT, "registration_struct", "Detailed registration Statistics", "" ) \
( REGISTER_TO_PROXY, "register_to_proxy", "Register to proxy enabled", "" ) \
( IS_REGISTERED, "registered", "Actually registered or not", "" ) \
( TIME_SINCE_REFRESH, "time_since_refresh", "Time since the last refresh", "" ) \
( TIME_REGISTERED_SECONDS, "time_registered_seconds", "Number of seconds since the NAP is registered", "x" ) \
( TIME_NOT_REGISTERED_SECONDS, "time_not_registered_seconds", "Number of seconds since the NAP is not registered", "x" ) \
/*!< Nap Status Fields */

 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.

== 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.

Example: @params = {:bridge => {:announcement_tone, "announcement.wav"}}

 

----

 Back to [[Routing script tutorial|Routing Script Tutorial]].

[[category:Needs revising]]

CAS R2 BACK DIGITS

2013-06-03T21:25:06Z

Alexandre Tremblay: Added dial plan extension

The default CAS R2 scripts included with Toolpack present a way to configure the sequence of tone requests sent by the backward end.

To learn how to set a CAS R2 variable in Toolpack, refer to the latest [[Web_Portal]] Tutorial Guide.

''The digit meanings in this article are the ones defined in the [[CAS R2 ITU specification]]. Digit meanings may vary depending on the variant specification.''

== Compelled sequence and information exchange ==
[[File:Casr2_fwd_bwd_diagram.png]]

In most implementations, the information available from CAS R2 signaling is exchanged through compelled sequences of tones. A compelled sequence consists into the forward end emitting a digit and the backward end responding with a second digit. The backward end confirms reception of the digit by stopping to emit, and waits for the forward end to do so as well to initiate the next sequence. In a way, the forward end is mostly passive: it responds to backward end requests, except when the requests cannot be met, or for the first sequence.

One way of representing such exchanges can be expressed as follows:
Called number: 4506558993
Forward : 4 5 0 6 5 5 8 9 9 3
Backward : 1 1 1 1 1 1 1 1 1 6

In the previous example, the forward end emits the first digit of the called number '4'. The backward end responds with the '1' digit, which request the next called number digit. The exchanges goes on until the backward end has received all the digits it needed, and closed the exchange with the '6' digit, which means the call was accepted.

The incoming end may (and will, most of the time) request the calling number and the calling party category as well:
Called number: 4506558993
Calling number: 4506559511
Calling party category: 1 (Subscriber without priority)
Forward : 4 5 0 6 5 5 8 9 9 3 1 4 5 0 6 5 5 9 5 1 1
Backward : 1 1 1 1 1 1 1 1 1 5 5 5 5 5 5 5 5 5 5 5 6

In this example, the backward end requested the calling party category (using digit '5') after having received all the called number digits. Still using digit '5', it then requested the calling number. Once it was fully received, the backward end accepted the call with digit '6'.

Here is a second example where the backward send an extra '5' when gathering the calling number (The picture also shows the different CAS R2 concepts and TelcoBridges Cas R2 script BACK_DIGITS variable):
Called number: 4506558993
Calling number: 4506559511
Calling party category: 2 (Subscriber without priority)

[[Image:Casr2_bwd_digit_string.png|widthpx|Backward digit string overview]]

Finally, here is a final example:
Called number: 4506558993
Calling number: 12345
Calling party category: 2 (Subscriber with priority)
Forward : 4 5 0 6 5 5 8 9 9 3 2 1 2 3 4 5 F 2
Backward : 1 1 1 1 1 1 1 1 1 5 5 5 5 5 5 5 3 6

In this final example, the backward end sent an extra extra '5' digit, and the backward end responded with 'F' to mark the end of the calling number. Then, the backward end sent digit '3' which represent two things: request of the category again, and also signify to the forward end that the next digit belongs to a different grouping (named "Group B"). Once the category was received, the backward end accepted the call with the '6'.

== The BACK_DIGITS variable ==

In the CAS R2 default scripts found in Toolpack, the BACK_DIGITS variable expresses the digits that must be emitted from the backward end (incoming CAS R2 call). A functioning BACK_DIGITS variable that complies with the second example of the last section would be:
BACK_DIGITS = "111111111555555536"

However, such a sequence has many problems. Firsthand, the called number may be invalid and the call should be refused. To let Toolpack handle the decision of accepting or refusing the call when in group B, the 'V' symbol is used. With the aid of tables found in the script, Toolpack will be able to find the appropriate digit to emit considering the acceptation or refusal of the call.
BACK_DIGITS = "11111111155555553V"

To simplify the variable contents, the 'A' symbol may be used. What it does is send the '5' digit until 'F' is received. The variable would be slimmer:
BACK_DIGITS = "1111111115A3V"

With such a string, the only thing to worry about is the length of the called number. If the called number length may vary, the number of '1's must match the longest possible number. In the case where the forward end receives a '1' when it has sent all its digits, the conventional way of handling this is for the forward end to not emit anything for a while, until the backward end detects this condition and requests something else. In this case, the default scripts in Toolpack will handle this case gracefully and skip the remaining '1's found in the BACK_DIGITS variable.

'''(2.7+ only)''' The 'G' symbol may be used to indicate that '1's should be sent until no digit or a 'F' is received. The BACK_DIGITS thus becomes the following:
BACK_DIGITS = "G5A3V"

'''(2.7.31+ only)''' The 'P' symbol may be used like the 'G' symbol, except that the number of called number digits is determined by the DIAL_PLAN variable, such as the following:
DIAL_PLAN =
{
-- Matching pattern Number of called number digits to request
{ "^12", 6 }, -- Expect 6 called number digits, must be
-- before "^1" because otherwise it wouldn't
-- be matched
{ "^1", 4 } -- Expect 4 digits total
-- If no matching pattern works, an unlimited number of called number digits
-- are requested until the remote equipment stops providing them.
};

[[category:Needs revising]]

CAS R2 BACK DIGITS

2013-04-22T21:07:39Z

Alexandre Tremblay: Added 'G' symbol from release 2.7

Test Call

2013-04-22T18:32:13Z

Alexandre Tremblay:

Starting with release [[TMG-CONTROL_Version_2.7|v2.7]] of [[Toolpack]], it is now possible to simulate incoming calls and issue outgoing calls through the [[Web Portal]] interface.

== Features ==

*'''Extended parameter testing''': It is often hard to reproduce a particular call scenario in a production environment. The Test Call interface provides every routing script parameter available to be customized, allowing to test every corner case possible. Calls can be initiated from and to every [[NAP|Network Access Point]]s available in the system, which lets the user quickly verify the behavior of calls to remote peers.
*'''Easier debugging''': The call flow of all supported protocols is available to the user as it occurs in real time. Audio can be played and recorded to enable the debugging of connection paths efficiently.
*'''Faster deployment''': With Test Call, any call can be instantly instantiated from and to any peer with different settings from the [[Web Portal]] interface. It is now possible to test a Toolpack system without using actual phones to start calls. Using incoming calls, the routing scripts can effectively be tested in an production environment without using an external equipment.

== Configuration ==

[[Web_Portal_Tutorial_Guide_v2.7#Test_Call|Toolpack v2.7: Test Call]]

Test Call

2013-04-22T18:29:00Z

Alexandre Tremblay:

Starting with release [[TMG-CONTROL_Version_2.7|v2.7]] of [[Toolpack]], it is now possible to simulate incoming calls and issue outgoing calls through the [[Web Portal]] interface.

== Features ==

*'''Extended parameter testing''': It is often hard to reproduce a particular call scenario in a production environment. The Test Call interface provides every routing script parameter available to be customized, allowing to test every corner case possible. Calls can be initiated from and to every [[NAP|Network Access Point]]s available in the system, which lets the user quickly verify the behavior of calls to remote peers.
*'''Easier debugging''': The call flow of all supported protocols is available to the user as it occurs in real time. Audio can be played and recorded to enabling the debugging of connection paths efficiently.
*'''Faster deployment''': With Test Call, any call can be instantly instantiated from and to any peer with different settings from the [[Web Portal]] interface. It is now possible to test a Toolpack system without using actual phones to start calls. Using incoming calls, the routing scripts can effectively be tested in an production environment without using an external equipment.

== Configuration ==

[[Web_Portal_Tutorial_Guide_v2.7#Test_Call|Toolpack v2.7: Test Call]]

Test Call

2013-04-22T18:24:22Z

Alexandre Tremblay: Added test call summary.

Starting with release [[TMG-CONTROL_Version_2.7|v2.7]] of [[Toolpack]], it is now possible to simulate incoming calls and issue outgoing calls through the [[Web Portal]] interface.

== Features ==

*'''Extended parameter testing''': It is often hard to reproduce a particular call scenario in a production environment. The Test Call interface provides every routing script parameter available to be customized, allowing to test every corner case possible. Calls can be initiated from and to every [[NAP|Network Access Point]]s available in the system, which lets the user quickly verify the behavior of calls to remote peers.
*'''Easier debugging''': The call flow of all supported protocols is available to the user as it occurs in real time. Audio can be played and recorded to enabling the debugging of connection paths efficiently.
*'''Faster deployment''': With Test Call, any call can be instantly instantiated from and to any peer with different settings from the [[Web Portal]] interface. It is now possible to test a Toolpack system without using actual phones to start calls.

== Configuration ==

[[Web_Portal_Tutorial_Guide_v2.7#Test_Call|Toolpack v2.7: Test Call]]

Toolpack:Configuring RADIUS authorization A

2013-03-07T21:03:15Z

Alexandre Tremblay: Added requires_radius_authorization?.

=== '''''Applies to version(s): v2.7.''''' ===
This page describes how to configure RADIUS authentication and authorization with [[Toolpack]].

1- Select '''Gateway -> Routing script''' from the navigation panel.

[[Image:RoutingScript_0_A.png|border]]

2- '''Edit''' your main script

[[Image:RoutingScript_2_A.png]]

3- Do the following operations in your script:

*'''At the top of the page'''
require 'radius_authorization'
*'''Following your main class definition'''
include RadiusAuthorization
*'''Add before filter in your main class'''
before_filter :method => :radius_authorization
*'''Optional: add the ''fill_authorization_attributes'' method'''
def fill_authorization_attributes(params, auth)
auth[:"User-Name"] = "bob"
...
end
*'''Optional: add the ''requires_radius_authorization?'' method to reduce the scope of the authorization:
def requires_radius_authorization?(params)
case params[:call][:called]
when /^123/
true
...
else
false
end
end
*'''Optional: add methods to handle the possible results of authorization: ''on_radius_authorization_accept'', ''on_radius_authorization_challenge'', ''on_radius_authorization_reject'' and ''on_radius_authorization_timeout'':'''
def on_radius_authorization_accept(params, auth)
log_trace :always, "Access-Accept: #{auth.inspect}"
end

def on_radius_authorization_challenge(params, auth)
log_trace :always, "Access-Challenge: #{auth.inspect}"
raise RoutingException, :call_rejected
end

def on_radius_authorization_reject(params, auth)
log_trace :always, "Access-Reject: #{auth.inspect}"
raise RoutingException, :call_rejected
end

def on_radius_authorization_timeout(params, auth)
log_trace :always, "Authorization Timeout"
raise RoutingException, :call_rejected
end

4- Click 'Save'

== Example ==
The following script configures RADIUS authorization with the default attributes (''User-Name'', ''Calling-Station-Id'' and ''Called-Station-Id''):
<pre>
require 'base_routing'
require 'radius_authorization' # <- Add this line here

class my_script < BaseRouting
include RadiusAuthorization # <- Add this line here

before_filter :method => :radius_authorization # <- Add this line here

route_match :call_field_name => :called
route_match :call_field_name => :calling
route_match :call_field_name => :nap
route_remap :call_field_name => :called, :route_field_name => :remapped_called
route_remap :call_field_name => :calling, :route_field_name => :remapped_calling
route_remap :call_field_name => :nap, :route_field_name => :remapped_nap
end

@@routing = my_script.new

def init_routes( routes )
@@routing.init routes
end

def route( call, nap_list )
@@routing.route call, nap_list
end
</pre>

== Advanced example ==
The following script configures RADIUS authorization with user-defined attributes, and prints attributes found in the Access-Accept message if it is received:
<pre>
require 'base_routing'
require 'radius_authorization' # <- Add this line here

class my_script < BaseRouting
include RadiusAuthorization # <- Add this line here

before_filter :method => :radius_authorization # <- Add this line here

def fill_authorization_attributes(params, auth) # <- Add this line here
call = params[:call] # <- Add this line here
auth[:"User-Name"] = "bob" # <- Add this line here
auth[:"User-Password"] = "hello" # <- Add this line here
auth[:"Calling-Station-Id"] = call[:calling] # <- Add this line here
auth[:"Called-Station-Id"] = call[:called] # <- Add this line here
end # <- Add this line here

def on_radius_authorization_accept(params, auth) # <- Add this line here
log_trace :always, "Access-Accept: #{auth.inspect}" # <- Add this line here
end # <- Add this line here

route_match :call_field_name => :called
route_match :call_field_name => :calling
route_match :call_field_name => :nap
route_remap :call_field_name => :called, :route_field_name => :remapped_called
route_remap :call_field_name => :calling, :route_field_name => :remapped_calling
route_remap :call_field_name => :nap, :route_field_name => :remapped_nap
end

@@routing = my_script.new

def init_routes( routes )
@@routing.init routes
end

def route( call, nap_list )
@@routing.route call, nap_list
end
</pre>

Toolpack:RADIUS CDR attributes C

2013-02-14T20:51:42Z

Alexandre Tremblay: Added Acct-Session-Time

{{DISPLAYTITLE:Toolpack:RADIUS CDR attributes}}
=== '''''Applies to version(s): v2.7''''' ===
This page contains the list of RADIUS attributes that are sent to a RADIUS accounting server.

From RFC 2865 and RFC 2866 (Accounting) :
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack CDR parameter
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 1 
| User-Name
| align="center" | string 
| IncomingNAP
| Incoming NAP name
|-
| align="center" | 4 
| NAS-IP-Address
| align="center" | IP address 
| -
| IP address of the TMedia generating the CDR record
|-
| align="center" | 30 
| Called-Sation-Id
| align="center" | string 
| Called Number
| Called party number
|-
| align="center" | 31 
| Calling-Station-Id
| align="center" | string 
| Calling Number
| Calling party number
|-
| align="center" | 32 
| NAS-Identifier
| align="center" | string 
| Application Name
| Application name of the CDR provider
|-
| align="center" | 40 
| Acct-Status-Type
| align="center" | integer 
| -
| Start or Stop
|-
| align="center" | 44 
| Acct-Session-Id
| align="center" | integer 
| Leg Id
| Call Leg Identifier
|-
| align="center" | 46 
| Acct-Session-Time
| align="center" | integer 
| Session duration
| Duration of the call, in seconds. Corresponds to h323-disconnect-time minus h323-connect-time.
|}

Attribute ID 26, Vendor ID: 9 (Cisco)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 2 
| Cisco-NAS-Port
| align="center" | string 
| NAP name
| Network Access Point name for the call leg
|-
| align="center" | 24 
| h323-conf-id
| align="center" | string 
| Unique Id
| Unique call identifier for the two initial legs (incoming and outgoing) - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 25 
| h323-setup-time
| align="center" | string 
| Start Time
| Represent the call leg setup time - Coordinated Universal Time (UTC)
|-
| align="center" | 26 
| h323-call-origin
| align="center" | string 
| Originator Name
| "originate" for an outgoing leg - "answer" for an incoming leg
|-
| align="center" | 27 
| h323-call-type
| align="center" | string 
| Protocol Type
| If protocol is SIP the value is "VOIP", otherwise it is "Telephony"
|-
| align="center" | 28 
| h323-connect-time
| align="center" | string 
| Connected Time
| Represent the call leg answer time (connect time) - Coordinated Universal Time (UTC)
|-
| align="center" | 29 
| h323-disconnect-time
| align="center" | string 
| EndTime
| Represent the call leg disconnect time - Coordinated Universal Time (UTC)
|-
| align="center" | 30 
| h323-disconnect-cause
| align="center" | string 
| Termination Reason
| Q.931 disconnect (1 to 160) cause, TB Toolpack system cause (200 to 300) and SIP cause (400 to 600)
|-
| align="center" | 35 
| h323-incoming-conf-id
| align="center" | string 
| Unique Id
| Contains the original h323-conf-id in case of call transfer for subsequent outgoing legs - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 115
| release-source
| align="center" | string 
| Termination Source
| "localLeg" if this leg terminate the call or "connectedLeg" if its the connected leg - We use a Cisco string field with our own value definition
|-
|}

Attribute ID 26, Vendor ID: 21776 (TelcoBridges)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 9 
| Telcob-ChargeIndicator
| align="center" | String 
| ChargeIndicator
| Represent the charge indicator value
|-
| align="center" | 10 
| Telcob-Protocol
| align="center" | String 
| Protocol
| Protocol used for this LegID. Ex. SIP, ISDN, etc.
|-
| align="center" | 11 
| Telcob-Codec
| align="center" | String 
| Codec
| Codec used for this LegID
|-
| align="center" | 12 
| Telcob-RemoteIP
| align="center" | IP address 
| RemoteIP
| RemoteIP used for the media. This is only for VOIP.
|-
| align="center" | 13 
| Telcob-RemotePort
| align="center" | Integer 
| RemotePort
| RTP Port of the remote peer. This is only for VOIP.
|-
| align="center" | 14 
| Telcob-TrunkName
| align="center" | String 
| TrunkName
| Name of the trunk (ex. "LS004400E1_0_00"). 
This attribute is only for non-VOIP.
|-
| align="center" | 15 
| Telcob-TimeslotNumber
| align="center" | Integer 
| TimeslotNumber
| This is the Timeslot Number. 
This attribute is only for non-VOIP.
|-
| align="center" | 16 
| Telcob-MediaInfo
| align="center" | String 
| MediaInfo
| String that represents the MediaInfo (ex. "PCMU@10.3.10.124:20054" for SIP and "LS_T1_BITS_00:1" for ISDN)
|-
| align="center" | 17 
| Telcob-StartTime
| align="center" | String 
| StartTime
| String that represents the StartTime of the call.
|-
| align="center" | 18 
| Telcob-ConnectedTime
| align="center" | String 
| ConnectedTime
| String that represents the time when the call was connected
|-
| align="center" | 19 
| Telcob-EndTime
| align="center" | String 
| EndTime
| String that represents when the call was completed.
|-
| align="center" | 20 
| Telcob-TerminationCause
| align="center" | Integer 
| TerminationCause
| Integer that represents the Termination Cause
|-
| align="center" | 21 
| Telcob-Other-Leg-Id
| align="center" | Integer 
| Other-Leg-Id
| Integer that represents the other LegID bridged for this call
|-
| align="center" | 22 
| Telcob-TerminationCauseString
| align="center" | String 
| TerminationCauseString
| String that represents the TerminationCause, but in a String format.
|-
| align="center" | 23 
| Telcob-TerminationSource
| align="center" | String 
| TerminationSource
| String that represents the Termination Source
|-
| align="center" | 24 
| Telcob-LocalSipIP*
| align="center" | String 
| LocalSipIP
| IP Address that represents the Local IP used for SIP.
|-
| align="center" | 25 
| Telcob-LocalSipPort*
| align="center" | Integer 
| LocalSipPort
| Integer that represents the Local port used for SIP.
|-
| align="center" | 26 
| Telcob-LocalMediaIP
| align="center" | IPAddr 
| LocalMediaIP
| IP Address that represents the Local IP used for the media.
|-
| align="center" | 27 
| Telcob-LocalMediaPort
| align="center" | String 
| LocalMediaPort
| Integer that represents the Local port used for the media.
|-
| align="center" | 28 
| Telcob-LocalMediaInfo
| align="center" | String 
| LocalMediaInfo
| String that represents the Local Media Info.(ex. "PCMU@10.3.10.124:20054")
Only for VOIP calls.
|-
| align="center" | 29 
| Telcob-RemoteMediaInfo
| align="center" | String 
| RemoteMediaInfo
| String that represents the Remote Media Info.(ex. "PCMU@10.3.10.124:20054")
This is the same as Telcob-MediaInfo. Only for VOIP calls.
|-
| align="center" | 30 
| Telcob-Alert-Time
| align="center" | Integer 
| AlertTime
| Time where the call has started ringing.
|-
| align="center" | 31 
| Telcob-Redirecting-Number
| align="center" | String 
| RedirectingNumber
| Redirecting number
|-
| align="center" | 32 
| Telcob-Original-Called-Number
| align="center" | String 
| OriginalCalledNumber
| Original called number
|-
| align="center" | 33 
| Telcob-Calling-Presentation
| align="center" | String 
| CallingPresentation
| Calling presentation: "Unspecified", "NotAvailable", "Allowed", "Restricted", "AddressRestricted" or "NameRestricted"
|}
* = Not implemented.

Toolpack:Configuring RADIUS authorization A

2013-01-25T20:33:43Z

Alexandre Tremblay: Raise :call_rejected instead of :no_route

=== '''''Applies to version(s): v2.7.''''' ===
This page describes how to configure RADIUS authentication and authorization with [[Toolpack]].

1- Select '''Gateway -> Routing script''' from the navigation panel.

[[Image:RoutingScript_0_A.png|border]]

2- '''Edit''' your main script

[[Image:RoutingScript_2_A.png]]

3- Do the following operations in your script:

*'''At the top of the page'''
require 'radius_authorization'
*'''Following your main class definition'''
include RadiusAuthorization
*'''Add before filter in your main class'''
before_filter :method => :radius_authorization
*'''Optional: add the ''fill_authorization_attributes'' method'''
def fill_authorization_attributes(params, auth)
auth[:"User-Name"] = "bob"
...
end
*'''Optional: add methods to handle the possible results of authorization: ''on_radius_authorization_accept'', ''on_radius_authorization_challenge'', ''on_radius_authorization_reject'' and ''on_radius_authorization_timeout'':'''
def on_radius_authorization_accept(params, auth)
log_trace :always, "Access-Accept: #{auth.inspect}"
end

def on_radius_authorization_challenge(params, auth)
log_trace :always, "Access-Challenge: #{auth.inspect}"
raise RoutingException, :call_rejected
end

def on_radius_authorization_reject(params, auth)
log_trace :always, "Access-Reject: #{auth.inspect}"
raise RoutingException, :call_rejected
end

def on_radius_authorization_timeout(params, auth)
log_trace :always, "Authorization Timeout"
raise RoutingException, :call_rejected
end

4- Click 'Save'

== Example ==
The following script configures RADIUS authorization with the default attributes (''User-Name'', ''Calling-Station-Id'' and ''Called-Station-Id''):
<pre>
require 'base_routing'
require 'radius_authorization' # <- Add this line here

class my_script < BaseRouting
include RadiusAuthorization # <- Add this line here

before_filter :method => :radius_authorization # <- Add this line here

route_match :call_field_name => :called
route_match :call_field_name => :calling
route_match :call_field_name => :nap
route_remap :call_field_name => :called, :route_field_name => :remapped_called
route_remap :call_field_name => :calling, :route_field_name => :remapped_calling
route_remap :call_field_name => :nap, :route_field_name => :remapped_nap
end

@@routing = my_script.new

def init_routes( routes )
@@routing.init routes
end

def route( call, nap_list )
@@routing.route call, nap_list
end
</pre>

== Advanced example ==
The following script configures RADIUS authorization with user-defined attributes, and prints attributes found in the Access-Accept message if it is received:
<pre>
require 'base_routing'
require 'radius_authorization' # <- Add this line here

class my_script < BaseRouting
include RadiusAuthorization # <- Add this line here

before_filter :method => :radius_authorization # <- Add this line here

def fill_authorization_attributes(params, auth) # <- Add this line here
call = params[:call] # <- Add this line here
auth[:"User-Name"] = "bob" # <- Add this line here
auth[:"User-Password"] = "hello" # <- Add this line here
auth[:"Calling-Station-Id"] = call[:calling] # <- Add this line here
auth[:"Called-Station-Id"] = call[:called] # <- Add this line here
end # <- Add this line here

def on_radius_authorization_accept(params, auth) # <- Add this line here
log_trace :always, "Access-Accept: #{auth.inspect}" # <- Add this line here
end # <- Add this line here

route_match :call_field_name => :called
route_match :call_field_name => :calling
route_match :call_field_name => :nap
route_remap :call_field_name => :called, :route_field_name => :remapped_called
route_remap :call_field_name => :calling, :route_field_name => :remapped_calling
route_remap :call_field_name => :nap, :route_field_name => :remapped_nap
end

@@routing = my_script.new

def init_routes( routes )
@@routing.init routes
end

def route( call, nap_list )
@@routing.route call, nap_list
end
</pre>

RADIUS

2013-01-21T19:16:25Z

Alexandre Tremblay: Added links for RADIUS CDR attribute remapping

Remote Authentication Dial In User Service, more popularly known as RADIUS, is used by telecom service providers for the purpose of authenticating, authorizing, and accounting for the use of services by subscribers. A RADIUS server is an application server that provides this functionality. It can take as input as well as output [[Call detail record]] (CDR) data.

 

== TelcoBridges and RADIUS ==

Starting with release [[Toolpack_version_2.3|v2.3]] of [[Toolpack]], explicit support for the accounting function of RADIUS is now offered. Previously, Toolpack stored [[Call detail record]] (CDR) data in a local database. Starting with Toolpack v2.3, CDR data is stored on a dedicated, external server running an implementation of the RADIUS standard. Configuration of the location of the RADIUS server is performed through the Toolpack web portal.

Starting with release [[TMG-CONTROL_Version_2.6|v2.6]] of [[Toolpack]], multiple RADIUS servers can now be configured for backup purposes.

Starting with release [[TMG-CONTROL_Version_2.7|v2.7]] of [[Toolpack]], calls can now be validated through a RADIUS server with authentication and authorization. The RADIUS server may also change routing parameters for calls.

[[File:Radius_High-level_drawing_v2.jpg]]

=== Prerequisites ===

In order to enable RADIUS functionality in Toolpack, you must have a RADIUS server already up and running. It is highly recommended that the RADIUS server software being running on a separate machine from the one running the Toolpack software. Before configuring Toolpack, you will need the IP address of the RADIUS server(s). You will need to specify a ‘secret key’ which will authenticate the Toolpack server so that it can send accounting, authentication and authorization data and to the RADIUS server and the RADIUS server will accept it.

=== Configuration ===

*[[Web_Portal_Tutorial_Guide_v2.6#CDR|Toolpack v2.6: RADIUS configuration]]
*[[Toolpack:Configuring_RADIUS_A|Toolpack v2.5: RADIUS configuration]]
*[[Toolpack:Configuring_RADIUS_A|Toolpack v2.4: RADIUS configuration]]
*[[Toolpack:Configuring_RADIUS_A|Toolpack v2.3: RADIUS configuration]]

=== Toolpack to RADIUS CDR attributes remapping ===
When Toolpack sends ''Access-Request'' messages to a RADIUS server, some specific attributes are included in the message. These attributes have been improved through Toolpack releases to better meet accounting services requirements.

*[[Toolpack:RADIUS_CDR_attributes_C|Toolpack v2.7: RADIUS CDR attributes]]
*[[Toolpack:RADIUS_CDR_attributes_B|Toolpack v2.6: RADIUS CDR attributes]]
*[[Toolpack:RADIUS_CDR_attributes_A|Toolpack v2.5 and earlier: RADIUS CDR attributes]]

== Dealing with incoherent CDR entries ==
In some situations (during HA switchover for example), some CDR entries may be lost.

The following guide lines provide information on how to deal with these corner cases:

[[Call_Detail_Records_Entry_Loss|Deal with CDR entries loss]]

== References ==

*[[Toolpack:Status_Menus:RADIUS_A|Radius status in TMG web portal]]
*[http://en.wikipedia.org/wiki/RADIUS Radius Wikipedia article]
*[http://www.ietf.org/rfc/rfc2866.txt Link to RFC 2866 – Radius Accounting]
*[http://www.freeradius.net FreeRADIUS website]

[[Category:Glossary]]

Toolpack:RADIUS CDR attributes C

2013-01-21T19:07:41Z

Alexandre Tremblay: User-Name is now the incoming NAP name

=== '''''Applies to version(s): v2.7''''' ===
This page contains the list of RADIUS attributes that are sent to a RADIUS accounting server.

From RFC 2865 and RFC 2866 (Accounting) :
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack CDR parameter
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 1 
| User-Name
| align="center" | string 
| IncomingNAP
| Incoming NAP name
|-
| align="center" | 4 
| NAS-IP-Address
| align="center" | IP address 
| -
| IP address of the TMedia generating the CDR record
|-
| align="center" | 30 
| Called-Sation-Id
| align="center" | string 
| Called Number
| Called party number
|-
| align="center" | 31 
| Calling-Station-Id
| align="center" | string 
| Calling Number
| Calling party number
|-
| align="center" | 32 
| NAS-Identifier
| align="center" | string 
| Application Name
| Application name of the CDR provider
|-
| align="center" | 40 
| Acct-Status-Type
| align="center" | integer 
| -
| Start or Stop
|-
| align="center" | 44 
| Acct-Session-Id
| align="center" | integer 
| Leg Id
| Call Leg Identifier
|}

Attribute ID 26, Vendor ID: 9 (Cisco)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 2 
| Cisco-NAS-Port
| align="center" | string 
| NAP name
| Network Access Point name for the call leg
|-
| align="center" | 24 
| h323-conf-id
| align="center" | string 
| Unique Id
| Unique call identifier for the two initial legs (incoming and outgoing) - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 25 
| h323-setup-time
| align="center" | string 
| Start Time
| Represent the call leg setup time - Coordinated Universal Time (UTC)
|-
| align="center" | 26 
| h323-call-origin
| align="center" | string 
| Originator Name
| "originate" for an outgoing leg - "answer" for an incoming leg
|-
| align="center" | 27 
| h323-call-type
| align="center" | string 
| Protocol Type
| If protocol is SIP the value is "VOIP", otherwise it is "Telephony"
|-
| align="center" | 28 
| h323-connect-time
| align="center" | string 
| Connected Time
| Represent the call leg answer time (connect time) - Coordinated Universal Time (UTC)
|-
| align="center" | 29 
| h323-disconnect-time
| align="center" | string 
| EndTime
| Represent the call leg disconnect time - Coordinated Universal Time (UTC)
|-
| align="center" | 30 
| h323-disconnect-cause
| align="center" | string 
| Termination Reason
| Q.931 disconnect (1 to 160) cause, TB Toolpack system cause (200 to 300) and SIP cause (400 to 600)
|-
| align="center" | 35 
| h323-incoming-conf-id
| align="center" | string 
| Unique Id
| Contains the original h323-conf-id in case of call transfer for subsequent outgoing legs - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 115
| release-source
| align="center" | string 
| Termination Source
| "localLeg" if this leg terminate the call or "connectedLeg" if its the connected leg - We use a Cisco string field with our own value definition
|-
|}

Attribute ID 26, Vendor ID: 21776 (TelcoBridges)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 9 
| Telcob-ChargeIndicator
| align="center" | String 
| ChargeIndicator
| Represent the charge indicator value
|-
| align="center" | 10 
| Telcob-Protocol
| align="center" | String 
| Protocol
| Protocol used for this LegID. Ex. SIP, ISDN, etc.
|-
| align="center" | 11 
| Telcob-Codec
| align="center" | String 
| Codec
| Codec used for this LegID
|-
| align="center" | 12 
| Telcob-RemoteIP
| align="center" | IP address 
| RemoteIP
| RemoteIP used for the media. This is only for VOIP.
|-
| align="center" | 13 
| Telcob-RemotePort
| align="center" | Integer 
| RemotePort
| RTP Port of the remote peer. This is only for VOIP.
|-
| align="center" | 14 
| Telcob-TrunkName
| align="center" | String 
| TrunkName
| Name of the trunk (ex. "LS004400E1_0_00"). 
This attribute is only for non-VOIP.
|-
| align="center" | 15 
| Telcob-TimeslotNumber
| align="center" | Integer 
| TimeslotNumber
| This is the Timeslot Number. 
This attribute is only for non-VOIP.
|-
| align="center" | 16 
| Telcob-MediaInfo
| align="center" | String 
| MediaInfo
| String that represents the MediaInfo (ex. "PCMU@10.3.10.124:20054" for SIP and "LS_T1_BITS_00:1" for ISDN)
|-
| align="center" | 17 
| Telcob-StartTime
| align="center" | String 
| StartTime
| String that represents the StartTime of the call.
|-
| align="center" | 18 
| Telcob-ConnectedTime
| align="center" | String 
| ConnectedTime
| String that represents the time when the call was connected
|-
| align="center" | 19 
| Telcob-EndTime
| align="center" | String 
| EndTime
| String that represents when the call was completed.
|-
| align="center" | 20 
| Telcob-TerminationCause
| align="center" | Integer 
| TerminationCause
| Integer that represents the Termination Cause
|-
| align="center" | 21 
| Telcob-Other-Leg-Id
| align="center" | Integer 
| Other-Leg-Id
| Integer that represents the other LegID bridged for this call
|-
| align="center" | 22 
| Telcob-TerminationCauseString
| align="center" | String 
| TerminationCauseString
| String that represents the TerminationCause, but in a String format.
|-
| align="center" | 23 
| Telcob-TerminationSource
| align="center" | String 
| TerminationSource
| String that represents the Termination Source
|-
| align="center" | 24 
| Telcob-LocalSipIP*
| align="center" | String 
| LocalSipIP
| IP Address that represents the Local IP used for SIP.
|-
| align="center" | 25 
| Telcob-LocalSipPort*
| align="center" | Integer 
| LocalSipPort
| Integer that represents the Local port used for SIP.
|-
| align="center" | 26 
| Telcob-LocalMediaIP
| align="center" | IPAddr 
| LocalMediaIP
| IP Address that represents the Local IP used for the media.
|-
| align="center" | 27 
| Telcob-LocalMediaPort
| align="center" | String 
| LocalMediaPort
| Integer that represents the Local port used for the media.
|-
| align="center" | 28 
| Telcob-LocalMediaInfo
| align="center" | String 
| LocalMediaInfo
| String that represents the Local Media Info.(ex. "PCMU@10.3.10.124:20054")
Only for VOIP calls.
|-
| align="center" | 29 
| Telcob-RemoteMediaInfo
| align="center" | String 
| RemoteMediaInfo
| String that represents the Remote Media Info.(ex. "PCMU@10.3.10.124:20054")
This is the same as Telcob-MediaInfo. Only for VOIP calls.
|-
| align="center" | 30 
| Telcob-Alert-Time
| align="center" | Integer 
| AlertTime
| Time where the call has started ringing.
|-
| align="center" | 31 
| Telcob-Redirecting-Number
| align="center" | String 
| RedirectingNumber
| Redirecting number
|-
| align="center" | 32 
| Telcob-Original-Called-Number
| align="center" | String 
| OriginalCalledNumber
| Original called number
|-
| align="center" | 33 
| Telcob-Calling-Presentation
| align="center" | String 
| CallingPresentation
| Calling presentation: "Unspecified", "NotAvailable", "Allowed", "Restricted", "AddressRestricted" or "NameRestricted"
|}
* = Not implemented.

Toolpack:RADIUS CDR attributes C

2013-01-21T19:06:32Z

Alexandre Tremblay: User-Name is now the incoming NAP name

=== '''''Applies to version(s): v2.7''''' ===
This page contains the list of RADIUS attributes that are sent to a RADIUS accounting server.

From RFC 2865 and RFC 2866 (Accounting) :
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack CDR parameter
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 1 
| User-Name
| align="center" | string 
| -
| Incoming NAP name
|-
| align="center" | 4 
| NAS-IP-Address
| align="center" | IP address 
| -
| IP address of the TMedia generating the CDR record
|-
| align="center" | 30 
| Called-Sation-Id
| align="center" | string 
| Called Number
| Called party number
|-
| align="center" | 31 
| Calling-Station-Id
| align="center" | string 
| Calling Number
| Calling party number
|-
| align="center" | 32 
| NAS-Identifier
| align="center" | string 
| Application Name
| Application name of the CDR provider
|-
| align="center" | 40 
| Acct-Status-Type
| align="center" | integer 
| -
| Start or Stop
|-
| align="center" | 44 
| Acct-Session-Id
| align="center" | integer 
| Leg Id
| Call Leg Identifier
|}

Attribute ID 26, Vendor ID: 9 (Cisco)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 2 
| Cisco-NAS-Port
| align="center" | string 
| NAP name
| Network Access Point name for the call leg
|-
| align="center" | 24 
| h323-conf-id
| align="center" | string 
| Unique Id
| Unique call identifier for the two initial legs (incoming and outgoing) - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 25 
| h323-setup-time
| align="center" | string 
| Start Time
| Represent the call leg setup time - Coordinated Universal Time (UTC)
|-
| align="center" | 26 
| h323-call-origin
| align="center" | string 
| Originator Name
| "originate" for an outgoing leg - "answer" for an incoming leg
|-
| align="center" | 27 
| h323-call-type
| align="center" | string 
| Protocol Type
| If protocol is SIP the value is "VOIP", otherwise it is "Telephony"
|-
| align="center" | 28 
| h323-connect-time
| align="center" | string 
| Connected Time
| Represent the call leg answer time (connect time) - Coordinated Universal Time (UTC)
|-
| align="center" | 29 
| h323-disconnect-time
| align="center" | string 
| EndTime
| Represent the call leg disconnect time - Coordinated Universal Time (UTC)
|-
| align="center" | 30 
| h323-disconnect-cause
| align="center" | string 
| Termination Reason
| Q.931 disconnect (1 to 160) cause, TB Toolpack system cause (200 to 300) and SIP cause (400 to 600)
|-
| align="center" | 35 
| h323-incoming-conf-id
| align="center" | string 
| Unique Id
| Contains the original h323-conf-id in case of call transfer for subsequent outgoing legs - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 115
| release-source
| align="center" | string 
| Termination Source
| "localLeg" if this leg terminate the call or "connectedLeg" if its the connected leg - We use a Cisco string field with our own value definition
|-
|}

Attribute ID 26, Vendor ID: 21776 (TelcoBridges)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 9 
| Telcob-ChargeIndicator
| align="center" | String 
| ChargeIndicator
| Represent the charge indicator value
|-
| align="center" | 10 
| Telcob-Protocol
| align="center" | String 
| Protocol
| Protocol used for this LegID. Ex. SIP, ISDN, etc.
|-
| align="center" | 11 
| Telcob-Codec
| align="center" | String 
| Codec
| Codec used for this LegID
|-
| align="center" | 12 
| Telcob-RemoteIP
| align="center" | IP address 
| RemoteIP
| RemoteIP used for the media. This is only for VOIP.
|-
| align="center" | 13 
| Telcob-RemotePort
| align="center" | Integer 
| RemotePort
| RTP Port of the remote peer. This is only for VOIP.
|-
| align="center" | 14 
| Telcob-TrunkName
| align="center" | String 
| TrunkName
| Name of the trunk (ex. "LS004400E1_0_00"). 
This attribute is only for non-VOIP.
|-
| align="center" | 15 
| Telcob-TimeslotNumber
| align="center" | Integer 
| TimeslotNumber
| This is the Timeslot Number. 
This attribute is only for non-VOIP.
|-
| align="center" | 16 
| Telcob-MediaInfo
| align="center" | String 
| MediaInfo
| String that represents the MediaInfo (ex. "PCMU@10.3.10.124:20054" for SIP and "LS_T1_BITS_00:1" for ISDN)
|-
| align="center" | 17 
| Telcob-StartTime
| align="center" | String 
| StartTime
| String that represents the StartTime of the call.
|-
| align="center" | 18 
| Telcob-ConnectedTime
| align="center" | String 
| ConnectedTime
| String that represents the time when the call was connected
|-
| align="center" | 19 
| Telcob-EndTime
| align="center" | String 
| EndTime
| String that represents when the call was completed.
|-
| align="center" | 20 
| Telcob-TerminationCause
| align="center" | Integer 
| TerminationCause
| Integer that represents the Termination Cause
|-
| align="center" | 21 
| Telcob-Other-Leg-Id
| align="center" | Integer 
| Other-Leg-Id
| Integer that represents the other LegID bridged for this call
|-
| align="center" | 22 
| Telcob-TerminationCauseString
| align="center" | String 
| TerminationCauseString
| String that represents the TerminationCause, but in a String format.
|-
| align="center" | 23 
| Telcob-TerminationSource
| align="center" | String 
| TerminationSource
| String that represents the Termination Source
|-
| align="center" | 24 
| Telcob-LocalSipIP*
| align="center" | String 
| LocalSipIP
| IP Address that represents the Local IP used for SIP.
|-
| align="center" | 25 
| Telcob-LocalSipPort*
| align="center" | Integer 
| LocalSipPort
| Integer that represents the Local port used for SIP.
|-
| align="center" | 26 
| Telcob-LocalMediaIP
| align="center" | IPAddr 
| LocalMediaIP
| IP Address that represents the Local IP used for the media.
|-
| align="center" | 27 
| Telcob-LocalMediaPort
| align="center" | String 
| LocalMediaPort
| Integer that represents the Local port used for the media.
|-
| align="center" | 28 
| Telcob-LocalMediaInfo
| align="center" | String 
| LocalMediaInfo
| String that represents the Local Media Info.(ex. "PCMU@10.3.10.124:20054")
Only for VOIP calls.
|-
| align="center" | 29 
| Telcob-RemoteMediaInfo
| align="center" | String 
| RemoteMediaInfo
| String that represents the Remote Media Info.(ex. "PCMU@10.3.10.124:20054")
This is the same as Telcob-MediaInfo. Only for VOIP calls.
|-
| align="center" | 30 
| Telcob-Alert-Time
| align="center" | Integer 
| AlertTime
| Time where the call has started ringing.
|-
| align="center" | 31 
| Telcob-Redirecting-Number
| align="center" | String 
| RedirectingNumber
| Redirecting number
|-
| align="center" | 32 
| Telcob-Original-Called-Number
| align="center" | String 
| OriginalCalledNumber
| Original called number
|-
| align="center" | 33 
| Telcob-Calling-Presentation
| align="center" | String 
| CallingPresentation
| Calling presentation: "Unspecified", "NotAvailable", "Allowed", "Restricted", "AddressRestricted" or "NameRestricted"
|}
* = Not implemented.

Toolpack:RADIUS CDR attributes C

2013-01-21T19:05:32Z

Alexandre Tremblay: Fixed spacing

=== '''''Applies to version(s): v2.7''''' ===
This page contains the list of RADIUS attributes that are sent to a RADIUS accounting server.

From RFC 2865 and RFC 2866 (Accounting) :
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack CDR parameter
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 1 
| User-Name
| align="center" | string 
| -
| For now this value is hardcoded to "100"
|-
| align="center" | 4 
| NAS-IP-Address
| align="center" | IP address 
| -
| IP address of the TMedia generating the CDR record
|-
| align="center" | 30 
| Called-Sation-Id
| align="center" | string 
| Called Number
| Called party number
|-
| align="center" | 31 
| Calling-Station-Id
| align="center" | string 
| Calling Number
| Calling party number
|-
| align="center" | 32 
| NAS-Identifier
| align="center" | string 
| Application Name
| Application name of the CDR provider
|-
| align="center" | 40 
| Acct-Status-Type
| align="center" | integer 
| -
| Start or Stop
|-
| align="center" | 44 
| Acct-Session-Id
| align="center" | integer 
| Leg Id
| Call Leg Identifier
|}

Attribute ID 26, Vendor ID: 9 (Cisco)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 2 
| Cisco-NAS-Port
| align="center" | string 
| NAP name
| Network Access Point name for the call leg
|-
| align="center" | 24 
| h323-conf-id
| align="center" | string 
| Unique Id
| Unique call identifier for the two initial legs (incoming and outgoing) - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 25 
| h323-setup-time
| align="center" | string 
| Start Time
| Represent the call leg setup time - Coordinated Universal Time (UTC)
|-
| align="center" | 26 
| h323-call-origin
| align="center" | string 
| Originator Name
| "originate" for an outgoing leg - "answer" for an incoming leg
|-
| align="center" | 27 
| h323-call-type
| align="center" | string 
| Protocol Type
| If protocol is SIP the value is "VOIP", otherwise it is "Telephony"
|-
| align="center" | 28 
| h323-connect-time
| align="center" | string 
| Connected Time
| Represent the call leg answer time (connect time) - Coordinated Universal Time (UTC)
|-
| align="center" | 29 
| h323-disconnect-time
| align="center" | string 
| EndTime
| Represent the call leg disconnect time - Coordinated Universal Time (UTC)
|-
| align="center" | 30 
| h323-disconnect-cause
| align="center" | string 
| Termination Reason
| Q.931 disconnect (1 to 160) cause, TB Toolpack system cause (200 to 300) and SIP cause (400 to 600)
|-
| align="center" | 35 
| h323-incoming-conf-id
| align="center" | string 
| Unique Id
| Contains the original h323-conf-id in case of call transfer for subsequent outgoing legs - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 115
| release-source
| align="center" | string 
| Termination Source
| "localLeg" if this leg terminate the call or "connectedLeg" if its the connected leg - We use a Cisco string field with our own value definition
|-
|}

Attribute ID 26, Vendor ID: 21776 (TelcoBridges)
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 9 
| Telcob-ChargeIndicator
| align="center" | String 
| ChargeIndicator
| Represent the charge indicator value
|-
| align="center" | 10 
| Telcob-Protocol
| align="center" | String 
| Protocol
| Protocol used for this LegID. Ex. SIP, ISDN, etc.
|-
| align="center" | 11 
| Telcob-Codec
| align="center" | String 
| Codec
| Codec used for this LegID
|-
| align="center" | 12 
| Telcob-RemoteIP
| align="center" | IP address 
| RemoteIP
| RemoteIP used for the media. This is only for VOIP.
|-
| align="center" | 13 
| Telcob-RemotePort
| align="center" | Integer 
| RemotePort
| RTP Port of the remote peer. This is only for VOIP.
|-
| align="center" | 14 
| Telcob-TrunkName
| align="center" | String 
| TrunkName
| Name of the trunk (ex. "LS004400E1_0_00"). 
This attribute is only for non-VOIP.
|-
| align="center" | 15 
| Telcob-TimeslotNumber
| align="center" | Integer 
| TimeslotNumber
| This is the Timeslot Number. 
This attribute is only for non-VOIP.
|-
| align="center" | 16 
| Telcob-MediaInfo
| align="center" | String 
| MediaInfo
| String that represents the MediaInfo (ex. "PCMU@10.3.10.124:20054" for SIP and "LS_T1_BITS_00:1" for ISDN)
|-
| align="center" | 17 
| Telcob-StartTime
| align="center" | String 
| StartTime
| String that represents the StartTime of the call.
|-
| align="center" | 18 
| Telcob-ConnectedTime
| align="center" | String 
| ConnectedTime
| String that represents the time when the call was connected
|-
| align="center" | 19 
| Telcob-EndTime
| align="center" | String 
| EndTime
| String that represents when the call was completed.
|-
| align="center" | 20 
| Telcob-TerminationCause
| align="center" | Integer 
| TerminationCause
| Integer that represents the Termination Cause
|-
| align="center" | 21 
| Telcob-Other-Leg-Id
| align="center" | Integer 
| Other-Leg-Id
| Integer that represents the other LegID bridged for this call
|-
| align="center" | 22 
| Telcob-TerminationCauseString
| align="center" | String 
| TerminationCauseString
| String that represents the TerminationCause, but in a String format.
|-
| align="center" | 23 
| Telcob-TerminationSource
| align="center" | String 
| TerminationSource
| String that represents the Termination Source
|-
| align="center" | 24 
| Telcob-LocalSipIP*
| align="center" | String 
| LocalSipIP
| IP Address that represents the Local IP used for SIP.
|-
| align="center" | 25 
| Telcob-LocalSipPort*
| align="center" | Integer 
| LocalSipPort
| Integer that represents the Local port used for SIP.
|-
| align="center" | 26 
| Telcob-LocalMediaIP
| align="center" | IPAddr 
| LocalMediaIP
| IP Address that represents the Local IP used for the media.
|-
| align="center" | 27 
| Telcob-LocalMediaPort
| align="center" | String 
| LocalMediaPort
| Integer that represents the Local port used for the media.
|-
| align="center" | 28 
| Telcob-LocalMediaInfo
| align="center" | String 
| LocalMediaInfo
| String that represents the Local Media Info.(ex. "PCMU@10.3.10.124:20054")
Only for VOIP calls.
|-
| align="center" | 29 
| Telcob-RemoteMediaInfo
| align="center" | String 
| RemoteMediaInfo
| String that represents the Remote Media Info.(ex. "PCMU@10.3.10.124:20054")
This is the same as Telcob-MediaInfo. Only for VOIP calls.
|-
| align="center" | 30 
| Telcob-Alert-Time
| align="center" | Integer 
| AlertTime
| Time where the call has started ringing.
|-
| align="center" | 31 
| Telcob-Redirecting-Number
| align="center" | String 
| RedirectingNumber
| Redirecting number
|-
| align="center" | 32 
| Telcob-Original-Called-Number
| align="center" | String 
| OriginalCalledNumber
| Original called number
|-
| align="center" | 33 
| Telcob-Calling-Presentation
| align="center" | String 
| CallingPresentation
| Calling presentation: "Unspecified", "NotAvailable", "Allowed", "Restricted", "AddressRestricted" or "NameRestricted"
|}
* = Not implemented.

Toolpack:RADIUS CDR attributes C

2013-01-21T19:04:41Z

Alexandre Tremblay: Added page

=== '''''Applies to version(s): v2.7''''' ===
This page contains the list of RADIUS attributes that are sent to a RADIUS accounting server.

From RFC 2865 and RFC 2866 (Accounting) :
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack CDR parameter
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 1 
| User-Name
| align="center" | string 
| -
| For now this value is hardcoded to "100"
|-
| align="center" | 4 
| NAS-IP-Address
| align="center" | IP address 
| -
| IP address of the TMedia generating the CDR record
|-
| align="center" | 30 
| Called-Sation-Id
| align="center" | string 
| Called Number
| Called party number
|-
| align="center" | 31 
| Calling-Station-Id
| align="center" | string 
| Calling Number
| Calling party number
|-
| align="center" | 32 
| NAS-Identifier
| align="center" | string 
| Application Name
| Application name of the CDR provider
|-
| align="center" | 40 
| Acct-Status-Type
| align="center" | integer 
| -
| Start or Stop
|-
| align="center" | 44 
| Acct-Session-Id
| align="center" | integer 
| Leg Id
| Call Leg Identifier
|}

Attribute ID 26, Vendor ID: 9 (Cisco)

{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 2 
| Cisco-NAS-Port
| align="center" | string 
| NAP name
| Network Access Point name for the call leg
|-
| align="center" | 24 
| h323-conf-id
| align="center" | string 
| Unique Id
| Unique call identifier for the two initial legs (incoming and outgoing) - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 25 
| h323-setup-time
| align="center" | string 
| Start Time
| Represent the call leg setup time - Coordinated Universal Time (UTC)
|-
| align="center" | 26 
| h323-call-origin
| align="center" | string 
| Originator Name
| "originate" for an outgoing leg - "answer" for an incoming leg
|-
| align="center" | 27 
| h323-call-type
| align="center" | string 
| Protocol Type
| If protocol is SIP the value is "VOIP", otherwise it is "Telephony"
|-
| align="center" | 28 
| h323-connect-time
| align="center" | string 
| Connected Time
| Represent the call leg answer time (connect time) - Coordinated Universal Time (UTC)
|-
| align="center" | 29 
| h323-disconnect-time
| align="center" | string 
| EndTime
| Represent the call leg disconnect time - Coordinated Universal Time (UTC)
|-
| align="center" | 30 
| h323-disconnect-cause
| align="center" | string 
| Termination Reason
| Q.931 disconnect (1 to 160) cause, TB Toolpack system cause (200 to 300) and SIP cause (400 to 600)
|-
| align="center" | 35 
| h323-incoming-conf-id
| align="center" | string 
| Unique Id
| Contains the original h323-conf-id in case of call transfer for subsequent outgoing legs - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 115
| release-source
| align="center" | string 
| Termination Source
| "localLeg" if this leg terminate the call or "connectedLeg" if its the connected leg - We use a Cisco string field with our own value definition
|-
|}

Attribute ID 26, Vendor ID: 21776 (TelcoBridges)

{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Vendor specific attribute ID
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | RADIUS IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 9 
| Telcob-ChargeIndicator
| align="center" | String 
| ChargeIndicator
| Represent the charge indicator value
|-
| align="center" | 10 
| Telcob-Protocol
| align="center" | String 
| Protocol
| Protocol used for this LegID. Ex. SIP, ISDN, etc.
|-
| align="center" | 11 
| Telcob-Codec
| align="center" | String 
| Codec
| Codec used for this LegID
|-
| align="center" | 12 
| Telcob-RemoteIP
| align="center" | IP address 
| RemoteIP
| RemoteIP used for the media. This is only for VOIP.
|-
| align="center" | 13 
| Telcob-RemotePort
| align="center" | Integer 
| RemotePort
| RTP Port of the remote peer. This is only for VOIP.
|-
| align="center" | 14 
| Telcob-TrunkName
| align="center" | String 
| TrunkName
| Name of the trunk (ex. "LS004400E1_0_00"). 
This attribute is only for non-VOIP.
|-
| align="center" | 15 
| Telcob-TimeslotNumber
| align="center" | Integer 
| TimeslotNumber
| This is the Timeslot Number. 
This attribute is only for non-VOIP.
|-
| align="center" | 16 
| Telcob-MediaInfo
| align="center" | String 
| MediaInfo
| String that represents the MediaInfo (ex. "PCMU@10.3.10.124:20054" for SIP and "LS_T1_BITS_00:1" for ISDN)
|-
| align="center" | 17 
| Telcob-StartTime
| align="center" | String 
| StartTime
| String that represents the StartTime of the call.
|-
| align="center" | 18 
| Telcob-ConnectedTime
| align="center" | String 
| ConnectedTime
| String that represents the time when the call was connected
|-
| align="center" | 19 
| Telcob-EndTime
| align="center" | String 
| EndTime
| String that represents when the call was completed.
|-
| align="center" | 20 
| Telcob-TerminationCause
| align="center" | Integer 
| TerminationCause
| Integer that represents the Termination Cause
|-
| align="center" | 21 
| Telcob-Other-Leg-Id
| align="center" | Integer 
| Other-Leg-Id
| Integer that represents the other LegID bridged for this call
|-
| align="center" | 22 
| Telcob-TerminationCauseString
| align="center" | String 
| TerminationCauseString
| String that represents the TerminationCause, but in a String format.
|-
| align="center" | 23 
| Telcob-TerminationSource
| align="center" | String 
| TerminationSource
| String that represents the Termination Source
|-
| align="center" | 24 
| Telcob-LocalSipIP*
| align="center" | String 
| LocalSipIP
| IP Address that represents the Local IP used for SIP.
|-
| align="center" | 25 
| Telcob-LocalSipPort*
| align="center" | Integer 
| LocalSipPort
| Integer that represents the Local port used for SIP.
|-
| align="center" | 26 
| Telcob-LocalMediaIP
| align="center" | IPAddr 
| LocalMediaIP
| IP Address that represents the Local IP used for the media.
|-
| align="center" | 27 
| Telcob-LocalMediaPort
| align="center" | String 
| LocalMediaPort
| Integer that represents the Local port used for the media.
|-
| align="center" | 28 
| Telcob-LocalMediaInfo
| align="center" | String 
| LocalMediaInfo
| String that represents the Local Media Info.(ex. "PCMU@10.3.10.124:20054")
Only for VOIP calls.
|-
| align="center" | 29 
| Telcob-RemoteMediaInfo
| align="center" | String 
| RemoteMediaInfo
| String that represents the Remote Media Info.(ex. "PCMU@10.3.10.124:20054")
This is the same as Telcob-MediaInfo. Only for VOIP calls.
|-
| align="center" | 30 
| Telcob-Alert-Time
| align="center" | Integer 
| AlertTime
| Time where the call has started ringing.
|-
| align="center" | 31 
| Telcob-Redirecting-Number
| align="center" | String 
| RedirectingNumber
| Redirecting number
|-
| align="center" | 32 
| Telcob-Original-Called-Number
| align="center" | String 
| OriginalCalledNumber
| Original called number
|-
| align="center" | 33 
| Telcob-Calling-Presentation
| align="center" | String 
| CallingPresentation
| Calling presentation: "Unspecified", "NotAvailable", "Allowed", "Restricted", "AddressRestricted" or "NameRestricted"
|}
* = Not implemented.

Toolpack:RADIUS CDR attributes B

2013-01-21T18:48:28Z

Alexandre Tremblay: Added page

=== '''''Applies to version(s): v2.6''''' ===
This page contains the list of RADIUS attributes that are sent to a RADIUS accounting server.

From RFC 2865 and RFC 2866 (Accounting) :
{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | AVP Id
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Radius IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 1 
| User-Name
| align="center" | string 
| -
| For now this value is hardcoded to "100"
|-
| align="center" | 4 
| NAS-IP-Address
| align="center" | IP address 
| -
| IP address of the TMedia generating the CDR record
|-
| align="center" | 30 
| Called-Sation-Id
| align="center" | string 
| Called Number
| Called party number
|-
| align="center" | 31 
| Calling-Station-Id
| align="center" | string 
| Calling Number
| Calling party number
|-
| align="center" | 32 
| NAS-Identifier
| align="center" | string 
| Application Name
| Application name of the CDR provider
|-
| align="center" | 40 
| Acct-Status-Type
| align="center" | integer 
| -
| Start or Stop
|-
| align="center" | 44 
| Acct-Session-Id
| align="center" | integer 
| Leg Id
| Call Leg Identifier
|}

AVP: 26 VendorID: 9 (Cisco)

{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | VSA Id
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Radius IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 2 
| Cisco-NAS-Port
| align="center" | string 
| NAP name
| Network Access Point name for the call leg
|-
| align="center" | 24 
| h323-conf-id
| align="center" | string 
| Unique Id
| Unique call identifier for the two initial legs (incoming and outgoing) - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 25 
| h323-setup-time
| align="center" | string 
| Start Time
| Represent the call leg setup time - Coordinated Universal Time (UTC)
|-
| align="center" | 26 
| h323-call-origin
| align="center" | string 
| Originator Name
| "answer" for an outgoing leg - "originate" for an incoming leg
|-
| align="center" | 27 
| h323-call-type
| align="center" | string 
| Protocol Type
| If protocol is SIP the value is "VOIP", otherwise it is "Telephony"
|-
| align="center" | 28 
| h323-connect-time
| align="center" | string 
| Connected Time
| Represent the call leg answer time (connect time) - Coordinated Universal Time (UTC)
|-
| align="center" | 29 
| h323-disconnect-time
| align="center" | string 
| EndTime
| Represent the call leg disconnect time - Coordinated Universal Time (UTC)
|-
| align="center" | 30 
| h323-disconnect-cause
| align="center" | string 
| Termination Reason
| Q.931 disconnect (1 to 160) cause, TB Toolpack system cause (200 to 300) and SIP cause (400 to 600)
|-
| align="center" | 35 
| h323-incoming-conf-id
| align="center" | string 
| Unique Id
| Contains the original h323-conf-id in case of call transfer for subsequent outgoing legs - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 115
| release-source
| align="center" | string 
| Termination Source
| "localLeg" if this leg terminate the call or "connectedLeg" if its the connected leg - We use a Cisco string field with our own value definition
|-
|}

AVP: 26 VendorID: 21776 (TelcoBridges)

{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | VSA Id
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Radius IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 9 
| Telcob-ChargeIndicator
| align="center" | String 
| ChargeIndicator
| Represent the charge indicator value
|-
| align="center" | 10 
| Telcob-Protocol
| align="center" | String 
| Protocol
| Protocol used for this LegID. Ex. SIP, ISDN, etc.
|-
| align="center" | 11 
| Telcob-Codec
| align="center" | String 
| Codec
| Codec used for this LegID
|-
| align="center" | 12 
| Telcob-RemoteIP
| align="center" | IP address 
| RemoteIP
| RemoteIP used for the media. This is only for VOIP.
|-
| align="center" | 13 
| Telcob-RemotePort
| align="center" | Integer 
| RemotePort
| RTP Port of the remote peer. This is only for VOIP.
|-
| align="center" | 14 
| Telcob-TrunkName
| align="center" | String 
| TrunkName
| Name of the trunk (ex. "LS004400E1_0_00"). 
This attribute is only for non-VOIP.
|-
| align="center" | 15 
| Telcob-TimeslotNumber
| align="center" | Integer 
| TimeslotNumber
| This is the Timeslot Number. 
This attribute is only for non-VOIP.
|-
| align="center" | 16 
| Telcob-MediaInfo
| align="center" | String 
| MediaInfo
| String that represents the MediaInfo (ex. "PCMU@10.3.10.124:20054" for SIP and "LS_T1_BITS_00:1" for ISDN)
|-
| align="center" | 17 
| Telcob-StartTime
| align="center" | String 
| StartTime
| String that represents the StartTime of the call.
|-
| align="center" | 18 
| Telcob-ConnectedTime
| align="center" | String 
| ConnectedTime
| String that represents the time when the call was connected
|-
| align="center" | 19 
| Telcob-EndTime
| align="center" | String 
| EndTime
| String that represents when the call was completed.
|-
| align="center" | 20 
| Telcob-TerminationCause
| align="center" | Integer 
| TerminationCause
| Integer that represents the Termination Cause
|-
| align="center" | 21 
| Telcob-Other-Leg-Id
| align="center" | Integer 
| Other-Leg-Id
| Integer that represents the other LegID bridged for this call
|-
| align="center" | 22 
| Telcob-TerminationCauseString
| align="center" | String 
| TerminationCauseString
| String that represents the TerminationCause, but in a String format.
|-
| align="center" | 23 
| Telcob-TerminationSource
| align="center" | String 
| TerminationSource
| String that represents the Termination Source
|-
| align="center" | 24 
| Telcob-LocalSipIP*
| align="center" | String 
| LocalSipIP
| IP Address that represents the Local IP used for SIP.
|-
| align="center" | 25 
| Telcob-LocalSipPort*
| align="center" | Integer 
| LocalSipPort
| Integer that represents the Local port used for SIP.
|-
| align="center" | 26 
| Telcob-LocalMediaIP
| align="center" | IPAddr 
| LocalMediaIP
| IP Address that represents the Local IP used for the media.
|-
| align="center" | 27 
| Telcob-LocalMediaPort
| align="center" | String 
| LocalMediaPort
| Integer that represents the Local port used for the media.
|-
| align="center" | 28 
| Telcob-LocalMediaInfo
| align="center" | String 
| LocalMediaInfo
| String that represents the Local Media Info.(ex. "PCMU@10.3.10.124:20054")
Only for VOIP calls.
|-
| align="center" | 29 
| Telcob-RemoteMediaInfo
| align="center" | String 
| RemoteMediaInfo
| String that represents the Remote Media Info.(ex. "PCMU@10.3.10.124:20054")
This is the same as Telcob-MediaInfo. Only for VOIP calls.
|}
* = Not implemented.

Toolpack:RADIUS CDR attributes A

2013-01-21T18:44:43Z

Alexandre Tremblay: Added page

=== '''''Applies to version(s): v2.5 and earlier''''' ===
This page contains the list of RADIUS attributes that are sent to a RADIUS accounting server.

{| cellpadding="5" border="1" class="wikitable"
|-
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | AVP Id
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Radius IETF param name
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Type
! width="210" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Toolpack param
! width="610" style="background: none repeat scroll 0% 0% rgb(239, 239, 239);" | Description
|-
| align="center" | 4 
| NAS-Identifier
| align="center" | string 
| Application Name
| Application name of the CDR provider
|-
| align="center" | 2 
| Acct-Session-Id
| align="center" | integer 
| Leg Id
| Call Leg Identifier
|-
| align="center" | 21
| Telcob-Other-Leg-Id
| align="center" | integer 
| Other Leg Id
| Call Leg Identifier of the other call leg joined with current call leg
|-
| align="center" | 9
| Telcob-ChargeIndicator
| align="center" | string 
| Charge indicator
| Represent the charge indicator value
|-
| align="center" | 5 
| Cisco-NAS-Port
| align="center" | string 
| NAP name
| Network Access Point name for the call leg
|-
| align="center" | 7 
| Called-Sation-Id
| align="center" | string 
| Called Number
| Called party number
|-
| align="center" | 6 
| Calling-Station-Id
| align="center" | string 
| Calling Number
| Calling party number
|-
| align="center" | 10 
| h323-call-type
| align="center" | string 
| Protocol Type
| If protocol is SIP the value is "VOIP", otherwise it is "Telephony"
|-
| align="center" | 17 
| h323-setup-time
| align="center" | string 
| Start Time
| Represent the call leg setup time - Coordinated Universal Time (UTC)
|-
| align="center" | 8 
| h323-call-origin
| align="center" | string 
| Originator Name
| "answer" for an outgoing leg - "originate" for an incoming leg
|-
| align="center" | 18 
| h323-connect-time
| align="center" | string 
|   Connected Time
| Represent the call leg answer time (connect time) - Coordinated Universal Time (UTC)
|-
| align="center" | 19 
| h323-disconnect-time
| align="center" | string 
|   EndTime
| Represent the call leg disconnect time - Coordinated Universal Time (UTC)
|-
| align="center" | 3 
| h323-conf-id
| align="center" | string 
|   Unique Id
| Unique call identifier for the two initial legs (incoming and outgoing) - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 
| h323-incoming-conf-id
| align="center" | string 
|   Unique Id
| Contains the original h323-conf-id in case of call transfer for subsequent outgoing legs - 128 bits integer formated as xxxxxxxx xxxxxxxx xxxxxxxx xxxxxxxx
|-
| align="center" | 1 
| User-Name
| align="center" | string 
| -
| For now this value is hardcoded to "100"
|-
| align="center" | 30 
| h323-disconnect-cause
| align="center" | string 
| Termination Reason
| Q.931 disconnect (1 to 160) cause, TB Toolpack system cause (200 to 300) and SIP cause (400 to 600)
|-
| align="center" | 115
| release-source
| align="center" | string 
| Termination Source
| "localLeg" if this leg terminate the call or "connectedLeg" if its the connected leg - We use a Cisco string field with our own value definition
|-
| align="center" | 40 
| Acct-Status-Type
| align="center" | integer 
| -
| Start or Stop
|}

Toolpack:Configuring RADIUS authorization A

2013-01-21T14:57:52Z

Alexandre Tremblay: Improved examples

=== '''''Applies to version(s): v2.7.''''' ===
This page describes how to configure RADIUS authentication and authorization with [[Toolpack]].

1- Select '''Gateway -> Routing script''' from the navigation panel.

[[Image:RoutingScript_0_A.png|border]]

2- '''Edit''' your main script

[[Image:RoutingScript_2_A.png]]

3- Do the following operations in your script:

*'''At the top of the page'''
require 'radius_authorization'
*'''Following your main class definition'''
include RadiusAuthorization
*'''Add before filter in your main class'''
before_filter :method => :radius_authorization
*'''Optional: add the ''fill_authorization_attributes'' method'''
def fill_authorization_attributes(params, auth)
auth[:"User-Name"] = "bob"
...
end
*'''Optional: add methods to handle the possible results of authorization: ''on_radius_authorization_accept'', ''on_radius_authorization_challenge'', ''on_radius_authorization_reject'' and ''on_radius_authorization_timeout'':'''
def on_radius_authorization_accept(params, auth)
log_trace :always, "Access-Accept: #{auth.inspect}"
end

def on_radius_authorization_challenge(params, auth)
log_trace :always, "Access-Challenge: #{auth.inspect}"
raise RoutingException, :no_route
end

def on_radius_authorization_reject(params, auth)
log_trace :always, "Access-Reject: #{auth.inspect}"
raise RoutingException, :no_route
end

def on_radius_authorization_timeout(params, auth)
log_trace :always, "Authorization Timeout"
raise RoutingException, :no_route
end

4- Click 'Save'

== Example ==
The following script configures RADIUS authorization with the default attributes (''User-Name'', ''Calling-Station-Id'' and ''Called-Station-Id''):
<pre>
require 'base_routing'
require 'radius_authorization' # <- Add this line here

class my_script < BaseRouting
include RadiusAuthorization # <- Add this line here

before_filter :method => :radius_authorization # <- Add this line here

route_match :call_field_name => :called
route_match :call_field_name => :calling
route_match :call_field_name => :nap
route_remap :call_field_name => :called, :route_field_name => :remapped_called
route_remap :call_field_name => :calling, :route_field_name => :remapped_calling
route_remap :call_field_name => :nap, :route_field_name => :remapped_nap
end

@@routing = my_script.new

def init_routes( routes )
@@routing.init routes
end

def route( call, nap_list )
@@routing.route call, nap_list
end
</pre>

== Advanced example ==
The following script configures RADIUS authorization with user-defined attributes, and prints attributes found in the Access-Accept message if it is received:
<pre>
require 'base_routing'
require 'radius_authorization' # <- Add this line here

class my_script < BaseRouting
include RadiusAuthorization # <- Add this line here

before_filter :method => :radius_authorization # <- Add this line here

def fill_authorization_attributes(params, auth) # <- Add this line here
call = params[:call] # <- Add this line here
auth[:"User-Name"] = "bob" # <- Add this line here
auth[:"User-Password"] = "hello" # <- Add this line here
auth[:"Calling-Station-Id"] = call[:calling] # <- Add this line here
auth[:"Called-Station-Id"] = call[:called] # <- Add this line here
end # <- Add this line here

def on_radius_authorization_accept(params, auth) # <- Add this line here
log_trace :always, "Access-Accept: #{auth.inspect}" # <- Add this line here
end # <- Add this line here

route_match :call_field_name => :called
route_match :call_field_name => :calling
route_match :call_field_name => :nap
route_remap :call_field_name => :called, :route_field_name => :remapped_called
route_remap :call_field_name => :calling, :route_field_name => :remapped_calling
route_remap :call_field_name => :nap, :route_field_name => :remapped_nap
end

@@routing = my_script.new

def init_routes( routes )
@@routing.init routes
end

def route( call, nap_list )
@@routing.route call, nap_list
end
</pre>

Toolpack:Configuring RADIUS authorization A

2013-01-21T14:45:45Z

Alexandre Tremblay: /* Example */

Adding a custom RADIUS dictionary A

2013-01-21T14:40:16Z

Alexandre Tremblay: moved Adding a custom RADIUS dictionary A to Toolpack:Adding a custom RADIUS dictionary A: Added Toolpack:

#REDIRECT [[Toolpack:Adding a custom RADIUS dictionary A]]

Toolpack:Adding a custom RADIUS dictionary A

2013-01-21T14:40:16Z

Alexandre Tremblay: moved Adding a custom RADIUS dictionary A to Toolpack:Adding a custom RADIUS dictionary A: Added Toolpack:

=== '''''Applies to version(s): v2.7.''''' ===
This page describes how to add a custom RADIUS dictionary to use vendor-specific RADIUS attributes.

1- Select '''Gateway -> File Db''' from the navigation panel.

[[Image:FileDB 0 A.png|border]]

2- In the '''RADIUS dictionaries''' section, click '''Import new RADIUS dictionary'''.

[[Image:RADIUS-file-db-dictionary-2_7.PNG|border]]

3- Click the '''Browse...''' button to select the RADIUS dictionary to add. When the file is selected, click '''Import'''.

[[Image:RADIUS-file-db-dictionary-import-2_7.PNG|border]]

4- Verify that the '''File was successfully imported.''' message appears.

Toolpack:Configuring RADIUS authorization A

2013-01-18T23:55:34Z

Alexandre Tremblay: Created page with "=== '''''Applies to version(s): v2.7.''''' === This page describes how to configure RADIUS authentication and authorization with Toolpack. 1- Select '''Gateway -> Routing..."

File:RADIUS-file-db-dictionary-import-2 7.PNG

2013-01-18T23:39:40Z

Alexandre Tremblay:

File:RADIUS-file-db-dictionary-2 7.PNG

2013-01-18T23:39:25Z

Alexandre Tremblay: uploaded a new version of "File:RADIUS-file-db-dictionary-2 7.PNG"

File:RADIUS-file-db-dictionary-2 7.PNG

2013-01-18T23:38:48Z

Alexandre Tremblay:

Toolpack:Adding a custom RADIUS dictionary A

2013-01-18T23:38:11Z

Alexandre Tremblay:

Toolpack:Adding a custom RADIUS dictionary A

2013-01-18T23:37:44Z

Alexandre Tremblay: Created page with "=== '''''Applies to version(s): v2.7.''''' === This page describes how to add a custom RADIUS dictionary to use vendor-specific RADIUS attributes. 1- Select '''Gateway -> Fil..."

=== '''''Applies to version(s): v2.7.''''' ===
This page describes how to add a custom RADIUS dictionary to use vendor-specific RADIUS attributes.

1- Select '''Gateway -> File Db''' from the navigation panel.

[[FileDB 0 A.png|border]]

2- In the '''RADIUS dictionaries''' section, click '''Import new RADIUS dictionary'''.

[[RADIUS-file-db-dictionary-2_7.PNG|border]]

3- Click the '''Browse...''' button to select the RADIUS dictionary to add. When the file is selected, click '''Import'''.

[[RADIUS-file-db-dictionary-import-2_7.PNG|border]]

4- Verify that the '''File was successfully imported.''' message appears.

Toolpack:Adding RADIUS server B

2013-01-18T20:57:53Z

Alexandre Tremblay:

=== '''''Applies to version(s): v2.7''''' ===
{{DISPLAYTITLE:Toolpack: Adding RADIUS Server(s)}}
Multiple RADIUS servers can be configured to provide RADIUS redundancy.

1- Select '''RADIUS''' from the navigation panel.

[[Image:Radius_Config_left_panel.jpg|border]]

2- In the '''Editing RADIUS client configuration''' page, select '''Create new RADIUS server'''

[[Image:RADIUS-client-create-server-2_7.PNG|border]]

3- Configure the RADIUS server with these parameters:
*'''Name''' to recognize the server.
*'''Server IP address''' is the IP address of the RADIUS server.
*'''Server UDP port''' is the UDP port of the RADIUS server.
*'''Timeout''' is the number of milliseconds to wait for a response from the RADIUS server before retrying.
*'''Retries''' is the number of times to send the same request again when the timeout is reached. See full details on functionality here: [[CDR redundancy]]
*'''Server type''' is the type of RADIUS server this configuration corresponds to. The choices are '''Accounting''' for CDR functionality and '''Authorization''' for authentication and authorization.
*'''Use polling (Status-Server)''' dictates if polling with the Status-Server message is supported by this server. When enabled, this option allows the RADIUS client to recognize quickly and efficiently if the RADIUS server is available or not.
*'''Server secret''' is the passphrase to authenticate the TMG on the RADIUS server.

[[Image:RADIUS-server-configuration-2_7.PNG|border]]

4- You can configure multiple RADIUS servers to have RADIUS redundancy. Each entry can have different settings.

[[Image:RADIUS-client-configuration-2_7.PNG|border]]

File:RADIUS-server-configuration-2 7.PNG

2013-01-18T20:57:12Z

Alexandre Tremblay:

Toolpack:Adding RADIUS server B

2013-01-18T20:57:01Z

Alexandre Tremblay:

=== '''''Applies to version(s): v2.7''''' ===
{{DISPLAYTITLE:Toolpack: Adding RADIUS Server(s)}}
Multiple RADIUS servers can be configured to provide RADIUS redundancy.

1- Select '''RADIUS''' from the navigation panel.

[[Image:Radius_Config_left_panel.jpg|border]]

2- In the '''Editing RADIUS client configuration''' page, select '''Create new RADIUS server'''

[[Image:RADIUS-client-create-server-2_7.PNG|border]]

3- Configure the RADIUS server with these parameters:
*'''Name''' to recognize the server.
*'''Server IP address''' is the IP address of the RADIUS server.
*'''Server UDP port''' is the UDP port of the RADIUS server.
*'''Timeout''' is the number of milliseconds to wait for a response from the RADIUS server before retrying.
*'''Retries''' is the number of times to send the same request again when the timeout is reached. See full details on functionality here: [[CDR redundancy]]
*'''Server type''' is the type of RADIUS server this configuration corresponds to. The choices are '''Accounting''' for CDR functionality and '''Authorization''' for authentication and authorization.
*'''Use polling (Status-Server)''' dictates if polling with the Status-Server message is supported by this server. When enabled, this option allows the RADIUS client to recognize quickly and efficiently if the RADIUS server is available or not.
*'''Server secret''' is the passphrase to authenticate the TMG on the RADIUS server.

[[Image:RADIUS-server-configuration-2_7.PNG|border]]

4- You can configure multiple RADIUS servers to have RADIUS redundancy. Each entry can have different timeouts and retries.

[[Image:RADIUS-client-configuration-2_7.PNG|border]]

Toolpack:Adding RADIUS server A

2013-01-18T20:49:20Z

Alexandre Tremblay:

=== '''''Applies to version(s): v2.6''''' ===
{{DISPLAYTITLE:Toolpack: Adding RADIUS Server(s)}}
Multiple RADIUS servers can be configured to provide RADIUS redundancy.

1- Select '''RADIUS''' from the navigation panel.

[[Image:Radius_Config_left_panel.jpg|border]]

2- In the '''Editing RADIUS client configuration''' page, select '''Create new RADIUS server'''

[[Image:Radius_0_1_A.png|border]]

3- Configure the RADIUS server with these parameters:
*'''Name''' to recognize the server.
*'''RADIUS server address and port''' is the IP address and port of the RADIUS server we want to access. The format is x.x.x.x for the IP and you can add the port with a colon if it is different than the default port 1813 (for example 10.10.10.10:12345)
*'''RADIUS timeout''' is the request timeout in milliseconds for each request to this server.
*'''RADIUS retries''' is the number of retries the RADIUS client will try before falling back to other methods. See full details on functionality here: [[CDR redundancy|CDR redundancy]]
*'''Server secret''' is the passphrase to authenticate the TMG on the RADIUS server.

[[Image:Radius_client_configuration.jpg|border]]

4- You can configure multiple RADIUS servers to have RADIUS redundancy. Each entry can have different settings.

[[Image:Radius_Configuration_2_6.jpg|border]]

Toolpack:Adding RADIUS server A

2013-01-18T20:48:45Z

Alexandre Tremblay:

=== '''''Applies to version(s): v2.6''''' ===
{{DISPLAYTITLE:Toolpack: Adding RADIUS Server(s)}}
Multiple RADIUS servers can be configured to provide RADIUS redundancy.

1- Select '''RADIUS''' from the navigation panel.

[[Image:Radius_Config_left_panel.jpg|border]]

2- In the '''Editing RADIUS client configuration''' page, select '''Create new RADIUS server'''

[[Image:Radius_0_1_A.png|border]]

3- Configure the RADIUS server with these parameters:
*'''Name''' to recognize the server.
*'''RADIUS server address and port''' is the IP address and port of the RADIUS server we want to access. The format is x.x.x.x for the IP and you can add the port with a colon if it is different than the default port 1813 (for example 10.10.10.10:12345)
*'''RADIUS timeout''' is the request timeout in milliseconds for each request to this server.
*'''RADIUS retries''' is the number of retries the RADIUS client will try before falling back to other methods. See full details on functionality here: [[CDR redundancy|CDR redundancy]]
*'''Server secret''' is the passphrase to authenticate the TMG on the RADIUS server.

[[Image:Radius_old_image.jpg|border]]

4- You can configure multiple RADIUS servers to have RADIUS redundancy. Each entry can have different settings.

[[Image:Radius_Configuration_2_6.jpg|border]]

File:RADIUS-client-create-server-2 7.PNG

2013-01-18T19:17:19Z

Alexandre Tremblay:

Toolpack:Adding RADIUS server B

2013-01-18T19:17:13Z

Alexandre Tremblay:

File:RADIUS-client-configuration-2 7.PNG

2013-01-18T19:15:38Z

Alexandre Tremblay:

Toolpack:Adding RADIUS server B

2013-01-18T19:15:28Z

Alexandre Tremblay: Created page with "=== '''''Applies to version(s): v2.7''''' === {{DISPLAYTITLE:Toolpack: Adding RADIUS Server(s)}} Multiple RADIUS servers can be configured to provide RADIUS redundancy. 1- S..."

Toolpack:Configuring RADIUS client B

2013-01-18T19:02:31Z

Alexandre Tremblay:

=== '''''Applies to version(s): v2.7''''' ===
{{DISPLAYTITLE:Toolpack: Configuring RADIUS}}
This page describes how to configure a TMG system to communicate with RADIUS servers. RADIUS requests will be sent from the TMG management port (mgmt0) or from the [[Virtual IP Interfaces|Virtual IP Interfaces]] (if they are configured).

1- Select '''RADIUS''' from the navigation panel.

[[Image:Radius_Config_left_panel.jpg|border]]

2- In the '''Editing RADIUS client configuration''' page, you can modify some general parameters:
*'''Timezone''' will use the local timezone or the GMT timezone in the RADIUS attributes
*'''Time format''' describes how to display the time in human readable format in the RADIUS attributes. This is based on the [http://php.net/manual/en/function.strftime.php strftime function]
*In Advanced Parameters you have:
**'''Maximum accounting concurrent records''' is the maximum number of simultaneous accounting records to be sent to the RADIUS servers. This improves the RADIUS performance of the system: it allows the RADIUS client to deal with servers that have high latency relative to the number of request/responses per second to be achieved. Default value is 10 records.
**'''Maximum accounting queued records''' is the maximum number of accounting records to queue when the maximum number of pending records has been reached. When the maximum number of queued messages is reached, CDR records will be dropped. To prevent this, make sure the system supports [[CDR redundancy|CDR redundancy]]. Default value is 12000 records.
**'''Maximum authorization concurrent requests''' is the maximum number of simultaneous authorization requests to be sent to the RADIUS servers. This improves the RADIUS performance of the system: it allows the RADIUS client to deal with servers that have high latency relative to the number of request/responses per second to be achieved. Default value is 200 requests.
**'''Maximum authorization queued requests''' is the maximum number of authorization requests to queue when the maximum number of pending requests has been reached. When the maximum number of queued messages is reached, authorization requests will be dropped. Default value is 12000 requests.
**'''Additional dictionary''' is an additional dictionary that provides the definition of custom RADIUS attributes.
** Click '''Save''' to save your changes.

[[Image:RADIUS-client-config-2_7.PNG|border]]