[asterisk-commits] tilghman: trunk r43444 - in /trunk: UPGRADE.txt
main/manager.c main/pbx.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Thu Sep 21 13:01:54 MST 2006
Author: tilghman
Date: Thu Sep 21 15:01:54 2006
New Revision: 43444
URL: http://svn.digium.com/view/asterisk?rev=43444&view=rev
Log:
Remove 1.4 changes from UPGRADE.txt, remove deprecated callerid field, remove deprecated SetGlobalVar app
Modified:
trunk/UPGRADE.txt
trunk/main/manager.c
trunk/main/pbx.c
Modified: trunk/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/trunk/UPGRADE.txt?rev=43444&r1=43443&r2=43444&view=diff
==============================================================================
--- trunk/UPGRADE.txt (original)
+++ trunk/UPGRADE.txt Thu Sep 21 15:01:54 2006
@@ -1,424 +1,3 @@
Information for Upgrading From Previous Asterisk Releases
=========================================================
-Build Process (configure script):
-
-Asterisk now uses an autoconf-generated configuration script to learn how it
-should build itself for your system. As it is a standard script, running:
-
-$ ./configure --help
-
-will show you all the options available. This script can be used to tell the
-build process what libraries you have on your system (if it cannot find them
-automatically), which libraries you wish to have ignored even though they may
-be present, etc.
-
-You must run the configure script before Asterisk will build, although it will
-attempt to automatically run it for you with no options specified; for most
-users, that will result in a similar build to what they would have had before
-the configure script was added to the build process (except for having to run
-'make' again after the configure script is run). Note that the configure script
-does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
-when your system configuration changes or you wish to build Asterisk with
-different options.
-
-Build Process (module selection):
-
-The Asterisk source tree now includes a basic module selection and build option
-selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
-In this tool, you can disable building of modules that you don't care about,
-turn on/off global options for the build and see which modules will not
-(and cannot) be built because your system does not have the required external
-dependencies installed.
-
-The resulting file from menuselect is called 'menuselect.makeopts'. Note that
-the resulting menuselect.makeopts file generally contains which modules *not*
-to build. The modules listed in this file indicate which modules have unmet
-dependencies, a present conflict, or have been disabled by the user in the
-menuselect interface. Compiler Flags can also be set in the menuselect
-interface. In this case, the resulting file contains which CFLAGS are in use,
-not which ones are not in use.
-
-If you would like to save your choices and have them applied against all
-builds, the file can be copied to '~/.asterisk.makeopts' or
-'/etc/asterisk.makeopts'.
-
-Build Process (Makefile targets):
-
-The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
-is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
-in the menuselect tool.
-
-It is now possible to run most make targets against a single subdirectory; from
-the top level directory, for example, 'make channels' will run 'make all' in the
-'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.
-
-Sound (prompt) and Music On Hold files:
-
-Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
-use with Asterisk have been replaced with new versions produced from high quality
-master recordings, and are available in three languages (English, French and
-Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
-In addition, the music on hold files provided by FreePlay Music are now available
-in the same five formats, but no longer available in MP3 format.
-
-The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
-(as were supplied with previous releases) and the FreePlay MOH files in WAV format.
-All of the other variations can be installed by running 'make menuselect' and
-selecting the packages you wish to install; when you run 'make install', those
-packages will be downloaded and installed along with the standard files included
-in the tarball.
-
-If for some reason you expect to not have Internet access at the time you will be
-running 'make install', you can make your package selections using menuselect and
-then run 'make sounds' to download (only) the sound packages; this will leave the
-sound packages in the 'sounds' subdirectory to be used later during installation.
-
-WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
-instead of the alternate-language files being stored in subdirectories underneath
-the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
-etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
-language itself, then places all the sound files for that language under that
-directory and its subdirectories. This is the layout that will be created if you
-select non-English languages to be installed via menuselect, HOWEVER Asterisk does
-not default to this layout and will not find the files in the places it expects them
-to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
-/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
-installed.
-
-PBX Core:
-
-* The (very old and undocumented) ability to use BYEXTENSION for dialing
- instead of ${EXTEN} has been removed.
-
-* Builtin (res_features) transfer functionality attempts to use the context
- defined in TRANSFER_CONTEXT variable of the transferer channel first. If
- not set, it uses the transferee variable. If not set in any channel, it will
- attempt to use the last non macro context. If not possible, it will default
- to the current context.
-
-* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
- if your dialplan relies on the ability to 'run off the end' of an extension
- and wait for a new extension without using WaitExten() to accomplish that,
- you will need set autofallthrough to 'no' in your extensions.conf file.
-
-Command Line Interface:
-
-* 'show channels concise', designed to be used by applications that will parse
- its output, previously used ':' characters to separate fields. However, some
- of those fields can easily contain that character, making the output not
- parseable. The delimiter has been changed to '!'.
-
-Applications:
-
-* In previous Asterisk releases, many applications would jump to priority n+101
- to indicate some kind of status or error condition. This functionality was
- marked deprecated in Asterisk 1.2. An option to disable it was provided with
- the default value set to 'on'. The default value for the global priority
- jumping option is now 'off'.
-
-* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
- AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
- and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
- been removed in this version. You should use the equivalent dialplan
- function in places where you have previously used one of these applications.
-
-* The application SetGlobalVar has been deprecated. You should replace uses
- of this application with the following combination of Set and GLOBAL():
- Set(GLOBAL(name)=value). You may also access global variables exclusively by
- using the GLOBAL() dialplan function, instead of relying on variable
- interpolation falling back to globals when no channel variable is set.
-
-* The application SetVar has been renamed to Set. The syntax SetVar was marked
- deprecated in version 1.2 and is no longer recognized in this version.
-
-* app_read has been updated to use the newer options codes, using "skip" or
- "noanswer" will not work. Use s or n. Also there is a new feature i, for
- using indication tones, so typing in skip would give you unexpected results.
-
-* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
-
-* The CONNECT event in the queue_log from app_queue now has a second field
- in addition to the holdtime field. It contains the unique ID of the
- queue member channel that is taking the call. This is useful when trying
- to link recording filenames back to a particular call from the queue.
-
-* The old/current behavior of app_queue has a serial type behavior
- in that the queue will make all waiting callers wait in the queue
- even if there is more than one available member ready to take
- calls until the head caller is connected with the member they
- were trying to get to. The next waiting caller in line then
- becomes the head caller, and they are then connected with the
- next available member and all available members and waiting callers
- waits while this happens. This cycle continues until there are
- no more available members or waiting callers, whichever comes first.
- The new behavior, enabled by setting autofill=yes in queues.conf
- either at the [general] level to default for all queues or
- to set on a per-queue level, makes sure that when the waiting
- callers are connecting with available members in a parallel fashion
- until there are no more available members or no more waiting callers,
- whichever comes first. This is probably more along the lines of how
- one would expect a queue should work and in most cases, you will want
- to enable this new behavior. If you do not specify or comment out this
- option, it will default to "no" to keep backward compatability with the old
- behavior.
-
-* The app_queue application now has the ability to use MixMonitor to
- record conversations queue members are having with queue callers. Please
- see configs/queues.conf.sample for more information on this option.
-
-* The app_queue application strategy called 'roundrobin' has been deprecated
- for this release. Users are encouraged to use 'rrmemory' instead, since it
- provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
- 'rrmemory' will be renamed 'roundrobin'.
-
-* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
- the 'm' option now provides the functionality of "initially muted".
- In practice, most existing dialplans using the 'm' flag should not notice
- any difference, unless the keypad menu is enabled, allowing the user
- to unmute themsleves.
-
-* ast_play_and_record would attempt to cancel the recording if a DTMF
- '0' was received. This behavior was not documented in most of the
- applications that used ast_play_and_record and the return codes from
- ast_play_and_record weren't checked for properly.
- ast_play_and_record has been changed so that '0' no longer cancels a
- recording. If you want to allow DTMF digits to cancel an
- in-progress recording use ast_play_and_record_full which allows you
- to specify which DTMF digits can be used to accept a recording and
- which digits can be used to cancel a recording.
-
-* ast_app_messagecount has been renamed to ast_app_inboxcount. There is now a
- new ast_app_messagecount function which takes a single context/mailbox/folder
- mailbox specification and returns the message count for that folder only.
- This addresses the deficiency of not being able to count the number of
- messages in folders other than INBOX and Old.
-
-* The exit behavior of the AGI applications has changed. Previously, when
- a connection to an AGI server failed, the application would cause the channel
- to immediately stop dialplan execution and hangup. Now, the only time that
- the AGI applications will cause the channel to stop dialplan execution is
- when the channel itself requests hangup. The AGI applications now set an
- AGISTATUS variable which will allow you to find out whether running the AGI
- was successful or not.
-
- Previously, there was no way to handle the case where Asterisk was unable to
- locally execute an AGI script for some reason. In this case, dialplan
- execution will continue as it did before, but the AGISTATUS variable will be
- set to "FAILURE".
-
- A locally executed AGI script can now exit with a non-zero exit code and this
- failure will be detected by Asterisk. If an AGI script exits with a non-zero
- exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
- "SUCCESS".
-
-* app_voicemail: The ODBC_STORAGE capability now requires the extended table format
- previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
- your table format using the schema provided in doc/odbcstorage.txt
-
-* app_waitforsilence: Fixes have been made to this application which changes the
- default behavior with how quickly it returns. You can maintain "old-style" behavior
- with the addition/use of a third "timeout" parameter.
- Please consult the application documentation and make changes to your dialplan
- if appropriate.
-
-Manager:
-
-* After executing the 'status' manager action, the "Status" manager events
- included the header "CallerID:" which was actually only the CallerID number,
- and not the full CallerID string. This header has been renamed to
- "CallerIDNum". For compatibility purposes, the CallerID parameter will remain
- until after the release of 1.4, when it will be removed. Please use the time
- during the 1.4 release to make this transition.
-
-* The AgentConnect event now has an additional field called "BridgedChannel"
- which contains the unique ID of the queue member channel that is taking the
- call. This is useful when trying to link recording filenames back to
- a particular call from the queue.
-
-* app_userevent has been modified to always send Event: UserEvent with the
- additional header UserEvent: <userspec>. Also, the Channel and UniqueID
- headers are not automatically sent, unless you specify them as separate
- arguments. Please see the application help for the new syntax.
-
-* app_meetme: Mute and Unmute events are now reported via the Manager API.
- Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
- are easier to use than "Action Command:". The MeetMeStopTalking event has
- also been deprecated in favor of the already existing MeetmeTalking event
- with a "Status" of "on" or "off" added.
-
-Variables:
-
-* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
- ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
- and ${LANGUAGE} have all been deprecated in favor of their related dialplan
- functions. You are encouraged to move towards the associated dialplan
- function, as these variables will be removed in a future release.
-
-* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
- adjustable from cdr.conf, instead of recompiling.
-
-* OSP applications exports several new variables, ${OSPINHANDLE},
- ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
- ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
-
-* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
- created channel. This variables holds the channel name of the transferer.
-
-* The dial plan variable PRI_CAUSE will be removed from future versions
- of Asterisk.
- It is replaced by adding a cause value to the hangup() application.
-
-Functions:
-
-* The function ${CHECK_MD5()} has been deprecated in favor of using an
- expression: $[${MD5(<string>)} = ${saved_md5}].
-
-* The 'builtin' functions that used to be combined in pbx_functions.so are
- now built as separate modules. If you are not using 'autoload=yes' in your
- modules.conf file then you will need to explicitly load the modules that
- contain the functions you want to use.
-
-* The ENUMLOOKUP() function with the 'c' option (for counting the number of
- records), but the lookup fails to match any records, the returned value will
- now be "0" instead of blank.
-
-* The REALTIME() function is now available in version 1.4 and app_realtime has
- been deprecated in favor of the new function. app_realtime will be removed
- completely with the version 1.6 release so please take the time between
- releases to make any necessary changes
-
-* The QUEUEAGENTCOUNT() function has been deprecated in favor of
- QUEUE_MEMBER_COUNT().
-
-The IAX2 channel:
-
-* The "mailboxdetail" option has been deprecated. Previously, if this option
- was not enabled, the 2 byte MSGCOUNT information element would be set to all
- 1's to indicate there there is some number of messages waiting. With this
- option enabled, the number of new messages were placed in one byte and the
- number of old messages are placed in the other. This is now the default
- (and the only) behavior.
-
-The SIP channel:
-
-* The "incominglimit" setting is replaced by the "call-limit" setting in
- sip.conf.
-
-* OSP support code is removed from SIP channel to OSP applications. ospauth
- option in sip.conf is removed to osp.conf as authpolicy. allowguest option
- in sip.conf cannot be set as osp anymore.
-
-* The Asterisk RTP stack has been changed in regards to RFC2833 reception
- and transmission. Packets will now be sent with proper duration instead of all
- at once. If you are receiving calls from a pre-1.4 Asterisk installation you
- will want to turn on the rfc2833compensate option. Without this option your
- DTMF reception may act poorly.
-
-* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
- in coming versions of Asterisk. Please use the dialplan function
- SIPCHANINFO(useragent) instead.
-
-* The ALERT_INFO dialplan variable is deprecated and will be removed
- in coming versions of Asterisk. Please use the dialplan application
- sipaddheader() to add the "Alert-Info" header to the outbound invite.
-
-The Zap channel:
-
-* Support for MFC/R2 has been removed, as it has not been functional for some
- time and it has no maintainer.
-
-The Agent channel:
-
-* Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
- it provided can be done using dialplan logic, without requiring additional
- channel and module locks (which frequently caused deadlocks). An example of
- how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.
-
-The G726-32 codec:
-
-* It has been determined that previous versions of Asterisk used the wrong codeword
- packing order for G726-32 data. This version supports both available packing orders,
- and can transcode between them. It also now selects the proper order when
- negotiating with a SIP peer based on the codec name supplied in the SDP. However,
- there are existing devices that improperly request one order and then use another;
- Sipura and Grandstream ATAs are known to do this, and there may be others. To
- be able to continue to use these devices with this version of Asterisk and the
- G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
- to sip.conf, so that Asterisk can use the packing order expected by the device (even
- though it requested a different order). In addition, the internal format number for
- G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
- result of this is that this version of Asterisk will be able to interoperate over
- IAX2 with older versions of Asterisk, as long as this version is told to allow
- 'g726aal2' instead of 'g726' as the codec for the call.
-
-Installation:
-
-* On BSD systems, the installation directories have changed to more "FreeBSDish"
- directories. On startup, Asterisk will look for the main configuration in
- /usr/local/etc/asterisk/asterisk.conf
- If you have an old installation, you might want to remove the binaries and
- move the configuration files to the new locations. The following directories
- are now default:
- ASTLIBDIR /usr/local/lib/asterisk
- ASTVARLIBDIR /usr/local/share/asterisk
- ASTETCDIR /usr/local/etc/asterisk
- ASTBINDIR /usr/local/bin/asterisk
- ASTSBINDIR /usr/local/sbin/asterisk
-
-Music on Hold:
-
-* The music on hold handling has been changed in some significant ways in hopes
- to make it work in a way that is much less confusing to users. Behavior will
- not change if the same configuration is used from older versions of Asterisk.
- However, there are some new configuration options that will make things work
- in a way that makes more sense.
-
- Previously, many of the channel drivers had an option called "musicclass" or
- something similar. This option set what music on hold class this channel
- would *hear* when put on hold. Some people expected (with good reason) that
- this option was to configure what music on hold class to play when putting
- the bridged channel on hold. This option has now been deprecated.
-
- Two new music on hold related configuration options for channel drivers have
- been introduced. Some channel drivers support both options, some just one,
- and some support neither of them. Check the sample configuration files to see
- which options apply to which channel driver.
-
- The "mohsuggest" option specifies which music on hold class to suggest to the
- bridged channel when putting them on hold. The only way that this class can
- be overridden is if the bridged channel has a specific music class set that
- was done in the dialplan using Set(CHANNEL(musicclass)=something).
-
- The "mohinterpret" option is similar to the old "musicclass" option. It
- specifies which music on hold class this channel would like to listen to when
- put on hold. This music class is only effective if this channel has no music
- class set on it from the dialplan and the bridged channel putting this one on
- hold had no "mohsuggest" setting.
-
- The IAX2 and Zap channel drivers have an additional feature for the
- "mohinterpret" option. If this option is set to "passthrough", then these
- channel drivers will pass through the HOLD message in signalling instead of
- starting music on hold on the channel. An example for how this would be
- useful is in an enterprise network of Asterisk servers. When one phone on one
- server puts a phone on a different server on hold, the remote server will be
- responsible for playing the hold music to its local phone that was put on
- hold instead of the far end server across the network playing the music.
-
-CDR Records:
-
-* The behavior of the "clid" field of the CDR has always been that it will
- contain the callerid ANI if it is set, or the callerid number if ANI was not
- set. When using the "callerid" option for various channel drivers, some
- would set ANI and some would not. This has been cleared up so that all
- channel drivers set ANI. If you would like to change the callerid number
- on the channel from the dialplan and have that change also show up in the
- CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).
-
-API:
-
-* There are some API functions that were not previously prefixed with the 'ast_'
- prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
- have a module that uses the services provided by res_adsi, res_odbc, or
- res_agi, you will need to add ast_ prefixes to the functions that you call
- from those modules.
Modified: trunk/main/manager.c
URL: http://svn.digium.com/view/asterisk/trunk/main/manager.c?rev=43444&r1=43443&r2=43444&view=diff
==============================================================================
--- trunk/main/manager.c (original)
+++ trunk/main/manager.c Thu Sep 21 15:01:54 2006
@@ -1430,7 +1430,6 @@
"Event: Status\r\n"
"Privilege: Call\r\n"
"Channel: %s\r\n"
- "CallerID: %s\r\n" /* This parameter is deprecated and will be removed post-1.4 */
"CallerIDNum: %s\r\n"
"CallerIDName: %s\r\n"
"Account: %s\r\n"
@@ -1445,7 +1444,6 @@
"\r\n",
c->name,
S_OR(c->cid.cid_num, "<unknown>"),
- S_OR(c->cid.cid_num, "<unknown>"),
S_OR(c->cid.cid_name, "<unknown>"),
c->accountcode,
ast_state2str(c->_state), c->context,
@@ -1455,7 +1453,6 @@
"Event: Status\r\n"
"Privilege: Call\r\n"
"Channel: %s\r\n"
- "CallerID: %s\r\n" /* This parameter is deprecated and will be removed post-1.4 */
"CallerIDNum: %s\r\n"
"CallerIDName: %s\r\n"
"Account: %s\r\n"
@@ -1465,7 +1462,6 @@
"%s"
"\r\n",
c->name,
- S_OR(c->cid.cid_num, "<unknown>"),
S_OR(c->cid.cid_num, "<unknown>"),
S_OR(c->cid.cid_name, "<unknown>"),
c->accountcode,
@@ -1595,12 +1591,10 @@
"Exten: %s\r\n"
"Reason: %d\r\n"
"Uniqueid: %s\r\n"
- "CallerID: %s\r\n" /* This parameter is deprecated and will be removed post-1.4 */
"CallerIDNum: %s\r\n"
"CallerIDName: %s\r\n",
in->idtext, in->tech, in->data, in->context, in->exten, reason,
chan ? chan->uniqueid : "<null>",
- S_OR(in->cid_num, "<unknown>"),
S_OR(in->cid_num, "<unknown>"),
S_OR(in->cid_name, "<unknown>")
);
Modified: trunk/main/pbx.c
URL: http://svn.digium.com/view/asterisk/trunk/main/pbx.c?rev=43444&r1=43443&r2=43444&view=diff
==============================================================================
--- trunk/main/pbx.c (original)
+++ trunk/main/pbx.c Thu Sep 21 15:01:54 2006
@@ -225,7 +225,6 @@
static int pbx_builtin_progress(struct ast_channel *, void *);
static int pbx_builtin_congestion(struct ast_channel *, void *);
static int pbx_builtin_busy(struct ast_channel *, void *);
-static int pbx_builtin_setglobalvar(struct ast_channel *, void *);
static int pbx_builtin_noop(struct ast_channel *, void *);
static int pbx_builtin_gotoif(struct ast_channel *, void *);
static int pbx_builtin_gotoiftime(struct ast_channel *, void *);
@@ -303,6 +302,13 @@
"Otherwise, this application will wait until the calling channel hangs up.\n"
},
+ { "ExecIfTime", pbx_builtin_execiftime,
+ "Conditional application execution based on the current time",
+ " ExecIfTime(<times>|<weekdays>|<mdays>|<months>?appname[|appargs]):\n"
+ "This application will execute the specified dialplan application, with optional\n"
+ "arguments, if the current time matches the given time specification.\n"
+ },
+
{ "Goto", pbx_builtin_goto,
"Jump to a particular priority, extension, or context",
" Goto([[context|]extension|]priority): This application will cause the\n"
@@ -331,11 +337,14 @@
"in the dialplan if the current time matches the given time specification.\n"
},
- { "ExecIfTime", pbx_builtin_execiftime,
- "Conditional application execution based on the current time",
- " ExecIfTime(<times>|<weekdays>|<mdays>|<months>?appname[|appargs]):\n"
- "This application will execute the specified dialplan application, with optional\n"
- "arguments, if the current time matches the given time specification.\n"
+ { "ImportVar", pbx_builtin_importvar,
+ "Import a variable from a channel into a new variable",
+ " ImportVar(newvar=channelname|variable): This application imports a variable\n"
+ "from the specified channel (as opposed to the current one) and stores it as\n"
+ "a variable in the current channel (the channel that is calling this\n"
+ "application). Variables created by this application have the same inheritance\n"
+ "properties as those created with the Set application. See the documentation for\n"
+ "Set for more information.\n"
},
{ "Hangup", pbx_builtin_hangup,
@@ -375,6 +384,20 @@
"tone to the user.\n"
},
+ { "SayAlpha", pbx_builtin_saycharacters,
+ "Say Alpha",
+ " SayAlpha(string): This application will play the sounds that correspond to\n"
+ "the letters of the given string.\n"
+ },
+
+ { "SayDigits", pbx_builtin_saydigits,
+ "Say Digits",
+ " SayDigits(digits): This application will play the sounds that correspond\n"
+ "to the digits of the given number. This will use the language that is currently\n"
+ "set for the channel. See the LANGUAGE function for more information on setting\n"
+ "the language for the channel.\n"
+ },
+
{ "SayNumber", pbx_builtin_saynumber,
"Say Number",
" SayNumber(digits[,gender]): This application will play the sounds that\n"
@@ -383,36 +406,10 @@
"LANGUAGE function for more information on setting the language for the channel.\n"
},
- { "SayDigits", pbx_builtin_saydigits,
- "Say Digits",
- " SayDigits(digits): This application will play the sounds that correspond\n"
- "to the digits of the given number. This will use the language that is currently\n"
- "set for the channel. See the LANGUAGE function for more information on setting\n"
- "the language for the channel.\n"
- },
-
- { "SayAlpha", pbx_builtin_saycharacters,
- "Say Alpha",
- " SayAlpha(string): This application will play the sounds that correspond to\n"
- "the letters of the given string.\n"
- },
-
{ "SayPhonetic", pbx_builtin_sayphonetic,
"Say Phonetic",
" SayPhonetic(string): This application will play the sounds from the phonetic\n"
"alphabet that correspond to the letters in the given string.\n"
- },
-
- { "SetAMAFlags", pbx_builtin_setamaflags,
- "Set the AMA Flags",
- " SetAMAFlags([flag]): This application will set the channel's AMA Flags for\n"
- " billing purposes.\n"
- },
-
- { "SetGlobalVar", pbx_builtin_setglobalvar,
- "Set a global variable to a given value",
- " SetGlobalVar(variable=value): This application sets a given global variable to\n"
- "the specified value.\n"
},
{ "Set", pbx_builtin_setvar,
@@ -429,14 +426,10 @@
" (applies only to variables, not functions)\n"
},
- { "ImportVar", pbx_builtin_importvar,
- "Import a variable from a channel into a new variable",
- " ImportVar(newvar=channelname|variable): This application imports a variable\n"
- "from the specified channel (as opposed to the current one) and stores it as\n"
- "a variable in the current channel (the channel that is calling this\n"
- "application). Variables created by this application have the same inheritance\n"
- "properties as those created with the Set application. See the documentation for\n"
- "Set for more information.\n"
+ { "SetAMAFlags", pbx_builtin_setamaflags,
+ "Set the AMA Flags",
+ " SetAMAFlags([flag]): This application will set the channel's AMA Flags for\n"
+ " billing purposes.\n"
},
{ "Wait", pbx_builtin_wait,
@@ -5861,31 +5854,6 @@
return(0);
}
-/*! \todo XXX overwrites data ? */
-static int pbx_builtin_setglobalvar(struct ast_channel *chan, void *data)
-{
- char *name;
- char *stringp = data;
- static int dep_warning = 0;
-
- if (ast_strlen_zero(data)) {
- ast_log(LOG_WARNING, "Ignoring, since there is no variable to set\n");
- return 0;
- }
-
- name = strsep(&stringp, "=");
-
- if (!dep_warning) {
- dep_warning = 1;
- ast_log(LOG_WARNING, "SetGlobalVar is deprecated. Please use Set(GLOBAL(%s)=%s) instead.\n", name, stringp);
- }
-
- /*! \todo XXX watch out, leading whitespace ? */
- pbx_builtin_setvar_helper(NULL, name, stringp);
-
- return(0);
-}
-
static int pbx_builtin_noop(struct ast_channel *chan, void *data)
{
return 0;
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