TBwiki - User contributions [en]

User talk:Danny Bouthot

2020-10-07T17:24:16Z

Cbilodeau: Welcome!

'''Welcome to ''TBwiki''!'''
We hope you will contribute much and well.
You will probably want to read the [[Help:Contents|help pages]].
Again, welcome and have fun! [[User:Cbilodeau|Cbilodeau]] 13:24, 7 October 2020 (EDT)

User:Danny Bouthot

2020-10-07T17:24:15Z

Cbilodeau: Creating user page with biography of new user.

Passionate software designer with a demonstrated history working in the computer software industry. Skilled in Python, C, C++, and people management. Strong people skills with a bachalor's degree in electrical engineering with a minor in computer science.

Team player always ready help and to learn new skills.

Passionate software designer with a demonstrated history working in the computer software industry. Skilled in Python, C, C++, and people management. Strong people skills with a bachalor's degree in electrical engineering with a minor in computer science.

Team player always ready help and to learn new skills.
Passionate software designer with a demonstrated history working in the computer software industry. Skilled in Python, C, C++, and people management. Strong people skills with a bachalor's degree in electrical engineering with a minor in computer science.

Team player always ready help and to learn new skills.
Passionate software designer with a demonstrated history working in the computer software industry. Skilled in Python, C, C++, and people management. Strong people skills with a bachalor's degree in electrical engineering with a minor in computer science.

Team player always ready help and to learn new skills.
Passionate software designer with a demonstrated history working in the computer software industry. Skilled in Python, C, C++, and people management. Strong people skills with a bachalor's degree in electrical engineering with a minor in computer science.

Team player always ready help and to learn new skills.
Passionate software designer with a demonstrated history working in the computer software industry. Skilled in Python, C, C++, and people management. Strong people skills with a bachalor's degree in electrical engineering with a minor in computer science.

Team player always ready help and to learn new skills.

User talk:Marc-André Gauthier

2020-09-29T19:00:14Z

Cbilodeau: Welcome!

User:Marc-André Gauthier

2020-09-29T19:00:14Z

Cbilodeau: Creating user page with biography of new user.

Until recently, the prevailing view assumed lorem ipsum was born as a nonsense text. “It's not Latin, though it looks like it, and it actually says nothing,” Before & After magazine answered a curious reader, “Its ‘words’ loosely approximate the frequency with which letters occur in English, which is why at a glance it looks pretty real.”

As Cicero would put it, “Um, not so fast.”

The placeholder text, beginning with the line “Lorem ipsum dolor sit amet, consectetur adipiscing elit”, looks like Latin because in its youth, centuries ago, it was Latin.

Richard McClintock, a Latin scholar from Hampden-Sydney College, is credited with discovering the source behind the ubiquitous filler text. In seeing a sample of lorem ipsum, his interest was piqued by consectetur—a genuine, albeit rare, Latin word. Consulting a Latin dictionary led McClintock to a passage from De Finibus Bonorum et Malorum (“On the Extremes of Good and Evil”), a first-century B.C. text from the Roman philosopher Cicero.

H.248

2020-09-21T19:44:49Z

Cbilodeau: reformulate confusing sentence

Also known as Megaco, H.248 is a protocol for controlling media gateways via a [[Softswitch|softswitch]]/[[Softswitch|Media Gateway Controller (MGC)]].

 

== TelcoBridges and H.248 ==

[[Toolpack]] now supports the H.248 protocol.

As indicated in the diagram below, the H.248 stack is a module in the media gateway application residing on the application server. It communicates with the various Toolpack classes (CAFClass, [[CMC library|CMC Library]], CMC Class) as required via a new [[CAF: Working With Call Legs|Media Leg API]]. In turn, the media gateway application communicates with the softswitch / media gateway controller via the H.248 protocol.

 [[Image:H.248 schematic.jpg|H.248 schematic.jpg]]

 

=== H.248 package support ===

[[Toolpack]] supports the following packages.

*'''H.248.1'''
**Protocol v2
**IPSec connection support
**Support UDP (2944 or user-defined)

 

*'''H.248.1 - Basic packages'''
**Generic
**Base Root
**Tone generator (for extension purposes only)
**Tone detection (for extension purposes only)
**Basic DTMF generation
**DTMF detection
**Call Progress Tones Generator (Ringing, Busy, Congestion and Warning tones supported)
**Basic continuity package
**Network package
**RTP package
**TDM Circuit package (only echo-cancellation)

 

*'''H.248.2 Facsimile, text conversation and call discrimination packages'''
**Fax/TextPhone/Modem tones detection (fax and data only)
**Call type discrimination package (fax and data only)
**IP Fax (Event: Fax Connection Change - end of fax session, call terminating)

 

*'''H.248.4 Transport over Stream Control Transmission Protocol (SCTP)'''
** Partial support
 

*'''H.248.7 Generic announcement package'''
**Generic announcement package (Fix Announcement play only)

 

*'''H.248.10 Media gateway resource congestion handling package'''
**Congestion handling package

 

*'''H.248.11 Media gateway overload control package'''
**Overload control package

 

*'''H.248.13 Quality alert ceasing package'''
**Quality alert ceasing package

 

*'''H.248.14 Inactivity timer package'''
**Inactivity timer package

 

*'''H.248.16 Enhanced digit collection packages and procedures'''
**Extended DTMF detection package
**Enhanced DTMF detection package

 

*'''H.248.23 Enhanced alerting packages'''
**Enhanced alerting package
**Analogue Display Signalling package
 

*'''H.248.25 Basic CAS packages'''
**Basic CAS package
**Basic CAS addressing
 

*'''H.248.27 Supplemental tones packages'''
**Carrier tones generation package
**Diagnostic tones generation package
 

*'''H.248.28 International CAS packages'''
**International CAS
**CAS blocking

 

*'''H.248.29 International CAS compelled register signaling packages'''
** Annex A.1 only in 'enbloc' compelled mode (International CAS Compelled package - No digitmap control: Digitmaps are preconfigured on MG)

 

*'''Q.1950 packages'''
**Basic call progress tones generator package
**Expanded call progress tones generator package
**Basic services tones generator package

 

== Additional details ==

*Support for text mode only

== Configuration ==
*[[Toolpack:Protocol_Stack_Settings_D#H.248_MEGACO|Web Portal v3.0: H.248 Configuration]]
*[[Toolpack:Protocol_Stack_Settings_C#H.248_MEGACO|Web Portal v2.10: H.248 Configuration]]
*[[Toolpack:Protocol_Stack_Settings_B#H.248_MEGACO|Web Portal v2.9: H.248 Configuration]]
*[[Toolpack:Protocol_Stack_Settings_A#H.248_MEGACO|Web Portal v2.8: H.248 Configuration]]
<div class="mw-collapsible mw-collapsed" data-collapsetext="other versions" data-expandtext="Click here for other versions" style="width: 400px;">
*[[Web_Portal_Tutorial_Guide_v2.7#H.248|Web Portal v2.7: H.248 Configuration]]
*[[Toolpack_v2.5:Creating_an_H.248_Configuration|Configuring H.248 on TMG devices (v2.6 and before)]]
</div>

== References ==

*[http://en.wikipedia.org/wiki/Megaco Wikipedia article]

[[Category:Glossary]]
[[Category:Tmedia_Features|H.248/Megaco]]
[[Category:Revise on Major]]

Sip headers manipulation

2020-08-07T17:29:31Z

Cbilodeau: sort headers alphabetically

The behavior 'SIP headers manipulation' allows the SBC users to freely add ,remove and modify the sip headers when SBC forwards the received invite, 18x and 200 sip messages (there are limitations, please refer to the limitations section below).

== Configuration: ==
To use the behavior, you need to select the option "Enable SIP headers manipulation" under the path "Gateway -> Advanced Parameters" and use the filter script "sip_headers_manipulation.rb" in routing script setting.

== Usage: ==
The manipulation actions are based on what are defined in "sip_headers_manipulation.rb" dynamically or statically. In sip_headers_manipulation.rb, there is filter_sip_headers_manipulation that should be applied as 'after filter' in main script and need to use call[:manipulation] to save what you want SBC to do regarding headers manipulation. An example is :

def filter_sip_headers_manipulation params

# The following 4 lines are needed to initialize the hash table
call = params[ :call ]
call[:manipulation] = {}
call[:manipulation][:invite] = {}
call[:manipulation]["18x"] = {}
call[:manipulation]["200"] = {}

# Will remove "P-Charging-Vector" header from the the invite if exists
call[:manipulation][:invite]["P-Charging-Vector"] = "todelete"
# Will add or modify "banana" and set its value to "red" header from the invite
call[:manipulation][:invite][:banana] = "red"
# Will add or modify "salad" and set its value to "blue" header from the invite
call[:manipulation][:invite][:salad] = "blue"

# Will add or modify "P-Charging-Vector" and set its value to "PCV" header from the 18x
call[:manipulation]["18x"]["P-Charging-Vector"] = "PCV"
# Will add or modify "banana" and set its value to "yellow" header from the 18x
call[:manipulation]["18x"][:banana] = "yellow"
# Will add or modify "salad" and set its value to "green" header from the 18x
call[:manipulation]["18x"][:salad] = "green"

call[:manipulation]["200"]["P-Charging-Vector"] = "PCV"
call[:manipulation]["200"][:banana] = "yellow"
call[:manipulation]["200"][:TEST] = "STXIE"

params
end

== Limitations: ==

# 'SIP headers manipulation' can only affect SIP headers who are not already modified/generated by the other SBC behaviors, see example below.
# Header values are viewed as a constant string, the routing script cannot pass variable to SBC.
# If the manipulated header is already present, it will be overwritten.
# Manipulate headers hash can only be set once per call (on the incoming invite message). It is not possible to update the hash later during the call.
# SIP header in Re-invite and 200 OK attached to Re-invite cannot be affected by this behavior.

=== Headers that cannot be used in 'SIP headers manipulation': ===
Accept
Accept-Contact
Accept-Encoding
Accept-Language
Alert-Info
Allow
Allow-Events
Also
Anonymity
Authentication-Info
Authorization
Call-ID
Call-Info
Contact
Content-Disposition
Content-Encoding
Content-Language
Content-Length
Content-Type
CSeq
Date
Encryption
Error-Info
Event
Expires
From
In-Reply-To
Max-Forwards
MIME-version
Min-Expires
Min-SE
Organization
P-Asserted-Identity
P-Media-Authorization
P-Preferred-Identity
Priority
Privacy
Proxy-Authenticate
Proxy-Authorization
Proxy-Require
RAck
Reason
Record-Route
Refer-To
Referred-By
Reject-Contact
Remote-Party-ID
Replaces
Reply-To
Request-Disposition
Require
Response-Key
Retry-After
Route
RPID-Privacy
RSeq
Security-Client
Security-Server
Security-Verify
Server
Service-Route
Session-Expires
Subject
Subscription-State
Supported
Timestamp
To
Unsupported
User-Agen
Via
Warning
WWW-Authenticate

Sip headers manipulation

2020-08-07T16:37:12Z

Cbilodeau: fix format

The behavior 'SIP headers manipulation' allows the SBC users to freely add ,remove and modify the sip headers when SBC forwards the received invite, 18x and 200 sip messages (there are limitations, please refer to the limitations section below).

== Configuration: ==
To use the behavior, you need to select the option "Enable SIP headers manipulation" under the path "Gateway -> Advanced Parameters" and use the filter script "sip_headers_manipulation.rb" in routing script setting.

== Usage: ==
The manipulation actions are based on what are defined in "sip_headers_manipulation.rb" dynamically or statically. In sip_headers_manipulation.rb, there is filter_sip_headers_manipulation that should be applied as 'after filter' in main script and need to use call[:manipulation] to save what you want SBC to do regarding headers manipulation. An example is :

def filter_sip_headers_manipulation params

# The following 4 lines are needed to initialize the hash table
call = params[ :call ]
call[:manipulation] = {}
call[:manipulation][:invite] = {}
call[:manipulation]["18x"] = {}
call[:manipulation]["200"] = {}

# Will remove "P-Charging-Vector" header from the the invite if exists
call[:manipulation][:invite]["P-Charging-Vector"] = "todelete"
# Will add or modify "banana" and set its value to "red" header from the invite
call[:manipulation][:invite][:banana] = "red"
# Will add or modify "salad" and set its value to "blue" header from the invite
call[:manipulation][:invite][:salad] = "blue"

# Will add or modify "P-Charging-Vector" and set its value to "PCV" header from the 18x
call[:manipulation]["18x"]["P-Charging-Vector"] = "PCV"
# Will add or modify "banana" and set its value to "yellow" header from the 18x
call[:manipulation]["18x"][:banana] = "yellow"
# Will add or modify "salad" and set its value to "green" header from the 18x
call[:manipulation]["18x"][:salad] = "green"

call[:manipulation]["200"]["P-Charging-Vector"] = "PCV"
call[:manipulation]["200"][:banana] = "yellow"
call[:manipulation]["200"][:TEST] = "STXIE"

params
end

== Limitations: ==

# 'SIP headers manipulation' can only affect SIP headers who are not already modified/generated by the other SBC behaviors, see example below.
# Header values are viewed as a constant string, the routing script cannot pass variable to SBC.
# If the manipulated header is already present, it will be overwritten.
# Manipulate headers hash can only be set once per call (on the incoming invite message). It is not possible to update the hash later during the call.
# SIP header in Re-invite and 200 OK attached to Re-invite cannot be affected by this behavior.

=== Headers that cannot be used in 'SIP headers manipulation': ===
- Accept
- Accept-Encoding
- Accept-Language
- Alert-Info
- Allow
- Authorization
- Authentication-Info
- Also
- Call-ID
- Call-Info
- Contact
- Content-Disposition
- Content-Encoding
- Content-Language
- Content-Length
- Content-Type
- CSeq
- Date
- Encryption
- Error-Info
- Expires
- From
- In-Reply-To
- Max-Forwards
- MIME-version
- Min-Expires
- Organization
- Priority
- Proxy-Authenticate
- Proxy-Authorization
- Proxy-Require
- Record-Route
- Reply-To
- Require
- Response-Key
- Retry-After
- Route
- Server
- Subject
- Supported
- Timestamp
- To
- Unsupported
- User-Agen
- Via
- RAck
- RSeq
- Warning
- WWW-Authenticate
- Event
- Allow-Events
- Refer-To
- Referred-By
- Replaces
- Session-Expires
- Min-SE
- Request-Disposition
- Accept-Contact
- Reject-Contact
- Anonymity
- RPID-Privacy
- Remote-Party-ID
- Subscription-State
- Security-Client
- Security-Server
- Security-Verify
- Reason
- P-Media-Authorization
- Privacy
- P-Asserted-Identity
- P-Preferred-Identity
- Service-Route

User talk:Song tao xie

2020-08-05T16:42:40Z

Cbilodeau: Welcome!

User:Song tao xie

2020-08-05T16:42:40Z

Cbilodeau: Creating user page with biography of new user.

Your e-mail address will be sent a confirmation message once this request is submitted. Please respond by clicking on the confirmation link provided by the e-mail. Also, your password will be e-mailed to you when your account is created. Your e-mail address will be sent a confirmation message once this request is submitted. Please respond by clicking on the confirmation link provided by the e-mail. Also, your password will be e-mailed to you when your account is created.

File:800 Series Installation Guide Issue5.0.pdf

2020-05-27T20:32:47Z

Cbilodeau: uploaded a new version of "File:800 Series Installation Guide Issue5.0.pdf": upload 5.0L remove DC cables that we don't supply anymore

Upload new installation guide for 800 series

Routing script tutorial:Mini Development Guide

2020-05-27T14:56:26Z

Cbilodeau: /* ENUM Query */ reformat responses_list

== Call object ==

=== Get ===

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

called_number = caf_call.get :called

=== List_params ===

This function is used to retrieve the list of supported call parameters. For example to extract all the possible call parameters 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 the gateway application will use to bridge the incoming call leg. If more than one outgoing route is "accepted", the gateway will try them one by one in the same order that 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 "[[Routing_script_tutorial:Mini_Development_Guide#Reason_values|Reason values]]".

=== Script parameters protocol mapping ===

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

called_number = call[:called]

For information on how to use and remap call parameters, see [[Toolpack:_How_to_Use_RegEx_in_Remapped_Called_and_Calling_Number_Mask | How to use regex in Remapped Called and Calling Number Mask]]

{| 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 
| 
|-
| original_session_id 
| N/A 
| N/A 
| N/A 
| N/A 
| Original Session ID (before call transfer or redirections) 
| 
|-
| 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_sip_host 
| N/A 
| N/A 
| N/A 
| SIP:From - host (domain or IP) 
| For example : The 'telcobridges.com' in From: "Cullen Jennings" <sip:fluffy@telcobridges.com> 
| 
|-
| calling_sip_port 
| N/A 
| N/A 
| N/A 
| SIP:From - port 
| For example : The '6060' in From: "Cullen Jennings" <sip:fluffy@telcobridges.com:6060> 
| 
|-
| calling_noa 
| Q931: 'Calling party number' IE - Type of number 
| N/A 
| Q763: 'Calling party number' IE - nature of address indicator (*) 
| N/A 
| * In ANSI SS7 LNP networks, the IE 'generic address parameter' is used (when present) instead 
| 
|-
| calling_npi 
| Q931: 'Calling party number' IE - Numbering plan identification 
| N/A 
| Q763: 'Calling party number' IE - numbering plan indicator (*) 
| N/A 
| * In ANSI SS7 LNP networks, the IE 'generic address parameter' is used (when present) instead 
| 
|-
| calling_display 
| Q931: 'Display' IE - Display information 
Q931: 'Facility CNAM' IE when presentation is allowed for DMS/NI2 variants 
| N/A 
| Q763
ITU97: 'Display information' IE - display information
ANSI95: 'Generic name' IE - display information
| SIP:From - display-name 
| 
| 
|-
| calling_display_type 
| Q931: 'Display' IE - Display information (present and/or first byte) 
| N/A 
| Q763: 'Display information' IE - present or not 
| N/A 
| 
| 
|-
| calling_presentation 
| Q931: 'Calling party number' IE - Presentation indicator 
| N/A 
| Q763: 'Calling party number' IE - address presentation restricted indicator 
|
SIP:From - display-name (displays 'anonymous' or not)

SIP:Remote-party-id - privacy 

| 
| 
|-
| calling_screening 
| Q931: 'Calling party number' IE - Screening indicator 
| N/A 
| Q763: 'Calling party number' IE - screening 
| SIP:Remote-party-id - screen 
| 
| 
|-
| calling_category 
| N/A 
| Call party category (Group A) 
| Q763: 'Calling party's category' IE - calling party's category 
|
SIP:From - cpc

SIP:P-asserted-identity - cpc 

| 
| 
|-
| calling_subscriber
(Generic Number / NDS) 
| Q931: 2nd 'Calling party number' IE - Number digits 
| N/A 
| Q763: Generic number IE with type 'additional calling party number' - Number digits 
|
SIP:P-asserted-identity - userinfo

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

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

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

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

SIP:Remote-party-id - user-info 
| 
| 
|-
| private_display 
| Q931: 'Facility CNAM' IE when presentation is restricted for DMS/NI2 variants 
| N/A 
| N/A 
|
SIP:P-asserted-identity - display-name 

SIP:Remote-party-id - display-name 

| 
| 
|-
| private_display_type 
| N/A 
| N/A 
| N/A 
| N/A 
| Indicate presence or not of the private calling information 
| 
|-
| private_address 
| N/A 
| N/A 
| N/A 
|
SIP:P-asserted-identity - userinfo

SIP:Remote-party-id - user-info 

| For example : The 'fluffy' in P-Asserted-Identity: "Cullen Jennings" <sip:fluffy@telcobridges.com> 
| 
|-
| private_address_sip_host 
| N/A 
| N/A 
| N/A 
|
SIP:P-asserted-identity - host (domain or IP)

SIP:Remote-party-id - host (domain or IP) 

| For example : The 'telcobridges.com' in P-Asserted-Identity: "Cullen Jennings" <sip:fluffy@telcobridges.com> 
| 
|-
| private_address_sip_port 
| N/A 
| N/A 
| N/A 
|
SIP:P-asserted-identity - port

SIP:Remote-party-id - port 

| For example : The '6060' in P-Asserted-Identity: "Cullen Jennings" <sip:fluffy@telcobridges.com:6060> 
| 
|-
| preferred_id_forward_enabled 
| N/A 
| N/A 
| N/A 
| N/A 
| Overwrite default P-Preferred-Identity header forwarding behavior from incoming to outgoing leg 
| 
|-
| preferred_id 
| N/A 
| N/A 
| N/A 
|
SIP:P-Preferred-Identity - userinfo

| For example : The 'fluffy' in P-Preferred-Identity: "Cullen Jennings" <sip:fluffy@telcobridges.com> 
| 
|-
| preferred_id_display_name 
| N/A 
| N/A 
| N/A 
|
SIP:P-Preferred-Identity - display name

| For example : The 'Cullen Jennings' in P-Preferred-Identity: "Cullen Jennings" <sip:fluffy@telcobridges.com> 
| 
|-
| preferred_id_sip_host 
| N/A 
| N/A 
| N/A 
|
SIP:P-Preferred-Identity - host (domain or IP)

| For example : The 'telcobridges.com' in P-Preferred-Identity: "Cullen Jennings" <sip:fluffy@telcobridges.com> 
| 
|-
| preferred_id_sip_port 
| N/A 
| N/A 
| N/A 
|
SIP:P-Preferred-Identity - port

| For example : The '6060' in P-Preferred-Identity: "Cullen Jennings" <sip:fluffy@telcobridges.com:6060> 
| 
|-
| 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_sip_host 
| N/A 
| N/A 
| N/A 
| SIP:To - host 
| For example : The 'telcobridges.com' in To: "Cullen Jennings" <sip:fluffy@telcobridges.com> 
| 
|-
| called_sip_port 
| N/A 
| N/A 
| N/A 
| SIP:To - port number 
| For example : The '6060' in To: "Cullen Jennings" <sip:fluffy@telcobridges.com:6060> 
| 
|-
| called_noa 
| Q931: 'Called party number' IE - Type of number 
| N/A 
| Q763: 'Called party number' IE - nature of address indicator 
| N/A 
| 
| 
|-
| called_npi 
| Q931: 'Called party number' IE - Numbering plan identification 
| N/A 
| Q763: 'Called party number' IE - numbering plan indicator 
| N/A 
| 
| 
|-
| charge_number 
| N/A 
| N/A 
| ANSI: 'Charge number' IE - address signals 
| N/A 
| 
| 
|-
| charge_number_noa 
| N/A 
| N/A 
| ANSI: 'Charge number' IE - nature of address indicator 
| N/A 
| 
| 
|-
| charge_number_npi 
| N/A 
| N/A 
| ANSI: 'Charge number' IE - numbering plan indicator 
| N/A 
| 
| 
|-
| redirecting_number_forward_enabled 
| N/A 
| N/A 
| N/A 
| N/A 
| Overwrite default redirecting number and original called number forwarding behavior from incoming to outgoing leg 
| 
|-
| redirecting_number 
| Q931: 'Redirecting number' 1st IE - Number digits 
| N/A 
| Q763: 'Redirecting number' IE - address signals 
| SIP:Diversion (2nd header) - display-name 
| 
| 
|-
| redirecting_number_noa 
| Q931: 'Redirecting number' 1st IE - Type of number 
| N/A 
| Q763: 'Redirecting number' IE - nature of address indicator 
| N/A 
| 
| 
|-
| redirecting_number_npi 
| Q931: 'Redirecting number' 1st IE - Numbering plan identification 
| N/A 
| Q763: 'Redirecting number' IE - numbering plan indicator 
| N/A 
| 
| 
|-
| redirecting_number_presentation 
| Q931: 'Redirecting number' 1st IE - Presentation indicator 
| N/A 
| Q763: 'Redirecting number' IE - address presentation restricted indicator 
| SIP:Diversion (2nd header) - diversion-privacy 
| 
| 
|-
| redirecting_number_indicator 
| N/A 
| N/A 
| Q763: 'Redirection information' IE - redirecting indicator 
| N/A 
| 
| 
|-
| redirecting_number_reason 
| Q931: 'Redirecting number' 1st IE - Reason for redirection 
| N/A 
| Q763: 'Redirection information' IE - redirecting reason 
| SIP:Diversion (2nd header) - diversion-reason 
| 
| 
|-
| redirecting_number_counter 
| N/A 
| N/A 
| Q763: 'Redirection information' IE - redirection counter 
| SIP:Diversion (2nd header) - diversion-counter 
| 
| 
|-
| original_called_number
(OCN) 
| Q931: 'Redirecting number' 2nd IE - Number digits 
| N/A 
| Q763: 'Redirection number' IE - address signals 
| SIP:Diversion  (1st header) - display-name 
| 
| 
|-
| original_called_number_noa 
| Q931: 'Redirecting number' 2nd IE - Type of number 
| N/A 
| Q763: 'Redirection number' IE - nature of address indicator 
| N/A 
| 
| 
|-
| original_called_number_npi 
| Q931: 'Redirecting number' 2nd IE - Numbering plan identification 
| N/A 
| Q763: 'Redirection number' IE - numbering plan indicator 
| N/A 
| 
| 
|-
| original_called_number_presentation 
| Q931: 'Redirecting number' 2nd IE - Presentation indicator 
| N/A 
| Q763: 'Redirection number' IE - address presentation restricted indicator 
| SIP:Diversion (1st header) - diversion-privacy 
| 
| 
|-
| original_called_number_reason 
| Q931: 'Redirecting number' 2nd IE - Reason for redirection 
| N/A 
| Q763: 'Redirection information' IE - original redirection reason 
| SIP:Diversion (1st header) - diversion-reason 
| 
| 
|-
| original_called_number_counter 
| N/A 
| N/A 
| N/A 
| SIP:Diversion (1st header) - diversion-counter 
| 
| 
|-
| ported_number_npdi 
| N/A 
| N/A 
| Q763: 'Generic number' IE - with qualifier=Ported number is present 
| SIP:RequestURI - npdi=yes is present 
| Only valid if SIP/SS7 supports LNP 
| 
|-
| ported_number 
| N/A 
| N/A 
| Q763: 'Generic number' IE - address signals with qualifier=Ported number 
| SIP:RequestURI - to user part when rn is present 
| rn is stored in the called number 
| 
|-
| ported_number_noa 
| N/A 
| N/A 
| Q763: 'Generic number' IE - nature of address indicator with qualifier=Ported number 
| N/A 
| Only valid if SIP/SS7 supports LNP 
| 
|-
| ported_number_npi 
| N/A 
| N/A 
| Q763: 'Generic number' IE - numbering plan indicator with qualifier=Ported number 
| N/A 
| Only valid if SIP/SS7 supports LNP 
| 
|-
| oli
(Originating line information) 
| 5ESS Codeset 6 OLI - Value 
| N/A 
| ANSI: 'Originating line information' IE - OLI 
|
SIP:From - oli

SIP:P-asserted-identity - oli 

| 
| 
|-
| request_uri 
| N/A 
| N/A 
| N/A 
| Complete Request URI string 
| 
| 
|-
| request_uri_forward_enabled 
| N/A 
| N/A 
| N/A 
| N/A 
| Overwrite default URI forwarding behavior from incoming to outgoing leg 
| 
|-
| sip_header 
| N/A 
| N/A 
| N/A 
| Any header 
| Requires option 'Enable SIP Custom Headers' in Profiles->SIP 
| 2.7.63 
|-
| nap
(Network Access Point) 
| N/A 
| N/A 
| N/A 
| N/A 
| Incoming leg NAP name (read-only) 
| 
|-
| type_of_network_identification 
| Q931: 'Transit network selection' IE - Type of network identification 
| N/A 
| Q763: 'Transit network selection' IE - Type of network identification 
| N/A 
| 
| 2.7 
|-
| network_identification 
| Q931: 'Transit network selection' IE - Network identification 
| N/A 
| Q763: 'Transit network selection' IE - Network identification 
| SIP: Request-Line - cic 
| 
| 2.7 
|-
| network_identification_plan 
| Q931: 'Transit network selection' IE - Network identification plan 
| N/A 
| Q763: 'Transit network selection' IE - Network identification plan 
| N/A 
| 
| 2.7 
|-
| location_number_forward_enabled 
| N/A 
| N/A 
| N/A 
| N/A 
| Overwrite default location number forwarding behavior from incoming to outgoing leg 
| 2.7 
|-
| location_number 
| N/A 
| N/A 
| Q763: 'Location number' IE - address signals 
| N/A 
| 
| 2.7 
|-
| location_number_noa 
| N/A 
| N/A 
| Q763: 'Location number' IE - nature of address indicator 
| N/A 
| 
| 2.7 
|-
| location_number_npi 
| N/A 
| N/A 
| Q763: 'Location number' IE - numbering plan indicator 
| N/A 
| 
| 2.7 
|-
| location_number_presentation 
| N/A 
| N/A 
| Q763: 'Location number' IE - presentation restricted indicator 
| N/A 
| 
| 2.7 
|-
| location_number_screening 
| N/A 
| N/A 
| Q763: 'Location number' IE - screening 
| N/A 
| 
| 2.7 
|-
| mlpp_forward_enabled 
| N/A 
| N/A 
| N/A 
| N/A 
| A script needs to set this to true if it wants to overwrite MLPP information in the outgoing leg. Otherwise, profile relay 'outgoing mode' applies automatically. 
| 2.7 
|-
| mlpp_look_for_busy 
| N/A 
| N/A 
| Q763: 'MLPP precedence' IE - look ahead for busy 
| N/A 
| 
| 2.7 
|-
| mlpp_precedence_level 
| N/A 
| N/A 
| Q763: 'MLPP precedence' IE - precedence level 
| SIP:Resource-Priority - q735 
| 
| 2.7 
|-
| mlpp_network_identity 
| N/A 
| N/A 
| Q763: 'MLPP precedence' IE - network identity 
| N/A 
| 
| 2.7 
|-
| mlpp_service_domain 
| N/A 
| N/A 
| Q763: 'MLPP precedence' IE - MLPP service domain 
| N/A 
| 
| 2.7 
|-
| isub_forward_enabled 
| N/A 
| N/A 
| N/A 
| N/A 
| Overwrite default ISUB forwarding behavior from incoming to outgoing leg 
| 3.0.138 
|-
| called_isub 
| Q931: 'Called party subaddress' IE - subaddress information 
| N/A 
| Q763: 'Access transport' IE 
| SIP:To - isub parameter 
| 
| 2.7 
|-
| called_isub_type 
| Q931: 'Called party subaddress' IE - type of subaddress 
| N/A 
| Q763: 'Access transport' IE 
| SIP:To - isub-encoding parameter 
| 
| 2.7 
|-
| calling_isub 
| Q931: 'Calling party subaddress' IE - subaddress information 
| N/A 
| Q763: 'Access transport' IE 
| SIP:From - isub 
| 
| 2.7 
|-
| calling_isub_type 
| Q931: 'Callinf party subaddress' IE - type of subaddress 
| N/A 
| Q763: 'Access transport' IE 
| SIP:From - isub-encoding 
| 
| 2.7 
|-
| ss7_fci_default 
| N/A 
| N/A 
| Default forward call indicator (FCI) value. 
| N/A 
| Toolpack will overwrite FCI bits A, D, F, I and M with appropriate values according to call conditions 
| 2.7 
|-
| ss7_fci_force_mask 
| N/A 
| N/A 
| Mask to select bits from ss7_fci_default that must be forced. 
| N/A 
| Bits from ss7_fci_default which corresponding bit in ss7_fci_force_mask is set will be forced, and no more controlled by Toolpack 
| 2.7 
|-
| ss7_bci_default 
| N/A 
| N/A 
| Default backward call indicator (BCI) value. 
| N/A 
| Toolpack will overwrite BCI bits AB, I, K, M and N with appropriate values according to call conditions 
| 2.7 
|-
| ss7_bci_force_mask 
| N/A 
| N/A 
| Mask to select bits from ss7_bci_default that must be forced. 
| N/A 
| Bits from ss7_bci_default which corresponding bit in ss7_bci_force_mask is set will be forced, and no more controlled by Toolpack 
| 2.7 
|-
| tdm_ls_name_forward_enabled 
| N/A 
| N/A 
| N/A 
| N/A 
| Enable line service and timeslot selection to create the outgoing leg. tdm_ls_name and tdm_timeslot_nb must be defined along with tdm_ls_name_forward_enabled 
| 3.0 
|-
| tdm_ls_name
(Line Service or T1/E1 trunk) 
| Incoming leg line service name 
| Incoming leg line service name 
| Incoming leg line service name 
| N/A 
| if tdm_ls_name_forward_enabled is set, try to use this line service name to create outgoing leg 
| 2.7 
|-
| tdm_timeslot_nb 
| Incoming leg timeslot number 
| Incoming leg timeslot number 
| Incoming leg timeslot number 
| N/A 
| if tdm_ls_name_forward_enabled is set, try to use this timeslot number to create outgoing leg 
| 2.7 
|-
| rtp_local_addr 
| N/A 
| N/A 
| N/A 
| Incoming leg local SDP IP address 
| (read-only) 
| 2.7 
|-
| rtp_local_port 
| N/A 
| N/A 
| N/A 
| Incoming leg local SDP IP port 
| (read-only) 
| 2.7 
|-
| rtp_remote_addr 
| N/A 
| N/A 
| N/A 
| Incoming leg remote SDP IP address 
| (read-only) 
| 2.7 
|-
| rtp_remote_port 
| N/A 
| N/A 
| N/A 
| Incoming leg remote SDP IP port 
| (read-only) 
| 2.7 
|-
| ss7_cot_enabled 
| N/A 
| N/A 
| Requests SS7 in-call continuity test for this outgoing SS7 call 
| N/A 
| Toolpack will request a continuity test on the timeslot before making the outgoing call. If COT fails, the call will be dropped (then another route may be attempted) 
| 2.8 
|-
| reverse_charging_indication 
| Incoming leg Reverse charging indication IE present 
| N/A 
| N/A 
| N/A 
| If set in routing script, will add Reverse charging indication IE in outgoing leg (also use reverse_charging_indication_forward_enabled) 
| 2.8.12 
|-
| reverse_charging_indication_forward_enabled 
| N/A 
| N/A 
| N/A 
| N/A 
| Enable forwarding of reverse charging indication from incoming to outgoing leg 
| 2.8.12 
|-
| sip_call_id 
| N/A 
| N/A 
| N/A 
| Incoming leg SIP Call-Id 
| (read-only) 
| 2.9.112 / 3.0.131 
|-
| sip_local_addr 
| N/A 
| N/A 
| N/A 
| Incoming leg local SIP IP address 
| (read-only) 
| 2.8.13 
|-
| sip_local_port 
| N/A 
| N/A 
| N/A 
| Incoming leg local SIP port 
| (read-only) 
| 2.8.13 
|-
| sip_remote_addr 
| N/A 
| N/A 
| N/A 
| Incoming leg remote SIP IP address 
| (read-only) 
| 2.8.13 
|-
| sip_remote_port 
| N/A 
| N/A 
| N/A 
| Incoming leg remote SIP port 
| (read-only) 
| 2.8.13 
|-
| acli 
| N/A 
| N/A 
| 'Additional Calling Party Information' IE - address signals.
A new ISUP parameter is defined under national option to carry the actual calling party number of the ported subscriber (N2) in the IAM message across the Network. 
| N/A 
| (read-only) 
| 3.0.143.2 
|-
| acli_nao 
| N/A 
| N/A 
| 'Additional Calling Party Information' IE - nature of address indicator.
A new ISUP parameter is defined under national option to carry the actual calling party number of the ported subscriber (N2) in the IAM message across the Network. 
| N/A 
| (read-only) 
| 3.0.143.2 
|-
| acli_npi 
| N/A 
| N/A 
| 'Additional Calling Party Information' IE - numbering plan indicator.
A new ISUP parameter is defined under national option to carry the actual calling party number of the ported subscriber (N2) in the IAM message across the Network. 
| N/A 
| (read-only) 
| 3.0.143.2 
|-
| acli_presentation 
| N/A 
| N/A 
| 'Additional Calling Party Information' IE - address presentation restricted indicator.
A new ISUP parameter is defined under national option to carry the actual calling party number of the ported subscriber (N2) in the IAM message across the Network. 
| N/A 
| (read-only) 
| 3.0.143.2 
|-
| acli_screening 
| N/A 
| N/A 
| 'Additional Calling Party Information' IE - screening indicator.
A new ISUP parameter is defined under national option to carry the actual calling party number of the ported subscriber (N2) in the IAM message across the Network. 
| N/A 
| (read-only) 
| 3.0.143.2 
|}

 

=== Noa values ===
The text below represents the value normally used by routing script. 
Incase it's required to use a value that's not defined in the text values below, a integer can be provided and will be used "as-is" in the signaling message. 
Example numeric values for the SS7 protocol are shown in parenthesis. 
 
*<tt>unknown_number (2 or 0x2)</tt>
*<tt>international_number (4 or 0x4)</tt>
*<tt>national_number (3 or 0x3)</tt>
*<tt>subscriber_number (1 or 0x1)</tt>
*<tt>network_specific (5 or 0x5)</tt>
*<tt>network_routing_national_format (7 or 0x7)</tt>
*<tt>network_routing_international_format (8 or 0x8)</tt>
*<tt>abbreviated_number (6 or 0x6)</tt>
*<tt>subscriber_number_operator_requested (113 or 0x71)</tt>
*<tt>national_number_operator_requested (114 or 0x72)</tt>
*<tt>international_number_operator_requested (115 or 0x73)</tt>
*<tt>no_number_present_operator_requested (116 or 0x74)</tt>
*<tt>no_number_present_cut_through_call_to_carrier (117 or 0x75)</tt>
*<tt>test_line_test_code (119 or 0x77)</tt>
*<tt>non_unique_subscriber_number (113 or 0x71)</tt>
*<tt>non_unique_national_number (115 or 0x73)</tt>
*<tt>non_unique_international_number (116 or 0x74)</tt>
*<tt>call_950_number (118 or 0x76)</tt>
*<tt>special_number (115 or 0x73)</tt>
*<tt>national_number_with_transit_network_selection (116 or 0x74)</tt>
*<tt>international_number_with_transit_network_selection (117 or 0x75)</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 for Calling number, Calling Subscriber (Generic Number), Redirecting Number, Original Called Number (OCN) and Location Number ===
The text below represents the value normally used by routing script. 
Incase it's required to use a value that's not defined in the text values below, a integer can be provided and will be used "as-is" in the signaling message. 
Example numeric values for the SS7 protocol are shown in parenthesis. 
 
*<tt>unspecified</tt>
*<tt>not_available (0x2)</tt>
*<tt>allowed (0x0)</tt>
*<tt>restricted (0x1)</tt>
*<tt>addr_restricted</tt>
*<tt>name_restricted</tt>

=== Calling Party Category ===
The text below represents the value normally used by routing script. 
In case it's required to use a value that's not defined in the text values below, a integer can be provided and will be used "as-is" in the signaling message. 

Mapping from routing script to SS7/CAS R2/SIP
{| cellspacing="1" cellpadding="1" border="1"
|-
! scope="col" | '''Routing Script string '''
! scope="col" | '''SS7 raw value '''
! scope="col" | '''R2 CAS scripts''' 
! scope="col" | '''Default Rx CAS''' 
! scope="col" | '''default Tx CAS''' 
! scope="col" | '''SIP "cpc=" '''
|-
| subscriber 
| 0xa 
| CATEGORY_SUBSCRIBER 
| 1 and 7 
| 7 
| ordinary 
|-
| subscriber_with_priority 
| 0xb 
| CATEGORY_SUBSCRIBER_WITH_PRIORITY 
| 2 and 9 
| 2 
| priority 
|-
| operator_french 
| 0x1 
| CATEGORY_OPERATOR_FRENCH 
| 5 
| 5 
| operator 
|-
| operator_english 
| 0x2 
| CATEGORY_OPERATOR_ENGLISH (5) 
| 5 
| 5 
| operator 
|-
| operator_german 
| 0x3 
| CATEGORY_OPERATOR_GERMAN (5) 
| 5 
| 5 
| operator 
|-
| operator_russian 
| 0x4 
| CATEGORY_OPERATOR_RUSSIAN (5) 
| 5 
| 5 
| operator 
|-
| operator_spanish 
| 0x5 
| CATEGORY_OPERATOR_SPANISH (5) 
| 5 
| 5 
| operator 
|-
| data 
| 0xc 
| CATEGORY_DATA 
| 6 and 8 
| 6 
| datacall 
|-
| test 
| 0xd 
| CATEGORY_TEST 
| 3 
| 3 
| test 
|-
| payphone 
| 0xf 
| CATEGORY_PAYPHONE 
| none 
| 7 
| payphone 
|-
| unknown 
| 0x0 
| CATEGORY_UNKNOWN 
| 4, 11 to 15 
| 7 
| unknown 
|-
| unspecified 
| 0xa 
| invalid 
| none 
| none 
| invalid 
|}
[[CAS_R2_scripting#Category_meanings|Link to calling party categories used in CAS R2 scripts]]

=== Screening values for Calling number, Calling Subscriber (Generic Number), and Location Number ===
The text below represents the value normally used by routing script. 
Incase it's required to use a value that's not defined in the text values below, a integer can be provided and will be used "as-is" in the signaling message. 
Example numeric values for the SS7 protocol are shown in parenthesis. 
 
*<tt>unspecified</tt>
*<tt>no (0x0)</tt>
*<tt>pass (0x1)</tt>
*<tt>fail (0x2)</tt>
*<tt>network_provided (0x3)</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, Original Called Number and Diversion Reason ===

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 (SIP: user-busy)</tt>
*<tt>no_reply (SIP: no-answer)</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.

=== Information Transfer Capability values ===

information_transfer_capability:

*<tt>digital</tt>
*<tt>restricted_digital</tt>
*<tt>digital_with_tones</tt>
*<tt>speech</tt>
*<tt>3_1_khz_audio</tt>
*<tt>video</tt>

=== redirecting_number_forward_enabled values ===

Controls forwarding or discarding of redirecting number (SIP: diversion header) to the 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 the input of the routing script depends on the "Forward redirecting number" parameter in the "Advanced" section of the Gateway configuration page of the Web Portal. The script may change this value to override the Gateway configuration.

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

=== request_uri ===

Enables access to the Request-Line URI. 

For example, if the Request-Line is:
<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 ====

This call parameter 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 default value for this parameters is false. 

==== sip_scheme ====
(Available in Toolpack 3.1+)
This call parameters indicates the scheme (generally "sip" or "sips") of the incoming call.

This also allows to control the scheme used for the outgoing call (regardless if request_uri_forward_enabled is used or not)

Note: sips scheme must only be used on TLS NAPs (will cause call routing failure if NAP has only UDP or TCP transport types).

=== sip_header values ===
Contains custom sip headers from the inbound call leg. Any custom sip header can be added to an outgoing call leg: 

'''Note:'''

The SIP header is in string format.

'''string format:'''
<pre>call[ :sip_header ] = "P-my-custom-header:value1 \nP-my-custom-header2:value2 \nP-my-custom-header3:value3"
</pre>
(Note: \n above are actual newline characters, not '\' followed by 'n')

* PCAP sample: [[File:TB_Custom_SIP_Headers.pcap]]

List of sip headers that will not appear in call[:sip_header] since they are already processed by the SIP stack:
<pre>
Accept Error-Info Remote-Party-ID
Accept-Contact Event Replaces
Accept-Encoding Expires Reply-To
Accept-Language From Request-Disposition
Alert-Info In-Reply-To Subject
Allow Max-Forwards Subscription-State
Allow-Events MIME-version Supported
Also Min-Expires Timestamp
Anonymity Min-SE To
Authorization Organization Unsupported
Authentication-Info Path User-Agent
Call-ID Priority Via
Call-Info Privacy Warning
Contact Proxy-Authenticate WWW-Authenticate
Content-Disposition Proxy-Authorization Require
Content-Encoding Proxy-Require Response-Key
Content-Language P-Media-Authorization Retry-After
Content-Length P-Preferred-Identity RPID-Privacy
Content-Type P-Asserted-Identity Route
CSeq RAck RSeq
RAck Reason Security-Client
Reason Record-Route Security-Server
Date Refer-To Security-Verify
Diversion Referred-By Server
Encryption Reject-Contact Service-Route
Session-Expires
</pre>

=== sip header parameters ===
The routing script can read (and modify) some SIP header parameters (user parameters, URI parameters or header parameters) from some SIP headers (To, From, P-Asserted-Identity, Remote-Party-ID, Contact).

==== Available parameters ====
* '''call[ :calling_parameters ]''' (SIP "From" header)
* '''call[ :called_parameters ]''' (SIP "To" header)
* '''call[ :private_address_parameters ]''' (SIP "P-Asserted-Identity" or "Remote-Party-ID" header)
* '''call[ :preferred_id_parameters ]''' (SIP "P-Preferred-Identity" header)
* '''call[ :contact_parameters ]''' (SIP "Contact" header)

These parameters (if present) contain a hash with 3 keys: user_param, uri_param and header_param.
Each of this key points to a string that contains all the parameters found in the corresponding SIP header.
* '''User parameters''' (parameters between the user name/number and the host). Example <sip:alice;'''param=value'''@somewhere.com>
* '''URI parameters''' (parameters at the end of the URI). Example <sip:alice@somewhere.com;'''param=value'''>
* '''Header parameters''' (outside the URI). Example <sip:alice@somewhere.com>;'''param=value'''
Example to print all parameters of SIP "To" header:
call[ :called_parameters ].inspect -> '{ :user_param => "name1=value1;name2=value2", :uri_param => "name=value", :header_param => "name=value;example_param_without_value" }'
Example to modify (replace) the URI parameters of SIP "To" header:
call[ :called_parameters ][ :uri_param ] = "user=phone"

==== Exceptions ====
'''Note:''' Some parameters are reported as their own call attribute (oli, isub, cpc, transport) so they have the same representation for all protocols (SS7, IDSN, SIP). They won't appear present in the generic SIP header parameters structures above.

==== Forwarding from inbound to outbound call ====

==== Legacy behavior ====
(For base_routing version 1.32 or older)
By default, the parameters are not forwarded in a SIP to SIP call flow. The parameters '''will be forwarded''' when:
* accessed (read) from either the inbound or outbound call parameters
* written in either the inbound or outbound call parameters

==== Current behavior ====
(For Toolpack 3.0.118+, with base_routing version 1.33+)
SIP headers host and parameters are forwarded by default.

A route attribute "forward_sip_domain" (along with filter script "forward_sip_domain.rb") will control, per route, if SIP headers host+parameters must be forwarded.

==== Example usage ====
Example to print the user parameters:
<pre>
if call[:calling_parameters]
puts "user parameters = #{call[:calling_parameters][:user_param].inspect}"
end
</pre>

Example SIP "From" header:
From:<sip:123456782;test1=val1;test2=val2@something.com;test3=val3;test4=val4>;test5=val5;test6=val6
And the resulting content in the routing script:
call[:calling_parameters].inspect -> {:user_param=>"test1=val1;test2=val2", :uri_param=>"test3=val3;test4=val4", :header_param=>"test5=val5;test6=val6"}

Example to overwrite inbound leg calling parameters with new parameters for the outbound leg:
<pre>
call [:calling_parameters] = {
:user_param => "user_param7=7;user_param8=8",
:uri_param => "uri_param9=value9",
:header_param => "header_paramA=A" }
</pre>

Example to add user=phone and keep all other uri parameters.
<pre>
call[:calling_parameters] ||= {} # Create a hash if not already present
call[:calling_parameters][:uri_param] ||= "" # Create a string if not already present
call[:calling_parameters][:uri_param] += ";" if call[:calling_parameters][:uri_param] != ""
call[:calling_parameters][:uri_param] += "user=phone"
</pre>

=== MLPP Precedence values ===

mlpp_look_for_busy:

*<tt>allowed</tt>
*<tt>path_reserved</tt>
*<tt>not_allowed</tt>

mlpp_precedence_level:

*<tt>flash_override</tt>
*<tt>flash</tt>
*<tt>immediate</tt>
*<tt>priority</tt>
*<tt>routine</tt>

mlpp_network_identity:

3 digits value from 0 to 999

mlpp_service_domain:

24 bits value from 0 to 16777215

=== ISUB subaddress information values ===

called_isub_type:
calling_isub_type:

*<tt>nsap</tt>
*<tt>nsap_ia5</tt>
*<tt>nsap_bcd</tt>
*<tt>user</tt>

called_isub:
calling_isub:

Digits for the subaddress information.

=== Network Identification Plan ===

network_identification_plan:

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

=== Registered Users Information ===
Routing script can access information about registered users (when either the calling or called user is a known registered user).
When these fields are empty, it means that the calling/called (SIP from/to) does not correspond to a known registered user (routing script may still decide to route the call based on static routes).

Information for the called user:
params[:registered_user]
Information for the calling user:
params[:calling_registered_user]

These parameters are a hash of key/values that provide information about the contact.
{
:contact_list=>
[
{
:contact=>"<sip:user_name_or_number@hostname:7070;transport=UDP>", -> Full contact
:expires=>"60", -> Contact expiry time (seconds)
:host=>"hostname", -> host name from the contact header
:name=>"user_name_or_number", -> user name from the contact header
:nap_in=>"NAP_NAME", -> NAP that the contact has registered from
:port=>"7070", -> Port from the contact header
:transport=>"UDP" -> Transport type from the contact header
:q_value=>"0.00", -> Q-value for the contact (for contact ordering)
:src_host=>"10.0.0.10", -> Actual source IP address that the contact has registered from
:src_port=>"7070", -> Actual source port that the contact has registered from
:src_transport=>"UDP", -> Actual protocol that the contact has been registering with
}
]
}

== Route parameters ==

All route may have these parameters:

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

Example:

route[:remapped_nap]

Additionally it is possible to add dynamic route attributes in the web portal. These can be referenced by their name.
For example:
*priority
*weight

== Routing calls toward registered users ==
Static routes normally chose an outbound NAP to forward the call to.

But it's also possible to create routes which outbound NAP is dynamically chosen by matching a registered user (when using [[Sip_registration_forwarding|SIP registration forwarding]]).

More information can be found [[Sip_registration_forwarding#SIP_Calls_routing|here]] about the way to control the [[Sip_registration_forwarding#SIP_Calls_routing|priority of "dynamic" vs "static" routes]].

More information can be found [[#Registered_Users_Information|here]] about using routing scripts to access registered users information during call routing.

== Playing prompts announcements or tones ==

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

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

Example to play an announcement to incoming call (before routing outgoing call, regardless if a matching route is found or not):
bridge[:announcement_tone ] = "my_announcement.wav"

Example to play a ring-tone while the outgoing call is ringing:
bridge[:ring_tone] = "my_ring_tone.wav"

Example to play an audio file when outgoing call fails (no route, or outgoing call is refused):
bridge[:busy_tone] = "my_busy_tone.wav"

Example to play an audio file when call has reached the maximum allowed duration:
bridge[:disconnect_tone] = "your_account_balance_is_empty.wav"

=== Announcement file path format and options ===

All file plabyacks (:announcement_tone, :busy_tone, :ring_tone, :disconnect_tone) inside bridge parameters use 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 parameters:
* repeat: number of times to play the file (0 and 1 have the same result)
* start_off: Start offset in milliseconds
* end_off: End offset in milliseconds

Http and other path formats are described here: [[Customer_application_framework:play_audio_files#Play_path_format|Path format]]

==== Example 1 ====
The following example will play file1.wav once, and then play file2.wav in loop:
"file1.wav,file2.wav:-1"

==== Example 2 ====
The following example will play file1.wav from start offset of 1 second to end offset of 3 seconds, then twice file2.wav from second 5 to second 10.
"file1.wav:0:1000:3000,file2.wav:2:5000:10000"

==== Example 3 ====
The following example will play file1.wav once, ending at offset of 30 seconds.
"file1.wav:0:0:30000"

=== announcement_tone ===

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

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

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

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

===== announcement_code_detect =====
This option allows that the tone detection is enabled during the announcement play.

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

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

Code detect has multiple options, as shown in the following code:
code_detect = {
:type => :DTMF, # :DTMF or :MFR1 tone detection.
# Default is MFR1.
:prefix => "", # Prefix (digits) that is removed from collected digits.
# Default is empty.
:suffix => "", # Suffix (digits) that is removed from collected digits
# and causes routing script to be immediately called.
# Default is empty.
:suffix_removal => false, # Controls the removal of the suffix from the collected digit string that's reported to routing script.
# Default is false
:timeout => 0, # Inter-digit timeout (ms) after which collected digits are passed to the routing script.
# Use 0 for "no timeout".
# Default is 1000ms
:barge_in_interruption => true, # When enabled, playing announcement is stopped as soon as first digit is collected.
# Default is true.
:proceed_on_play_done => false, # When true: Outgoing call is made after announcement finishes playing.
# Routing script is not called again.
# When false: Outgoing call is never made.
# Digits are collected until timeout or suffix match,
# then routing script is called again.
# Default is false.
:cas_on_hook => false, # Specific for CAS-R1 calls. Makes CAS bits switch to "on-hook" when announcement finished playing
# (but the call is not "terminated" from Toolpack point of view)
# Default is false.
:cas_on_hook_delay => 0, # Duration of cas bits "on-hook" state.
# Only effective if cas_on_hook is set to true.
# Value of 0 stands for "infinite delay".
# Default is 0.
:repeat_delay => 0, # Delay between repetition of the announcement. The announcement will repeat
# itself every "repeat_delay" until a code is detected (suffix match or timetout).
# Value of 0 stands for "infinite delay" (no repeating).
# Default is 0.
}
'''Example 1''': Collect DTMF digits, and call routing script again with collected digits upon timeout or suffix match.
code_detect = { :type => :DTMF, :suffix => "#", :timeout => 5000 }
params[:bridge][:announcement_code_detect] = code_detect

'''Example 2''': Collect digits during the announcement (for CDR logs), then proceed (make outgoing call) after announcement finishes playing
code_detect = { :type => :DTMF, :timeout => 0, :barge_in_interruption => false, :proceed_on_play_done => true }
params[:bridge][:announcement_code_detect] = code_detect

==== Controlling what happens after announcement ====
The routing script can control what happens with the call after the announcement finishes playing:
* An outgoing call is made
* Incoming call is hung-up
* Do nothing (wait for the incoming call to hang-up)
===== An outgoing call is made =====
This happens when the script has returned matching routes (and did not raise RoutingException)

===== Incoming call is hung-up =====
This happens when the script returns no routes (in which case base_routing will raise RoutingException with cause :no_route).

It also happens when the script explicitly raises RoutingException.

The incoming call will be terminated with the specified cause.
For example
raise RoutingException, :temporary_failure
(See "Reason values" section in this page for list of available causes)

===== Do nothing (wait for the incoming call to hang-up) =====
If a filter raises RoutingException with code :ok, then the incoming call will not be terminated at the end of the announcement play.
Announcement digit collection will remain active if appropriate.
For example:
raise RoutingException, :ok

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

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

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

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

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

Call state from which ring tone is being played. Available values are:
* '''immediately''': Ring tone starts playing immediately on the incoming leg
* '''accepted''': Ring tone starts playing as soon as outgoing call is accepted
* '''callprogress''': Ring tone starts playing as soon as "call progress" is received on the outgoing call
* '''alerted''' (default): Ring tone starts playing only once outgoing call is alerted (but won't play if alert indicates early media from outgoing call)

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

==== The default (replicated) prompts folder ====
By default, when playing a prompt, Toolpack will look in the default prompts folder:
/lib/tb/toolpack/pkg/prompts
The root of this "prompts" directory is automatically replicated to secondary unit of redundant setups (1+1, N+1, redundant hosts). Sub-folders won't be replicated.

Any prompt play request without explicit file path will map to this folder. For example:
params[:bridge][:busy_tone] = "no_route.wav"
This will correspond to file /lib/tb/toolpack/pkg/prompts/no_route.wav

==== Relative file paths ====
Any file path that begins with "file://" is considered relative to the tbstreamserver application's working directory:
/lib/tb/toolpack/setup/12358/2.8/apps/tbstreamserver/
(Where "2.8" may be replaced by the current major version of your system)

For example:
params[:bridge][:busy_tone] = "file://my_folder/no_route.wav"
This will correspond to file /lib/tb/toolpack/setup/12358/2.8/apps/tbstreamserver/my_folder/no_route.wav

==== Absolute file paths ====
Absolute paths can also be provided.
For example:
params[:bridge][:busy_tone] = "file:///root/my_folder/no_route.wav"
This will correspond to file /root/my_folder/no_route.wav

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

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

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

bridge[ :record_incoming ] = ""

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

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

=== Record the outgoing call leg within incoming leg's recorded file (mixing) ===
[...]
route[ :record_outgoing ] = "@{MixWithIncoming}"
[...]

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

The paths can be absolute, or relative. When relative, they are relative to the "tbstreamserver" application working directory, for example:
/lib/tb/toolpack/setup/12358/2.7/apps/tbstreamserver/

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

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

Routing scripts can control if the UUI received from one leg through the call will be forwarded to the other call leg:
*uui_forward_enabled

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

=== UUI (user-to-user indication) values ===

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

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

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

=== uui_forward_enabled values ===

Controls forwarding or discarding of UUI to outgoing call leg.

Values for this parameter are "0", "1", "false" or "true.
* 0/false: UUI is not forwarded between call legs
* 1/true: UUI is forwarded between call legs

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

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

== ENUM Query ==
Starting with release 3.1, it is possible to issue ENUM Query requests to DNS servers from routing scripts. To do so, the params[:enum_query] object must be filled with the required ENUM Query attributes params[:enum_query][:fqdn] and [[#Refuse|an exception must be raised]] with reason :enum_query_required.

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

* ''ok'': The ENUM Query was successful.
* ''timeout'': The ENUM Query was not answered.

The params[:enum_query][:responses_list] field will contain a list of hash responses for each NAPTR records.

NAPTR records contain:
* '':uri''
* '':order''
* '':preference''
Example:
<code>
params[:enum_query][:responses_list]
[{:order=>"200", :preference=>"10", :uri=>"!^03111.*$!sip:123456782@example-2.com!"},
{:order=>"100", :preference=>"1", :uri=>"!^03222.*$!sip:123456782@example-3.com!"},
{:order=>"200", :preference=>"1", :uri=>"!^03111.*$!sip:123456782@exampler-1.com!"}]
</code>

== 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 ] = "0,1,2"
or
bridge[ :diversion_reason ] = "unknown,busy,no_reply"
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,_Original_Called_Number_and_Diversion_Reason|Redirecting number reason values]].

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

If the chosen [[Call transfer]] mode is to process requests locally, upon reception of a call transfer request (SIP REFER or ISDN Facility), routing script will be called once again, to select the routes for the new outgoing call (call transfer target).

=== How to route call transfer request ===
Routing of a call transfer request is done exactly like routing of a normal incoming call.
The routing script generally does not need any modification to support that.

In some cases, the routing script may want to use information related to the transfer request to perform routing, or to insert information in the outgoing call leg.
Additional information is provided to the routing script, allowing routing decisions using information from the call transfer request (SIP REFER or ISDN Facility).
See below...

=== params[ :call ] content during transfer request ===
When processing a call transfer request, the params[ :call ] hash contains the information from the inbound call (same as was passed to the routing script upon arrival of the inbound call)
call = params[ :call ] -> Information from original inbound call, with exception of call[ :called ]

One exception (convenient because it allows a unmodified routing script to process call transfer request the same way as any other routing request):
call[ :called ] -> Replaced by the called number from the call transfer request (also called "redirection number")
Complementary information:
call[ :original_called_number ] -> Contains the called number that was initially received from the incoming call, prior to call transfer request
call[ :redirecting_number ] -> Number of the call from which the call transfer request was received (generally equals to original_called_number)

These fields will also be included in the outgoing call made after routing:
* original called number and redirecting number are existing fields on SS7 and ISDN calls
* SIP "diversion" header is used for SIP calls

=== params[ :transfer ] content ===
(this if valid only for release 2.7.102 and above) 
When processing a call transfer request, information from the call transfer request message (SIP REFER, ISDN Facility) is provided in params[ :transfer ]:
transfer = params[ :transfer ]
The following field is always present:
transfer[ :original_nap ] -> Contains the NAP of the first call from which a call transfer request was received
transfer[ :redirecting_nap ] -> Contains the NAP of the call from which the current call transfer request was received
(same as :original_nap for the first call transfer, different for subsequent transfers)
Examples of other fields that may be present, when appropriate:
transfer[ :uui ] -> The UUI (user-to-user information) found in the call transfer request
transfer[ :sip_header ] -> Contains custom SIP headers from the call transfer request
transfer[ :request_uri ] -> Contains the SIP Request URI

These fields are 'read-only'. They will not be included in the outgoing call, as they represent the contents of the call transfer request, and not the outgoing call to be made.

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

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

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

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

To get more information, see:
*[[Routing_script_tutorial:SIP_Redirection_Contacts|SIP Redirection Contacts Parameters in a Call Flow]]

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

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

== Terminating calls ==
In release 2.8, it is now possible to terminate a call through the routing scripts. The [[#Reason values|reason code]] must be specified in <code>params[:bridge][:reason]</code>. The <code>:terminate</code> hash must be created and copied into <code>params</code>:
terminate = {}
params[:terminate] = terminate
The following fields can then be set in <code>:terminate</code>:
* <code>:sip_header</code>
* <code>:contacts</code> # list of contacts as described in the [[#Redirection|redirection]] section
* <code>:isup_raw</code>
* <code>:isup_raw_variant</code>
* <code>:redirecting_number</code>
* <code>:redirecting_number_noa</code>
* <code>:redirecting_number_npi</code>
* <code>:redirecting_number_presentation</code>
* <code>:redirecting_number_reason</code>
* <code>:redirecting_number_counter</code>
* <code>:redirecting_number_indicator</code>
* <code>:original_called_number</code>
* <code>:original_called_number_noa</code>
* <code>:original_called_number_npi</code>
* <code>:original_called_number_presentation</code>
* <code>:original_called_number_reason</code>
* <code>:original_called_number_counter</code>

== Reason values ==

Check here for Termination Reason Cause codes: 

[[Termination_cause_codes|Termination Reason Cause codes]]
 

Example to refuse an incoming call leg.
raise RoutingException, :no_route

Reason cause strings available inside routing scripts:

List of Q.850 reason causes:
:unallocated_number
:no_route_to_network
:no_route_to_destination
:send_special_tone
:misdialled_trunk_prefix
:channel_unacceptable
:call_awarded_in_established_channel
:preemption
:reattempt
:qor_ported_number
:normal_call_clearing
:user_busy
:no_user_responding
:no_answer_from_user
:subscriber_absent
:call_rejected
:number_changed
:redirection
:exchange_routing_error
:non_selected_user_clearing
:destination_out_of_order
:address_incomplete
:facility_rejected
:response_to_status_enquiry
:normal_unspecified
:no_circuit_available
:network_out_of_order
:frame_mode_out_of_service
:frame_mode_connection_operational
:temporary_failure
:switching_equipment_congestion
:access_information_discarded
:requested_circuit_not_available
:precedence_call_blocked
:resource_unavailable
:quality_of_service_not_available
:requested_facility_not_subscribed
:outgoing_calls_barred
:outgoing_calls_barred_within_cug
:incoming_calls_barred
:incoming_calls_barred_within_cug
:bearer_cap_not_authorized
:bearer_cap_not_available
:inconsistency_access_info
:service_not_available
:bearer_cap_not_implemented
:channel_type_not_implemented
:requested_facility_not_implemented
:only_restricted_digital_info
:service_not_implemented
:invalid_call_reference
:channel_does_not_exist
:call_identity_does_not_exist
:call_identity_in_use
:no_call_suspended
:call_has_been_cleared
:user_not_member_of_cug
:incompatible_destination
:non_existant_cug
:invalid_transit_network
:invalid_message_unspecified
:mandatory_ie_missing
:message_type_non_existent
:message_not_compatible_with_call_state
:ie_non_existent
:invalid_ie_content
:msg_not_compatible_with_call_state
:recovery_on_timer_expiry
:parameter_non_existent_passed_on
:message_with_non_recognized_parameters_discarded
:protocol_error
:interworking_unspecified

List of toolpack reason causes:

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

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

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

== Nap status ==

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

'''Notice:''' These values may change between major release.

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

 If the nap status is part of a substructure, the substructure will be a hash containing all subfield elements.

Example to access the signaling type and the number of incoming call count for the NAP of the current call:

incoming_nap = params[:naps][ params[:call][ :nap ].to_sym]
log_trace 4,"Incoming NAP parameters=" + incoming_nap.inspect
log_trace 4,"Incoming NAP signaling type=" + incoming_nap[:signaling_type].inspect
log_trace 4,"Incoming NAP call cnt=" + incoming_nap[:asr_stats_incoming_struct][:total_call_cnt].inspect

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

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

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

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

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

The routing script can save a recursive hash of attributes here:
params[:user_context]

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

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

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

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

=== Test parameters ===
==== @call_params ====

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

 

==== @nap_list ====

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

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

 

==== @params ====

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

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

 

----

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

User talk:Philippe Bouffard

2020-04-23T20:19:47Z

Cbilodeau: Welcome!

User:Philippe Bouffard

2020-04-23T20:19:47Z

Cbilodeau: Creating user page with biography of new user.

My name is Philippe Bouffard. I have been working mostly in Software Quality Assurance and Software development. I graduated from the University of Ottawa in 2001 from the Electrical Engineering program and stated my career in the aeronautic field. Since then, I worked also in railways and telecommunications. I'm also a marathon runner and a big music fan (I play bass, guitar and drums). Nice to meet you!

Voice codecs

2020-02-20T20:24:22Z

Cbilodeau: update ptime for g722

TelcoBridges products support the following voice codecs.

== Supported codecs ==
 

*'''Universal codecs'''
**[http://en.wikipedia.org/wiki/G.711 G.711] - Supported packet durations: 20,30,40,50 and 60 msec
**[http://en.wikipedia.org/wiki/G.723.1 G.723.1] - Supported packet durations: 30,60 and 90 msec
**[http://en.wikipedia.org/wiki/G.726 G.726] - Supported packet durations: 20,30,40,50 and 60 msec
**[http://en.wikipedia.org/wiki/G.729 G.729a/ab] - Supported packet durations: 20,30,40,50,60,70 and 80 msec
**[[Clearmode]] ([http://tools.ietf.org/html/rfc4040 rfc4040]) - Supported packet durations: 5,10,20,30,40,50,60 msec
**[http://en.wikipedia.org/wiki/T.38 T.38]

*'''Other codecs'''
**[http://en.wikipedia.org/wiki/G.722 G.722] - Supported packet durations: 5,10,20,30,40,50 and 60 msec
**[http://en.wikipedia.org/wiki/G.728 G.728] - Supported packet durations: 20,30,40,50 and 60 msec
**[http://en.wikipedia.org/wiki/G.729 G.729eg] - Supported packet durations: 20,30,40,50,60,70 and 80 msec
**[http://en.wikipedia.org/wiki/Internet_Low_Bit_Rate_Codec iLBC] - Supported packet durations: [13kbps] 30 and 60 msec, [15kpbs] 20 and 40 msec

*'''Wireless codecs''' '''1'''
**[http://en.wikipedia.org/wiki/Adaptive_multi-rate_compression AMR] - Supported packet durations: 20,40,60 and 80 msec
**[http://en.wikipedia.org/wiki/Full_Rate GSM-FR] - Supported packet durations: 20 and 30 msec
**[http://en.wikipedia.org/wiki/Enhanced_Full_Rate GSM-EFR] - Supported packet durations: 20,40 and 60 msec
**[http://en.wikipedia.org/wiki/EVRC EVRC] - Supported packet durations: 20,40 and 60 msec
**[http://en.wikipedia.org/wiki/QCELP QCELP] - Supported packet durations: 20,40 and 60 msec
**[[G.722.2 (AMR-WB)]] - Supported packet durations: 20,40,60 and 80 msec
 

== TMG media gateways codec capacity ==
 
{| cellpadding="5" border="1" class="wikitable" style="text-align: center"
|-
! width="40" rowspan="2" | Media Gateway
! width="400" colspan="6" | Universal codecs
! width="240" colspan="4" | Other codecs
! width="320" colspan="4" | Wireless codecs'''1'''
|-
! width="80" | G.711
! width="80" | G.723.1
! width="80" | G.726
! width="80" | G.729ab
! width="80" | T.38
! width="80" | clearmode (rfc4040)
! width="80" | G.722
! width="80" | G.728
! width="80" | G.729eg
! width="80" | iLBC
! width="80" | AMR
! width="80" | AMR-WB
! width="80" | GSM-FR/GSM-EFR
! width="80" | EVRC/QCELP
|-
| TMG800
| 512 || 508 || 512 || 512 || 508 || 512 || 236 || 232 || 256 || 300 || 364 || 236 || 428 || 380
|-
| TMG3200-TE16 TMG7800-TE16
| 512 || 508 || 512 || 512 || 508 || 512 || 236 || 232 || 256 || 300 || 364 || 236 || 428 || 380
|-
| TMG3200-TE32 TMG7800-TE32
| 1024 || 1016 || 1024 || 1024 || 1016 || 1024|| 472 || 464 || 512 || 600 || 728 || 472 || 856 || 760
|-
| TMG3200-TE48 TMG7800-TE48
| 1536 || 1524 || 1536 || 1536 || 1524 || 1536 || 708 || 696 || 768 || 900 || 1092 || 708 || 1284 || 1140
|-
| TMG3200-TE64 TMG7800-TE64 TMG3200-STM1 TMG7800-STM1
| 2048 || 2032 || 2048 || 2048 || 2032 || 2048 || 944 || 928 || 1024 || 1200 || 1456 || 944 || 1712 || 1520
|}

== Tdev development platform ==

=== TMP6400 Codec density ===
Codec density is comprised of two modes:
*Mixed Mode
*Fixed Mode
 

==== Mixed Mode ====

The following densities are achieved when a codec mix is dynamically negotiated.
{| cellpadding="5" border="1" class="wikitable" style="text-align: center"
|-
! width="150" rowspan="2" | Type of
VoIP mezzanine
! width="400" colspan="5" | Universal codecs
! width="240" colspan="4" | Other codecs
! width="320" colspan="4" | Wireless codecs'''1'''
|-
! width="80" | G.711
! width="80" | G.723.1
! width="80" | G.726
! width="80" | G.729ab
! width="80" | T.38
! width="80" | G.722
! width="80" | G.728
! width="80" | G.729eg
! width="80" | iLBC
! width="80" | AMR
! width="80" | AMR-WB
! width="80" | GSM-FR/GSM-EFR
! width="80" | EVRC/QCELP
|-
| TM-VOIP-1
| 341
| 127
| 169
| 154
| 127
| 59
| 58
| 64
| 75
| 91
| 59
| 107
| 95
|-
| TM-VOIP-2
| 682
| 254
| 338
| 308
| 254
| 118
| 116
| 128
| 150
| 182
| 118
| 214
| 190
|-
| TM-VOIP-3
| 1023
| 381
| 507
| 462
| 381
| 177
| 174
| 192
| 225
| 273
| 177
| 321
| 285
|-
| TM-VOIP-4
| 1364
| 508
| 676
| 616
| 508
| 236
| 232
| 256
| 300
| 364
| 236
| 428
| 380
|}

==== Fixed mode ====

The following densities are achieved when a codec mix is predefined.
{| cellpadding="5" border="1" class="wikitable" style="text-align: center"
|-
! width="150" rowspan="2" | Type of
VoIP mezzanine
! width="400" colspan="5" | Universal codecs
! width="240" colspan="4" | Other codecs
! width="320" colspan="4" | Wireless codecs'''1'''
|-
! width="80" | G.711
! width="80" | G.723.1
! width="80" | G.726
! width="80" | G.729ab
! width="80" | T.38
! width="80" | G.722
! width="80" | G.728
! width="80" | G.729eg
! width="80" | iLBC
! width="80" | AMR
! width="80" | AMR-WB
! width="80" | GSM-FR/GSM-EFR
! width="80" | EVRC/QCELP
|-
| TM-VOIP-1
| 404
| 128
| 200
| 168
| 128
| 60
| 66
| 72
| 84
| 102
| 60
| 120
| 96
|-
| TM-VOIP-2
| 808
| 256
| 400
| 336
| 256
| 120
| 132
| 144
| 168
| 204
| 120
| 240
| 192
|-
| TM-VOIP-3
| 1212
| 384
| 600
| 504
| 384
| 180
| 198
| 216
| 252
| 306
| 180
| 360
| 288
|-
| TM-VOIP-4
| 1616
| 512
| 800
| 672
| 512
| 240
| 264
| 288
| 336
| 408
| 240
| 480
| 384
|}

['''1'''] May require additional licenses

[[Category:Glossary]]
[[Category:Tmedia_Features|Voice Codecs]]
[[Category:Tdev_Features|Voice Codecs]]
[[Category:Revise on Major]]

TLS

2019-10-15T11:45:53Z

Cbilodeau: add reference link

Transport Layer Security (TLS) is cryptographic protocols designed to provide communications security over a computer network.
[[Image:tls_connection_handshake.png|thumb|TLS Handshake]]

== TelcoBridges and TLS ==
TLS allows TelcoBridges' SBC to secured SIP signaling messages over networks.

=== Supported TLS RFCs ===
TLS 1.2
*[https://www.rfc-editor.org/rfc/rfc5246.html RFC 5246]

== Configuration ==
{| 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;" | SBC
|-
| valign="top" |
*[[Toolpack:Tsbc_Protocol_Stack_Settings_3.1|v3.1: TLS Configuration]]
|}

== References ==
*[https://en.wikipedia.org/wiki/Transport_Layer_Security Transport Layer Security]
*[https://www.rfc-editor.org/rfc/rfc5246.html RFC 5246]
*[http://www.steves-internet-guide.com/ssl-certificates-explained/ SSL Certificate explained]

Toolpack:Tsbc TLS Profiles

2019-09-05T20:47:20Z

Cbilodeau: re-arrange page

=== '''''Applies to version(s): v3.1''''' ===
{{DISPLAYTITLE:Configuring TLS Profiles}}

Multiple TLS profiles can be created in [[Toolpack]]. A TLS profile is a collection of certificates and other settings that define the behavior of TLS connections. Multiple TLS profiles can be used to expose different certificates and options to different TLS connections in [[Toolpack]], athough one TLS profile is generally enough.

== Create TLS profile ==

1. Select '''TLS Profiles''' from the navigation panel

[[Image:ConfigureTlsProfile_0.png]]

2. Click ''' Create New TLS Profile'''

[[Image:ConfigureTlsProfile_1.png]]

3. Configure TLS Profile parameters

* Provide a name that is meaningful to you.
* Check if client authentication is required. Enabling this checkbox means that incoming connections require the remote side to present a or many trusted certificate(s) (later selected after creation).
* Select a local certificate (either one imported earlier, or the "Default" certificate which is automatically generated by Toolpack and is also used for HTTPS)
* Click "Create"

[[Image:ConfigureTlsProfile_2.png|500px]]

* Select one or multiple "trusted" certificates, only necessary if "Require peer authentication" is checked.

[[Image:ConfigureTlsProfile_3.png|500px]]

== List of Parameters ==

*[[Parameter: Name|Name]]
*[[Parameter: Require client authentication|Require client authentication]]
*[[Parameter: Local certificate|Local certificate]]

Toolpack:Tsbc Protocol Stack Settings 3.1

2019-09-05T20:40:20Z

Cbilodeau: change optional tag

{{DISPLAYTITLE:FreeSBC/ProSBC Web Portal 3.1: Protocol Stacks}}

'''Applies to FreeSBC and Tmedia products'''

=Protocol Stacks=

===SSL Certification===
'''Applies to when SIP is used over TLS'''
{| cellpadding="5" border="1" class="wikitable"
|-
! width="250" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | Certificates (optional)
! width="250" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | TLS Profiles
|-
| valign="top" |
*[[TLS_Create_Key_Certificate|Create local Key/Certificate]]
*[[Create_A_SIP_Certificate_A|Create a SIP Certificate]]

| valign="top" |
*[[Toolpack:Tsbc_TLS_Profiles|Create a TLS Profile]]
|}

==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 Certificates and TLS Profiles
! 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 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 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 4a) Configure Network Access Points (NAPs)
! 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 4b) Configure Network Access Points (NAPs)
|-
| valign="top" |
*[[Toolpack:Creating_an_IP_Port_Range_SBC A|Create an IP port range]]

| valign="top" |
*[[Create_A_SIP_Certificate_A|Create a SIP Certificate]]
*[[Toolpack:Tsbc_TLS_Profiles|Create a TLS Profile]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Stack_SBC A|Create a SIP stack]]
*[[Toolpack:Creating_a_SIP_Transport_Server_SBC A|Create a SIP transport server]]

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

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

| valign="top" |
*[[Toolpack:Allocating_a_SIP_Open_Network_Access_Point_(NAP)_SBC A|Allocate an open SIP NAP]]

|}

===SIP Register===

{| 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 SIP Domain
! width="250" 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 Register Filtering Rule (Optional)
|-
| valign="top" |
*[[Toolpack:Creating_a_SIP_Domain_SBC B|Create a SIP domain]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_A|Create a filtering rule]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Condition_A|Add a filtering rule condition]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Action_A|Add a filtering rule action]]
|}

===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 D|Enable 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 D|Enable 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
|-
| valign="top" |
*[[Enabling_Remote_NAT_Traversal_C|Enable Remote NAT Traversal]]
|}
</div>

<div class="mw-collapsible mw-collapsed" data-collapsetext="DNS" data-expandtext="DNS" 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;" | DNS
|-
| valign="top" |
*[[Toolpack:Configuring_DNS_Parameters_C|Enable DNS]]
|}
</div>

===SIP Service Affecting Parameters===

<div class="mw-collapsible mw-collapsed" data-collapsetext="SIP Service Affecting Parameters" data-expandtext="SIP Service Affecting Parameters" style="width: 400px;">
When modifying the following parameters, the SIP stack will be deallocated and reallocated, resulting in a loss of active calls. Be careful when changing these parameters: 

'''Quirks'''
* All Parameters
'''Header Parameters'''
*Domain Name
*Add rPort
'''DNS Parameters'''
*Enabled
*Use Cache
*Cache Expire Time
*Address and Port
</div>

Create A SIP Certificate A

2019-09-05T20:39:48Z

Cbilodeau: re-order article

=== '''''Applies to version(s): v3.1''''' ===
{{DISPLAYTITLE:Configuring Certificates}}

Certificates are used to provide secure connections, such as HTTPs (secure connection to the web portal) or [[FreeSBC]] secure SIP calls (SIP over TLS).

This article describes how to import certificates. These certificates are later grouped into [[Toolpack:Tsbc_TLS_Profiles|TLS Profiles]], which is the first step in configuring secure SIP on [[FreeSBC]].

== Upload Local Certificates ==
Toolpack already contains, by default, a unique self-signed certificate (unique for each Toolpack system, shared for 1+1 hosts). This certificate is used for HTTPs, and can also be used for TLS.
Note:If you want create your own local certificate, [[Toolpack:Certificates|use can use openssl to create a self-signed certificate.]]

In case you want to use a custom local certificate, proceed as follows:

1. Select '''Certificates''' from the navigation panel:

[[Image:ConfigureCertificates_0.png|150px]]

2. Click ''' Create New Certificate ''' ===

[[Image:ConfigureCertificates_1.png|250px]]

3. Enters (but using "Local" certificate type) to import the local certificate.
* Provide a name that is meaningful to you.
* Select "Local" certificate.
* Drag-and-drop (or copy-paste) the certificate's text content into the appropriate text box
* Click "Create"

[[Image:ConfigureCertificates_3.png|500px]]

4. Make sure to copy the private key in the following directory:
/lib/tb/toolpack/pkg/ssl_certificate/

You can use ssh (command-line or using a tool like Filezilla) to upload the private key to the unit on the following path.
We recommend to use the same name for the key and certificate (except the extension of course):
Example:
scp my_local_cert.key root@MyFreeSbcHostName:/lib/tb/toolpack/pkg/ssl_certificate/

== Upload Trusted Certificate ==

1. Select '''Certificates''' from the navigation panel:

[[Image:ConfigureCertificates_0.png|150px]]

2. Click ''' Create New Certificate ''' ===

[[Image:ConfigureCertificates_1.png|250px]]

3. Configure the certificate parameters:
* Provide a name that is meaningful to you.
* Select "Trusted" certificate to import a remote party's certificate to trust.
* Drag-and-drop (or copy-paste) the certificate's text content into the appropriate text box
* Click "Create"

[[Image:ConfigureCertificates_2.png|500px]]

== List of Parameters ==

*[[Parameter: Name|Name]]
*[[Parameter: Certificate Type|Certificate Type]]

TLS Create Key Certificate

2019-09-05T20:24:37Z

Cbilodeau: Redirected page to Toolpack:Certificates

#REDIRECT [[Toolpack:Certificates]]

Toolpack:Tsbc Protocol Stack Settings 3.1

2019-09-05T20:19:55Z

Cbilodeau: add local certificate link

{{DISPLAYTITLE:FreeSBC/ProSBC Web Portal 3.1: Protocol Stacks}}

'''Applies to FreeSBC and Tmedia products'''

=Protocol Stacks=

===SSL Certification===
'''Applies to when SIP is used over TLS'''
{| cellpadding="5" border="1" class="wikitable"
|-
! width="250" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | Certificates
! width="250" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | TLS Profiles
|-
| valign="top" |
*[[TLS_Create_Key_Certificate|Create local Key/Certificate(optional)]]
*[[Create_A_SIP_Certificate_A|Create a SIP Certificate]]

| valign="top" |
*[[Toolpack:Tsbc_TLS_Profiles|Create a TLS Profile]]
|}

==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 Certificates and TLS Profiles
! 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 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 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 4a) Configure Network Access Points (NAPs)
! 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 4b) Configure Network Access Points (NAPs)
|-
| valign="top" |
*[[Toolpack:Creating_an_IP_Port_Range_SBC A|Create an IP port range]]

| valign="top" |
*[[Create_A_SIP_Certificate_A|Create a SIP Certificate]]
*[[Toolpack:Tsbc_TLS_Profiles|Create a TLS Profile]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Stack_SBC A|Create a SIP stack]]
*[[Toolpack:Creating_a_SIP_Transport_Server_SBC A|Create a SIP transport server]]

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

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

| valign="top" |
*[[Toolpack:Allocating_a_SIP_Open_Network_Access_Point_(NAP)_SBC A|Allocate an open SIP NAP]]

|}

===SIP Register===

{| 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 SIP Domain
! width="250" 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 Register Filtering Rule (Optional)
|-
| valign="top" |
*[[Toolpack:Creating_a_SIP_Domain_SBC B|Create a SIP domain]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_A|Create a filtering rule]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Condition_A|Add a filtering rule condition]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Action_A|Add a filtering rule action]]
|}

===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 D|Enable 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 D|Enable 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
|-
| valign="top" |
*[[Enabling_Remote_NAT_Traversal_C|Enable Remote NAT Traversal]]
|}
</div>

<div class="mw-collapsible mw-collapsed" data-collapsetext="DNS" data-expandtext="DNS" 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;" | DNS
|-
| valign="top" |
*[[Toolpack:Configuring_DNS_Parameters_C|Enable DNS]]
|}
</div>

===SIP Service Affecting Parameters===

<div class="mw-collapsible mw-collapsed" data-collapsetext="SIP Service Affecting Parameters" data-expandtext="SIP Service Affecting Parameters" style="width: 400px;">
When modifying the following parameters, the SIP stack will be deallocated and reallocated, resulting in a loss of active calls. Be careful when changing these parameters: 

'''Quirks'''
* All Parameters
'''Header Parameters'''
*Domain Name
*Add rPort
'''DNS Parameters'''
*Enabled
*Use Cache
*Cache Expire Time
*Address and Port
</div>

TLS Create Key Certificate

2019-09-05T20:17:20Z

Cbilodeau: fix typo

Although there are many ways to create TLS keys and certificates, we will show how to create self-signed key and certificate.

== Create self-signed key and certificate ==

openssl req -new -newkey rsa:4096 -x509 -sha256 -days <Number of days> -nodes -out <name>.crt –keyout <name>.key

* <Number of days>: Certificate/Key valid duration
* <name>: Name of key or certificate (we recommend using the same name for both)

TLS Create Key Certificate

2019-09-05T20:16:52Z

Cbilodeau: creation

File:Tls connection handshake.png

2019-09-04T17:00:38Z

Cbilodeau: uploaded a new version of "File:Tls connection handshake.png": remove optional messages

TLS

2019-09-04T15:51:03Z

Cbilodeau: adjust image in page

File:Tls connection handshake.png

2019-09-04T15:42:45Z

Cbilodeau:

TLS

2019-09-04T15:42:31Z

Cbilodeau: add tls connection image

Transport Layer Security (TLS) is cryptographic protocols designed to provide communications security over a computer network.

== TelcoBridges and TLS ==
TLS allows TelcoBridges' SBC to secured SIP signaling messages over networks.

[[Image:tls_connection_handshake.png]]

=== Supported TLS RFCs ===
TLS 1.2
*[https://www.rfc-editor.org/rfc/rfc5246.html RFC 5246]

== Configuration ==
{| 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;" | SBC
|-
| valign="top" |
*[[Toolpack:Tsbc_Protocol_Stack_Settings_3.1|v3.1: TLS Configuration]]
|}

== References ==
*[https://en.wikipedia.org/wiki/Transport_Layer_Security Transport Layer Security]
*[https://www.rfc-editor.org/rfc/rfc5246.html RFC 5246]

TLS

2019-09-04T14:58:15Z

Cbilodeau: create

Transport Layer Security (TLS) is cryptographic protocols designed to provide communications security over a computer network.

== TelcoBridges and TLS ==
TLS allows TelcoBridges' SBC to secured SIP signaling messages over networks.

=== Supported TLS RFCs ===
TLS 1.2
*[https://www.rfc-editor.org/rfc/rfc5246.html RFC 5246]

== Configuration ==
{| 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;" | SBC
|-
| valign="top" |
*[[Toolpack:Tsbc_Protocol_Stack_Settings_3.1|v3.1: TLS Configuration]]
|}

== References ==
*[https://en.wikipedia.org/wiki/Transport_Layer_Security Transport Layer Security]
*[https://www.rfc-editor.org/rfc/rfc5246.html RFC 5246]

Use Cases:Configuring Tmedia Gateways for M3UA Shared PC

2019-08-02T16:03:54Z

Cbilodeau: fix text align to top

{{DISPLAYTITLE:Use Case: Configuring Tmedia Gateways for Shared PC}}
=== '''''Applies to version(s): v3.0''''' ===

In this scenario . . . 

{|
| valign="top" |
[[File:Diagram_M3UA_Shared PC.jpg|800px]]
|}
 

====Configure Tmedia Gateway 1 for M3UA SGP====
{| colspan="4" border="1" class="wikitable"
! width="400" style="background: none repeat scroll 0% 0% rgb(0, 83, 139);" | Configure MTP2
! width="400" style="background: none repeat scroll 0% 0% rgb(0, 83, 139);" | Configure MTP3
! width="400" style="background: none repeat scroll 0% 0% rgb(0, 83, 139);" | Configure IP Interface
! width="400" style="background: none repeat scroll 0% 0% rgb(0, 83, 139);" | Configure SCTP
! width="400" style="background: none repeat scroll 0% 0% rgb(0, 83, 139);" | Configure M3UA
|-style="vertical-align:top;"
|
*[[Toolpack:M3UA_Use_Case_A:Configure_MTP2_A|Configure MTP2]]
|
*[[Toolpack:M3UA_Use_Case_A:Configure_MTP3_A|Configure MTP3]]
**
**
|
*[[Toolpack:M3UA_Use_Case_A:Configure_IP_Interface_A|Configure IP Interface]]
**
**
|
*[[Toolpack:M3UA_Use_Case_A:Configure_SCTP_A|Configure SCTP]]
**
**
|
*[[Toolpack:M3UA_Use_Case_A:Configure_M3UA_A|Configure M3UA]]
**
**
|}

====Configure Tmedia Gateway 2 for M3UA ASP====
{| colspan="4" border="1" class="wikitable"
! width="400" style="background: none repeat scroll 0% 0% rgb(0, 83, 139);" | Configure IP Interface
! width="400" style="background: none repeat scroll 0% 0% rgb(0, 83, 139);" | Configure M3UA
! width="400" style="background: none repeat scroll 0% 0% rgb(0, 83, 139);" | Configure ISUP
! width="400" style="background: none repeat scroll 0% 0% rgb(0, 83, 139);" | Configure NAP
|-
|
*Configuration Guidelines:
*[[Toolpack:M3UA_Use_Case_B:Configure_IP_Interface_A|Configure IP Interface]]
|
*Configuration Guidelines:
*[[Toolpack:M3UA_Use_Case_B:Configure_M3UA_A|Configure M3UA]]
**
**
|
*Configuration Guidelines:
*[[Toolpack:M3UA_Use_Case_B:Configure_ISUP_A|Configure ISUP]]
**
**
|
*Configuration Guidelines:
*[[Toolpack:M3UA_Use_Case_B:Configure_NAP_A|Configure NAP]]
**
**
|}

File:3200 Series Installation Guide Issue6.0.pdf

2019-07-11T14:24:56Z

Cbilodeau: uploaded a new version of "File:3200 Series Installation Guide Issue6.0.pdf": upgrade to version 6.0i

File:TSG3200 datasheet.pdf

2019-07-10T18:30:33Z

Cbilodeau: uploaded a new version of "File:TSG3200 datasheet.pdf": update to version 1.7

File:TSG800.pdf

2019-07-10T18:10:31Z

Cbilodeau: uploaded a new version of "File:TSG800.pdf": update to version 6.6

add datasheet

User talk:Benoit Fournier

2019-06-14T19:24:28Z

Cbilodeau: Welcome!

User:Benoit Fournier

2019-06-14T19:24:28Z

Cbilodeau: Creating user page with biography of new user.

Lorem ipsum dolor sit amet, consectetur adipiscing elit, sed do eiusmod tempor incididunt ut labore et dolore magna aliqua. Ut enim ad minim veniam, quis nostrud exercitation ullamco laboris nisi ut aliquip ex ea commodo consequat. Duis aute irure dolor in reprehenderit in voluptate velit esse cillum dolore eu fugiat nulla pariatur. Excepteur sint occaecat cupidatat non proident, sunt in culpa qui officia deserunt mollit anim id est laborum.

Toolpack:Creating a SIP Domain SBC A

2019-06-07T14:11:48Z

Cbilodeau: add steps to include domain to nap and ordered registrar

=== '''''Applies to version(s): v3.0''''' ===
{{DISPLAYTITLE:Creating a SIP Domain}}
A SIP domain represents a grouping of devices (or users) that can communicate with one another.
You must configure SIP Registration Domain for your system. The first step in doing so is to create a SIP Domain:

1- Click '''SIP Domain''' in the navigation panel

[[Image:SIP_Domain_NavigationMenu.png]]

2- Click '''Create New Domain'''

[[Image:Create_SIP_Domain.png]]

3- Create the new Domain:

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain (domain can be a FQDN or an IP address)
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expire''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. Higher value are replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request are being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request are being refused passed this threshold.
** Select the Registration '''Registrar Selection Mode''' to the registrar:
*** '''Active/Standby''': The active registrar is the one with status up and the lowest priority (automatic fallback).
** Enable the '''Forward After Switchover''': After a switchover, the first register refresh request from each user will be forwarded to the active registrar.
* Click '''Create'''

[[Image:Create_SIP_Domain1.png]]

4- Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:Create_SIP_Domain2.png]]
 
[[Image:Create_SIP_Domain3.png]]

5- Add the domain to one or multiple NAPs from which users are allowed to register for this domain
* Click '''NAPs''' in the navigation panel
* Click on a NAP to bind the SIP Domain to
* Add the SIP Domain to the NAP (bottom of the page):
[[Image:bind_domain_to_nap.png]]

6- Multiple registrars (registration redundancy)
Multiple registrars can be created per domain, for redundancy.
* Registrars are ordered by priority
* Users' registration requests are forwarded to the available registrar with the highest priority in the list
* SIP 'options' polling is used to determine which registrars are available or not

[[Image:multiple_registrars.png]]

== Create routes for registered users ==
In most cases, TBSC will be required to route SIP calls (SIP Invite) to registered users. That's done by [[Sip_registration_forwarding#SIP_Calls_routing|creating appropriate routes]].

Toolpack:Creating a SIP Domain SBC A

2019-06-07T14:03:39Z

Cbilodeau: revert changes to reflect 3.0

Toolpack:Tsbc Protocol Stack Settings 3.1

2019-06-07T13:58:54Z

Cbilodeau: change link for 3.1

{{DISPLAYTITLE:FreeSBC Web Portal 3.1: Protocol Stacks}}

'''Applies to FreeSBC and Tmedia products'''

=Protocol Stacks=

===SSL Certification===
'''Applies to when SIP is used over TLS'''
{| 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;" | Certificates
! width="250" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | TLS Profiles
|-
| valign="top" |
*[[Create_A_SIP_Certificate_A|Create a SIP Certificate]]

| valign="top" |
*[[Toolpack:Tsbc_TLS_Profiles|Create a TLS Profile]]
|}

==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 Certificates and TLS Profiles
! 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 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 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 4a) Configure Network Access Points (NAPs)
! 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 4b) Configure Network Access Points (NAPs)
|-
| valign="top" |
*[[Toolpack:Creating_an_IP_Port_Range_SBC A|Create an IP port range]]

| valign="top" |
*[[Create_A_SIP_Certificate_A|Create a SIP Certificate]]
*[[Toolpack:Tsbc_TLS_Profiles|Create a TLS Profile]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Stack_SBC A|Create a SIP stack]]
*[[Toolpack:Creating_a_SIP_Transport_Server_SBC A|Create a SIP transport server]]

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

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

| valign="top" |
*[[Toolpack:Allocating_a_SIP_Open_Network_Access_Point_(NAP)_SBC A|Allocate an open SIP NAP]]

|}

===SIP Register===

{| 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 SIP Domain
! width="250" 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 Register Filtering Rule (Optional)
|-
| valign="top" |
*[[Toolpack:Creating_a_SIP_Domain_SBC B|Create a SIP domain]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_A|Create a filtering rule]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Condition_A|Add a filtering rule condition]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Action_A|Add a filtering rule action]]
|}

===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 D|Enable 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 D|Enable 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
|-
| valign="top" |
*[[Enabling_Remote_NAT_Traversal_C|Enable Remote NAT Traversal]]
|}
</div>

<div class="mw-collapsible mw-collapsed" data-collapsetext="DNS" data-expandtext="DNS" 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;" | DNS
|-
| valign="top" |
*[[Toolpack:Configuring_DNS_Parameters_C|Enable DNS]]
|}
</div>

===SIP Service Affecting Parameters===

<div class="mw-collapsible mw-collapsed" data-collapsetext="SIP Service Affecting Parameters" data-expandtext="SIP Service Affecting Parameters" style="width: 400px;">
When modifying the following parameters, the SIP stack will be deallocated and reallocated, resulting in a loss of active calls. Be careful when changing these parameters: 

'''Quirks'''
* All Parameters
'''Header Parameters'''
*Domain Name
*Add rPort
'''DNS Parameters'''
*Enabled
*Use Cache
*Cache Expire Time
*Address and Port
</div>

Toolpack:Creating a SIP Domain SBC B

2019-06-07T13:58:07Z

Cbilodeau: create page for 3.1

=== '''''Applies to version(s): v3.1''''' ===

{{DISPLAYTITLE:Creating a SIP Domain}}
In a FreeSBC or Tmedia configuration, a "SIP Domain" represents a domain name to which users can register. It also defines registrars to which FreeSBC/Tmedia will forward registration requests.

The creation of SIP domains is performed as follows:

1. Select '''SIP Domain''' in the navigation panel

[[Image:Create_SIP_DomainNavMenu.png]]

2. Click '''Create New SIP Regisration Domain'''

[[Image:Create_SIP_Domain.png]]

3. Configure the SIP Registration Domain

[[Image:Create_SIP_Domain1.png]]

* Enter a configuration '''Name''' for this domain.
* Enter a '''Domain Name''' for the SIP Registration Domain. A domain can be a FQDN or an IP address.
* Set the number '''Maximum Registered Users''' for this domain
* Set the Expires value used by SBC when the remote device doesn't supply one ('''Default Contact Expiration''')
* Select '''Routing Method''' the system will use to route calls to registered users (if enabled in routing scripts).
** '''Register source''': Sends SIP Invite to the registering source IP address.
** '''Contact''': Sends SIP Invite to the 'contact' from the Register message.
* Set the '''Default Contact Expiration''', this value will be used when no ''Expires'' value is supplied by the user agent.
* Set the '''Minimum Contact Expiration''', this is the minimum ''Expires'' value that can be supplied by a user agent. Lower values will be rejected with a 423 'Interval too brief' response.
* Set the '''Maximum Contact Expiration''', this is the maximum ''Expires'' value that can be supplied by a user agent. Higher value are replaced by this parameter.
* Forwarding Parameters:
** Select the Registration '''Forwarding Mode''' to the registrar:
*** '''Contact Remapping''': Changes the user and the IP address.
*** '''Contact Passthrough''': Doesn't change anything. Enables devices to be contacted directly without going through the SBC.
** Set '''Minimum Registrar Expiration''', this is the minimum ''Expires'' value sent by the SBC to the registrar. If a user agent 'Expires' value is greater than this parameter, the SBC will do rate adaptation between the user agent and the Registrar.
** Set '''Maximum Pending Register Forward''', the maximum number of simultaneous pending register requests allowed for this domain. New REGISTER request are being refused passed this threshold.
** Set '''Maximum Simultaneous Register Forward''', the maximum number of simultaneous active register requests allowed for this domain. New REGISTER request are being refused passed this threshold.
** Select the Registration '''Registrar Selection Mode''' to the registrar:
*** '''Active/Standby''': The active registrar is the one with status up and the lowest priority (automatic fallback).
** Enable the '''Forward After Switchover''': After a switchover, the first register refresh request from each user will be forwarded to the active registrar.
* Click '''Create'''

4. Verify that the '''Sip registration domain was successfully created''' message appears

[[Image:Create_SIP_Domain2.png]]
 
[[Image:Create_SIP_Domain3.png]]

5. Create one or multiple SIP registrars for the SIP domain
* Click the SIP domain that was just created
* Click '''Create New SIP Registration Registrar'''

[[Image:Create_SIP_registrar.png]]

6. Enter the name of SIP Registration Registrar
* Select a NAP
* Click '''Create'''

[[Image:Sip5.png]]

'''Multiple registrars (registration redundancy)'''
Multiple registrars can be created per domain, for redundancy.
* Registrars are ordered by priority
* Users' registration requests are forwarded to the available registrar with the highest priority in the list
* SIP 'options' polling is used to determine which registrars are available or not

[[Image:multiple_registrars.png]]

'''Create routes for registered users'''

In most cases, FreeSBC will be required to route SIP calls (SIP Invite) to registered users. This is done by [[Sip_registration_forwarding#SIP_Calls_routing|creating appropriate routes]].

Toolpack:Tsbc Protocol Stack Settings 3.1

2019-06-07T13:57:29Z

Cbilodeau: remove registrar configuration

{{DISPLAYTITLE:FreeSBC Web Portal 3.1: Protocol Stacks}}

'''Applies to FreeSBC and Tmedia products'''

=Protocol Stacks=

===SSL Certification===
'''Applies to when SIP is used over TLS'''
{| 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;" | Certificates
! width="250" style="background: rgb(239, 239, 239) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;" | TLS Profiles
|-
| valign="top" |
*[[Create_A_SIP_Certificate_A|Create a SIP Certificate]]

| valign="top" |
*[[Toolpack:Tsbc_TLS_Profiles|Create a TLS Profile]]
|}

==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 Certificates and TLS Profiles
! 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 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 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 4a) Configure Network Access Points (NAPs)
! 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 4b) Configure Network Access Points (NAPs)
|-
| valign="top" |
*[[Toolpack:Creating_an_IP_Port_Range_SBC A|Create an IP port range]]

| valign="top" |
*[[Create_A_SIP_Certificate_A|Create a SIP Certificate]]
*[[Toolpack:Tsbc_TLS_Profiles|Create a TLS Profile]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Stack_SBC A|Create a SIP stack]]
*[[Toolpack:Creating_a_SIP_Transport_Server_SBC A|Create a SIP transport server]]

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

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

| valign="top" |
*[[Toolpack:Allocating_a_SIP_Open_Network_Access_Point_(NAP)_SBC A|Allocate an open SIP NAP]]

|}

===SIP Register===

{| 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 SIP Domain
! width="250" 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 Register Filtering Rule (Optional)
|-
| valign="top" |
*[[Toolpack:Creating_a_SIP_Domain_SBC A|Create a SIP domain]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_A|Create a filtering rule]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Condition_A|Add a filtering rule condition]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Action_A|Add a filtering rule action]]
|}

===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 D|Enable 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 D|Enable 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
|-
| valign="top" |
*[[Enabling_Remote_NAT_Traversal_C|Enable Remote NAT Traversal]]
|}
</div>

<div class="mw-collapsible mw-collapsed" data-collapsetext="DNS" data-expandtext="DNS" 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;" | DNS
|-
| valign="top" |
*[[Toolpack:Configuring_DNS_Parameters_C|Enable DNS]]
|}
</div>

===SIP Service Affecting Parameters===

<div class="mw-collapsible mw-collapsed" data-collapsetext="SIP Service Affecting Parameters" data-expandtext="SIP Service Affecting Parameters" style="width: 400px;">
When modifying the following parameters, the SIP stack will be deallocated and reallocated, resulting in a loss of active calls. Be careful when changing these parameters: 

'''Quirks'''
* All Parameters
'''Header Parameters'''
*Domain Name
*Add rPort
'''DNS Parameters'''
*Enabled
*Use Cache
*Cache Expire Time
*Address and Port
</div>

Toolpack:Creating a SIP Registrar SBC A

2019-06-07T13:55:30Z

Cbilodeau: remove 3.1 support

=== '''''Applies to version(s): v3.0''''' ===
{{DISPLAYTITLE:Creating a SIP Registrar}}
A SIP registrar represents a SIP endpoint that provide a location service.
You must configure SIP Registrar for your system. The first step in doing so is to select your SIP Domain:

1- Click on your domain in the '''SIP Domain List'''

[[Image:Select_SIP_Domain.png]]

2- Click '''Create New SIP Registration Registrar'''

[[Image:Create_New_SIP_Registrar.png]]

3- Create the new SIP Registration Registrar

* Enter a '''Name''' for the SIP Registration Registrar
* Enter an '''Address''' to reach the registrar
* Select the registrar destination '''Port'''
* Select which local interface will be use to send REGISTER message in '''SIP Transport Server'''
* '''Poll Registrar''' enabled by default. Sends an OPTIONS message to determine registrar availability.
* '''Polling Interval''' in seconds. Time between each OPTIONS sending message.
* '''Response Timeout''' in second. Maximum waiting time for a response to a registrar polling.
* Click '''Create'''

[[Image:Create_New_SIP_Registrar3.png]]

4- Verify that the '''Sip registration registrar was successfully created''' message appears

[[Image:Create_New_SIP_Registrar2.png]]

User talk:Alan Percy

2019-06-05T14:25:26Z

Cbilodeau: Welcome!

User:Alan Percy

2019-06-05T14:25:26Z

Cbilodeau: Creating user page with biography of new user.

Marketing
MediaWiki is a free software open source wiki package written in PHP, originally for use on Wikipedia. It is now also used by several other projects of the non-profit Wikimedia Foundation and by many other wikis, including this website, the home of MediaWiki.

MediaWiki is a free software open source wiki package written in PHP, originally for use on Wikipedia. It is now also used by several other projects of the non-profit Wikimedia Foundation and by many other wikis, including this website, the home of MediaWiki.

MediaWiki is a free software open source wiki package written in PHP, originally for use on Wikipedia. It is now also used by several other projects of the non-profit Wikimedia Foundation and by many other wikis, including this website, the home of MediaWiki.

Toolpack prerequisite Windows XP 2-5

2019-05-01T11:54:35Z

Cbilodeau: /* Toolpack System Database */

This page shows the Toolpack 2-5, 2-6, 2-7 prerequisites for Windows XP.

== Toolpack Prerequisite ==

'''WARNINGS:'''
* Do not run toolpack with a mySQL server version other than 5.1. There are known problems with other versions and Toolpack.
* Do not run toolpack with a ODBC connector other than 3.51, as other versions (such as 5.2) are known to cause memory leak or other problems running Toolpack.

=== Toolpack Basic packages ===

The software listed below is required by the host machines.

1) Go to TelcoBridges download site and get:
http://download.distribution.telcobridges.com/3rdparty/prerequisite/bin/windows/iconv-1.9.2.win32.zip
http://download.distribution.telcobridges.com/3rdparty/prerequisite/bin/windows/libxml2-2.7.3.win32.zip
http://download.distribution.telcobridges.com/3rdparty/prerequisite/bin/windows/zlib-1.2.3.win32.zip
2) Decompress and copy the DLL files to C:\Windows\System32
iconv.dll
libxml2.dll
zlib1.dll

==== MSVC8 Redistributable ====
If you plan on using TBToolPack-x.y.zz-msvc8-Setup.exe, then you need to apply all windows updates from MicroSoft AND install the following:

[http://download.distribution.telcobridges.com/3rdparty/prerequisite/bin/windows/vcredist_x86.exe Microsoft MSVC8 Redistributables 8.0.50727.4053 after kb973923 automatic update]

[http://download.distribution.telcobridges.com/3rdparty/prerequisite/bin/windows/vcredist_msdev2005_x86_KB2467175.exe Microsoft MSVC8 Redistributables 8.0.50727.5592 after kb2467175 automatic update]

=== Toolpack System Database ===

The Toolpack system database is used by the main host and requires mySQL database server version 5.1, and ODBC driver 3.51. It can be downloaded from the following location:

1) Go to TelcoBridges download site and get:
http://download.distribution.telcobridges.com/3rdparty/prerequisite/bin/windows/mysql-5.1.56-win32.msi


'''*** WARNING: Do not run toolpack with a mySQL server version other than 5.1. There are known problems with other versions and Toolpack.'''


2) Install MySQL 5.1

*Choose complete installation
*Run MySQL Server Instance Configuration Wizard. (You should be prompted to do so after selecting complete installation.)
*Choose standard configuration
*Install as a Windows service and check, Launch Automatically.
*Choose Include Bin Directory in Windows PATH.
*Enter a User Name and Password, and note them down on paper as you will need them later in the installation process.
*Enable access from remote machines if you plan to run Tmedia Toolpack on more than one host machine.

3) Create new database user (tbdb) with all privilege access
*Log in mysql and create a new user with priviledge

mysql -u root
CREATE USER '[DatabaseUsername]'@'%' IDENTIFIED BY '[DatabasePassword]';
GRANT ALL privileges ON *.* TO '[DatabaseUsername]'@'%' IDENTIFIED BY '[DatabasePassword]';
exit

Example

mysql -u root
CREATE USER 'tbdb'@'%' IDENTIFIED BY 'tbdbpw';
GRANT ALL privileges ON *.* TO 'tbdb'@'%' IDENTIFIED BY 'tbdbpw';
exit

4) Install Mysql ODBC driver
*Get and install Mysql ODBC connector at:
http://download.distribution.telcobridges.com/3rdparty/prerequisite/bin/windows/mysql-connector-odbc-3.51.27-win32.zip
*Install Mysql ODBC connector


'''*** WARNING: Do not run toolpack with a ODBC connector other than 3.51, as other versions (such as 5.2) are known to cause memory leak or other problems running Toolpack.'''


5) Enable mysql database replication
*Open configuration file at location similar to c:\Program Files\MySQL\MySQL Server 5.1\my.ini
*Add 2 new variable in the mysql config file, and make sure it is in the [mysqld] section:
log-bin
server-id=[UniqueId] <--- [UniqueId] must be a non-zero value, and must be unique when used one each host of an HA system

Add max_allowed_packet to increase the maximum size of binary blobs in the database.
Make sure it is in the [mysqld] section AND [mysqldump] section (create the [mysqldump] section if it doesn't exist)
max_allowed_packet = 200MB

Example:
[mysqld]
port=3306
basedir="C:/Program Files/MySQL/MySQL Server 5.1/"
datadir="C:/Documents and Settings/All Users/Application Data/MySQL/MySQL Server 5.1/Data/"
default-character-set=latin1
default-storage-engine=INNODB
# Enable mysql database replication
log-bin
server-id=132
max_allowed_packet = 200MB
#
sql-mode="STRICT_TRANS_TABLES,NO_AUTO_CREATE_USER,NO_ENGINE_SUBSTITUTION"
max_connections=100

# section [mysqld_safe]
table_cache=256
tmp_table_size=18M
thread_cache_size=8

[mysqldump]
max_allowed_packet = 200MB

6) Disable virus scanning

It is '''mandatory''' to disable the anti-virus software’s on access scan on the MYSQL database directory (generally C:\Program Files\MySQL\MySQL Server 5.0\data or C:\ProgramData\MySQL\MySQL Server 5.1\data) and the Windows temporary directory(typically identified by the TEMP environmental variable), as this will prevent database backup operation to complete properly upon Toolpack System Update.

7) Restart the mysql service
net stop MySQL
net start MySQL
*Warning: You may have a different service name in your system (ex. MySQL51, MySQL50, etc.). To open the service control panel, run:
services.msc

=== Toolpack System Web Portal ===

1) Install Ruby
*Go to TelcoBridges download site and get:
http://download.distribution.telcobridges.com/3rdparty/prerequisite/bin/windows/ruby186-27_rc2.exe
*Install Ruby 1.8.6, check the option "Enable RubyGems", make sure the ''Add Ruby Executable to your path'' option is checked
2) Install Mongrel using the following commands.
*gem install mongrel --include-dependencies
*choose option 4. mongrel 1.1.5 (x86-mswin32-60)
3) Install additional ruby gems
* gem install json

=== Toolpack Stream Server ===

The Tmedia Toolpack Stream Server requires WinPCAP 4.0 or later. It can be downloaded from the following location:

1) Go to http://www.winpcap.org/ and download WinPcap_4_0_2.exe

2) Install WinPCAP 4.0

=== Toolpack Development Prerequisite ===

==== Cygwin ====
Telcobridges source package is compiled using unix Make application. For this you need to install Cygwin (Linux API layer for Windows).

1) install the latest release of cygwin.
http://www.cygwin.com/

2) The "make" package must be manually selected, it is not part of the standard cygwin installation.
*In the installer view, change to the "Full" view, search for the "make" package, click to install the latest version.

==== MSDEV ====
The Makefiles uses Visual Studio C++ compiler and linker. '''The result compiled code has no dependency on the cygwin.dll library.'''

The following Microsoft Visual Studio product could be be installed:
* Visual Studio 6.0 (1998) - MSVC6
* Visual Studio 2005 - MSVC8

==== Folder "BashPrompt" Menu item ====
This small registry entry will add a folder command to the Explorer application. This helper command can be used by right-clicking on a folder where you want to launch a compilation, it will open a console window and setup the proper compiling environment.

For Visual Studio 6.0 user, create a file named "BashPrompt1998.reg" with the following content:

<pre>
Windows Registry Editor Version 5.00
[HKEY_CLASSES_ROOT\Folder\shell\BashPrompt1998]
[HKEY_CLASSES_ROOT\Folder\shell\BashPrompt1998\command]
@="cmd.exe /k pushd \"%L\" && \"C:\\Program Files\\Microsoft Visual Studio\\VC98\\Bin\\VCVARS32.BAT\" && bash"
</pre>

*Make sure the path is the same as your installation path.
*Double click on the file to install in the registry.

For Visual Studio 2005 user, create a file named "BashPrompt2005.reg" with the following content:

<pre>
Windows Registry Editor Version 5.00
[HKEY_CLASSES_ROOT\Folder\shell\BashPrompt2005]
[HKEY_CLASSES_ROOT\Folder\shell\BashPrompt2005\command]
@="cmd.exe /k pushd \"%L\" && \"C:\\Program Files\\Microsoft Visual Studio 8\\VC\\Bin\\VCVARS32.BAT\" && bash"
</pre>

*Make sure the path is the same as your installation path.
*Double click on the file to install in the registry.

==== Environment variable ====
To finalize the compiling environment, you must set the following system variable environment ('''Start->Settings->Control Panel->System->Advanced Tab->Environment variables'''):

1) add the following path to the "PATH" environment variable, each is separated by a semi-column (;)
* C:\cygwin\bin;
* C:\Ruby\bin;

2) add the following variable:
{| cellpadding="5" border="1" class="wikitable"
|-
! width="80" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | Name
! width="80" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | Value
|-
! width="80" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | MAKE_MODE
! width="80" style="background: none repeat scroll 0% 0% rgb(255, 255, 255);" | unix
|}

== Validation Guide (Optional) ==
If you would like to verify your installation go to the [[Toolpack_Installation:Validate_Toolpack_Installation_2-5|Toolpack Validation Guide]]

== Toolpack System installation ==

Now that the prerequisite are installed, now it is time to install the [[Toolpack_installer_2-4|Toolpack system]]

Create A DNS Local Entry A

2019-04-18T11:48:04Z

Cbilodeau: add missing steps

=== '''''Applies to version(s): v3.1''''' ===
{{DISPLAYTITLE:Creating a DNS Local Entry}}

'''Note: ''' This article describes the creation of optional DNS local entries.

You can create one or more DNS Local Entries. This is useful when you want to bypass the DNS server. This is usually done for troubleshooting.

1- Click '''DNS''' in the navigation panel

[[Image:dns_group_0.png]] 

2- Select a DNS group, from the '''DNS group list'''

[[Image:dns_entry_0.png]] 

3- Create a new DNS local entry for the DNS server.
*Click '''Local Entries'''.

[[Image:dns_entry_1.png]] 

4- Click '''Create New DNS local entry'''.

[[Image:dns_entry_2.png]] 

5- Configure the DNS local entry.
*Enter a name for the DNS local entry
*Enter an FQDN.

*Click '''Create'''.

[[Image:dns_entry_3.png]] 

6- Verify that the '''DNS local entry was successfully created''' message is displayed.

[[Image:dns_entry_4.png]] 

7- Once the DNS Local entry is created you add one or more DNS local entry records to it.
*Click '''Create New DNS local entry record'''.

[[Image:dns_entry_5.png]] 

8- Configure the DNS local entry record.
*Enter a name for the DNS local entry record.
*Enter an IP address.
*Enter a port number.
*Enter a service name.
*Click '''Create'''.

[[Image:dns_entry_6.png]] 

The DNS local entry with its accompanying records are displayed in the DNS Local Entries list.

[[Image:dns_entry_9.png]] 
==List of Parameters==

'''DNS Local Entries'''
* [[Parameter: Name|Name]]
* [[Parameter: FQDN|FQDN]]
* [[Parameter: Enables|Enabled]]

'''DNS Local Entry Records'''
* [[Parameter: IP Address|IP Address]]
* [[Parameter: Port|Port]]
* [[Parameter: Service|Service]]

Create A DNS Local Entry A

2019-04-16T19:31:54Z

Cbilodeau: removed extra steps

=== '''''Applies to version(s): v3.1''''' ===
{{DISPLAYTITLE:Creating a DNS Local Entry}}

'''Note: ''' This article describes the creation of optional DNS local entries.

You can create one or more DNS Local Entries. This is useful when you want to bypass the DNS server. This is usually done for troubleshooting.

1- Click '''DNS''' in the navigation panel

[[Image:dns_group_0.png]] 

2- To create a DNS group, click '''Create New DNS Group'''

[[Image:dns_entry_0.png]] 

3- Create a new DNS local entry for the DNS server.
*Click '''Local Entries'''.

[[Image:dns_entry_1.png]] 

4- Click '''Create New DNS local entry'''.

[[Image:dns_entry_2.png]] 

5- Configure the DNS local entry.
*Enter a name for the DNS local entry
*Enter an FQDN.

*Click '''Create'''.

[[Image:dns_entry_3.png]] 

6- Verify that the '''DNS local entry was successfully created''' message is displayed.

[[Image:dns_entry_4.png]] 

7- Once the DNS Local entry is created you add one or more DNS local entry records to it.
*Click '''Create New DNS local entry record'''.

[[Image:dns_entry_5.png]] 

8- Configure the DNS local entry record.
*Enter a name for the DNS local entry record.
*Enter an IP address.
*Enter a port number.
*Enter a service name.
*Click '''Create'''.

[[Image:dns_entry_6.png]] 

14- Verify that the '''DNS local entry record was successfully created''' message is displayed.

[[Image:dns_entry_7.png]] 

15- Click '''Save'''

[[Image:dns_entry_8.png]] 

The DNS local entry with its accompanying records are displayed in the DNS Local Entries list.

[[Image:dns_entry_9.png]] 

==List of Parameters==

'''DNS Local Entries'''
* [[Parameter: Name|Name]]
* [[Parameter: FQDN|FQDN]]
* [[Parameter: Enables|Enabled]]

'''DNS Local Entry Records'''
* [[Parameter: IP Address|IP Address]]
* [[Parameter: Port|Port]]
* [[Parameter: Service|Service]]

Sbc Tutorial Guide v3.1

2019-04-12T17:18:53Z

Cbilodeau: add dns in list

{{DISPLAYTITLE:FreeSBC 3.1 Web Portal Tutorial Guide}}

__NOTOC__
=Introduction=
The Web Portal Tutorial provides you with a scenario-based approach to configuring and monitoring the status of [[FreeSBC|FreeSbc]] systems, using the Web Portal configuration tool. Complete general installation configuration steps, before you begin configuring your specific application.

==Prerequisites==
[[FreeSBC|FreeSBC]] devices must be installed as described in their respective [[Product_Installation_SBC|installation guides]].

==Web Portal Categories==
Web Portal configuration and status topics are grouped into the following major categories. It is recommended that you follow this order.

*System [[Toolpack:Tsbc System Settings 3.1|(Configuration)]] [[Toolpack:Tsbc System Status 3.1|(Status)]]
**Host Role
**SBC
**Access and User Management
**Database Backup
**SNMP
**Configuration
**Minor Package Upgrade
**Software Licenses
**Statistics
**Audit Logs

*IP Network [[Toolpack:Tsbc IP Network Settings 3.1|(Configuration)]] [[Toolpack:Tsbc_IP_Network_Status_3.1|(Status)]]
**IP Network
**DNS
**NAT
**Transcoding Units

*Protocol Stack [[Toolpack:Tsbc_Protocol_Stack_Settings_3.1|(Configuration)]] [[Toolpack:Tsbc_Protocol_Stack_Status_3.1|(Status)]]
**Certificates and TLS Profiles
**SIP
**Calls
**NAP

*CDR [[Toolpack:Tsbc_CDR_Settings_3.1|(Configuration)]]
**Text CDR
**RADIUS [[Toolpack:Tsbc_CDR_Status_3.1|(Status)]]

*Fax [[Toolpack:Tsbc_Fax_Settings_3.1|(Configuration)]]
**FAX passthrough (silence suppression)
**FAX T.38
**FAX passthrough (NSE)
**FAX passthrough (VBD)

*Call Route [[Toolpack:Tsbc_Call Routes_Settings_3.1|(Configuration)]]
**Static Route
**Routing Script
**Label Routing
**RADIUS Authorization and Authentication [[Toolpack:Tsbc_CDR_Status_3.1|(Status)]]

*Lawful Intercept [[Toolpack:Tsbc Lawful Interception 3.1|(Configuration)]] [[(Status)]]

*Troubleshooting [[Toolpack:Tsbc Troubleshooting 3.1|(Configuration)]]
**CallTrace
**Test Call
**Target Recording
**Ethernet Capture

*Use Cases [[Toolpack:Tsbc Use_Cases A|(Configuration)]]
**IP Network
**Redirect
**Transcoding Unit

[[category:Toolpack]]

Create A DNS Group A

2019-04-04T18:26:18Z

Cbilodeau: fix typo

=== '''''Applies to version(s): v3.1''''' ===
{{DISPLAYTITLE:Creating a DNS Group}}

'''Note: ''' This article describes how to configure a DNS Group, which is not the same as creating a [[Toolpack:set_DNS_A|DNS server group for the Linux host]].

You can create one or more DNS Groups to manage the DNS Server traffic of Fully Qualified Domain Names (FQDN).

1- Click '''DNS''' in the navigation panel

[[Image:dns_group_0.png]] 

2- Click '''Create New DNS Group'''

[[Image:dns_group_1.png]] 

3- Enter a name for the DNS Group.
* Click '''Create'''

[[Image:dns_group_2.png]] 

4- Verify that the '''"DNS group was successfully created"''' message appears

[[Image:dns_group_3.png]] 

5- Create a DNS Server for the DNS Group.
*Click '''Create New DNS Server'''.

[[Image:dns_group_4.png]] 

6- Configure the DNS Server.
*Enter a name for the DNS server.
*Enter an IP address for the server.
*Select a Local IP interface. This is the interface through which the DNS requests will be transmitted.

*Click '''Create'''.

[[Image:dns_group_5.png]] 

7- Verify that the '''"DNS server was successfully created"''' message appears

[[Image:dns_group_6.png]] 

8- Select an IP interface from the list of available interfaces. The interfaces that are added will have
their FQDN requests resolved by this DNS group.

*Click [[Image:dns_group_8.png]] to move the available interface over to the '''Current''' list.

[[Image:dns_group_7.png]] 

*Click '''Create'''.

The DNS Group with the new DNS Server and DNS Local entries are displayed.

[[Image:dns_group_d.png]] 

==List of Parameters==

* [[Parameter: Name|Name]]
* [[Parameter: Enables|Enabled]]
* [[Parameter: Use Cache|Use Cache]]
* [[Parameter: Cache Expire Time|Cache Expire Time]]
* [[Parameter: Query Timer|Query Timer]]
* [[Parameter: Query Retry Count|Query Retry Count]]
 

<div class="mw-collapsible mw-collapsed" data-collapsetext="DNS Server" data-expandtext="DNS Server" style="width: 400px;">
{| class="wikitable"
|-
* [[Parameter: Name|Name]]
* [[Parameter: Enabled|Enabled]]
* [[Parameter: Server Address|Server Address]]
* [[Parameter: Local IP Interface|Local IP Interface]]

'''Advanced Parameters'''
* [[Parameter: Server DNS Port|Server DNS Port]]
* [[Parameter: TOS to use in IP header|TOS to use in IP header]]
* [[Parameter: Protocol|Protocol]]
* [[Parameter: Local Port|Local Port]]
|}
</div>

<div class="mw-collapsible mw-collapsed" data-collapsetext="IP Interface" data-expandtext="IP Interface" style="width: 400px;">
{| class="wikitable"
|-
* [[Parameter: Interface name|Interface name]]
* [[Parameter: Services to use|Services to use]]
* [[Parameter: Host VLAN|Host VLAN]]
* [[Parameter: Use DHCP|Use DHCP]]
|}
</div>

Toolpack:Tsbc TLS Profiles

2019-03-22T15:38:00Z

Cbilodeau: fix syntax

=== '''''Applies to version(s): v3.1''''' ===
{{DISPLAYTITLE:Configuring TLS Profiles}}

Multiple TLS profiles can be created in [[Toolpack]]. A TLS profile is a collection of certificates and other settings that define the behavior of TLS connections. Multiple TLS profiles can be used to expose different certificates and options to different TLS connections in [[Toolpack]], athough one TLS profile is generally enough.

== Create TLS profile ==

1. Select '''TLS Profiles''' from the navigation panel

[[Image:ConfigureTlsProfile_0.png]]

2. Click ''' Create New TLS Profile'''

[[Image:ConfigureTlsProfile_1.png]]

3. Configure TLS Profile parameters

* Provide a name that is meaningful to you.
* Decide if client authentication is required. This means that incoming connections require the remote side to present a trusted certificate.
* Select a local certificate (either one imported earlier, or the "Default" certificate which is automatically generated by Toolpack and is also used for HTTPS)
* Click "Create"
* Select one or multiple "trusted" certificates, which correspond to remote equipment's certificates (or root certificates from which remote certificates are signed)

[[Image:ConfigureTlsProfile_2.png|500px]]

[[Image:ConfigureTlsProfile_3.png|500px]]

== List of Parameters ==

*[[Parameter: Name|Name]]
*[[Parameter: Require client authentication|Require client authentication]]
*[[Parameter: Local certificate|Local certificate]]

Toolpack:Tsbc Protocol Stack Settings 3.0

2019-03-22T15:26:02Z

Cbilodeau: remove security section - not available for 3.0

{{DISPLAYTITLE:FreeSBC Web Portal 3.0: Protocol Stacks}}

'''Applies to FreeSBC and Tmedia products'''

=Protocol Stacks=

==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) 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 4a) Configure Network Access Points (NAPs)
! 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 4b) Configure Network Access Points (NAPs)
|-
| valign="top" |
*[[Toolpack:Creating_an_IP_Port_Range_SBC A|Create an IP port range]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Stack_SBC A|Create a SIP stack]]
*[[Toolpack:Creating_a_SIP_Transport_Server_SBC A|Create a SIP transport server]]

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

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

| valign="top" |
*[[Toolpack:Allocating_a_SIP_Open_Network_Access_Point_(NAP)_SBC A|Allocate an open SIP NAP]]

|}

===SIP Register===

{| 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 SIP Domain
! width="250" 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 Register Filtering Rule (Optional)
|-
| valign="top" |
*[[Toolpack:Creating_a_SIP_Domain_SBC A|Create a SIP domain]]
*[[Toolpack:Creating_a_SIP_Registrar_SBC A|Create a SIP registrar]]

| valign="top" |
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_A|Create a filtering rule]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Condition_A|Add a filtering rule condition]]
*[[Toolpack:Creating_a_SIP_Register_Filtering_Rule_Action_A|Add a filtering rule action]]
|}

===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 D|Enable 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 D|Enable 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
|-
| valign="top" |
*[[Enabling_Remote_NAT_Traversal_C|Enable Remote NAT Traversal]]
|}
</div>

<div class="mw-collapsible mw-collapsed" data-collapsetext="DNS" data-expandtext="DNS" 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;" | DNS
|-
| valign="top" |
*[[Toolpack:Configuring_DNS_Parameters_C|Enable DNS]]
|}
</div>

===SIP Service Affecting Parameters===

<div class="mw-collapsible mw-collapsed" data-collapsetext="SIP Service Affecting Parameters" data-expandtext="SIP Service Affecting Parameters" style="width: 400px;">
When modifying the following parameters, the SIP stack will be deallocated and reallocated, resulting in a loss of active calls. Be careful when changing these parameters: 

'''Quirks'''
* All Parameters
'''Header Parameters'''
*Domain Name
*Add rPort
'''DNS Parameters'''
*Enabled
*Use Cache
*Cache Expire Time
*Address and Port
</div>

User talk:Marc Paquette

2019-02-15T17:00:02Z

Cbilodeau: Welcome!

User:Marc Paquette

2019-02-15T17:00:02Z

Cbilodeau: Creating user page with biography of new user.

Lorem Ipsum is simply dummy text of the printing and typesetting industry. Lorem Ipsum has been the industry's standard dummy text ever since the 1500s, when an unknown printer took a galley of type and scrambled it to make a type specimen book. It has survived not only five centuries, but also the leap into electronic typesetting, remaining essentially unchanged. It was popularised in the 1960s with the release of Letraset sheets containing Lorem Ipsum passages, and more recently with desktop publishing software like Aldus PageMaker including versions of Lorem Ipsum.

User talk:Songtao xie

2019-02-15T16:51:33Z

Cbilodeau: Welcome!