From 7051fbefc6d82faee85a2cfaf937039b48ea3d3f Mon Sep 17 00:00:00 2001 From: Sean Bright Date: Fri, 29 May 2026 20:07:18 +0000 Subject: [PATCH] docs: Remove leading blank lines from all code blocks --- .../Assigning-Agents-to-Queues.md | 4 -- .../Controlling-the-way-Queues-Call-Agents.md | 2 - .../Queue-Pre-Acknowledgement-Messages.md | 1 - .../Basic-SLA-Configuration-Example.md | 2 - .../Mobile-Channel-Dialplan-Hints.md | 1 - .../Skinny/Skinny-call-logging.md | 2 - .../Introduction-to-the-Unistim-channel.md | 1 - .../Advanced-pbx_lua-Topics.md | 5 --- .../Dialplan-to-Lua-Reference.md | 15 ------- .../Lua-Dialplan-Examples.md | 2 - .../Lua-Dialplan-Hints.md | 1 - .../Channel-Variables/Case-Sensitivity.md | 9 ----- .../Followme-Realtime.md | 2 - .../Distributed-Device-State/Corosync.md | 10 ----- .../PostgreSQL-CDR-Backend.md | 3 -- .../RADIUS-CDR-Backend.md | 7 ---- .../MSSQL-CEL-Backend.md | 11 ----- .../PostgreSQL-CEL-Backend.md | 2 - .../RADIUS-CEL-Backend.md | 7 ---- ...stalling-and-Configuring-CyberMegaPhone.md | 1 - .../Digium-General-Peering-Agreement.md | 1 - .../About-the-Sounds-Tools.md | 1 - .../Packet-Loss-Concealment-PLC/PLC-Tips.md | 1 - .../Configuration-of-users.conf.md | 2 - .../Creating-Phone-Profiles.md | 1 - ...he-Hoard-Memory-Allocator-with-Asterisk.md | 1 - .../Asterisk-Channel-Data-Stores.md | 2 - .../Modules.md | 5 --- .../Using-the-Configuration-Framework.md | 3 -- .../Locking-in-Asterisk.md | 2 - .../Measuring-SIP-Channel-Performance.md | 4 -- .../Media-Architecture-Proposal.md | 5 --- .../AMI-Event-Documentation.md | 8 ---- ...ion-parsing-with-the-Config-Options-API.md | 2 - .../Unistim-channel-improvements.md | 1 - .../Asterisk-11-Projects/Who-Hung-Up.md | 1 - .../app_macro-Deprecation.md | 4 -- .../Asterisk-Client-Library-Generator.md | 1 - .../Asterisk-12-API-Improvements/index.md | 1 - .../Asterisk-12-Bridging-API.md | 40 ------------------- .../Asterisk-12-Bridging-Project/index.md | 5 --- .../Asterisk-on-OpenSolaris.md | 1 - .../Tests-with-SIPp/Basic-Test-with-SIPp.md | 2 - 43 files changed, 182 deletions(-) diff --git a/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Assigning-Agents-to-Queues.md b/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Assigning-Agents-to-Queues.md index 3101233881..8921891af1 100644 --- a/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Assigning-Agents-to-Queues.md +++ b/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Assigning-Agents-to-Queues.md @@ -10,7 +10,6 @@ Because a single agent can make themselves available to more than one queue, the ##### Agents Log In and Out ``` - context queues-loginout { 6092 => { Answer(); @@ -31,7 +30,6 @@ context queues-loginout { In the above contexts, the agents dial 6092 to log into their queues, and they dial 6093 to log out of their queues. The agent is prompted for their agent number, and if they are logging in, their passcode, and then they are transferred to the proper extension in the queues-manip context. The queues-manip context does all the actual work: ``` - context queues-manip { // Raquel Squelch _[IO]6121 => { @@ -69,7 +67,6 @@ Also note that Rock will be among the first to be called in the sales-general qu The call to queue-success() gives some feedback to the agent as they log in and out, that the process has completed. ``` - macro queue-success(exten) { if( ${queue-announce-success} > 0 ) { switch(${exten:0:1}) { @@ -89,7 +86,6 @@ macro queue-success(exten) { The queue-addremove macro is defined in this manner: ``` - macro queue-addremove(queuename,penalty,exten) { switch(${exten:0:1}) { case I: // Login diff --git a/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Controlling-the-way-Queues-Call-Agents.md b/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Controlling-the-way-Queues-Call-Agents.md index 99f470bf4c..f5df10248e 100644 --- a/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Controlling-the-way-Queues-Call-Agents.md +++ b/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Controlling-the-way-Queues-Call-Agents.md @@ -6,7 +6,6 @@ pageid: 5243037 Notice in the above, that the commands to manipulate agents in queues have "@agents" in their arguments. This is a reference to the agents context: ``` - context agents { // General sales queue 8010 => { @@ -51,7 +50,6 @@ The 8010, 8011, and 8013 extensions are purely for transferring incoming callers Here is the callagent macro, note that if a person in the queue is called, but does not answer, then they are automatically removed from the queue. ``` - macro callagent(device,exten) { if( ${GROUP_COUNT(${exten}@agents)}=0 ) { Set(OUTBOUND_GROUP_ONCE=${exten}@agents); diff --git a/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Queue-Pre-Acknowledgement-Messages.md b/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Queue-Pre-Acknowledgement-Messages.md index 4582e60b14..178f3f0130 100644 --- a/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Queue-Pre-Acknowledgement-Messages.md +++ b/docs/Configuration/Applications/Asterisk-Queues/Configuring-Call-Queues-with-AEL/Queue-Pre-Acknowledgement-Messages.md @@ -6,7 +6,6 @@ pageid: 5243040 If you would like to have a pre acknowledge message with option to reject the message you can use the following dialplan Macro as a base with the 'M' dial argument. ``` - [macro-screen] exten=>s,1,Wait(.25) exten=>s,2,Read(ACCEPT,screen-callee-options,1) diff --git a/docs/Configuration/Applications/Shared-Line-Appearances-SLA/SLA-Configuration-Examples/Basic-SLA-Configuration-Example.md b/docs/Configuration/Applications/Shared-Line-Appearances-SLA/SLA-Configuration-Examples/Basic-SLA-Configuration-Example.md index c29f6c420b..8ae52f7e81 100644 --- a/docs/Configuration/Applications/Shared-Line-Appearances-SLA/SLA-Configuration-Examples/Basic-SLA-Configuration-Example.md +++ b/docs/Configuration/Applications/Shared-Line-Appearances-SLA/SLA-Configuration-Examples/Basic-SLA-Configuration-Example.md @@ -8,7 +8,6 @@ This is an example of the most basic SLA setup. It uses the automatic dialplan g sla.conf: ``` - [line1] type=trunk device=DAHDI/1 @@ -38,7 +37,6 @@ device=SIP/station3 With this configuration, the dialplan is generated automatically. The first DAHDI channel should have its context set to "line1" and the second should be set to "line2" in dahdi.conf. In sip.conf, station1, station2, and station3 should all have their context set to "sla_stations". ``` - [line1] exten => s,1,SLATrunk(line1) diff --git a/docs/Configuration/Channel-Drivers/Mobile-Channel/Mobile-Channel-Dialplan-Hints.md b/docs/Configuration/Channel-Drivers/Mobile-Channel/Mobile-Channel-Dialplan-Hints.md index 76f0fb59e2..1965d1f02d 100644 --- a/docs/Configuration/Channel-Drivers/Mobile-Channel/Mobile-Channel-Dialplan-Hints.md +++ b/docs/Configuration/Channel-Drivers/Mobile-Channel/Mobile-Channel-Dialplan-Hints.md @@ -6,6 +6,5 @@ pageid: 4817207 chan_mobile supports 'device status' so you can do somthing like ``` - exten => 1234,hint,SIP/30&Mobile/dave&Mobile/blackberry ``` diff --git a/docs/Configuration/Channel-Drivers/Skinny/Skinny-call-logging.md b/docs/Configuration/Channel-Drivers/Skinny/Skinny-call-logging.md index 030e28cf6b..2d02db61e8 100644 --- a/docs/Configuration/Channel-Drivers/Skinny/Skinny-call-logging.md +++ b/docs/Configuration/Channel-Drivers/Skinny/Skinny-call-logging.md @@ -12,7 +12,6 @@ If a call is not placed to a channel they will not be recorded in the log. eg a Example (This will produce a logged call): ``` - exten => 100,1,NoOp exten => 100,n,Progress exten => 100,n,Ringing @@ -22,7 +21,6 @@ exten => 100,n,VoicemailMain(${CALLERID(num)@mycontext,s) Example (This will not): ``` - exten => 100,1,NoOp exten => 100,n,VoicemailMain(${CALLERID(num)@mycontext,s) ``` diff --git a/docs/Configuration/Channel-Drivers/Unistim/Introduction-to-the-Unistim-channel.md b/docs/Configuration/Channel-Drivers/Unistim/Introduction-to-the-Unistim-channel.md index c3b89d834e..915acd5614 100644 --- a/docs/Configuration/Channel-Drivers/Unistim/Introduction-to-the-Unistim-channel.md +++ b/docs/Configuration/Channel-Drivers/Unistim/Introduction-to-the-Unistim-channel.md @@ -44,7 +44,6 @@ The line=> entry in unistim.conf does not add an extension in asterisk by defaul If you have this entry on unistim.conf : ``` - [violet] device=006038abcdef line => 102 diff --git a/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Advanced-pbx_lua-Topics.md b/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Advanced-pbx_lua-Topics.md index db67f4bfd9..31839bbfdf 100644 --- a/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Advanced-pbx_lua-Topics.md +++ b/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Advanced-pbx_lua-Topics.md @@ -15,7 +15,6 @@ The `extensions.lua` file is loaded into memory once when the pbx_lua module is This is probably a bad idea. ``` - -- my fancy extensions.lua extensions = {} @@ -37,7 +36,6 @@ The `extensions` Table The `extensions` table is a standard lua table and can be defined however you like. The pbx_lua module loads and sorts the table when it is needed. The keys in the table are context names and each value is another lua table containing extensions. Each key in the context table is an extension name and each value is an extension function. ``` - extensions = { context_table = { extension1 = function() @@ -74,7 +72,6 @@ Apps, Functions, and Variables channel variable: var is the placeholder object ``` - var = channel.my_variable var:set("my value") value = var:get("my value") @@ -85,7 +82,6 @@ value = var:get("my value") dialplan function: fax_modems is the placeholder object ``` - fax_modems = channel.FAXOPT("module") -- the function arguments are stored in the placeholder @@ -99,7 +95,6 @@ value = fax_modems:get() dialplan application: dial is the placeholder object ``` - dial = app.dial -- the only thing we can do with it is execute it diff --git a/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Dialplan-to-Lua-Reference.md b/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Dialplan-to-Lua-Reference.md index 43f532ef65..b35ca7b024 100644 --- a/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Dialplan-to-Lua-Reference.md +++ b/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Dialplan-to-Lua-Reference.md @@ -16,7 +16,6 @@ extensions.conf --------------- ``` - [users] exten => _1XX,1,Dial(SIP/${EXTEN}) @@ -29,7 +28,6 @@ extensions.lua -------------- ``` - extensions = {} extensions.users = {} @@ -51,7 +49,6 @@ extensions.conf --------------- ``` - [users] exten => 100,1,Noop exten => 100,n,Dial("SIP/100") @@ -71,7 +68,6 @@ extensions.lua -------------- ``` - extensions = { users = { [100] = function() @@ -100,7 +96,6 @@ extensions.conf --------------- ``` - exten => 100,1,Noop exten => 100,n,Set(i=0) exten => 100,n,While($[i < 10]) @@ -114,7 +109,6 @@ extensions.lua -------------- ``` - i = 0 while i < 10 do app.verbose("i = " .. i) @@ -130,7 +124,6 @@ extensions.conf --------------- ``` - exten => 100,1,Set(my_variable=my_value) exten => 100,n,Verbose(my_variable = ${my_variable}) ``` @@ -141,7 +134,6 @@ extensions.lua -------------- ``` - channel.my_variable = "my_value" app.verbose("my_variable = " .. channel.my_variable:get()) ``` @@ -155,7 +147,6 @@ extensions.conf --------------- ``` - exten => 100,1,Dial("SIP/100",,m) ``` @@ -165,7 +156,6 @@ extensions.lua -------------- ``` - app.dial("SIP/100", nil, "m") ``` @@ -180,7 +170,6 @@ extensions.conf --------------- ``` - [macro-dial] exten => s,1,Noop exten => s,n,Dial(${ARG1}) @@ -195,7 +184,6 @@ extensions.lua -------------- ``` - extensions = {} extensions.default = {} @@ -219,7 +207,6 @@ extensions.conf --------------- ``` - [default] exten => 100,1,Goto(102,1) @@ -233,7 +220,6 @@ extensions.lua -------------- ``` - extensions = {} extensions.default = {} @@ -254,7 +240,6 @@ end In Asterisk 1.8, use return ``` - function extension_function(c, e) return app.goto("default", "100", 1) diff --git a/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Lua-Dialplan-Examples.md b/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Lua-Dialplan-Examples.md index 4d59965844..aa683c272b 100644 --- a/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Lua-Dialplan-Examples.md +++ b/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Lua-Dialplan-Examples.md @@ -15,7 +15,6 @@ Instead of defining every extension inline, you can use this method to create a extensions.lua ``` - -- this function serves as an extension function directly function call_user(c, user) app.dial("SIP/" .. user, 60) @@ -54,7 +53,6 @@ In this example, we use a fancy function to register extensions. extensions.lua ``` - function register(context, extension, func) if not extensions then extensions = {} diff --git a/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Lua-Dialplan-Hints.md b/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Lua-Dialplan-Hints.md index a34b25f8f5..f76ad70c6f 100644 --- a/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Lua-Dialplan-Hints.md +++ b/docs/Configuration/Dialplan/Lua-Dialplan-Configuration/Lua-Dialplan-Hints.md @@ -10,7 +10,6 @@ In Asterisk 10 dialplan hints can be specified in `extensions.lua` in a manner s extensions.lua ``` - hints = { default = { ["100"] = "SIP/100"; diff --git a/docs/Configuration/Dialplan/Variables/Channel-Variables/Case-Sensitivity.md b/docs/Configuration/Dialplan/Variables/Channel-Variables/Case-Sensitivity.md index e71755c38f..d12996d7c0 100644 --- a/docs/Configuration/Dialplan/Variables/Channel-Variables/Case-Sensitivity.md +++ b/docs/Configuration/Dialplan/Variables/Channel-Variables/Case-Sensitivity.md @@ -33,7 +33,6 @@ This is best illustrated through the following examples In this example, the user retrieves a value from the AstDB and then uses it as the destination for a `Dial` command. ``` - [default] exten => 1000,1,Set(DEST=${DB(egg/salad)}) same => n,Dial(${DEST},15) @@ -42,7 +41,6 @@ exten => 1000,1,Set(DEST=${DB(egg/salad)}) Since the `DEST` variable is set and evaluated in the dialplan, its evaluation is case-insensitive. Thus the following would be equivalent: ``` - exten => 1000,1,Set(DEST=${DB(egg/salad)}) same => n,Dial(${dest},15) ``` @@ -50,7 +48,6 @@ exten => 1000,1,Set(DEST=${DB(egg/salad)}) As would this: ``` - exten => 1000,1,Set(DeSt=${DB(egg/salad)}) same => n,Dial(${dEsT},15) ``` @@ -60,14 +57,12 @@ exten => 1000,1,Set(DeSt=${DB(egg/salad)}) In this example, the user wishes to use a built-in variable in order to determine the destination for a call. ``` - exten => _X.,1,Dial(SIP/${EXTEN}) ``` Since the variable `EXTEN` is a built-in variable, the following would **not** be equivalent: ``` - exten => _X.,1,Dial(SIP/${exten}) ``` @@ -78,7 +73,6 @@ The lowercase `exten` variable would evaluate to an empty string since no previo In this example, the user wishes to suggest to the SIP channel driver what codec to use on the call. ``` - exten => 1000,Set(SIP_CODEC=g729) same => n,Dial(SIP/1000,15) ``` @@ -86,7 +80,6 @@ same => n,Dial(SIP/1000,15) `SIP_CODEC` is set in the dialplan, but it gets evaluated inside of Asterisk, so the evaluation is case-sensitive. Thus the following dialplan would not be equivalent: ``` - exten => 1000,Set(sip_codec=g729) same => n,Dial(SIP/1000,15) ``` @@ -94,7 +87,6 @@ exten => 1000,Set(sip_codec=g729) This can lead to some rather confusing situations. Consider that a user wrote the following dialplan. He intended to set the variable `SIP_CODEC` but instead made a typo: ``` - exten => 1000,Set(SIP_CODEc=g729) same => n,Dial(SIP/1000,15) ``` @@ -102,7 +94,6 @@ exten => 1000,Set(SIP_CODEc=g729) As has already been discussed, this is not equivalent to using `SIP_CODEC`. The user looks over his dialplan and does not notice the typo. As a way of debugging, he decides to place a `NoOp` in the dialplan: ``` - exten => 1000,Set(SIP_CODEc=g729) same => n,NoOp(${SIP_CODEC}) same => n,Dial(SIP/1000,15) diff --git a/docs/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/Followme-Realtime.md b/docs/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/Followme-Realtime.md index f3c5eedf65..c717f019f4 100644 --- a/docs/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/Followme-Realtime.md +++ b/docs/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/Followme-Realtime.md @@ -8,7 +8,6 @@ Followme is now realtime-enabled. To use, you must define two backend data structures, with the following fields: ``` - followme: name Name of this followme entry. Specified when invoking the FollowMe application in the dialplan. This field is the only one which is @@ -30,7 +29,6 @@ followme: sorry_prompt Normally, "Unable to locate party". ``` ``` - followme_numbers: name Name of this followme entry. Must match the name above. ordinal An integer, specifying the order in which these numbers will be diff --git a/docs/Configuration/Interfaces/Distributed-Device-State/Corosync.md b/docs/Configuration/Interfaces/Distributed-Device-State/Corosync.md index 142d7adbec..c4f6a4bb32 100644 --- a/docs/Configuration/Interfaces/Distributed-Device-State/Corosync.md +++ b/docs/Configuration/Interfaces/Distributed-Device-State/Corosync.md @@ -30,14 +30,12 @@ Using Corosync together with res_corosync allows events to be shared amongst a l Debian / Ubuntu ``` - apt-get install corosync corosync-dev ``` Red Hat / Fedora ``` - yum install corosync corosynclib corosynclib-devel ``` @@ -46,21 +44,18 @@ yum install corosync corosynclib corosynclib-devel To create an authentication key for secure communications between your nodes you need to do this on, what will be, the active node. ``` - corosync-keygen ``` This creates a key in /etc/corosync/authkey. ``` - asterisk_active:~# scp /etc/corosync/authkey asterisk_standby: ``` Now, on the standby node, you'll need to stick the authkey in it's new home and fix it's permissions / ownership. ``` - asterisk_standby:~# mv ~/authkey /etc/corosync/authkey asterisk_standby:~# chown root:root /etc/corosync/authkey asterisk_standby:~# chmod 400 /etc/corosync/authkey @@ -131,7 +126,6 @@ The interface section under the totem block defines the communication path(s) to * ###### Start Corosync ``` - service corosync start ``` @@ -142,7 +136,6 @@ service corosync start In your Asterisk source directory: ``` - ./configure make make install @@ -151,7 +144,6 @@ make install * ###### /etc/asterisk/res_corosync.conf ``` - ; ; Sample configuration file for res_corosync. ; @@ -192,7 +184,6 @@ In the general section of the res_corosync.conf file we are specifying which eve If everything is set up correctly, you should see this output after executing a 'corosync show members' on the Asterisk CLI. ``` - *CLI> corosync show members ============================================================= @@ -209,7 +200,6 @@ If everything is set up correctly, you should see this output after executing a After starting Corosync and Asterisk on your second node, the 'corosync show members' output should look something like this: ``` - *CLI> corosync show members ============================================================= diff --git a/docs/Configuration/Reporting/Call-Detail-Records-CDR/CDR-Storage-Backends/PostgreSQL-CDR-Backend.md b/docs/Configuration/Reporting/Call-Detail-Records-CDR/CDR-Storage-Backends/PostgreSQL-CDR-Backend.md index 08dc314afa..fe6817957a 100644 --- a/docs/Configuration/Reporting/Call-Detail-Records-CDR/CDR-Storage-Backends/PostgreSQL-CDR-Backend.md +++ b/docs/Configuration/Reporting/Call-Detail-Records-CDR/CDR-Storage-Backends/PostgreSQL-CDR-Backend.md @@ -12,7 +12,6 @@ Here is a sample: /etc/asterisk/cdr_pgsql.conf ``` - ; Sample Asterisk config file for CDR logging to PostgresSQL [global] hostname=localhost @@ -26,7 +25,6 @@ table=cdr Now create a table in postgresql for your cdrs ``` - CREATE TABLE cdr ( calldate timestamp NOT NULL , clid varchar (80) NOT NULL , @@ -52,7 +50,6 @@ CREATE TABLE cdr ( The following columns can also be defined: ``` - peeraccount varchar(20) NOT NULL linkedid varchar(150) NOT NULL sequence int NOT NULL diff --git a/docs/Configuration/Reporting/Call-Detail-Records-CDR/CDR-Storage-Backends/RADIUS-CDR-Backend.md b/docs/Configuration/Reporting/Call-Detail-Records-CDR/CDR-Storage-Backends/RADIUS-CDR-Backend.md index 39c5734b51..18f95eafb3 100644 --- a/docs/Configuration/Reporting/Call-Detail-Records-CDR/CDR-Storage-Backends/RADIUS-CDR-Backend.md +++ b/docs/Configuration/Reporting/Call-Detail-Records-CDR/CDR-Storage-Backends/RADIUS-CDR-Backend.md @@ -18,14 +18,12 @@ From ##### Untar the source tarball: ``` - root@localhost:/usr/local/src# tar xvfz radiusclient-ng-0.5.2.tar.gz ``` ##### Compile and install the library: ``` - root@localhost:/usr/local/src# cd radiusclient-ng-0.5.2 root@localhost:/usr/local/src/radiusclient-ng-0.5.2#./configure root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make @@ -39,14 +37,12 @@ By default all the configuration files of the radiusclient library will be in /u File "radiusclient.conf" Open the file and find lines containing the following: ``` - authserver localhost ``` This is the hostname or IP address of the RADIUS server used for authentication. You will have to change this unless the server is running on the same host as your Asterisk PBX. ``` - acctserver localhost ``` @@ -63,7 +59,6 @@ You need to configure a shared secret for each server you have configured in rad Each line contains hostname of a RADIUS server and shared secret used in communication with that server. The two values are separated by white spaces. Configure shared secrets for every RADIUS server you are going to use. ``` - File "dictionary" ``` @@ -84,7 +79,6 @@ $INCLUDE /path/to/dictionary.digium ##### Untar, configure, build, and install the server: ``` - root@localhost:/usr/local/src# tar xvfz freeradius-1.1.1.tar.gz root@localhost:/usr/local/src# cd freeradius-1.1.1 root@localhost"/usr/local/src/freeradius-1.1.1# ./configure @@ -105,7 +99,6 @@ File /usr/local/etc/raddb/clients.conf contains description of RADIUS clients th Example: ``` - client myhost { secret = mysecret shortname = foo } ``` diff --git a/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/MSSQL-CEL-Backend.md b/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/MSSQL-CEL-Backend.md index d446ca6a3f..2558182edc 100644 --- a/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/MSSQL-CEL-Backend.md +++ b/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/MSSQL-CEL-Backend.md @@ -21,21 +21,18 @@ The following provide some examples known to get asterisk working with mssql. ### Compile, configure, and install the latest unixODBC package: ``` - tar -zxvf unixODBC-2.2.9.tar.gz && cd unixODBC-2.2.9 && ./configure --sysconfdir=/etc --prefix=/usr --disable-gui && make && make install ``` ### Compile, configure, and install the latest FreeTDS package: ``` - tar -zxvf freetds-0.62.4.tar.gz && cd freetds-0.62.4 && ./configure --prefix=/usr --with-tdsver=7.0 \ --with-unixodbc=/usr/lib && make && make install ``` ### Compile, or recompile, asterisk so that it will now add support for cel_odbc. ``` - make clean && ./configure --with-odbc && make update && make && make install ``` @@ -46,7 +43,6 @@ These are working examples from my system. You will need to modify for your setu /etc/odbcinst.ini ``` - [FreeTDS] Description = FreeTDS ODBC driver for MSSQL Driver = /usr/lib/libtdsodbc.so @@ -57,7 +53,6 @@ FileUsage = 1 /etc/odbc.ini ``` - [MSSQL-asterisk] description = Asterisk ODBC for MSSQL driver = FreeTDS @@ -88,7 +83,6 @@ These are working samples from my system. You will need to modify for your setup /etc/asterisk/cel_odbc.conf ``` - [global] dsn=MSSQL-asterisk username=voipdbuser @@ -99,7 +93,6 @@ loguniqueid=yes ### And finally, create the 'cel' table in your mssql database. ``` - CREATE TABLE cel ( [eventtype] [varchar] (30) NOT NULL , [eventtime] [datetime] NOT NULL , @@ -128,14 +121,12 @@ Start asterisk in verbose mode, you should see that asterisk logs a connection t ### Compile, configure, and install the latest FreeTDS package: ``` - tar -zxvf freetds-0.62.4.tar.gz && cd freetds-0.62.4 && ./configure --prefix=/usr --with-tdsver=7.0 make && make install ``` ### Compile, or recompile, asterisk so that it will now add support for cel_tds. ``` - make clean && ./configure --with-tds && make update && make && make install ``` @@ -154,7 +145,6 @@ These are working samples from my system. You will need to modify for your setup /etc/asterisk/cel_tds.conf ``` - [global] hostname=192.168.1.25 port=1433 @@ -167,7 +157,6 @@ charset=BIG5 ### And finally, create the 'cel' table in your mssql database. ``` - CREATE TABLE cel ( [eventtype] [varchar] (30) NULL , [eventtime] [datetime] NULL , diff --git a/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/PostgreSQL-CEL-Backend.md b/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/PostgreSQL-CEL-Backend.md index 9069fb8b83..ad1317ac22 100644 --- a/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/PostgreSQL-CEL-Backend.md +++ b/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/PostgreSQL-CEL-Backend.md @@ -12,7 +12,6 @@ Here is a sample: /etc/asterisk/cel_pgsql.conf ``` - ; Sample Asterisk config file for CEL logging to PostgresSQL [global] hostname=localhost @@ -26,7 +25,6 @@ table=cel Now create a table in postgresql for your cels ``` - CREATE TABLE cel ( id serial , eventtype varchar (30) NOT NULL , diff --git a/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/RADIUS-CEL-Backend.md b/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/RADIUS-CEL-Backend.md index 42221dd6c0..d079ff9112 100644 --- a/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/RADIUS-CEL-Backend.md +++ b/docs/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/RADIUS-CEL-Backend.md @@ -18,14 +18,12 @@ From ##### Untar the source tarball: ``` - root@localhost:/usr/local/src# tar xvfz radiusclient-ng-0.5.2.tar.gz ``` ##### Compile and install the library: ``` - root@localhost:/usr/local/src# cd radiusclient-ng-0.5.2 root@localhost:/usr/local/src/radiusclient-ng-0.5.2#./configure root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make @@ -39,14 +37,12 @@ By default all the configuration files of the radiusclient library will be in /u File "radiusclient.conf" Open the file and find lines containing the following: ``` - authserver localhost ``` This is the hostname or IP address of the RADIUS server used for authentication. You will have to change this unless the server is running on the same host as your Asterisk PBX. ``` - acctserver localhost ``` @@ -63,7 +59,6 @@ You need to configure a shared secret for each server you have configured in rad Each line contains hostname of a RADIUS server and shared secret used in communication with that server. The two values are separated by white spaces. Configure shared secrets for every RADIUS server you are going to use. ``` - File "dictionary" ``` @@ -84,7 +79,6 @@ $INCLUDE /path/to/dictionary.digium ##### Untar, configure, build, and install the server: ``` - root@localhost:/usr/local/src# tar xvfz freeradius-1.1.1.tar.gz root@localhost:/usr/local/src# cd freeradius-1.1.1 root@localhost"/usr/local/src/freeradius-1.1.1# ./configure @@ -105,7 +99,6 @@ File /usr/local/etc/raddb/clients.conf contains description of RADIUS clients th Example: ``` - client myhost { secret = mysecret shortname = foo } ``` diff --git a/docs/Configuration/WebRTC/Installing-and-Configuring-CyberMegaPhone.md b/docs/Configuration/WebRTC/Installing-and-Configuring-CyberMegaPhone.md index 1fad8a18be..1b79c51036 100644 --- a/docs/Configuration/WebRTC/Installing-and-Configuring-CyberMegaPhone.md +++ b/docs/Configuration/WebRTC/Installing-and-Configuring-CyberMegaPhone.md @@ -64,7 +64,6 @@ Now check that the configuration was applied. From the Asterisk CLI... Asterisk CLI ``` - *CLI> http show status HTTP Server Status: Prefix: diff --git a/docs/Deployment/Distributed-Universal-Number-Discovery-DUNDi/Digium-General-Peering-Agreement.md b/docs/Deployment/Distributed-Universal-Number-Discovery-DUNDi/Digium-General-Peering-Agreement.md index 1d416ce8a4..666eb8884c 100644 --- a/docs/Deployment/Distributed-Universal-Number-Discovery-DUNDi/Digium-General-Peering-Agreement.md +++ b/docs/Deployment/Distributed-Universal-Number-Discovery-DUNDi/Digium-General-Peering-Agreement.md @@ -4,7 +4,6 @@ pageid: 4259957 --- ``` - DIGIUM GENERAL PEERING AGREEMENT (TM) Version 1.0.0, September 2004 Copyright (C) 2004 Digium, Inc. diff --git a/docs/Deployment/Internationalization-and-Localization/Asterisk-Sounds-Packages/About-the-Sounds-Tools.md b/docs/Deployment/Internationalization-and-Localization/Asterisk-Sounds-Packages/About-the-Sounds-Tools.md index f7bcd560cd..f3cf2f48ea 100644 --- a/docs/Deployment/Internationalization-and-Localization/Asterisk-Sounds-Packages/About-the-Sounds-Tools.md +++ b/docs/Deployment/Internationalization-and-Localization/Asterisk-Sounds-Packages/About-the-Sounds-Tools.md @@ -14,7 +14,6 @@ The audiofilter application is used to "tune" the sound files in such a way that The audiofilter application is used by the 'converter' script located in the scripts subdirectory of repotools/sound_tools. The values being passed to the audiofilter application is as follows: ``` - audiofilter -n 0.86916 -1.73829 0.86916 -d 1.00000 -1.74152 0.77536 ``` diff --git a/docs/Deployment/Packet-Loss-Concealment-PLC/PLC-Tips.md b/docs/Deployment/Packet-Loss-Concealment-PLC/PLC-Tips.md index 5984e17013..7c79ee5ced 100644 --- a/docs/Deployment/Packet-Loss-Concealment-PLC/PLC-Tips.md +++ b/docs/Deployment/Packet-Loss-Concealment-PLC/PLC-Tips.md @@ -6,7 +6,6 @@ pageid: 5243118 One of the restrictions mentioned is that PLC will only be used when two audio channels are bridged together. Through the use of Local channels, you can create a bridge even if the call is, for all intents and purposes, one-legged. By using a combination of the /n and /j suffixes for a Local channel, one can ensure that the Local channel is not optimized out of the talk path and that a jitter buffer is applied to the Local channel as well. Consider the following simple dialplan: ``` - [example] exten => 1,1,Playback(tt-weasels) exten => 2,1,Dial(Local/1@example/nj) diff --git a/docs/Deployment/Phone-Provisioning-in-Asterisk/Configuration-of-users.conf.md b/docs/Deployment/Phone-Provisioning-in-Asterisk/Configuration-of-users.conf.md index ca618599ef..8f6b0a874a 100644 --- a/docs/Deployment/Phone-Provisioning-in-Asterisk/Configuration-of-users.conf.md +++ b/docs/Deployment/Phone-Provisioning-in-Asterisk/Configuration-of-users.conf.md @@ -14,7 +14,6 @@ There are only two settings in the general section of users.conf that apply to p To enable auto-provisioning of a phone, the user in users.conf needs to have: ``` - ... autoprov=yes macaddress=deadbeef4dad @@ -24,7 +23,6 @@ profile=polycom The profile is optional if a default_profile is set in phoneprov.conf. The following is a sample users.conf entry, with the template variables commented next to the settings: ``` - [6001] callwaiting = yes context = numberplan-custom-1 diff --git a/docs/Deployment/Phone-Provisioning-in-Asterisk/Creating-Phone-Profiles.md b/docs/Deployment/Phone-Provisioning-in-Asterisk/Creating-Phone-Profiles.md index a28421ae0a..2412080c4e 100644 --- a/docs/Deployment/Phone-Provisioning-in-Asterisk/Creating-Phone-Profiles.md +++ b/docs/Deployment/Phone-Provisioning-in-Asterisk/Creating-Phone-Profiles.md @@ -6,7 +6,6 @@ pageid: 5243051 A phone profile is basically a list of files that a particular group of phones needs to function. For most phone types there are files that are identical for all phones (firmware, for instance) as well as a configuration file that is specific to individual phones. res_phoneprov breaks these two groups of files into static files and dynamic files, respectively. A sample profile: ``` - [polycom] staticdir => configs/ mime_type => text/xml diff --git a/docs/Development/Debugging/Using-the-Hoard-Memory-Allocator-with-Asterisk.md b/docs/Development/Debugging/Using-the-Hoard-Memory-Allocator-with-Asterisk.md index d8630d0c75..3a7f7aa2d7 100644 --- a/docs/Development/Debugging/Using-the-Hoard-Memory-Allocator-with-Asterisk.md +++ b/docs/Development/Debugging/Using-the-Hoard-Memory-Allocator-with-Asterisk.md @@ -18,7 +18,6 @@ Run ./configure in the root of the asterisk source directory, passing the **--wi For example: ``` - ./configure --with-hoard=/usr/src/hoard-371/src/ ``` diff --git a/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Asterisk-Channel-Data-Stores.md b/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Asterisk-Channel-Data-Stores.md index f6793830e7..c01604efac 100644 --- a/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Asterisk-Channel-Data-Stores.md +++ b/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Asterisk-Channel-Data-Stores.md @@ -15,7 +15,6 @@ If the data store is not freed by said application though, a callback to a destr ##### A datastore info structure ``` - static const struct example_datastore { .type = "example", .destroy = callback_destroy @@ -41,7 +40,6 @@ This is a needed structure that contains information about a datastore, it's use Full Example: ``` - void callback_destroy(void *data) { ast_free(data); diff --git a/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Modules.md b/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Modules.md index 9cf6eb81f3..1efb39912f 100644 --- a/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Modules.md +++ b/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Modules.md @@ -9,7 +9,6 @@ Load Module ----------- ``` - static int load_module(): ``` @@ -18,7 +17,6 @@ Module loading including tests for configuration or dependencies. This function Example: ``` - /*! * \brief Load the module * @@ -48,7 +46,6 @@ Unload Module ------------- ``` - static int unload_module(): ``` @@ -57,7 +54,6 @@ The module will soon be unloaded. If any channels are using your features, you s Example: ``` - /*! * \brief Unload Module */ @@ -76,7 +72,6 @@ Module Information ------------------ ``` - AST_MODULE_INFO_STANDARD(keystr, desc); ... or ... AST_MODULE_INFO(keystr, flags_to_set, desc, load_func, unload_func, reload_func, load_pri); diff --git a/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Using-the-Configuration-Framework.md b/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Using-the-Configuration-Framework.md index 86dca75b64..8a437f2263 100644 --- a/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Using-the-Configuration-Framework.md +++ b/docs/Development/Reference-Information/Asterisk-Framework-and-API-Examples/Using-the-Configuration-Framework.md @@ -50,7 +50,6 @@ Our configuration file `my_module.conf` may look something like this: my_module.conf ``` - [general] foobar = True foo = 1 @@ -239,7 +238,6 @@ First, let's assume that our module is loaded and running with the configuration A Bad my_module.conf ``` - [general] foobar = False foo = Oh snap I'm not an integer @@ -255,7 +253,6 @@ Well, since the variables will most likely be parsed in the order that they appe Resulting Values in the Module after the Failed Load ``` - foobar = False foo = 1 bar = Some string value diff --git a/docs/Development/Reference-Information/Other-Reference-Information/Locking-in-Asterisk.md b/docs/Development/Reference-Information/Other-Reference-Information/Locking-in-Asterisk.md index cc4995706f..21c9073b05 100644 --- a/docs/Development/Reference-Information/Other-Reference-Information/Locking-in-Asterisk.md +++ b/docs/Development/Reference-Information/Other-Reference-Information/Locking-in-Asterisk.md @@ -10,7 +10,6 @@ Asterisk is a heavily multithreaded application. It makes extensive use of locki When more that one lock is involved in a given code path, there is the potential for deadlocks. A deadlock occurs when a thread is stuck waiting for a resource that it will never acquire. Here is a classic example of a deadlock: ``` - Thread 1 Thread 2 ------------ ------------ Holds Lock A Holds Lock B @@ -75,7 +74,6 @@ For this reason, just calling unlock() once does not guarantee that the lock is An alternative, but still incorrect, construct is widely used in the asterisk code to try and improve the situation: ``` - while (trylock(ast_channel) == FAILURE) { unlock(pvt); usleep(1); /* yield to other thread */ diff --git a/docs/Development/Reference-Information/Other-Reference-Information/Measuring-SIP-Channel-Performance.md b/docs/Development/Reference-Information/Other-Reference-Information/Measuring-SIP-Channel-Performance.md index 9933144a1e..4193ef6e37 100644 --- a/docs/Development/Reference-Information/Other-Reference-Information/Measuring-SIP-Channel-Performance.md +++ b/docs/Development/Reference-Information/Other-Reference-Information/Measuring-SIP-Channel-Performance.md @@ -21,7 +21,6 @@ The version of sipp that I used was sipp-2.0.1; however, I have reason to believ On the asterisk machine, I included the following in my extensions.ael file: ``` - context test11 { s => { @@ -46,7 +45,6 @@ Basically, incoming SIP calls are answered, and the demo-instruct sound file is The sip.conf file has this entry: ``` - [asterisk02] type=friend context=test11 @@ -66,7 +64,6 @@ I learned to split the testing into two modes: one for just call call processing Call processing power is measured with extremely short duration calls: ``` - ./sipp -sn uac 192.168.134.252 -s 12 -d 100 -l 256 ``` @@ -77,7 +74,6 @@ Sipp starts at 10 calls/sec, and you can slowly increase the speed by hitting '\ Simultaneous calls can be measured with very long duration calls: ``` - ./sipp -sn uac 192.168.134.252 -s 12 -d 100000 -l 270 ``` diff --git a/docs/Development/Roadmap/Asterisk-10-Projects/Media-Architecture-Proposal.md b/docs/Development/Roadmap/Asterisk-10-Projects/Media-Architecture-Proposal.md index 81ef126a32..5689dfcb60 100644 --- a/docs/Development/Roadmap/Asterisk-10-Projects/Media-Architecture-Proposal.md +++ b/docs/Development/Roadmap/Asterisk-10-Projects/Media-Architecture-Proposal.md @@ -38,7 +38,6 @@ The ast_format structure on an ast_frame has a slightly different behavior than ## The Ast Format API ``` - /*! \brief This structure contains the buffer used for format attribute */ struct ast_format_attr { uint8_t format_attr[AST_FORMATNEW_ATTR_SIZE]; @@ -829,7 +828,6 @@ If a channel driver is capable of negotiating more streams than can be represent Since Asterisk supports multiple protocols with various capabilities, all the auxiliary streams that can be used anywhere in Asterisk must be defined at compile time. This means when a channel driver is extended to make use of a new type of auxiliary stream, that stream must be defined with a stream id that uniquely represents it across the entire code base. This is the only way to keep the different types of auxiliary streams and what they are used for consistent across all modules. ``` - Example 1: Chan_sip is extended to make use of up to four video and audio streams per call. This sort of functionality has never been done before so six new auxiliary streams must be defined for the three new video and three new audio streams. enum ast_channel_stream_id { @@ -873,7 +871,6 @@ struct ast_channel_stream { }; ``` ``` - /*! \brief stream identifier structure. Present on both ast_frame * and ast_channel_stream structure. */ @@ -958,7 +955,6 @@ type=silk samplerates=8000,12000,16000,24000 ``` ``` - /* Define a peer using only the narrow band custom SILK format definitio */ [sip_peer] type=friend @@ -977,7 +973,6 @@ res=vga,svga framerate=30 ``` ``` - /* Define a peer using the new h264_custom1 custom format type */ [sip_peer] type=friend diff --git a/docs/Development/Roadmap/Asterisk-11-Projects/AMI-Event-Documentation.md b/docs/Development/Roadmap/Asterisk-11-Projects/AMI-Event-Documentation.md index b3455115f7..94d17240a0 100644 --- a/docs/Development/Roadmap/Asterisk-11-Projects/AMI-Event-Documentation.md +++ b/docs/Development/Roadmap/Asterisk-11-Projects/AMI-Event-Documentation.md @@ -46,7 +46,6 @@ Two new CLI commands have been added: Example output of both commands is shown below. ``` - *CLI> manager show events Events: -------------------- -------------------- -------------------- @@ -62,7 +61,6 @@ Events: QueueMemberRinginuse QueueMemberStatus UserEvent ``` ``` - *CLI> manager show event Dial Event: Dial [Synopsis] @@ -123,7 +121,6 @@ AMI Event documentation behaves a bit differently then other Asterisk documentat 1. Event documentation can be built directly from the macros that raise the AMI events. This includes manager_event, ast_manager_event, and ast_manager_event_multichan. Because of this, AMI event documentation is typically co-located with the macro call that raises the event. Note that in the example below, only the DialStatus field is explicitly defined; however, the generated AMI event documentation will include all fields found in the ast_manager_event call. ``` - /*** DOCUMENTATION Raised when a dial action has ended. @@ -144,7 +141,6 @@ AMI Event documentation behaves a bit differently then other Asterisk documentat 2. Each instance of an AMI event can be documented. This is particularly useful when the same event can have different fields, e.g., Dial, PeerStatus, etc. Even if the event has the same fields across all instances, it is also useful to document why the event is raised in the tag. Because each instance of an AMI event should be documented, a post-processing script aggregates the various XML fragments that match the same event name under a single tag. Fields that are shared across instances of the same event are combined and only need to be documented a single time. In the example below, the SubEvent field is only documented once, but the full documentation for the field will be displayed for both instances of the Dial event, as both instances of the event contain that field. In contrast to that, only the second instance of the event contains the DialStatus field; hence, only that instance will contain that field. ``` - /*** DOCUMENTATION Raised when a dial action has started. @@ -175,7 +171,6 @@ AMI Event documentation behaves a bit differently then other Asterisk documentat 4. Because pre- and post-processing scripts are involved, some burden on having a well-formed XML fragment is lifted from the documenter. Often, the fields in an event are self-explanatory, or are documented significantly in other AMI events. When that is the case, documentation for the event may only consist of a field and one or two parameters - in which case, the element is inferred for the parameters. ``` - /*** DOCUMENTATION Raised when a dial action has ended. @@ -205,7 +200,6 @@ XML Schema The following are the changes to the XML DTD schema used to validate the generated XML documentation. An example of a generated XML fragment for the Dial event is also shown below. ``` - @@ -214,7 +208,6 @@ The following are the changes to the XML DTD schema used to validate the generat ``` ``` - Raised when a dial action has started. @@ -262,7 +255,6 @@ Source Comments * Event documentation **MUST** be within a documentation comment block (shown below), regardless of its location within an implementation file. ``` - /*** DOCUMENTATION .... * */ diff --git a/docs/Development/Roadmap/Asterisk-11-Projects/Configuration-parsing-with-the-Config-Options-API.md b/docs/Development/Roadmap/Asterisk-11-Projects/Configuration-parsing-with-the-Config-Options-API.md index 1d4221603d..1282a35d2b 100644 --- a/docs/Development/Roadmap/Asterisk-11-Projects/Configuration-parsing-with-the-Config-Options-API.md +++ b/docs/Development/Roadmap/Asterisk-11-Projects/Configuration-parsing-with-the-Config-Options-API.md @@ -8,7 +8,6 @@ pageid: 19008617 Asterisk uses a standard config file format that is essentially: ``` - [context] variable=value ``` @@ -28,7 +27,6 @@ aco_type - A mapping between categories in a config file and user-defined object category - A section of a config field denoted by a bracketed name. A category named "general" might look like: ``` - [general] variable1 = value variable2 = value2 diff --git a/docs/Development/Roadmap/Asterisk-11-Projects/Unistim-channel-improvements.md b/docs/Development/Roadmap/Asterisk-11-Projects/Unistim-channel-improvements.md index 9e9f09256a..b25d6b63f2 100644 --- a/docs/Development/Roadmap/Asterisk-11-Projects/Unistim-channel-improvements.md +++ b/docs/Development/Roadmap/Asterisk-11-Projects/Unistim-channel-improvements.md @@ -30,7 +30,6 @@ Using multiple line on Unistim phones was on of the most desired functions users Phone with two lines defined ``` - [violet] device=006038abcdef callerid = "Violet" <660100> ; Individual callerid used for each line diff --git a/docs/Development/Roadmap/Asterisk-11-Projects/Who-Hung-Up.md b/docs/Development/Roadmap/Asterisk-11-Projects/Who-Hung-Up.md index 3b08ee7942..46326948a9 100644 --- a/docs/Development/Roadmap/Asterisk-11-Projects/Who-Hung-Up.md +++ b/docs/Development/Roadmap/Asterisk-11-Projects/Who-Hung-Up.md @@ -17,7 +17,6 @@ Example ======= ``` - [foo] exten => s,1,Dial(SIP/bar) diff --git a/docs/Development/Roadmap/Asterisk-11-Projects/app_macro-Deprecation.md b/docs/Development/Roadmap/Asterisk-11-Projects/app_macro-Deprecation.md index 021a348665..d8fc445cf7 100644 --- a/docs/Development/Roadmap/Asterisk-11-Projects/app_macro-Deprecation.md +++ b/docs/Development/Roadmap/Asterisk-11-Projects/app_macro-Deprecation.md @@ -22,7 +22,6 @@ In channel configurations, 'cc_callback_sub' should be used instead of 'cc_callb Assuming a SIP configuration of the following: ``` - [general] limitonpeers=yes udpbindaddr = 127.0.0.2 @@ -54,7 +53,6 @@ callcounter=yes And dialplan: ``` - [to-bob] exten => 1234,1,NoOp exten => 1234,n,Dial(SIP/bob) @@ -85,7 +83,6 @@ The primary difference between the app_macro-based callbacks and the new app_sta Starting dialplan execution at 'test,100,1', the following is a working example of the app_stack-based CLI callback using local channels. ``` - [test] exten => 100,1,NoOp exten => 100,n,Set(CONNECTED_LINE_SEND_SUB=callback,s,1) @@ -128,7 +125,6 @@ The primary difference between the app_macro-based callbacks and the new app_sta Starting dialplan execution at 'test,100,1', the following is a working example of the app_stack-based RI callback using local channels. ``` - [test] exten => 100,1,NoOp exten => 100,n,Set(REDIRECTING_SEND_SUB=callback,s,1) diff --git a/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/Asterisk-Client-Library-Generator.md b/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/Asterisk-Client-Library-Generator.md index 75857884d2..c688214fba 100644 --- a/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/Asterisk-Client-Library-Generator.md +++ b/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/Asterisk-Client-Library-Generator.md @@ -67,7 +67,6 @@ Project location and directory structure This project is located [on GitHub](https://github.com/asterisk/asterisk_rest_libraries). Inside asterisk_rest_libraries ``` - asterisk_rest_libraries generate_library.py api.py diff --git a/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/index.md b/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/index.md index 4d0d1ee1b4..5c07df75a9 100644 --- a/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/index.md +++ b/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/index.md @@ -257,7 +257,6 @@ Please see the [API docs|http://doxygen.asterisk.org/trunk/d8/d9c/stasis__app_8h The `app\_stasis.so` module simply exports the `res\_stasis.so` functionality as a dialplan application. This allows you to send channels to a Stasis application from within the dialplan. ``` - ; Send channel to the 'Queue' application, with the args 'enqueue,sales' exten => 7001,1,Stasis(Queue,enqueue,sales) ``` diff --git a/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-Bridging-Project/Asterisk-12-Bridging-API.md b/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-Bridging-Project/Asterisk-12-Bridging-API.md index 052844adb8..bbdf01f85a 100644 --- a/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-Bridging-Project/Asterisk-12-Bridging-API.md +++ b/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-Bridging-Project/Asterisk-12-Bridging-API.md @@ -19,7 +19,6 @@ Enumerations A bridge technology uses this when it registers to inform the framework which capabilities it can provide. When a bridge is created, the creator of the bridge can specify capabilities that it knows it will need for that bridge and the framework will choose the best technology that matches those capabilities. Additionally, when conditions in a bridge change such that the technology can no longer meet all of the conditions, the framework will use the capabilities to pick a new technology for the bridge. ``` - /*! \brief Capabilities for a bridge technolog */ enum ast_bridge_capability { /*! Bridge is only capable of mixing 2 channel */ @@ -44,7 +43,6 @@ enum ast_bridge_capability { The current state of an `ast_bridge_channel` object. ``` - /*! \brief State information about a bridged channe */ enum ast_bridge_channel_state { /*! Waiting for a signal (Channel in the bridge */ @@ -65,7 +63,6 @@ enum ast_bridge_channel_state { The possible conditions that a `ast_bridge_technology` write operation can return. ``` - /*! \brief Return values for bridge technology write functio */ enum ast_bridge_write_result { /*! Bridge technology wrote out frame fin */ @@ -82,7 +79,6 @@ enum ast_bridge_write_result { The framework uses this enum to determine the action it is supposed to execute on a channel in a bridge. ``` - enum ast_bridge_action_type { /*! Bridged channel is to detect a feature hoo */ AST_BRIDGE_ACTION_FEATURE, @@ -102,7 +98,6 @@ enum ast_bridge_action_type { For bridges that support video, the supported ways in which a bridge can choose the video source. ``` - enum ast_bridge_video_mode_type { /*! Video is not allowed in the bridg */ AST_BRIDGE_VIDEO_MODE_NONE = 0, @@ -122,7 +117,6 @@ Structures Some bridging technologies support advanced talk optimizations/detection operations. This structure provides configuration information for those technologies. ``` - /*! * \brief Structure specific to bridge technologies capable of * performing talking optimizations. @@ -145,7 +139,6 @@ struct ast_bridge_tech_optimizations { The `ast_bridge_channel` object maintain the state of an `ast_channel` in a bridge. ``` - /*! * \brief Structure that contains information regarding a channel in a bridge */ @@ -189,7 +182,6 @@ struct ast_bridge_channel { For bridges that support video that are in AST_BRIDGE_VIDEO_MODE_SINGLE_SRC mode, this structure is used to pass information about the video source. ``` - /*! This is used for both SINGLE_SRC mode to set what channel * should be the current single video fee */ struct ast_bridge_video_single_src_data { @@ -203,7 +195,6 @@ struct ast_bridge_video_single_src_data { For bridges that support video that are in AST_BRIDGE_VIDEO_MODE_TALKER_SRC mode, this structure is used to pass information about the source of audio and the possible video sources. ``` - /*! This is used for both SINGLE_SRC_TALKER mode to set what channel * should be the current single video fee */ struct ast_bridge_video_talker_src_data { @@ -221,7 +212,6 @@ struct ast_bridge_video_talker_src_data { This structure acts as a wrapper around the various possible video mode channel information objects. ``` - struct ast_bridge_video_mode { enum ast_bridge_video_mode_type mode; /* Add data for all the video modes here */ @@ -237,7 +227,6 @@ struct ast_bridge_video_mode { The main bridging type, the `ast_bridge` type defines how a bridge behaves, what technology it uses to perform the operations on the channels in the bridge, what channels are in the bridge, the state of the bridge, and more. Operations on a bridge are performed on an instance of `ast_bridge`. ``` - /*! * \brief Structure that contains information about a bridge */ @@ -294,7 +283,6 @@ Functions on ast_bridge Create a new instance of `ast_bridge` with the requested capabilities. ``` - /*! * \brief Create a new bridge * @@ -322,7 +310,6 @@ struct ast_bridge *ast_bridge_new(uint32_t capabilities, int flags); Lock the bridge. While locked, the state of the bridge cannot be changed by external entities. Internal entities may become blocked as well when they need to change the state of the bridge. ``` - /*! * \brief Lock the bridge. * @@ -342,7 +329,6 @@ static inline void _ast_bridge_lock(struct ast_bridge *bridge, const char *file, Unlock the bridge. ``` - /*! * \brief Unlock the bridge. * @@ -362,7 +348,6 @@ static inline void _ast_bridge_unlock(struct ast_bridge *bridge, const char *fil Determine if the Bridging Framework can create a bridge with the requested capabilities. ``` - /*! * \brief See if it is possible to create a bridge * @@ -388,7 +373,6 @@ int ast_bridge_check(uint32_t capabilities); Explicitly destroy a bridge. Note that a self managing bridge will automatically destroy itself when no more channels are in the bridge. ``` - /*! * \brief Destroy a bridge * @@ -423,7 +407,6 @@ It is up to the caller of the function to decide what happens next. In general, * Perform an `ast_async_goto` on the channel ``` - /*! * \brief Join (blocking) a channel to a bridge * @@ -475,7 +458,6 @@ When the channel leaves the bridge the channel will: * hangup ``` - /*! * \brief Impart (non-blocking) a channel onto a bridge * @@ -529,7 +511,6 @@ int ast_bridge_impart(struct ast_bridge *bridge, struct ast_channel *chan, struc Remove a previously imparted `ast_channel` from the bridge. ``` - /*! * \brief Depart a channel from a bridge * @@ -560,7 +541,6 @@ int ast_bridge_depart(struct ast_bridge *bridge, struct ast_channel *chan); Remove any channel from the bridge. ``` - /*! * \brief Remove a channel from a bridge * @@ -591,7 +571,6 @@ int ast_bridge_remove(struct ast_bridge *bridge, struct ast_channel *chan); Merge two bridges together. ``` - /*! * \brief Merge two bridges together * @@ -621,7 +600,6 @@ int ast_bridge_merge(struct ast_bridge *bridge0, struct ast_bridge *bridge1); Suspend a channel from the bridge. Channels that are suspended from the bridge are no longer manipulated by threads in the bridge and can be safely accessed by non-bridge threads. Channels in a bridge **must** be suspended prior to manipulation by external threads. ``` - /*! * \brief Suspend a channel temporarily from a bridge * @@ -652,7 +630,6 @@ int ast_bridge_suspend(struct ast_bridge *bridge, struct ast_channel *chan); Unsuspend a previously suspended channel, returning control of it back to the bridge's threads. ``` - /*! * \brief Unsuspend a channel from a bridge * @@ -687,7 +664,6 @@ Change the state of a bridged channel. [//]: # (end-warning) ``` - /*! * \brief Change the state of a bridged channel * @@ -714,7 +690,6 @@ void ast_bridge_change_state(struct ast_bridge_channel *bridge_channel, enum ast If a bridging technology supports the multimix capability, set the mixing sampling rate. ``` - /*! * \brief Adjust the internal mixing sample rate of a bridge * used during multimix mode. @@ -733,7 +708,6 @@ void ast_bridge_set_internal_sample_rate(struct ast_bridge *bridge, unsigned int If a bridging technology supports the multimix capability, set the mixing interval. ``` - /*! * \brief Adjust the internal mixing interval of a bridge used * during multimix mode. @@ -750,7 +724,6 @@ void ast_bridge_set_mixing_interval(struct ast_bridge *bridge, unsigned int mixi If a bridging technology supports video, set the single video source to feed to all participants. ``` - /*! * \brief Set a bridge to feed a single video source to all participants. */ @@ -762,7 +735,6 @@ void ast_bridge_set_single_src_video_mode(struct ast_bridge *bridge, struct ast_ If a bridging technology supports video, set the video mode to use the current talker. ``` - /*! * \brief Set the bridge to pick the strongest talker supporting * video as the single source video feed @@ -775,7 +747,6 @@ void ast_bridge_set_talker_src_video_mode(struct ast_bridge *bridge); Inform a video capable bridging technology about the talk energy and frame information for a specific channel. ``` - /*! * \brief Update information about talker energy for talker src video mode. */ @@ -787,7 +758,6 @@ void ast_bridge_update_talker_src_video_mode(struct ast_bridge *bridge, struct a Get the number of video sources in the bridge. ``` - /*! * \brief Returns the number of video sources currently active in the bridge */ @@ -799,7 +769,6 @@ int ast_bridge_number_video_src(struct ast_bridge *bridge); Return whether or not a channel is a video source. ``` - /*! * \brief Determine if a channel is a video src for the bridge * @@ -816,7 +785,6 @@ int ast_bridge_is_video_src(struct ast_bridge *bridge, struct ast_channel *chan) Remove a channel from being the video source. ``` - /*! * \brief remove a channel as a source of video for the bridge. */ @@ -834,7 +802,6 @@ Enumerations An enumeration that specifies for a registered bridging technology the preference the Bridging Framework should assign when picking between technologies with equivalent capabilities. ``` - /*! \brief Preference for choosing the bridge technolog */ enum ast_bridge_preference { /*! Bridge technology should have high precedence over other bridge technologie */ @@ -854,7 +821,6 @@ Structs The interface that defines a bridging technology. ``` - /*! * \brief Structure that is the essence of a bridge technology */ @@ -907,7 +873,6 @@ Functions Register a technology with the Bridging Framework. ``` - /*! * \brief Register a bridge technology for use * @@ -938,7 +903,6 @@ int __ast_bridge_technology_register(struct ast_bridge_technology *technology, s Unregister a technology with the Bridging Framework. ``` - /*! * \brief Unregister a bridge technology from use * @@ -965,7 +929,6 @@ int ast_bridge_technology_unregister(struct ast_bridge_technology *technology); Notify the Bridging Framework that a channel has a frame waiting. ``` - /*! * \brief Feed notification that a frame is waiting on a channel into the bridging core * @@ -993,7 +956,6 @@ void ast_bridge_handle_trip(struct ast_bridge *bridge, struct ast_bridge_channel Notify the Bridging Framework that a channel has started talking. ``` - /*! * \brief Lets the bridging indicate when a bridge channel has stopped or started talking. * @@ -1015,7 +977,6 @@ void ast_bridge_notify_talking(struct ast_bridge_channel *bridge_channel, int st Suspend a bridging technology from consideration by the Bridging Framework. ``` - /*! * \brief Suspend a bridge technology from consideration * @@ -1039,7 +1000,6 @@ void ast_bridge_technology_suspend(struct ast_bridge_technology *technology); Unsuspend a bridging technology from consideration by the Bridging Framework. ``` - /*! * \brief Unsuspend a bridge technology * diff --git a/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-Bridging-Project/index.md b/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-Bridging-Project/index.md index 70d10b8841..16695d69d3 100644 --- a/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-Bridging-Project/index.md +++ b/docs/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-Bridging-Project/index.md @@ -28,7 +28,6 @@ A channel masquerade is a fundamental yet incredibly confusing concept in Asteri A comment from ast_do_masquerade ``` - /* XXX This operation is a bit odd. We're essentially putting the guts of * the clone channel into the original channel. Start by killing off the * original channel's backend. While the features are nice, which is the @@ -361,7 +360,6 @@ Bridge channel hooks can move the bridge channel between bridges. This would be A way to implement the toggle between A and C parties is to have an atxfer bridge subclass. Set up the links this way: ``` - A -- B1 --Local@special/b -- Batxfer1 -- B -- Batxfer2 -- Local@special/b -- B2 -- C ``` @@ -489,7 +487,6 @@ Bridge technologies are an embedded object of the abstract bridge class. Park, Queue, ConfBridge could be derivative classes of the abstract bridge class. ``` - class ast_bridge { join(struct ast_channel *chan); depart(struct ast_channel *chan); @@ -528,7 +525,6 @@ class ast_bridge_channel { #### Locking precedence order: ``` - bridges ao2_container | ast_bridge @@ -687,7 +683,6 @@ channel private #### CLI commands: ``` - bridge show all List all current bridges by bridge-id and number of channels in the bridge. diff --git a/docs/Getting-Started/Installing-Asterisk/Installing-Asterisk-on-Non-Linux-Operating-Systems/Asterisk-on-OpenSolaris.md b/docs/Getting-Started/Installing-Asterisk/Installing-Asterisk-on-Non-Linux-Operating-Systems/Asterisk-on-OpenSolaris.md index f9e7dadd5c..5aab4e4eca 100644 --- a/docs/Getting-Started/Installing-Asterisk/Installing-Asterisk-on-Non-Linux-Operating-Systems/Asterisk-on-OpenSolaris.md +++ b/docs/Getting-Started/Installing-Asterisk/Installing-Asterisk-on-Non-Linux-Operating-Systems/Asterisk-on-OpenSolaris.md @@ -47,7 +47,6 @@ Despite the above two possibilities, there is a workaround to make Asterisk comp * When running make, specify the use of the OpenLDAP headers like this: ``` - "make LDAP_INCLUDE=-I/usr/include/openldap" ``` diff --git a/docs/Test-Suite-Documentation/Test-Development/Tests-with-SIPp/Basic-Test-with-SIPp.md b/docs/Test-Suite-Documentation/Test-Development/Tests-with-SIPp/Basic-Test-with-SIPp.md index 434d12342d..b6ac57bf28 100644 --- a/docs/Test-Suite-Documentation/Test-Development/Tests-with-SIPp/Basic-Test-with-SIPp.md +++ b/docs/Test-Suite-Documentation/Test-Development/Tests-with-SIPp/Basic-Test-with-SIPp.md @@ -17,7 +17,6 @@ SIPp Scenario ------------- ``` - @@ -49,7 +48,6 @@ Asterisk Configuration Since the SIPp scenario is attempting a REGISTER request for v4-in, we must have a corresponding sip.conf configuration file defining the peer. ``` - [general] bindaddr=[::]:5060