Merged revisions 157706 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
  r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines
  
  make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines
  
  also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
........


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@157738 f38db490-d61c-443f-a65b-d21fe96a405b

6 files changed, 485 insertions, 383 deletions

diff --git a/UPGRADE-1.6.txt b/UPGRADE-1.6.txt
new file mode 100644
index 000000000..8fb0a0f71
--- /dev/null
+++ b/UPGRADE-1.6.txt
@@ -0,0 +1,250 @@
+=========================================================
+=== Information for upgrading from Asterisk 1.4 to 1.6
+===
+===
+=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
+=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
+=== UPGRADE.txt     -- Upgrade info for 1.4 to 1.6
+=========================================================
+
+AEL:
+
+* Macros are now implemented underneath with the Gosub() application.
+  Heaven Help You if you wrote code depending on any aspect of this!
+  Previous to 1.6, macros were implemented with the Macro() app, which
+  provided a nice feature of auto-returning. The compiler will do its
+  best to insert a Return() app call at the end of your macro if you did
+  not include it, but really, you should make sure that all execution
+  paths within your macros end in "return;".
+
+* The conf2ael program is 'introduced' in this release; it is in a rather
+  crude state, but deemed useful for making a first pass at converting
+  extensions.conf code into AEL. More intelligence will come with time.
+
+Core:
+
+* The 'languageprefix' option in asterisk.conf is now deprecated, and
+  the default sound file layout for non-English sounds is the 'new
+  style' layout introduced in Asterisk 1.4 (and used by the automatic
+  sound file installer in the Makefile).
+
+* The ast_expr2 stuff has been modified to handle floating-point numbers.
+  Numbers of the format D.D are now acceptable input for the expr parser, 
+  Where D is a string of base-10 digits. All math is now done in "long double",
+  if it is available on your compiler/architecture. This was half-way between
+  a bug-fix (because the MATH func returns fp by default), and an enhancement.
+  Also, for those counting on, or needing, integer operations, a series of
+  'functions' were also added to the expr language, to allow several styles
+  of rounding/truncation, along with a set of common floating point operations,
+  like sin, cos, tan, log, pow, etc. The ability to call external functions
+  like CDR(), etc. was also added, without having to use the ${...} notation.
+ 
+* The delimiter passed to applications has been changed to the comma (','), as
+  that is what people are used to using within extensions.conf.  If you are
+  using realtime extensions, you will need to translate your existing dialplan
+  to use this separator.  To use a literal comma, you need merely to escape it
+  with a backslash ('\').  Another possible side effect is that you may need to
+  remove the obscene level of backslashing that was necessary for the dialplan
+  to work correctly in 1.4 and previous versions.  This should make writing
+  dialplans less painful in the future, albeit with the pain of a one-time
+  conversion.  If you would like to avoid this conversion immediately, set
+  pbx_realtime=1.4 in the [compat] section of asterisk.conf.  After
+  transitioning, set pbx_realtime=1.6 in the same section.
+
+* For the same purpose as above, you may set res_agi=1.4 in the [compat]
+  section of asterisk.conf to continue to use the '|' delimiter in the EXEC
+  arguments of AGI applications.  After converting to use the ',' delimiter,
+  change this option to res_agi=1.6.
+
+* The logger.conf option 'rotatetimestamp' has been deprecated in favor of
+  'rotatestrategy'.  This new option supports a 'rotate' strategy that more
+  closely mimics the system logger in terms of file rotation.
+
+* The concise versions of various CLI commands are now deprecated. We recommend
+  using the manager interface (AMI) for application integration with Asterisk.
+
+Voicemail:
+
+* The voicemail configuration values 'maxmessage' and 'minmessage' have
+  been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
+  to make them more distinguishable from 'maxmsgs', which sets folder
+  size.  The old variables will continue to work in this version, albeit
+  with a deprecation warning.
+
+* If you use any interface for modifying voicemail aside from the built in
+  dialplan applications, then the option "pollmailboxes" *must* be set in
+  voicemail.conf for message waiting indication (MWI) to work properly.  This
+  is because Voicemail notification is now event based instead of polling
+  based.  The channel drivers are no longer responsible for constantly manually
+  checking mailboxes for changes so that they can send MWI information to users.
+  Examples of situations that would require this option are web interfaces to
+  voicemail or an email client in the case of using IMAP storage.
+
+Applications:
+
+
+* ChanIsAvail() now has a 't' option, which allows the specified device
+  to be queried for state without consulting the channel drivers. This
+  performs mostly a 'ChanExists' sort of function.
+
+* ChannelRedirect() will not terminate the channel that fails to do a
+  channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
+  will reflect if the attempt was successful of not.
+
+* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
+  and is now deprecated.
+
+* DISA()'s fifth argument is now an options argument.  If you have previously
+  used 'NOANSWER' in this argument, you'll need to convert that to the new
+  option 'n'.
+
+* Macro() is now deprecated.  If you need subroutines, you should use the
+  Gosub()/Return() applications.  To replace MacroExclusive(), we have
+  introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK().  You may use
+  these functions in any location where you desire to ensure that only one
+  channel is executing that path at any one time.  The Macro() applications
+  are deprecated for performance reasons.  However, since Macro() has been
+  around for a long time and so many dialplans depend heavily on it, for the
+  sake of backwards compatibility it will not be removed .  It is also worth
+  noting that using both Macro() and GoSub() at the same time is _heavily_
+  discouraged.
+
+* Read() now sets a READSTATUS variable on exit.  It does NOT automatically
+  return -1 (and hangup) anymore on error.  If you want to hangup on error,
+  you need to do so explicitly in your dialplan.
+
+* Privacy() no longer uses privacy.conf, so any options must be specified
+  directly in the application arguments.
+
+* MusicOnHold application now has duration parameter which allows specifying
+  timeout in seconds.
+
+* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.
+
+* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
+  instead.
+
+* The arguments in ExecIf changed a bit, to be more like other applications.
+  The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)).
+
+* The behavior of the Set application now depends upon a compatibility option,
+  set in asterisk.conf.  To use the old 1.4 behavior, which allowed Set to take
+  multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf.  To
+  use the new behavior, which permits variables to be set with embedded commas,
+  set app_set=1.6 in [compat] in asterisk.conf.  Note that you can have both
+  behaviors at the same time, if you switch to using MSet if you want the old
+  behavior.
+
+Dialplan Functions:
+
+* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
+  more information, issue a "show function QUEUE_MEMBER" from the CLI.
+
+CDR:
+
+* The cdr_sqlite module has been marked as deprecated in favor of
+  cdr_sqlite3_custom.  It will potentially be removed from the tree
+  after Asterisk 1.6 is released.
+
+* The cdr_odbc module now uses res_odbc to manage its connections.  The
+  username and password parameters in cdr_odbc.conf, therefore, are no
+  longer used.  The dsn parameter now points to an entry in res_odbc.conf.
+
+* The uniqueid field in the core Asterisk structure has been changed from a
+  maximum 31 character field to a 149 character field, to account for all
+  possible values the systemname prefix could be.  In the past, if the
+  systemname was too long, the uniqueid would have been truncated.
+
+* The cdr_tds module now supports all versions of FreeTDS that contain
+  the db-lib frontend.  It will also now log the userfield variable if
+  the target database table contains a column for it.
+
+Formats:
+
+* format_wav: The GAIN preprocessor definition and source code that used it
+  is removed.  This change was made in response to user complaints of
+  choppiness or the clipping of loud signal peaks.  To increase the volume
+  of voicemail messages, use the 'volgain' option in voicemail.conf
+
+Channel Drivers:
+
+* SIP: a small upgrade to support the "Record" button on the SNOM360,
+  which sends a sip INFO message with a "Record: on" or "Record: off" 
+  header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
+  requests (by default, via '*1'), then the user-configured dialpad sequence
+  is generated, and recording can be started and stopped via this button. The
+  file names and formats are all controlled via the normal mechanisms. If the
+  user has not configured the automon feature, the normal "415 Unsupported media type"
+  is returned, and nothing is done.
+
+* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
+  Asterisk, but will be removed in the following version. Please use the groupcount functions
+  in the dialplan to enforce call limits. The "limitonpeer" configuration option is
+  now renamed to "counteronpeer".
+
+* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
+  These are used only before registration to call a peer with the uri 
+	sip:defaultuser@defaultip
+  The "username" setting still work, but is deprecated and will not work in 
+  the next version of Asterisk.
+
+* chan_local.c: the comma delimiter inside the channel name has been changed to a
+  semicolon, in order to make the Local channel driver compatible with the comma
+  delimiter change in applications.
+
+* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
+  to be compatible with settings in sip.conf. The "tos" and "cos" configuration
+  is deprecated and will stop working in the next release of Asterisk.
+
+* Console: A new console channel driver, chan_console, has been added to Asterisk.
+  This new module can not be loaded at the same time as chan_alsa or chan_oss.  The
+  default modules.conf only loads one of them (chan_oss by default).  So, unless you
+  have modified your modules.conf to not use the autoload option, then you will need
+  to modify modules.conf to add another "noload" line to ensure that only one of
+  these three modules gets loaded.
+
+* DAHDI: The chan_zap module that supported PSTN interfaces using
+  Zaptel has been renamed to chan_dahdi, and only supports the DAHDI
+  telephony driver package for PSTN interfaces. See the
+  Zaptel-to-DAHDI.txt file for more details on this transition.
+
+* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over
+  the method of stripping digits in the dialplan using variable substring syntax.
+
+Configuration:
+
+* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
+  lowcost and other is not acceptable now. Look into qos.tex for description of 
+  this parameter.
+
+* queues.conf: the queue-lessthan sound file option is no longer available, and the
+  queue-round-seconds option no longer takes '1' as a valid parameter.
+
+Manager:
+
+* Manager has been upgraded to version 1.1 with a lot of changes. 
+  Please check doc/manager_1_1.txt for information
+
+* The IAXpeers command output has been changed to more closely resemble the
+  output of the SIPpeers command.
+
+* cdr_manager now reports at the "cdr" level, not at "call"  You may need to
+   change your manager.conf to add the level to existing AMI users, if they
+   want to see the CDR events generated.
+
+* The Originate command now requires the Originate write permission.  For
+   Originate with the Application parameter, you need the additional System
+   privilege if you want to do anything that calls out to a subshell.
+
+iLBC Codec:
+
+* Previously, the Asterisk source code distribution included the iLBC
+  encoder/decoder source code, from Global IP Solutions
+  (http://www.gipscorp.com). This code is not licensed for
+  distribution, and thus has been removed from the Asterisk source
+  code distribution. If you wish to use codec_ilbc to support iLBC
+  channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
+  script to download the source and put it in the proper place in
+  the Asterisk build tree. Once that is done you can follow your normal
+  steps of building Asterisk. You will need to run 'menuselect' and enable
+  the iLBC codec in the 'Codec  Translators' category.
diff --git a/UPGRADE.txt b/UPGRADE.txt
index 8fb0a0f71..0f04671d8 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -1,250 +1,27 @@
-=========================================================
-=== Information for upgrading from Asterisk 1.4 to 1.6
+===========================================================
+=== Information for upgrading between Asterisk 1.6 versions
 ===
 ===
 === UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
 === UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE.txt     -- Upgrade info for 1.4 to 1.6
-=========================================================
-
-AEL:
-
-* Macros are now implemented underneath with the Gosub() application.
-  Heaven Help You if you wrote code depending on any aspect of this!
-  Previous to 1.6, macros were implemented with the Macro() app, which
-  provided a nice feature of auto-returning. The compiler will do its
-  best to insert a Return() app call at the end of your macro if you did
-  not include it, but really, you should make sure that all execution
-  paths within your macros end in "return;".
-
-* The conf2ael program is 'introduced' in this release; it is in a rather
-  crude state, but deemed useful for making a first pass at converting
-  extensions.conf code into AEL. More intelligence will come with time.
-
-Core:
-
-* The 'languageprefix' option in asterisk.conf is now deprecated, and
-  the default sound file layout for non-English sounds is the 'new
-  style' layout introduced in Asterisk 1.4 (and used by the automatic
-  sound file installer in the Makefile).
-
-* The ast_expr2 stuff has been modified to handle floating-point numbers.
-  Numbers of the format D.D are now acceptable input for the expr parser, 
-  Where D is a string of base-10 digits. All math is now done in "long double",
-  if it is available on your compiler/architecture. This was half-way between
-  a bug-fix (because the MATH func returns fp by default), and an enhancement.
-  Also, for those counting on, or needing, integer operations, a series of
-  'functions' were also added to the expr language, to allow several styles
-  of rounding/truncation, along with a set of common floating point operations,
-  like sin, cos, tan, log, pow, etc. The ability to call external functions
-  like CDR(), etc. was also added, without having to use the ${...} notation.
- 
-* The delimiter passed to applications has been changed to the comma (','), as
-  that is what people are used to using within extensions.conf.  If you are
-  using realtime extensions, you will need to translate your existing dialplan
-  to use this separator.  To use a literal comma, you need merely to escape it
-  with a backslash ('\').  Another possible side effect is that you may need to
-  remove the obscene level of backslashing that was necessary for the dialplan
-  to work correctly in 1.4 and previous versions.  This should make writing
-  dialplans less painful in the future, albeit with the pain of a one-time
-  conversion.  If you would like to avoid this conversion immediately, set
-  pbx_realtime=1.4 in the [compat] section of asterisk.conf.  After
-  transitioning, set pbx_realtime=1.6 in the same section.
-
-* For the same purpose as above, you may set res_agi=1.4 in the [compat]
-  section of asterisk.conf to continue to use the '|' delimiter in the EXEC
-  arguments of AGI applications.  After converting to use the ',' delimiter,
-  change this option to res_agi=1.6.
-
-* The logger.conf option 'rotatetimestamp' has been deprecated in favor of
-  'rotatestrategy'.  This new option supports a 'rotate' strategy that more
-  closely mimics the system logger in terms of file rotation.
-
-* The concise versions of various CLI commands are now deprecated. We recommend
-  using the manager interface (AMI) for application integration with Asterisk.
-
-Voicemail:
-
-* The voicemail configuration values 'maxmessage' and 'minmessage' have
-  been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
-  to make them more distinguishable from 'maxmsgs', which sets folder
-  size.  The old variables will continue to work in this version, albeit
-  with a deprecation warning.
-
-* If you use any interface for modifying voicemail aside from the built in
-  dialplan applications, then the option "pollmailboxes" *must* be set in
-  voicemail.conf for message waiting indication (MWI) to work properly.  This
-  is because Voicemail notification is now event based instead of polling
-  based.  The channel drivers are no longer responsible for constantly manually
-  checking mailboxes for changes so that they can send MWI information to users.
-  Examples of situations that would require this option are web interfaces to
-  voicemail or an email client in the case of using IMAP storage.
-
-Applications:
-
-
-* ChanIsAvail() now has a 't' option, which allows the specified device
-  to be queried for state without consulting the channel drivers. This
-  performs mostly a 'ChanExists' sort of function.
-
-* ChannelRedirect() will not terminate the channel that fails to do a
-  channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
-  will reflect if the attempt was successful of not.
-
-* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
-  and is now deprecated.
-
-* DISA()'s fifth argument is now an options argument.  If you have previously
-  used 'NOANSWER' in this argument, you'll need to convert that to the new
-  option 'n'.
-
-* Macro() is now deprecated.  If you need subroutines, you should use the
-  Gosub()/Return() applications.  To replace MacroExclusive(), we have
-  introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK().  You may use
-  these functions in any location where you desire to ensure that only one
-  channel is executing that path at any one time.  The Macro() applications
-  are deprecated for performance reasons.  However, since Macro() has been
-  around for a long time and so many dialplans depend heavily on it, for the
-  sake of backwards compatibility it will not be removed .  It is also worth
-  noting that using both Macro() and GoSub() at the same time is _heavily_
-  discouraged.
-
-* Read() now sets a READSTATUS variable on exit.  It does NOT automatically
-  return -1 (and hangup) anymore on error.  If you want to hangup on error,
-  you need to do so explicitly in your dialplan.
-
-* Privacy() no longer uses privacy.conf, so any options must be specified
-  directly in the application arguments.
-
-* MusicOnHold application now has duration parameter which allows specifying
-  timeout in seconds.
-
-* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.
-
-* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
-  instead.
-
-* The arguments in ExecIf changed a bit, to be more like other applications.
-  The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)).
-
-* The behavior of the Set application now depends upon a compatibility option,
-  set in asterisk.conf.  To use the old 1.4 behavior, which allowed Set to take
-  multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf.  To
-  use the new behavior, which permits variables to be set with embedded commas,
-  set app_set=1.6 in [compat] in asterisk.conf.  Note that you can have both
-  behaviors at the same time, if you switch to using MSet if you want the old
-  behavior.
-
-Dialplan Functions:
-
-* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
-  more information, issue a "show function QUEUE_MEMBER" from the CLI.
-
-CDR:
-
-* The cdr_sqlite module has been marked as deprecated in favor of
-  cdr_sqlite3_custom.  It will potentially be removed from the tree
-  after Asterisk 1.6 is released.
-
-* The cdr_odbc module now uses res_odbc to manage its connections.  The
-  username and password parameters in cdr_odbc.conf, therefore, are no
-  longer used.  The dsn parameter now points to an entry in res_odbc.conf.
-
-* The uniqueid field in the core Asterisk structure has been changed from a
-  maximum 31 character field to a 149 character field, to account for all
-  possible values the systemname prefix could be.  In the past, if the
-  systemname was too long, the uniqueid would have been truncated.
-
-* The cdr_tds module now supports all versions of FreeTDS that contain
-  the db-lib frontend.  It will also now log the userfield variable if
-  the target database table contains a column for it.
-
-Formats:
-
-* format_wav: The GAIN preprocessor definition and source code that used it
-  is removed.  This change was made in response to user complaints of
-  choppiness or the clipping of loud signal peaks.  To increase the volume
-  of voicemail messages, use the 'volgain' option in voicemail.conf
-
-Channel Drivers:
-
-* SIP: a small upgrade to support the "Record" button on the SNOM360,
-  which sends a sip INFO message with a "Record: on" or "Record: off" 
-  header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
-  requests (by default, via '*1'), then the user-configured dialpad sequence
-  is generated, and recording can be started and stopped via this button. The
-  file names and formats are all controlled via the normal mechanisms. If the
-  user has not configured the automon feature, the normal "415 Unsupported media type"
-  is returned, and nothing is done.
-
-* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
-  Asterisk, but will be removed in the following version. Please use the groupcount functions
-  in the dialplan to enforce call limits. The "limitonpeer" configuration option is
-  now renamed to "counteronpeer".
-
-* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
-  These are used only before registration to call a peer with the uri 
-	sip:defaultuser@defaultip
-  The "username" setting still work, but is deprecated and will not work in 
-  the next version of Asterisk.
-
-* chan_local.c: the comma delimiter inside the channel name has been changed to a
-  semicolon, in order to make the Local channel driver compatible with the comma
-  delimiter change in applications.
-
-* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
-  to be compatible with settings in sip.conf. The "tos" and "cos" configuration
-  is deprecated and will stop working in the next release of Asterisk.
-
-* Console: A new console channel driver, chan_console, has been added to Asterisk.
-  This new module can not be loaded at the same time as chan_alsa or chan_oss.  The
-  default modules.conf only loads one of them (chan_oss by default).  So, unless you
-  have modified your modules.conf to not use the autoload option, then you will need
-  to modify modules.conf to add another "noload" line to ensure that only one of
-  these three modules gets loaded.
-
-* DAHDI: The chan_zap module that supported PSTN interfaces using
-  Zaptel has been renamed to chan_dahdi, and only supports the DAHDI
-  telephony driver package for PSTN interfaces. See the
-  Zaptel-to-DAHDI.txt file for more details on this transition.
-
-* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over
-  the method of stripping digits in the dialplan using variable substring syntax.
-
-Configuration:
-
-* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
-  lowcost and other is not acceptable now. Look into qos.tex for description of 
-  this parameter.
-
-* queues.conf: the queue-lessthan sound file option is no longer available, and the
-  queue-round-seconds option no longer takes '1' as a valid parameter.
-
-Manager:
-
-* Manager has been upgraded to version 1.1 with a lot of changes. 
-  Please check doc/manager_1_1.txt for information
-
-* The IAXpeers command output has been changed to more closely resemble the
-  output of the SIPpeers command.
-
-* cdr_manager now reports at the "cdr" level, not at "call"  You may need to
-   change your manager.conf to add the level to existing AMI users, if they
-   want to see the CDR events generated.
-
-* The Originate command now requires the Originate write permission.  For
-   Originate with the Application parameter, you need the additional System
-   privilege if you want to do anything that calls out to a subshell.
-
-iLBC Codec:
-
-* Previously, the Asterisk source code distribution included the iLBC
-  encoder/decoder source code, from Global IP Solutions
-  (http://www.gipscorp.com). This code is not licensed for
-  distribution, and thus has been removed from the Asterisk source
-  code distribution. If you wish to use codec_ilbc to support iLBC
-  channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
-  script to download the source and put it in the proper place in
-  the Asterisk build tree. Once that is done you can follow your normal
-  steps of building Asterisk. You will need to run 'menuselect' and enable
-  the iLBC codec in the 'Codec  Translators' category.
+=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
+===========================================================
+
+From 1.6.0.1 to 1.6.0.2:
+
+* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
+  API calls were added in 1.6.0, so that modules that provide multiple
+  AGI commands could register/unregister them all with a single
+  step. However, these API calls were not implemented properly, and did
+  not allow the caller to know whether registration or unregistration
+  succeeded or failed. They have been redefined to now return success
+  or failure, but this means any code using these functions will need
+  be recompiled after upgrading to a version of Asterisk containing
+  these changes. In addition, the source code using these functions
+  should be reviewed to ensure it can properly react to failure
+  of registration or unregistration of its API commands.
+
+* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
+  to better match what it really does, and the argument order has been
+  changed to be consistent with other API calls that perform similar
+  operations.
diff --git a/apps/app_stack.c b/apps/app_stack.c
index 841562536..a6333d8d4 100644
--- a/apps/app_stack.c
+++ b/apps/app_stack.c
@@ -406,11 +406,11 @@ static int handle_gosub(struct ast_channel *chan, AGI *agi, int argc, char **arg
 		/* Lookup the priority label */
 		if ((priority = ast_findlabel_extension(chan, argv[1], argv[2], argv[3], chan->cid.cid_num)) < 0) {
 			ast_log(LOG_ERROR, "Priority '%s' not found in '%s@%s'\n", argv[3], argv[2], argv[1]);
-			ast_agi_fdprintf(chan, agi->fd, "200 result=-1 Gosub label not found\n");
+			ast_agi_send(agi->fd, chan, "200 result=-1 Gosub label not found\n");
 			return RESULT_FAILURE;
 		}
 	} else if (!ast_exists_extension(chan, argv[1], argv[2], priority, chan->cid.cid_num)) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=-1 Gosub label not found\n");
+		ast_agi_send(agi->fd, chan, "200 result=-1 Gosub label not found\n");
 		return RESULT_FAILURE;
 	}
 
@@ -421,7 +421,7 @@ static int handle_gosub(struct ast_channel *chan, AGI *agi, int argc, char **arg
 
 	if (!(theapp = pbx_findapp("Gosub"))) {
 		ast_log(LOG_ERROR, "Gosub() cannot be found in the list of loaded applications\n");
-		ast_agi_fdprintf(chan, agi->fd, "503 result=-2 Gosub is not loaded\n");
+		ast_agi_send(agi->fd, chan, "503 result=-2 Gosub is not loaded\n");
 		return RESULT_FAILURE;
 	}
 
@@ -455,19 +455,19 @@ static int handle_gosub(struct ast_channel *chan, AGI *agi, int argc, char **arg
 			struct ast_pbx *pbx = chan->pbx;
 			/* Suppress warning about PBX already existing */
 			chan->pbx = NULL;
-			ast_agi_fdprintf(chan, agi->fd, "100 result=0 Trying...\n");
+			ast_agi_send(agi->fd, chan, "100 result=0 Trying...\n");
 			ast_pbx_run(chan);
-			ast_agi_fdprintf(chan, agi->fd, "200 result=0 Gosub complete\n");
+			ast_agi_send(agi->fd, chan, "200 result=0 Gosub complete\n");
 			if (chan->pbx) {
 				ast_free(chan->pbx);
 			}
 			chan->pbx = pbx;
 		} else {
-			ast_agi_fdprintf(chan, agi->fd, "200 result=%d Gosub failed\n", res);
+			ast_agi_send(agi->fd, chan, "200 result=%d Gosub failed\n", res);
 		}
 		ast_free(gosub_args);
 	} else {
-		ast_agi_fdprintf(chan, agi->fd, "503 result=-2 Memory allocation failure\n");
+		ast_agi_send(agi->fd, chan, "503 result=-2 Memory allocation failure\n");
 		return RESULT_FAILURE;
 	}
 
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index e7c5ebb04..a4cec6310 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4487,7 +4487,7 @@ static int update_call_counter(struct sip_pvt *fup, int event)
 	int outgoing = fup->outgoing_call;
 	struct sip_user *u = NULL;
 	struct sip_peer *p = NULL;
-	ast_mutex_t *pu_lock;
+	ast_mutex_t *pu_lock = NULL;
 
 	ast_debug(3, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
 
diff --git a/include/asterisk/agi.h b/include/asterisk/agi.h
index 609e98815..10c88e632 100644
--- a/include/asterisk/agi.h
+++ b/include/asterisk/agi.h
@@ -61,11 +61,53 @@ typedef struct agi_command {
 #define AGI_WEAK
 #endif
 
-int AGI_WEAK ast_agi_fdprintf(struct ast_channel *chan, int fd, char *fmt, ...);
+/*!
+ * \brief
+ *
+ * Sends a string of text to an application connected via AGI.
+ *
+ * \param fd The file descriptor for the AGI session (from struct agi_state)
+ * \param chan Pointer to an associated Asterisk channel, if any
+ * \param fmt printf-style format string
+ * \return 0 for success, -1 for failure
+ *
+ */
+int AGI_WEAK ast_agi_send(int fd, struct ast_channel *chan, char *fmt, ...) __attribute__((format(printf, 3, 4)));
 int AGI_WEAK ast_agi_register(struct ast_module *mod, agi_command *cmd);
 int AGI_WEAK ast_agi_unregister(struct ast_module *mod, agi_command *cmd);
-void AGI_WEAK ast_agi_register_multiple(struct ast_module *mod, agi_command *cmd, int len);
-void AGI_WEAK ast_agi_unregister_multiple(struct ast_module *mod, agi_command *cmd, int len);
+
+/*!
+ * \brief
+ *
+ * Registers a group of AGI commands, provided as an array of struct agi_command
+ * entries.
+ *
+ * \param mod Pointer to the module_info structure for the module that is registering the commands
+ * \param cmd Pointer to the first entry in the array of commands
+ * \param len Length of the array (use the ARRAY_LEN macro to determine this easily)
+ * \return 0 on success, -1 on failure
+ *
+ * \note If any command fails to register, all commands previously registered during the operation
+ * will be unregistered. In other words, this function registers all the provided commands, or none
+ * of them.
+ */
+int AGI_WEAK ast_agi_register_multiple(struct ast_module *mod, struct agi_command *cmd, unsigned int len);
+
+/*!
+ * \brief
+ *
+ * Unregisters a group of AGI commands, provided as an array of struct agi_command
+ * entries.
+ *
+ * \param mod Pointer to the module_info structure for the module that is unregistering the commands
+ * \param cmd Pointer to the first entry in the array of commands
+ * \param len Length of the array (use the ARRAY_LEN macro to determine this easily)
+ * \return 0 on success, -1 on failure
+ *
+ * \note If any command fails to unregister, this function will continue to unregister the
+ * remaining commands in the array; it will not reregister the already-unregistered commands.
+ */
+int AGI_WEAK ast_agi_unregister_multiple(struct ast_module *mod, struct agi_command *cmd, unsigned int len);
 
 #if defined(__cplusplus) || defined(c_plusplus)
 }
diff --git a/res/res_agi.c b/res/res_agi.c
index 2df15c4d8..14d80d969 100644
--- a/res/res_agi.c
+++ b/res/res_agi.c
@@ -114,7 +114,7 @@ static agi_command *find_command(char *cmds[], int exact);
 AST_THREADSTORAGE(agi_buf);
 #define AGI_BUF_INITSIZE 256
 
-int ast_agi_fdprintf(struct ast_channel *chan, int fd, char *fmt, ...)
+int ast_agi_send(int fd, struct ast_channel *chan, char *fmt, ...)
 {
 	int res = 0;
 	va_list ap;
@@ -581,7 +581,7 @@ static enum agi_result launch_netscript(char *agiurl, char *argv[], int *fds, in
 		}
 	}
 
-	if (ast_agi_fdprintf(NULL, s, "agi_network: yes\n") < 0) {
+	if (ast_agi_send(s, NULL, "agi_network: yes\n") < 0) {
 		if (errno != EINTR) {
 			ast_log(LOG_WARNING, "Connect to '%s' failed: %s\n", agiurl, strerror(errno));
 			close(s);
@@ -592,7 +592,7 @@ static enum agi_result launch_netscript(char *agiurl, char *argv[], int *fds, in
 	/* If we have a script parameter, relay it to the fastagi server */
 	/* Script parameters take the form of: AGI(agi://my.example.com/?extension=${EXTEN}) */
 	if (!ast_strlen_zero(script))
-		ast_agi_fdprintf(NULL, s, "agi_network_script: %s\n", script);
+		ast_agi_send(s, NULL, "agi_network_script: %s\n", script);
 
 	ast_debug(4, "Wow, connected!\n");
 	fds[0] = s;
@@ -743,40 +743,40 @@ static void setup_env(struct ast_channel *chan, char *request, int fd, int enhan
 
 	/* Print initial environment, with agi_request always being the first
 	   thing */
-	ast_agi_fdprintf(chan, fd, "agi_request: %s\n", request);
-	ast_agi_fdprintf(chan, fd, "agi_channel: %s\n", chan->name);
-	ast_agi_fdprintf(chan, fd, "agi_language: %s\n", chan->language);
-	ast_agi_fdprintf(chan, fd, "agi_type: %s\n", chan->tech->type);
-	ast_agi_fdprintf(chan, fd, "agi_uniqueid: %s\n", chan->uniqueid);
-	ast_agi_fdprintf(chan, fd, "agi_version: %s\n", ast_get_version());
+	ast_agi_send(fd, chan, "agi_request: %s\n", request);
+	ast_agi_send(fd, chan, "agi_channel: %s\n", chan->name);
+	ast_agi_send(fd, chan, "agi_language: %s\n", chan->language);
+	ast_agi_send(fd, chan, "agi_type: %s\n", chan->tech->type);
+	ast_agi_send(fd, chan, "agi_uniqueid: %s\n", chan->uniqueid);
+	ast_agi_send(fd, chan, "agi_version: %s\n", ast_get_version());
 
 	/* ANI/DNIS */
-	ast_agi_fdprintf(chan, fd, "agi_callerid: %s\n", S_OR(chan->cid.cid_num, "unknown"));
-	ast_agi_fdprintf(chan, fd, "agi_calleridname: %s\n", S_OR(chan->cid.cid_name, "unknown"));
-	ast_agi_fdprintf(chan, fd, "agi_callingpres: %d\n", chan->cid.cid_pres);
-	ast_agi_fdprintf(chan, fd, "agi_callingani2: %d\n", chan->cid.cid_ani2);
-	ast_agi_fdprintf(chan, fd, "agi_callington: %d\n", chan->cid.cid_ton);
-	ast_agi_fdprintf(chan, fd, "agi_callingtns: %d\n", chan->cid.cid_tns);
-	ast_agi_fdprintf(chan, fd, "agi_dnid: %s\n", S_OR(chan->cid.cid_dnid, "unknown"));
-	ast_agi_fdprintf(chan, fd, "agi_rdnis: %s\n", S_OR(chan->cid.cid_rdnis, "unknown"));
+	ast_agi_send(fd, chan, "agi_callerid: %s\n", S_OR(chan->cid.cid_num, "unknown"));
+	ast_agi_send(fd, chan, "agi_calleridname: %s\n", S_OR(chan->cid.cid_name, "unknown"));
+	ast_agi_send(fd, chan, "agi_callingpres: %d\n", chan->cid.cid_pres);
+	ast_agi_send(fd, chan, "agi_callingani2: %d\n", chan->cid.cid_ani2);
+	ast_agi_send(fd, chan, "agi_callington: %d\n", chan->cid.cid_ton);
+	ast_agi_send(fd, chan, "agi_callingtns: %d\n", chan->cid.cid_tns);
+	ast_agi_send(fd, chan, "agi_dnid: %s\n", S_OR(chan->cid.cid_dnid, "unknown"));
+	ast_agi_send(fd, chan, "agi_rdnis: %s\n", S_OR(chan->cid.cid_rdnis, "unknown"));
 
 	/* Context information */
-	ast_agi_fdprintf(chan, fd, "agi_context: %s\n", chan->context);
-	ast_agi_fdprintf(chan, fd, "agi_extension: %s\n", chan->exten);
-	ast_agi_fdprintf(chan, fd, "agi_priority: %d\n", chan->priority);
-	ast_agi_fdprintf(chan, fd, "agi_enhanced: %s\n", enhanced ? "1.0" : "0.0");
+	ast_agi_send(fd, chan, "agi_context: %s\n", chan->context);
+	ast_agi_send(fd, chan, "agi_extension: %s\n", chan->exten);
+	ast_agi_send(fd, chan, "agi_priority: %d\n", chan->priority);
+	ast_agi_send(fd, chan, "agi_enhanced: %s\n", enhanced ? "1.0" : "0.0");
 
 	/* User information */
-	ast_agi_fdprintf(chan, fd, "agi_accountcode: %s\n", chan->accountcode ? chan->accountcode : "");
-	ast_agi_fdprintf(chan, fd, "agi_threadid: %ld\n", (long)pthread_self());
+	ast_agi_send(fd, chan, "agi_accountcode: %s\n", chan->accountcode ? chan->accountcode : "");
+	ast_agi_send(fd, chan, "agi_threadid: %ld\n", (long)pthread_self());
 
 	/* Send any parameters to the fastagi server that have been passed via the agi application */
 	/* Agi application paramaters take the form of: AGI(/path/to/example/script|${EXTEN}) */
 	for(count = 1; count < argc; count++)
-		ast_agi_fdprintf(chan, fd, "agi_arg_%d: %s\n", count, argv[count]);
+		ast_agi_send(fd, chan, "agi_arg_%d: %s\n", count, argv[count]);
 
 	/* End with empty return */
-	ast_agi_fdprintf(chan, fd, "\n");
+	ast_agi_send(fd, chan, "\n");
 }
 
 static int handle_answer(struct ast_channel *chan, AGI *agi, int argc, char *argv[])
@@ -787,7 +787,7 @@ static int handle_answer(struct ast_channel *chan, AGI *agi, int argc, char *arg
 	if (chan->_state != AST_STATE_UP)
 		res = ast_answer(chan);
 
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -800,7 +800,7 @@ static int handle_waitfordigit(struct ast_channel *chan, AGI *agi, int argc, cha
 	if (sscanf(argv[3], "%d", &to) != 1)
 		return RESULT_SHOWUSAGE;
 	res = ast_waitfordigit_full(chan, to, agi->audio, agi->ctrl);
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -819,7 +819,7 @@ static int handle_sendtext(struct ast_channel *chan, AGI *agi, int argc, char *a
 	   parsing, then here, add a newline at the end of the string
 	   before sending it to ast_sendtext --DUDE */
 	res = ast_sendtext(chan, argv[2]);
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -832,17 +832,15 @@ static int handle_recvchar(struct ast_channel *chan, AGI *agi, int argc, char *a
 
 	res = ast_recvchar(chan,atoi(argv[2]));
 	if (res == 0) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%d (timeout)\n", res);
+		ast_agi_send(agi->fd, chan, "200 result=%d (timeout)\n", res);
 		return RESULT_SUCCESS;
 	}
 	if (res > 0) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+		ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 		return RESULT_SUCCESS;
 	}
-	else {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%d (hangup)\n", res);
-		return RESULT_FAILURE;
-	}
+	ast_agi_send(agi->fd, chan, "200 result=%d (hangup)\n", res);
+	return RESULT_FAILURE;
 }
 
 static int handle_recvtext(struct ast_channel *chan, AGI *agi, int argc, char *argv[])
@@ -854,10 +852,10 @@ static int handle_recvtext(struct ast_channel *chan, AGI *agi, int argc, char *a
 
 	buf = ast_recvtext(chan,atoi(argv[2]));
 	if (buf) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1 (%s)\n", buf);
+		ast_agi_send(agi->fd, chan, "200 result=1 (%s)\n", buf);
 		ast_free(buf);
 	} else {	
-		ast_agi_fdprintf(chan, agi->fd, "200 result=-1\n");
+		ast_agi_send(agi->fd, chan, "200 result=-1\n");
 	}
 	return RESULT_SUCCESS;
 }
@@ -879,9 +877,9 @@ static int handle_tddmode(struct ast_channel *chan, AGI *agi, int argc, char *ar
 		x = 1;
 	res = ast_channel_setoption(chan, AST_OPTION_TDD, &x, sizeof(char), 0);
 	if (res != RESULT_SUCCESS)
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 	else
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+		ast_agi_send(agi->fd, chan, "200 result=1\n");
 	return RESULT_SUCCESS;
 }
 
@@ -895,7 +893,7 @@ static int handle_sendimage(struct ast_channel *chan, AGI *agi, int argc, char *
 	res = ast_send_image(chan, argv[2]);
 	if (!ast_check_hangup(chan))
 		res = 0;
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -932,7 +930,7 @@ static int handle_controlstreamfile(struct ast_channel *chan, AGI *agi, int argc
 	
 	res = ast_control_streamfile(chan, argv[3], fwd, rev, stop, pause, NULL, skipms, NULL);
 	
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
@@ -954,7 +952,7 @@ static int handle_streamfile(struct ast_channel *chan, AGI *agi, int argc, char
 		return RESULT_SHOWUSAGE;
 
 	if (!(fs = ast_openstream(chan, argv[2], chan->language))) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%d endpos=%ld\n", 0, sample_offset);
+		ast_agi_send(agi->fd, chan, "200 result=%d endpos=%ld\n", 0, sample_offset);
 		return RESULT_SUCCESS;
 	}
 
@@ -982,7 +980,7 @@ static int handle_streamfile(struct ast_channel *chan, AGI *agi, int argc, char
 		/* Stop this command, don't print a result line, as there is a new command */
 		return RESULT_SUCCESS;
 	}
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d endpos=%ld\n", res, sample_offset);
+	ast_agi_send(agi->fd, chan, "200 result=%d endpos=%ld\n", res, sample_offset);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -1009,7 +1007,7 @@ static int handle_getoption(struct ast_channel *chan, AGI *agi, int argc, char *
 	}
 
 	if (!(fs = ast_openstream(chan, argv[2], chan->language))) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%d endpos=%ld\n", 0, sample_offset);
+		ast_agi_send(agi->fd, chan, "200 result=%d endpos=%ld\n", 0, sample_offset);
 		ast_log(LOG_WARNING, "Unable to open %s\n", argv[2]);
 		return RESULT_SUCCESS;
 	}
@@ -1047,7 +1045,7 @@ static int handle_getoption(struct ast_channel *chan, AGI *agi, int argc, char *
 			res=0;
 	}
 
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d endpos=%ld\n", res, sample_offset);
+	ast_agi_send(agi->fd, chan, "200 result=%d endpos=%ld\n", res, sample_offset);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -1067,7 +1065,7 @@ static int handle_saynumber(struct ast_channel *chan, AGI *agi, int argc, char *
 	res = ast_say_number_full(chan, num, argv[3], chan->language, argc > 4 ? argv[4] : NULL, agi->audio, agi->ctrl);
 	if (res == 1)
 		return RESULT_SUCCESS;
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -1083,7 +1081,7 @@ static int handle_saydigits(struct ast_channel *chan, AGI *agi, int argc, char *
 	res = ast_say_digit_str_full(chan, argv[2], argv[3], chan->language, agi->audio, agi->ctrl);
 	if (res == 1) /* New command */
 		return RESULT_SUCCESS;
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -1097,7 +1095,7 @@ static int handle_sayalpha(struct ast_channel *chan, AGI *agi, int argc, char *a
 	res = ast_say_character_str_full(chan, argv[2], argv[3], chan->language, agi->audio, agi->ctrl);
 	if (res == 1) /* New command */
 		return RESULT_SUCCESS;
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -1112,7 +1110,7 @@ static int handle_saydate(struct ast_channel *chan, AGI *agi, int argc, char *ar
 	res = ast_say_date(chan, num, argv[3], chan->language);
 	if (res == 1)
 		return RESULT_SUCCESS;
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -1127,7 +1125,7 @@ static int handle_saytime(struct ast_channel *chan, AGI *agi, int argc, char *ar
 	res = ast_say_time(chan, num, argv[3], chan->language);
 	if (res == 1)
 		return RESULT_SUCCESS;
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -1161,7 +1159,7 @@ static int handle_saydatetime(struct ast_channel *chan, AGI *agi, int argc, char
 	if (res == 1)
 		return RESULT_SUCCESS;
 
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -1175,7 +1173,7 @@ static int handle_sayphonetic(struct ast_channel *chan, AGI *agi, int argc, char
 	res = ast_say_phonetic_str_full(chan, argv[2], argv[3], chan->language, agi->audio, agi->ctrl);
 	if (res == 1) /* New command */
 		return RESULT_SUCCESS;
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 	return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
 }
 
@@ -1198,11 +1196,11 @@ static int handle_getdata(struct ast_channel *chan, AGI *agi, int argc, char *ar
 	if (res == 2)			/* New command */
 		return RESULT_SUCCESS;
 	else if (res == 1)
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%s (timeout)\n", data);
+		ast_agi_send(agi->fd, chan, "200 result=%s (timeout)\n", data);
 	else if (res < 0 )
-		ast_agi_fdprintf(chan, agi->fd, "200 result=-1\n");
+		ast_agi_send(agi->fd, chan, "200 result=-1\n");
 	else
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%s\n", data);
+		ast_agi_send(agi->fd, chan, "200 result=%s\n", data);
 	return RESULT_SUCCESS;
 }
 
@@ -1212,7 +1210,7 @@ static int handle_setcontext(struct ast_channel *chan, AGI *agi, int argc, char
 	if (argc != 3)
 		return RESULT_SHOWUSAGE;
 	ast_copy_string(chan->context, argv[2], sizeof(chan->context));
-	ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+	ast_agi_send(agi->fd, chan, "200 result=0\n");
 	return RESULT_SUCCESS;
 }
 	
@@ -1221,7 +1219,7 @@ static int handle_setextension(struct ast_channel *chan, AGI *agi, int argc, cha
 	if (argc != 3)
 		return RESULT_SHOWUSAGE;
 	ast_copy_string(chan->exten, argv[2], sizeof(chan->exten));
-	ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+	ast_agi_send(agi->fd, chan, "200 result=0\n");
 	return RESULT_SUCCESS;
 }
 
@@ -1238,7 +1236,7 @@ static int handle_setpriority(struct ast_channel *chan, AGI *agi, int argc, char
 	}
 
 	ast_explicit_goto(chan, NULL, NULL, pri);
-	ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+	ast_agi_send(agi->fd, chan, "200 result=0\n");
 	return RESULT_SUCCESS;
 }
 		
@@ -1314,12 +1312,12 @@ static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, char
 	if (!res)
 		res = ast_waitstream(chan, argv[4]);
 	if (res) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%d (randomerror) endpos=%ld\n", res, sample_offset);
+		ast_agi_send(agi->fd, chan, "200 result=%d (randomerror) endpos=%ld\n", res, sample_offset);
 	} else {
 		fs = ast_writefile(argv[2], argv[3], NULL, O_CREAT | O_WRONLY | (sample_offset ? O_APPEND : 0), 0, AST_FILE_MODE);
 		if (!fs) {
 			res = -1;
-			ast_agi_fdprintf(chan, agi->fd, "200 result=%d (writefile)\n", res);
+			ast_agi_send(agi->fd, chan, "200 result=%d (writefile)\n", res);
 			if (sildet)
 				ast_dsp_free(sildet);
 			return RESULT_FAILURE;
@@ -1339,14 +1337,14 @@ static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, char
 			res = ast_waitfor(chan, -1);
 			if (res < 0) {
 				ast_closestream(fs);
-				ast_agi_fdprintf(chan, agi->fd, "200 result=%d (waitfor) endpos=%ld\n", res,sample_offset);
+				ast_agi_send(agi->fd, chan, "200 result=%d (waitfor) endpos=%ld\n", res,sample_offset);
 				if (sildet)
 					ast_dsp_free(sildet);
 				return RESULT_FAILURE;
 			}
 			f = ast_read(chan);
 			if (!f) {
-				ast_agi_fdprintf(chan, agi->fd, "200 result=%d (hangup) endpos=%ld\n", -1, sample_offset);
+				ast_agi_send(agi->fd, chan, "200 result=%d (hangup) endpos=%ld\n", -1, sample_offset);
 				ast_closestream(fs);
 				if (sildet)
 					ast_dsp_free(sildet);
@@ -1361,7 +1359,7 @@ static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, char
 					ast_stream_rewind(fs, 200);
 					ast_truncstream(fs);
 					sample_offset = ast_tellstream(fs);
-					ast_agi_fdprintf(chan, agi->fd, "200 result=%d (dtmf) endpos=%ld\n", f->subclass, sample_offset);
+					ast_agi_send(agi->fd, chan, "200 result=%d (dtmf) endpos=%ld\n", f->subclass, sample_offset);
 					ast_closestream(fs);
 					ast_frfree(f);
 					if (sildet)
@@ -1406,7 +1404,7 @@ static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, char
 				ast_truncstream(fs);
 				sample_offset = ast_tellstream(fs);
 		}		
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%d (timeout) endpos=%ld\n", res, sample_offset);
+		ast_agi_send(agi->fd, chan, "200 result=%d (timeout) endpos=%ld\n", res, sample_offset);
 		ast_closestream(fs);
 	}
 
@@ -1433,7 +1431,7 @@ static int handle_autohangup(struct ast_channel *chan, AGI *agi, int argc, char
 		chan->whentohangup = time(NULL) + timeout;
 	else
 		chan->whentohangup = 0;
-	ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+	ast_agi_send(agi->fd, chan, "200 result=0\n");
 	return RESULT_SUCCESS;
 }
 
@@ -1444,7 +1442,7 @@ static int handle_hangup(struct ast_channel *chan, AGI *agi, int argc, char **ar
 	if (argc == 1) {
 		/* no argument: hangup the current channel */
 		ast_softhangup(chan,AST_SOFTHANGUP_EXPLICIT);
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+		ast_agi_send(agi->fd, chan, "200 result=1\n");
 		return RESULT_SUCCESS;
 	} else if (argc == 2) {
 		/* one argument: look for info on the specified channel */
@@ -1452,12 +1450,12 @@ static int handle_hangup(struct ast_channel *chan, AGI *agi, int argc, char **ar
 		if (c) {
 			/* we have a matching channel */
 			ast_softhangup(c,AST_SOFTHANGUP_EXPLICIT);
-			ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+			ast_agi_send(agi->fd, chan, "200 result=1\n");
 			ast_channel_unlock(c);
 			return RESULT_SUCCESS;
 		}
 		/* if we get this far no channel name matched the argument given */
-		ast_agi_fdprintf(chan, agi->fd, "200 result=-1\n");
+		ast_agi_send(agi->fd, chan, "200 result=-1\n");
 		return RESULT_SUCCESS;
 	} else {
 		return RESULT_SHOWUSAGE;
@@ -1496,7 +1494,7 @@ static int handle_exec(struct ast_channel *chan, AGI *agi, int argc, char **argv
 		ast_log(LOG_WARNING, "Could not find application (%s)\n", argv[1]);
 		res = -2;
 	}
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", res);
+	ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
 
 	/* Even though this is wrong, users are depending upon this result. */
 	return res;
@@ -1519,7 +1517,7 @@ static int handle_setcallerid(struct ast_channel *chan, AGI *agi, int argc, char
 		ast_set_callerid(chan, l, n, NULL);
 	}
 
-	ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+	ast_agi_send(agi->fd, chan, "200 result=1\n");
 	return RESULT_SUCCESS;
 }
 
@@ -1528,18 +1526,18 @@ static int handle_channelstatus(struct ast_channel *chan, AGI *agi, int argc, ch
 	struct ast_channel *c;
 	if (argc == 2) {
 		/* no argument: supply info on the current channel */
-		ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", chan->_state);
+		ast_agi_send(agi->fd, chan, "200 result=%d\n", chan->_state);
 		return RESULT_SUCCESS;
 	} else if (argc == 3) {
 		/* one argument: look for info on the specified channel */
 		c = ast_get_channel_by_name_locked(argv[2]);
 		if (c) {
-			ast_agi_fdprintf(chan, agi->fd, "200 result=%d\n", c->_state);
+			ast_agi_send(agi->fd, chan, "200 result=%d\n", c->_state);
 			ast_channel_unlock(c);
 			return RESULT_SUCCESS;
 		}
 		/* if we get this far no channel name matched the argument given */
-		ast_agi_fdprintf(chan, agi->fd, "200 result=-1\n");
+		ast_agi_send(agi->fd, chan, "200 result=-1\n");
 		return RESULT_SUCCESS;
 	} else {
 		return RESULT_SHOWUSAGE;
@@ -1551,7 +1549,7 @@ static int handle_setvariable(struct ast_channel *chan, AGI *agi, int argc, char
 	if (argv[3])
 		pbx_builtin_setvar_helper(chan, argv[2], argv[3]);
 
-	ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+	ast_agi_send(agi->fd, chan, "200 result=1\n");
 	return RESULT_SUCCESS;
 }
 
@@ -1571,9 +1569,9 @@ static int handle_getvariable(struct ast_channel *chan, AGI *agi, int argc, char
 	}
 
 	if (ret)
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1 (%s)\n", ret);
+		ast_agi_send(agi->fd, chan, "200 result=1 (%s)\n", ret);
 	else
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 
 	return RESULT_SUCCESS;
 }
@@ -1592,9 +1590,9 @@ static int handle_getvariablefull(struct ast_channel *chan, AGI *agi, int argc,
 	}
 	if (chan2) {
 		pbx_substitute_variables_helper(chan2, argv[3], tmp, sizeof(tmp) - 1);
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1 (%s)\n", tmp);
+		ast_agi_send(agi->fd, chan, "200 result=1 (%s)\n", tmp);
 	} else {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 	}
 	if (chan2 && (chan2 != chan))
 		ast_channel_unlock(chan2);
@@ -1613,7 +1611,7 @@ static int handle_verbose(struct ast_channel *chan, AGI *agi, int argc, char **a
 
 	ast_verb(level, "%s: %s\n", chan->data, argv[1]);
 	
-	ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+	ast_agi_send(agi->fd, chan, "200 result=1\n");
 	
 	return RESULT_SUCCESS;
 }
@@ -1627,9 +1625,9 @@ static int handle_dbget(struct ast_channel *chan, AGI *agi, int argc, char **arg
 		return RESULT_SHOWUSAGE;
 	res = ast_db_get(argv[2], argv[3], tmp, sizeof(tmp));
 	if (res) 
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 	else
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1 (%s)\n", tmp);
+		ast_agi_send(agi->fd, chan, "200 result=1 (%s)\n", tmp);
 
 	return RESULT_SUCCESS;
 }
@@ -1641,7 +1639,7 @@ static int handle_dbput(struct ast_channel *chan, AGI *agi, int argc, char **arg
 	if (argc != 5)
 		return RESULT_SHOWUSAGE;
 	res = ast_db_put(argv[2], argv[3], argv[4]);
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%c\n", res ? '0' : '1');
+	ast_agi_send(agi->fd, chan, "200 result=%c\n", res ? '0' : '1');
 	return RESULT_SUCCESS;
 }
 
@@ -1652,7 +1650,7 @@ static int handle_dbdel(struct ast_channel *chan, AGI *agi, int argc, char **arg
 	if (argc != 4)
 		return RESULT_SHOWUSAGE;
 	res = ast_db_del(argv[2], argv[3]);
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%c\n", res ? '0' : '1');
+	ast_agi_send(agi->fd, chan, "200 result=%c\n", res ? '0' : '1');
 	return RESULT_SUCCESS;
 }
 
@@ -1667,7 +1665,7 @@ static int handle_dbdeltree(struct ast_channel *chan, AGI *agi, int argc, char *
 	else
 		res = ast_db_deltree(argv[2], NULL);
 
-	ast_agi_fdprintf(chan, agi->fd, "200 result=%c\n", res ? '0' : '1');
+	ast_agi_send(agi->fd, chan, "200 result=%c\n", res ? '0' : '1');
 	return RESULT_SUCCESS;
 }
 
@@ -1702,7 +1700,7 @@ static char *handle_cli_agi_debug(struct ast_cli_entry *e, int cmd, struct ast_c
 
 static int handle_noop(struct ast_channel *chan, AGI *agi, int arg, char *argv[])
 {
-	ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+	ast_agi_send(agi->fd, chan, "200 result=0\n");
 	return RESULT_SUCCESS;
 }
 
@@ -1712,7 +1710,7 @@ static int handle_setmusic(struct ast_channel *chan, AGI *agi, int argc, char *a
 		ast_moh_start(chan, argc > 3 ? argv[3] : NULL, NULL);
 	else if (!strncasecmp(argv[2], "off", 3))
 		ast_moh_stop(chan);
-	ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+	ast_agi_send(agi->fd, chan, "200 result=0\n");
 	return RESULT_SUCCESS;
 }
 
@@ -1720,14 +1718,14 @@ static int handle_speechcreate(struct ast_channel *chan, AGI *agi, int argc, cha
 {
 	/* If a structure already exists, return an error */
         if (agi->speech) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 		return RESULT_SUCCESS;
 	}
 	
 	if ((agi->speech = ast_speech_new(argv[2], AST_FORMAT_SLINEAR)))
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+		ast_agi_send(agi->fd, chan, "200 result=1\n");
 	else
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 	
 	return RESULT_SUCCESS;
 }
@@ -1740,12 +1738,12 @@ static int handle_speechset(struct ast_channel *chan, AGI *agi, int argc, char *
 	
 	/* Check to make sure speech structure exists */
 	if (!agi->speech) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 		return RESULT_SUCCESS;
 	}
 	
 	ast_speech_change(agi->speech, argv[2], argv[3]);
-	ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+	ast_agi_send(agi->fd, chan, "200 result=1\n");
 	
 	return RESULT_SUCCESS;
 }
@@ -1755,9 +1753,9 @@ static int handle_speechdestroy(struct ast_channel *chan, AGI *agi, int argc, ch
 	if (agi->speech) {
 		ast_speech_destroy(agi->speech);
 		agi->speech = NULL;
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+		ast_agi_send(agi->fd, chan, "200 result=1\n");
 	} else {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 	}
 	
 	return RESULT_SUCCESS;
@@ -1769,14 +1767,14 @@ static int handle_speechloadgrammar(struct ast_channel *chan, AGI *agi, int argc
 		return RESULT_SHOWUSAGE;
 	
 	if (!agi->speech) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 		return RESULT_SUCCESS;
 	}
 	
 	if (ast_speech_grammar_load(agi->speech, argv[3], argv[4]))
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 	else
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+		ast_agi_send(agi->fd, chan, "200 result=1\n");
 	
 	return RESULT_SUCCESS;
 }
@@ -1787,14 +1785,14 @@ static int handle_speechunloadgrammar(struct ast_channel *chan, AGI *agi, int ar
 		return RESULT_SHOWUSAGE;
 	
 	if (!agi->speech) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 		return RESULT_SUCCESS;
 	}
 	
 	if (ast_speech_grammar_unload(agi->speech, argv[3]))
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 	else
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+		ast_agi_send(agi->fd, chan, "200 result=1\n");
 	
 	return RESULT_SUCCESS;
 }
@@ -1805,14 +1803,14 @@ static int handle_speechactivategrammar(struct ast_channel *chan, AGI *agi, int
 		return RESULT_SHOWUSAGE;
 	
 	if (!agi->speech) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 		return RESULT_SUCCESS;
 	}
 	
 	if (ast_speech_grammar_activate(agi->speech, argv[3]))
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 	else
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+		ast_agi_send(agi->fd, chan, "200 result=1\n");
 	
 	return RESULT_SUCCESS;
 }
@@ -1823,14 +1821,14 @@ static int handle_speechdeactivategrammar(struct ast_channel *chan, AGI *agi, in
 		return RESULT_SHOWUSAGE;
 	
 	if (!agi->speech) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 		return RESULT_SUCCESS;
 	}
 	
 	if (ast_speech_grammar_deactivate(agi->speech, argv[3]))
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 	else
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1\n");
+		ast_agi_send(agi->fd, chan, "200 result=1\n");
 	
 	return RESULT_SUCCESS;
 }
@@ -1870,7 +1868,7 @@ static int handle_speechrecognize(struct ast_channel *chan, AGI *agi, int argc,
 		return RESULT_SHOWUSAGE;
 	
 	if (!speech) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 		return RESULT_SUCCESS;
 	}
 	
@@ -1884,7 +1882,7 @@ static int handle_speechrecognize(struct ast_channel *chan, AGI *agi, int argc,
 	/* We want frames coming in signed linear */
 	old_read_format = chan->readformat;
 	if (ast_set_read_format(chan, AST_FORMAT_SLINEAR)) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+		ast_agi_send(agi->fd, chan, "200 result=0\n");
 		return RESULT_SUCCESS;
 	}
 	
@@ -1989,13 +1987,13 @@ static int handle_speechrecognize(struct ast_channel *chan, AGI *agi, int argc,
 			i++;
 		}
                 /* Print out */
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1 (speech) endpos=%ld results=%d %s\n", current_offset, i, tmp);
+		ast_agi_send(agi->fd, chan, "200 result=1 (speech) endpos=%ld results=%d %s\n", current_offset, i, tmp);
 	} else if (!strcasecmp(reason, "dtmf")) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1 (digit) digit=%c endpos=%ld\n", dtmf, current_offset);
+		ast_agi_send(agi->fd, chan, "200 result=1 (digit) digit=%c endpos=%ld\n", dtmf, current_offset);
 	} else if (!strcasecmp(reason, "hangup") || !strcasecmp(reason, "timeout")) {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=1 (%s) endpos=%ld\n", reason, current_offset);
+		ast_agi_send(agi->fd, chan, "200 result=1 (%s) endpos=%ld\n", reason, current_offset);
 	} else {
-		ast_agi_fdprintf(chan, agi->fd, "200 result=0 endpos=%ld\n", current_offset);
+		ast_agi_send(agi->fd, chan, "200 result=0 endpos=%ld\n", current_offset);
 	}
 	
 	return RESULT_SUCCESS;
@@ -2003,7 +2001,7 @@ static int handle_speechrecognize(struct ast_channel *chan, AGI *agi, int argc,
 
 static int handle_asyncagi_break(struct ast_channel *chan, AGI *agi, int argc, char *argv[])
 {
-	ast_agi_fdprintf(chan, agi->fd, "200 result=0\n");
+	ast_agi_send(agi->fd, chan, "200 result=0\n");
 	return AST_PBX_KEEPALIVE;
 }
 
@@ -2418,21 +2416,50 @@ int ast_agi_unregister(struct ast_module *mod, agi_command *cmd)
 	return unregistered;
 }
 
-void ast_agi_register_multiple(struct ast_module *mod, agi_command *cmd, int len)
+int ast_agi_register_multiple(struct ast_module *mod, struct agi_command *cmd, unsigned int len)
 {
-	int i;
+	unsigned int i, x = 0;
+
+	for (i = 0; i < len; i++) {
+		if (ast_agi_register(mod, cmd + i) == 1) {
+			x++;
+			continue;
+		}
 
-	for (i = 0; i < len; i++)
-		ast_agi_register(mod, cmd + i);
+		/* registration failed, unregister everything
+		   that had been registered up to that point
+		*/
+		for (; x > 0; x--) {
+			/* we are intentionally ignoring the
+			   result of ast_agi_unregister() here,
+			   but it should be safe to do so since
+			   we just registered these commands and
+			   the only possible way for unregistration
+			   to fail is if the command is not
+			   registered
+			*/
+			(void) ast_agi_unregister(mod, cmd + x - 1);
+		}
+		return -1;
+	}
 
+	return 0;
 }
 
-void ast_agi_unregister_multiple(struct ast_module *mod, agi_command *cmd, int len)
+int ast_agi_unregister_multiple(struct ast_module *mod, struct agi_command *cmd, unsigned int len)
 {
-	int i;
+	unsigned int i;
+	int res = 0;
 
-	for (i = 0; i < len; i++)
-		ast_agi_unregister(mod, cmd + i);
+	for (i = 0; i < len; i++) {
+		/* remember whether any of the unregistration
+		   attempts failed... there is no recourse if
+		   any of them do
+		*/
+		res |= ast_agi_unregister(mod, cmd + i);
+	}
+
+	return res;
 }
 
 static agi_command *find_command(char *cmds[], int exact)
@@ -2574,9 +2601,9 @@ static int agi_handle_command(struct ast_channel *chan, AGI *agi, char *buf, int
 				"Result: %s\r\n", chan->name, command_id, ami_cmd, resultcode, ami_res);
 		switch(res) {
 		case RESULT_SHOWUSAGE:
-			ast_agi_fdprintf(chan, agi->fd, "520-Invalid command syntax.  Proper usage follows:\n");
-			ast_agi_fdprintf(chan, agi->fd, c->usage);
-			ast_agi_fdprintf(chan, agi->fd, "520 End of proper usage.\n");
+			ast_agi_send(agi->fd, chan, "520-Invalid command syntax.  Proper usage follows:\n");
+			ast_agi_send(agi->fd, NULL, "%s", c->usage);
+			ast_agi_send(agi->fd, chan, "520 End of proper usage.\n");
 			break;
 		case AST_PBX_KEEPALIVE:
 			/* We've been asked to keep alive, so do so */
@@ -2588,7 +2615,7 @@ static int agi_handle_command(struct ast_channel *chan, AGI *agi, char *buf, int
 			return -1;
 		}
 	} else if ((c = find_command(argv, 0))) {
-		ast_agi_fdprintf(chan, agi->fd, "511 Command Not Permitted on a dead channel\n");
+		ast_agi_send(agi->fd, chan, "511 Command Not Permitted on a dead channel\n");
 		manager_event(EVENT_FLAG_CALL, "AGIExec",
 				"SubEvent: End\r\n"
 				"Channel: %s\r\n"
@@ -2597,7 +2624,7 @@ static int agi_handle_command(struct ast_channel *chan, AGI *agi, char *buf, int
 				"ResultCode: 511\r\n"
 				"Result: Command not permitted on a dead channel\r\n", chan->name, command_id, ami_cmd);
 	} else {
-		ast_agi_fdprintf(chan, agi->fd, "510 Invalid or unknown command\n");
+		ast_agi_send(agi->fd, chan, "510 Invalid or unknown command\n");
 		manager_event(EVENT_FLAG_CALL, "AGIExec",
 				"SubEvent: End\r\n"
 				"Channel: %s\r\n"
@@ -3018,7 +3045,10 @@ static struct ast_cli_entry cli_agi[] = {
 static int unload_module(void)
 {
 	ast_cli_unregister_multiple(cli_agi, sizeof(cli_agi) / sizeof(struct ast_cli_entry));
-	ast_agi_unregister_multiple(ast_module_info->self, commands, sizeof(commands) / sizeof(struct agi_command));
+	/* we can safely ignore the result of ast_agi_unregister_multiple() here, since it cannot fail, as
+	   we know that these commands were registered by this module and are still registered
+	*/
+	(void) ast_agi_unregister_multiple(ast_module_info->self, commands, ARRAY_LEN(commands));
 	ast_unregister_application(eapp);
 	ast_unregister_application(deadapp);
 	ast_manager_unregister("AGI");
@@ -3028,7 +3058,10 @@ static int unload_module(void)
 static int load_module(void)
 {
 	ast_cli_register_multiple(cli_agi, sizeof(cli_agi) / sizeof(struct ast_cli_entry));
-	ast_agi_register_multiple(ast_module_info->self, commands, sizeof(commands) / sizeof(struct agi_command));
+	/* we can safely ignore the result of ast_agi_register_multiple() here, since it cannot fail, as
+	   no other commands have been registered yet
+	*/
+	(void) ast_agi_register_multiple(ast_module_info->self, commands, ARRAY_LEN(commands));
 	ast_register_application(deadapp, deadagi_exec, deadsynopsis, descrip);
 	ast_register_application(eapp, eagi_exec, esynopsis, descrip);
 	ast_manager_register2("AGI", EVENT_FLAG_CALL, action_add_agi_cmd, "Add an AGI command to execute by Async AGI", mandescr_asyncagi);